From ad233a5f0f33a894f48c7d968ec207f46cbcae03 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 18 Nov 2013 17:47:15 +0100
Subject: ALSA: cs46xx: Move DSP blob out of kernel tree

... and load from the external firmware files.

The firmware binary blobs in cs46xx driver have been in a gray zone
regarding the license.  It's most likely should be OK, but still
unclear.  And, the size isn't that small, too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=10750
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig                  |    1 +
 sound/pci/cs46xx/cs46xx.h          |    5 +
 sound/pci/cs46xx/cs46xx_image.h    | 3468 ------------------------------------
 sound/pci/cs46xx/cs46xx_lib.c      |  265 ++-
 sound/pci/cs46xx/imgs/cwc4630.h    |  320 ----
 sound/pci/cs46xx/imgs/cwcasync.h   |  176 --
 sound/pci/cs46xx/imgs/cwcbinhack.h |   48 -
 sound/pci/cs46xx/imgs/cwcdma.asp   |  170 --
 sound/pci/cs46xx/imgs/cwcdma.h     |   68 -
 sound/pci/cs46xx/imgs/cwcsnoop.h   |   46 -
 10 files changed, 226 insertions(+), 4341 deletions(-)
 delete mode 100644 sound/pci/cs46xx/cs46xx_image.h
 delete mode 100644 sound/pci/cs46xx/imgs/cwc4630.h
 delete mode 100644 sound/pci/cs46xx/imgs/cwcasync.h
 delete mode 100644 sound/pci/cs46xx/imgs/cwcbinhack.h
 delete mode 100644 sound/pci/cs46xx/imgs/cwcdma.asp
 delete mode 100644 sound/pci/cs46xx/imgs/cwcdma.h
 delete mode 100644 sound/pci/cs46xx/imgs/cwcsnoop.h

(limited to 'sound')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 46ed9e8ae0fd..3a0504dfeb8a 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -254,6 +254,7 @@ config SND_CS46XX
 	tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x"
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
+	select FW_LOADER
 	help
 	  Say Y here to include support for Cirrus Logic CS4610/CS4612/
 	  CS4614/CS4615/CS4622/CS4624/CS4630/CS4280 chips.
diff --git a/sound/pci/cs46xx/cs46xx.h b/sound/pci/cs46xx/cs46xx.h
index fc339ef0a0ae..c49a082c378b 100644
--- a/sound/pci/cs46xx/cs46xx.h
+++ b/sound/pci/cs46xx/cs46xx.h
@@ -1716,9 +1716,14 @@ struct snd_cs46xx {
 	struct snd_pcm *pcm_rear;
 	struct snd_pcm *pcm_center_lfe;
 	struct snd_pcm *pcm_iec958;
+
+#define CS46XX_DSP_MODULES	5
+	struct dsp_module_desc *modules[CS46XX_DSP_MODULES];
 #else /* for compatibility */
 	struct snd_cs46xx_pcm *playback_pcm;
 	unsigned int play_ctl;
+
+	struct ba1_struct *ba1;
 #endif
 
 #ifdef CONFIG_PM_SLEEP
diff --git a/sound/pci/cs46xx/cs46xx_image.h b/sound/pci/cs46xx/cs46xx_image.h
deleted file mode 100644
index dc93f62db2c2..000000000000
--- a/sound/pci/cs46xx/cs46xx_image.h
+++ /dev/null
@@ -1,3468 +0,0 @@
-struct BA1struct {
-	struct {
-		unsigned long offset;
-		unsigned long size;
-	} memory[BA1_MEMORY_COUNT];
-	u32 map[BA1_DWORD_SIZE];
-};
-
-
-static struct BA1struct BA1Struct = {
-{{ 0x00000000, 0x00003000 },{ 0x00010000, 0x00003800 },{ 0x00020000, 0x00007000 }},
-{0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000163,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00200040,0x00008010,0x00000000,
-0x00000000,0x80000001,0x00000001,0x00060000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00900080,0x00000173,0x00000000,
-0x00000000,0x00000010,0x00800000,0x00900000,
-0xf2c0000f,0x00000200,0x00000000,0x00010600,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000163,0x330300c2,
-0x06000000,0x00000000,0x80008000,0x80008000,
-0x3fc0000f,0x00000301,0x00010400,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00b00000,0x00d0806d,0x330480c3,
-0x04800000,0x00000001,0x00800001,0x0000ffff,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
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-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x066a0600,0x06350070,0x0000929d,0x929d929d,
-0x00000000,0x0000735a,0x00000600,0x00000000,
-0x929d735a,0x8734abfe,0x00010000,0x735a735a,
-0xa431ac75,0xa431ac75,0xa431ac75,0xa431ac75,
-0x00000000,0x00000000,0x00000000,0x00000000,
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-0x00000000,0x00000000,0x0000804f,0x000000c3,
-0x05000000,0x00a00010,0x00000000,0x80008000,
-0x00000000,0x00000000,0x00000700,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000080,0x00a00000,0x0000809a,0x000000c2,
-0x07400000,0x00000000,0x80008000,0xffffffff,
-0x00c80028,0x00005555,0x00000000,0x000107a0,
-0x00c80028,0x000000c2,0x06800000,0x00000000,
-0x06e00080,0x00300000,0x000080bb,0x000000c9,
-0x07a00000,0x04000000,0x80008000,0xffffffff,
-0x00c80028,0x00005555,0x00000000,0x00000780,
-0x00c80028,0x000000c5,0xff800000,0x00000000,
-0x00640080,0x00c00000,0x00008197,0x000000c9,
-0x07800000,0x04000000,0x80008000,0xffffffff,
-0x00000000,0x00000000,0x00000000,0x00000000,
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-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000,
-0x00000000,0x00000000,0x00000000,0x00000000}
- };
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 1b66efd9b728..f18e5878f58b 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -54,7 +54,9 @@
 #include <linux/gameport.h>
 #include <linux/mutex.h>
 #include <linux/export.h>
-
+#include <linux/module.h>
+#include <linux/firmware.h>
+#include <linux/vmalloc.h>
 
 #include <sound/core.h>
 #include <sound/control.h>
@@ -330,13 +332,146 @@ int snd_cs46xx_download(struct snd_cs46xx *chip,
 	return 0;
 }
 
+static inline void memcpy_le32(void *dst, const void *src, unsigned int len)
+{
+#ifdef __LITTLE_ENDIAN
+	memcpy(dst, src, len);
+#else
+	u32 *_dst = dst;
+	const __le32 *_src = src;
+	len /= 4;
+	while (len-- > 0)
+		*_dst++ = le32_to_cpu(*_src++);
+#endif
+}
+
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 
-#include "imgs/cwc4630.h"
-#include "imgs/cwcasync.h"
-#include "imgs/cwcsnoop.h"
-#include "imgs/cwcbinhack.h"
-#include "imgs/cwcdma.h"
+static const char *module_names[CS46XX_DSP_MODULES] = {
+	"cwc4630", "cwcasync", "cwcsnoop", "cwcbinhack", "cwcdma"
+};
+
+MODULE_FIRMWARE("cs46xx/cwc4630");
+MODULE_FIRMWARE("cs46xx/cwcasync");
+MODULE_FIRMWARE("cs46xx/cwcsnoop");
+MODULE_FIRMWARE("cs46xx/cwcbinhack");
+MODULE_FIRMWARE("cs46xx/cwcdma");
+
+static void free_module_desc(struct dsp_module_desc *module)
+{
+	if (!module)
+		return;
+	kfree(module->module_name);
+	kfree(module->symbol_table.symbols);
+	if (module->segments) {
+		int i;
+		for (i = 0; i < module->nsegments; i++)
+			kfree(module->segments[i].data);
+		kfree(module->segments);
+	}
+}
+
+/* firmware binary format:
+ * le32 nsymbols;
+ * struct {
+ *	le32 address;
+ *	char symbol_name[DSP_MAX_SYMBOL_NAME];
+ *	le32 symbol_type;
+ * } symbols[nsymbols];
+ * le32 nsegments;
+ * struct {
+ *	le32 segment_type;
+ *	le32 offset;
+ *	le32 size;
+ *	le32 data[size];
+ * } segments[nsegments];
+ */
+
+static int load_firmware(struct snd_cs46xx *chip,
+			 struct dsp_module_desc **module_ret,
+			 const char *fw_name)
+{
+	int i, err;
+	unsigned int nums, fwlen, fwsize;
+	const __le32 *fwdat;
+	struct dsp_module_desc *module = NULL;
+	const struct firmware *fw;
+	char fw_path[32];
+
+	sprintf(fw_path, "cs46xx/%s", fw_name);
+	err = request_firmware(&fw, fw_path, &chip->pci->dev);
+	if (err < 0)
+		return err;
+	fwsize = fw->size / 4;
+	if (fwsize < 2) {
+		err = -EINVAL;
+		goto error;
+	}
+
+	err = -ENOMEM;
+	module = kzalloc(sizeof(*module), GFP_KERNEL);
+	if (!module)
+		goto error;
+	module->module_name = kstrdup(fw_name, GFP_KERNEL);
+	if (!module->module_name)
+		goto error;
+
+	fwlen = 0;
+	fwdat = (const __le32 *)fw->data;
+	nums = module->symbol_table.nsymbols = le32_to_cpu(fwdat[fwlen++]);
+	if (nums >= 40)
+		goto error_inval;
+	module->symbol_table.symbols =
+		kcalloc(nums, sizeof(struct dsp_symbol_entry), GFP_KERNEL);
+	if (!module->symbol_table.symbols)
+		goto error;
+	for (i = 0; i < nums; i++) {
+		struct dsp_symbol_entry *entry =
+			&module->symbol_table.symbols[i];
+		if (fwlen + 2 + DSP_MAX_SYMBOL_NAME / 4 > fwsize)
+			goto error_inval;
+		entry->address = le32_to_cpu(fwdat[fwlen++]);
+		memcpy(entry->symbol_name, &fwdat[fwlen], DSP_MAX_SYMBOL_NAME - 1);
+		fwlen += DSP_MAX_SYMBOL_NAME / 4;
+		entry->symbol_type = le32_to_cpu(fwdat[fwlen++]);
+	}
+
+	if (fwlen >= fwsize)
+		goto error_inval;
+	nums = module->nsegments = le32_to_cpu(fwdat[fwlen++]);
+	if (nums > 10)
+		goto error_inval;
+	module->segments =
+		kcalloc(nums, sizeof(struct dsp_segment_desc), GFP_KERNEL);
+	if (!module->segments)
+		goto error;
+	for (i = 0; i < nums; i++) {
+		struct dsp_segment_desc *entry = &module->segments[i];
+		if (fwlen + 3 > fwsize)
+			goto error_inval;
+		entry->segment_type = le32_to_cpu(fwdat[fwlen++]);
+		entry->offset = le32_to_cpu(fwdat[fwlen++]);
+		entry->size = le32_to_cpu(fwdat[fwlen++]);
+		if (fwlen + entry->size > fwsize)
+			goto error_inval;
+		entry->data = kmalloc(entry->size * 4, GFP_KERNEL);
+		if (!entry->data)
+			goto error;
+		memcpy_le32(entry->data, &fwdat[fwlen], entry->size * 4);
+		fwlen += entry->size;
+	}
+
+	*module_ret = module;
+	release_firmware(fw);
+	return 0;
+
+ error_inval:
+	err = -EINVAL;
+ error:
+	free_module_desc(module);
+	release_firmware(fw);
+	return err;
+}
 
 int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip,
                          unsigned long offset,
@@ -361,20 +496,63 @@ int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip,
 
 #else /* old DSP image */
 
-#include "cs46xx_image.h"
+struct ba1_struct {
+	struct {
+		u32 offset;
+		u32 size;
+	} memory[BA1_MEMORY_COUNT];
+	u32 map[BA1_DWORD_SIZE];
+};
+
+MODULE_FIRMWARE("cs46xx/ba1");
+
+static int load_firmware(struct snd_cs46xx *chip)
+{
+	const struct firmware *fw;
+	int i, size, err;
+
+	err = request_firmware(&fw, "cs46xx/ba1", &chip->pci->dev);
+	if (err < 0)
+		return err;
+	if (fw->size != sizeof(*chip->ba1)) {
+		err = -EINVAL;
+		goto error;
+	}
+
+	chip->ba1 = vmalloc(sizeof(*chip->ba1));
+	if (!chip->ba1) {
+		err = -ENOMEM;
+		goto error;
+	}
+
+	memcpy_le32(chip->ba1, fw->data, sizeof(*chip->ba1));
+
+	/* sanity check */
+	size = 0;
+	for (i = 0; i < BA1_MEMORY_COUNT; i++)
+		size += chip->ba1->memory[i].size;
+	if (size > BA1_DWORD_SIZE * 4)
+		err = -EINVAL;
+
+ error:
+	release_firmware(fw);
+	return err;
+}
 
 int snd_cs46xx_download_image(struct snd_cs46xx *chip)
 {
 	int idx, err;
-	unsigned long offset = 0;
+	unsigned int offset = 0;
+	struct ba1_struct *ba1 = chip->ba1;
 
 	for (idx = 0; idx < BA1_MEMORY_COUNT; idx++) {
-		if ((err = snd_cs46xx_download(chip,
-					       &BA1Struct.map[offset],
-					       BA1Struct.memory[idx].offset,
-					       BA1Struct.memory[idx].size)) < 0)
+		err = snd_cs46xx_download(chip,
+					  &ba1->map[offset],
+					  ba1->memory[idx].offset,
+					  ba1->memory[idx].size);
+		if (err < 0)
 			return err;
-		offset += BA1Struct.memory[idx].size >> 2;
+		offset += ba1->memory[idx].size >> 2;
 	}	
 	return 0;
 }
@@ -2798,6 +2976,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip)
 		cs46xx_dsp_spos_destroy(chip);
 		chip->dsp_spos_instance = NULL;
 	}
+	for (idx = 0; idx < CS46XX_DSP_MODULES; idx++)
+		free_module_desc(chip->modules[idx]);
+#else
+	vfree(chip->ba1);
 #endif
 	
 #ifdef CONFIG_PM_SLEEP
@@ -3067,6 +3249,11 @@ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip)
 int snd_cs46xx_start_dsp(struct snd_cs46xx *chip)
 {	
 	unsigned int tmp;
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+	int i;
+#endif
+	int err;
+
 	/*
 	 *  Reset the processor.
 	 */
@@ -3075,45 +3262,33 @@ int snd_cs46xx_start_dsp(struct snd_cs46xx *chip)
 	 *  Download the image to the processor.
 	 */
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-#if 0
-	if (cs46xx_dsp_load_module(chip, &cwcemb80_module) < 0) {
-		snd_printk(KERN_ERR "image download error\n");
-		return -EIO;
-	}
-#endif
-
-	if (cs46xx_dsp_load_module(chip, &cwc4630_module) < 0) {
-		snd_printk(KERN_ERR "image download error [cwc4630]\n");
-		return -EIO;
-	}
-
-	if (cs46xx_dsp_load_module(chip, &cwcasync_module) < 0) {
-		snd_printk(KERN_ERR "image download error [cwcasync]\n");
-		return -EIO;
-	}
-
-	if (cs46xx_dsp_load_module(chip, &cwcsnoop_module) < 0) {
-		snd_printk(KERN_ERR "image download error [cwcsnoop]\n");
-		return -EIO;
-	}
-
-	if (cs46xx_dsp_load_module(chip, &cwcbinhack_module) < 0) {
-		snd_printk(KERN_ERR "image download error [cwcbinhack]\n");
-		return -EIO;
-	}
-
-	if (cs46xx_dsp_load_module(chip, &cwcdma_module) < 0) {
-		snd_printk(KERN_ERR "image download error [cwcdma]\n");
-		return -EIO;
+	for (i = 0; i < CS46XX_DSP_MODULES; i++) {
+		err = load_firmware(chip, &chip->modules[i], module_names[i]);
+		if (err < 0) {
+			snd_printk(KERN_ERR "firmware load error [%s]\n",
+				   module_names[i]);
+			return err;
+		}
+		err = cs46xx_dsp_load_module(chip, chip->modules[i]);
+		if (err < 0) {
+			snd_printk(KERN_ERR "image download error [%s]\n",
+				   module_names[i]);
+			return err;
+		}
 	}
 
 	if (cs46xx_dsp_scb_and_task_init(chip) < 0)
 		return -EIO;
 #else
+	err = load_firmware(chip);
+	if (err < 0)
+		return err;
+
 	/* old image */
-	if (snd_cs46xx_download_image(chip) < 0) {
+	err = snd_cs46xx_download_image(chip);
+	if (err < 0) {
 		snd_printk(KERN_ERR "image download error\n");
-		return -EIO;
+		return err;
 	}
 
 	/*
diff --git a/sound/pci/cs46xx/imgs/cwc4630.h b/sound/pci/cs46xx/imgs/cwc4630.h
deleted file mode 100644
index 37c4f1318dc5..000000000000
--- a/sound/pci/cs46xx/imgs/cwc4630.h
+++ /dev/null
@@ -1,320 +0,0 @@
-/* generated from cwc4630.osp DO NOT MODIFY */
-
-#ifndef __HEADER_cwc4630_H__
-#define __HEADER_cwc4630_H__
-
-static struct dsp_symbol_entry cwc4630_symbols[] = {
-  { 0x0000, "BEGINADDRESS",0x00 },
-  { 0x8000, "EXECCHILD",0x03 },
-  { 0x8001, "EXECCHILD_98",0x03 },
-  { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
-  { 0x8008, "EXECSIBLING",0x03 },
-  { 0x800a, "EXECSIBLING_298",0x03 },
-  { 0x800b, "EXECSIBLING_2IND1",0x03 },
-  { 0x8010, "TIMINGMASTER",0x03 },
-  { 0x804f, "S16_CODECINPUTTASK",0x03 },
-  { 0x805e, "PCMSERIALINPUTTASK",0x03 },
-  { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
-  { 0x809a, "S16_MIX",0x03 },
-  { 0x80bb, "S16_UPSRC",0x03 },
-  { 0x813b, "MIX3_EXP",0x03 },
-  { 0x8164, "DECIMATEBYPOW2",0x03 },
-  { 0x8197, "VARIDECIMATE",0x03 },
-  { 0x81f2, "_3DINPUTTASK",0x03 },
-  { 0x820a, "_3DPRLGCINPTASK",0x03 },
-  { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
-  { 0x8242, "_3DOUTPUTTASK",0x03 },
-  { 0x82c4, "HRTF_MORPH_TASK",0x03 },
-  { 0x82c6, "WAIT4DATA",0x03 },
-  { 0x82fa, "PROLOGIC",0x03 },
-  { 0x8496, "DECORRELATOR",0x03 },
-  { 0x84a4, "STEREO2MONO",0x03 },
-  { 0x0070, "SPOSCB",0x02 },
-  { 0x0107, "TASKTREETHREAD",0x03 },
-  { 0x013c, "TASKTREEHEADERCODE",0x03 },
-  { 0x0145, "FGTASKTREEHEADERCODE",0x03 },
-  { 0x0169, "NULLALGORITHM",0x03 },
-  { 0x016d, "HFGEXECCHILD",0x03 },
-  { 0x016e, "HFGEXECCHILD_98",0x03 },
-  { 0x0170, "HFGEXECCHILD_PUSH1IND",0x03 },
-  { 0x0173, "HFGEXECSIBLING",0x03 },
-  { 0x0175, "HFGEXECSIBLING_298",0x03 },
-  { 0x0176, "HFGEXECSIBLING_2IND1",0x03 },
-  { 0x0179, "S16_CODECOUTPUTTASK",0x03 },
-  { 0x0194, "#CODE_END",0x00 },
-}; /* cwc4630 symbols */
-
-static u32 cwc4630_code[] = {
-/* BEGINADDRESS */
-/* 0000 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 0002 */ 0x00001705,0x00001400,0x000a411e,0x00001003,
-/* 0004 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 0006 */ 0x00009705,0x00001400,0x000a411e,0x00001003,
-/* 0008 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 000A */ 0x00011705,0x00001400,0x000a411e,0x00001003,
-/* 000C */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 000E */ 0x00019705,0x00001400,0x000a411e,0x00001003,
-/* 0010 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 0012 */ 0x00021705,0x00001400,0x000a411e,0x00001003,
-/* 0014 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 0016 */ 0x00029705,0x00001400,0x000a411e,0x00001003,
-/* 0018 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 001A */ 0x00031705,0x00001400,0x000a411e,0x00001003,
-/* 001C */ 0x00040730,0x00001002,0x000f619e,0x00001003,
-/* 001E */ 0x00039705,0x00001400,0x000a411e,0x00001003,
-/* 0020 */ 0x000fe19e,0x00001003,0x0009c730,0x00001003,
-/* 0022 */ 0x0008e19c,0x00001003,0x000083c1,0x00093040,
-/* 0024 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 0026 */ 0x00009705,0x00001400,0x000a211e,0x00001003,
-/* 0028 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 002A */ 0x00011705,0x00001400,0x000a211e,0x00001003,
-/* 002C */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 002E */ 0x00019705,0x00001400,0x000a211e,0x00001003,
-/* 0030 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 0032 */ 0x00021705,0x00001400,0x000a211e,0x00001003,
-/* 0034 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 0036 */ 0x00029705,0x00001400,0x000a211e,0x00001003,
-/* 0038 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 003A */ 0x00031705,0x00001400,0x000a211e,0x00001003,
-/* 003C */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
-/* 003E */ 0x00039705,0x00001400,0x000a211e,0x00001003,
-/* 0040 */ 0x0001a730,0x00001008,0x000e2730,0x00001002,
-/* 0042 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
-/* 0044 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
-/* 0046 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
-/* 0048 */ 0x00000000,0x00000000,0x000f619c,0x00001003,
-/* 004A */ 0x0007f801,0x000c0000,0x00000037,0x00001000,
-/* 004C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 004E */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0050 */ 0x00000000,0x000c0000,0x00000000,0x00000000,
-/* 0052 */ 0x0000373c,0x00001000,0x00000000,0x00000000,
-/* 0054 */ 0x000ee19c,0x00001003,0x0007f801,0x000c0000,
-/* 0056 */ 0x00000037,0x00001000,0x00000000,0x00000000,
-/* 0058 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 005A */ 0x00000000,0x00000000,0x0000273c,0x00001000,
-/* 005C */ 0x00000033,0x00001000,0x000e679e,0x00001003,
-/* 005E */ 0x00007705,0x00001400,0x000ac71e,0x00001003,
-/* 0060 */ 0x00087fc1,0x000c3be0,0x0007f801,0x000c0000,
-/* 0062 */ 0x00000037,0x00001000,0x00000000,0x00000000,
-/* 0064 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0066 */ 0x00000000,0x00000000,0x0000a730,0x00001003,
-/* 0068 */ 0x00000033,0x00001000,0x0007f801,0x000c0000,
-/* 006A */ 0x00000037,0x00001000,0x00000000,0x00000000,
-/* 006C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 006E */ 0x00000000,0x00000000,0x00000000,0x000c0000,
-/* 0070 */ 0x00000032,0x00001000,0x0000273d,0x00001000,
-/* 0072 */ 0x0004a730,0x00001003,0x00000f41,0x00097140,
-/* 0074 */ 0x0000a841,0x0009b240,0x0000a0c1,0x0009f040,
-/* 0076 */ 0x0001c641,0x00093540,0x0001cec1,0x0009b5c0,
-/* 0078 */ 0x00000000,0x00000000,0x0001bf05,0x0003fc40,
-/* 007A */ 0x00002725,0x000aa400,0x00013705,0x00093a00,
-/* 007C */ 0x0000002e,0x0009d6c0,0x0002ef8a,0x00000000,
-/* 007E */ 0x00040630,0x00001004,0x0004ef0a,0x000eb785,
-/* 0080 */ 0x0003fc8a,0x00000000,0x00000000,0x000c70e0,
-/* 0082 */ 0x0007d182,0x0002c640,0x00008630,0x00001004,
-/* 0084 */ 0x000799b8,0x0002c6c0,0x00031705,0x00092240,
-/* 0086 */ 0x00039f05,0x000932c0,0x0003520a,0x00000000,
-/* 0088 */ 0x00070731,0x0000100b,0x00010705,0x000b20c0,
-/* 008A */ 0x00000000,0x000eba44,0x00032108,0x000c60c4,
-/* 008C */ 0x00065208,0x000c2917,0x000486b0,0x00001007,
-/* 008E */ 0x00012f05,0x00036880,0x0002818e,0x000c0000,
-/* 0090 */ 0x0004410a,0x00000000,0x00048630,0x00001007,
-/* 0092 */ 0x00029705,0x000c0000,0x00000000,0x00000000,
-/* 0094 */ 0x00003fc1,0x0003fc40,0x000037c1,0x00091b40,
-/* 0096 */ 0x00003fc1,0x000911c0,0x000037c1,0x000957c0,
-/* 0098 */ 0x00003fc1,0x000951c0,0x000037c1,0x00000000,
-/* 009A */ 0x00003fc1,0x000991c0,0x000037c1,0x00000000,
-/* 009C */ 0x00003fc1,0x0009d1c0,0x000037c1,0x00000000,
-/* 009E */ 0x0001ccc1,0x000915c0,0x0001c441,0x0009d800,
-/* 00A0 */ 0x0009cdc1,0x00091240,0x0001c541,0x00091d00,
-/* 00A2 */ 0x0009cfc1,0x00095240,0x0001c741,0x00095c80,
-/* 00A4 */ 0x000e8ca9,0x00099240,0x000e85ad,0x00095640,
-/* 00A6 */ 0x00069ca9,0x00099d80,0x000e952d,0x00099640,
-/* 00A8 */ 0x000eaca9,0x0009d6c0,0x000ea5ad,0x00091a40,
-/* 00AA */ 0x0006bca9,0x0009de80,0x000eb52d,0x00095a40,
-/* 00AC */ 0x000ecca9,0x00099ac0,0x000ec5ad,0x0009da40,
-/* 00AE */ 0x000edca9,0x0009d300,0x000a6e0a,0x00001000,
-/* 00B0 */ 0x000ed52d,0x00091e40,0x000eeca9,0x00095ec0,
-/* 00B2 */ 0x000ee5ad,0x00099e40,0x0006fca9,0x00002500,
-/* 00B4 */ 0x000fb208,0x000c59a0,0x000ef52d,0x0009de40,
-/* 00B6 */ 0x00068ca9,0x000912c1,0x000683ad,0x00095241,
-/* 00B8 */ 0x00020f05,0x000991c1,0x00000000,0x00000000,
-/* 00BA */ 0x00086f88,0x00001000,0x0009cf81,0x000b5340,
-/* 00BC */ 0x0009c701,0x000b92c0,0x0009de81,0x000bd300,
-/* 00BE */ 0x0009d601,0x000b1700,0x0001fd81,0x000b9d80,
-/* 00C0 */ 0x0009f501,0x000b57c0,0x000a0f81,0x000bd740,
-/* 00C2 */ 0x00020701,0x000b5c80,0x000a1681,0x000b97c0,
-/* 00C4 */ 0x00021601,0x00002500,0x000a0701,0x000b9b40,
-/* 00C6 */ 0x000a0f81,0x000b1bc0,0x00021681,0x00002d00,
-/* 00C8 */ 0x00020f81,0x000bd800,0x000a0701,0x000b5bc0,
-/* 00CA */ 0x00021601,0x00003500,0x000a0f81,0x000b5f40,
-/* 00CC */ 0x000a0701,0x000bdbc0,0x00021681,0x00003d00,
-/* 00CE */ 0x00020f81,0x000b1d00,0x000a0701,0x000b1fc0,
-/* 00D0 */ 0x00021601,0x00020500,0x00020f81,0x000b1341,
-/* 00D2 */ 0x000a0701,0x000b9fc0,0x00021681,0x00020d00,
-/* 00D4 */ 0x00020f81,0x000bde80,0x000a0701,0x000bdfc0,
-/* 00D6 */ 0x00021601,0x00021500,0x00020f81,0x000b9341,
-/* 00D8 */ 0x00020701,0x000b53c1,0x00021681,0x00021d00,
-/* 00DA */ 0x000a0f81,0x000d0380,0x0000b601,0x000b15c0,
-/* 00DC */ 0x00007b01,0x00000000,0x00007b81,0x000bd1c0,
-/* 00DE */ 0x00007b01,0x00000000,0x00007b81,0x000b91c0,
-/* 00E0 */ 0x00007b01,0x000b57c0,0x00007b81,0x000b51c0,
-/* 00E2 */ 0x00007b01,0x000b1b40,0x00007b81,0x000b11c0,
-/* 00E4 */ 0x00087b01,0x000c3dc0,0x0007e488,0x000d7e45,
-/* 00E6 */ 0x00000000,0x000d7a44,0x0007e48a,0x00000000,
-/* 00E8 */ 0x00011f05,0x00084080,0x00000000,0x00000000,
-/* 00EA */ 0x00001705,0x000b3540,0x00008a01,0x000bf040,
-/* 00EC */ 0x00007081,0x000bb5c0,0x00055488,0x00000000,
-/* 00EE */ 0x0000d482,0x0003fc40,0x0003fc88,0x00000000,
-/* 00F0 */ 0x0001e401,0x000b3a00,0x0001ec81,0x000bd6c0,
-/* 00F2 */ 0x0002ef88,0x000e7784,0x00056f08,0x00000000,
-/* 00F4 */ 0x000d86b0,0x00001007,0x00008281,0x000bb240,
-/* 00F6 */ 0x0000b801,0x000b7140,0x00007888,0x00000000,
-/* 00F8 */ 0x0000073c,0x00001000,0x0007f188,0x000c0000,
-/* 00FA */ 0x00000000,0x00000000,0x00055288,0x000c555c,
-/* 00FC */ 0x0005528a,0x000c0000,0x0009fa88,0x000c5d00,
-/* 00FE */ 0x0000fa88,0x00000000,0x00000032,0x00001000,
-/* 0100 */ 0x0000073d,0x00001000,0x0007f188,0x000c0000,
-/* 0102 */ 0x00000000,0x00000000,0x0008c01c,0x00001003,
-/* 0104 */ 0x00002705,0x00001008,0x0008b201,0x000c1392,
-/* 0106 */ 0x0000ba01,0x00000000,
-/* TASKTREETHREAD */
-/* 0107 */ 0x00008731,0x00001400,0x0004c108,0x000fe0c4,
-/* 0109 */ 0x00057488,0x00000000,0x000a6388,0x00001001,
-/* 010B */ 0x0008b334,0x000bc141,0x0003020e,0x00000000,
-/* 010D */ 0x000986b0,0x00001008,0x00003625,0x000c5dfa,
-/* 010F */ 0x000a638a,0x00001001,0x0008020e,0x00001002,
-/* 0111 */ 0x0009a6b0,0x00001008,0x0007f301,0x00000000,
-/* 0113 */ 0x00000000,0x00000000,0x00002725,0x000a8c40,
-/* 0115 */ 0x000000ae,0x00000000,0x000e8630,0x00001008,
-/* 0117 */ 0x00000000,0x000c74e0,0x0007d182,0x0002d640,
-/* 0119 */ 0x000b8630,0x00001008,0x000799b8,0x0002d6c0,
-/* 011B */ 0x0000748a,0x000c3ec5,0x0007420a,0x000c0000,
-/* 011D */ 0x00062208,0x000c4117,0x000a0630,0x00001009,
-/* 011F */ 0x00000000,0x000c0000,0x0001022e,0x00000000,
-/* 0121 */ 0x0006a630,0x00001009,0x00000032,0x00001000,
-/* 0123 */ 0x000ca21c,0x00001003,0x00005a02,0x00000000,
-/* 0125 */ 0x0001a630,0x00001009,0x00000000,0x000c0000,
-/* 0127 */ 0x00000036,0x00001000,0x00000000,0x00000000,
-/* 0129 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 012B */ 0x00000000,0x00000000,0x0003a730,0x00001008,
-/* 012D */ 0x0007f801,0x000c0000,0x00000037,0x00001000,
-/* 012F */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0131 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0133 */ 0x0003a730,0x00001008,0x00000033,0x00001000,
-/* 0135 */ 0x0003a705,0x00001008,0x00007a01,0x000c0000,
-/* 0137 */ 0x000e6288,0x000d550a,0x0006428a,0x00000000,
-/* 0139 */ 0x00090730,0x0000100a,0x00000000,0x000c0000,
-/* 013B */ 0x00000000,0x00000000,
-/* TASKTREEHEADERCODE */
-/* 013C */ 0x0007aab0,0x00034880,0x000a8fb0,0x0000100b,
-/* 013E */ 0x00057488,0x00000000,0x00033b94,0x00081140,
-/* 0140 */ 0x000183ae,0x00000000,0x000a86b0,0x0000100b,
-/* 0142 */ 0x00022f05,0x000c3545,0x0000eb8a,0x00000000,
-/* 0144 */ 0x00042731,0x00001003,
-/* FGTASKTREEHEADERCODE */
-/* 0145 */ 0x0007aab0,0x00034880,0x00078fb0,0x0000100a,
-/* 0147 */ 0x00057488,0x00000000,0x00033b94,0x00081140,
-/* 0149 */ 0x000183ae,0x00000000,0x000b06b0,0x0000100b,
-/* 014B */ 0x00022f05,0x00000000,0x00007401,0x00091140,
-/* 014D */ 0x00048f05,0x000951c0,0x00042731,0x00001003,
-/* 014F */ 0x0000473d,0x00001000,0x000f19b0,0x000bbc47,
-/* 0151 */ 0x00080000,0x000bffc7,0x000fe19e,0x00001003,
-/* 0153 */ 0x00000000,0x00000000,0x0008e19c,0x00001003,
-/* 0155 */ 0x000083c1,0x00093040,0x00000f41,0x00097140,
-/* 0157 */ 0x0000a841,0x0009b240,0x0000a0c1,0x0009f040,
-/* 0159 */ 0x0001c641,0x00093540,0x0001cec1,0x0009b5c0,
-/* 015B */ 0x00000000,0x000fdc44,0x00055208,0x00000000,
-/* 015D */ 0x00010705,0x000a2880,0x0000a23a,0x00093a00,
-/* 015F */ 0x0003fc8a,0x000df6c5,0x0004ef0a,0x000c0000,
-/* 0161 */ 0x00012f05,0x00036880,0x00065308,0x000c2997,
-/* 0163 */ 0x000086b0,0x0000100b,0x0004410a,0x000d40c7,
-/* 0165 */ 0x00000000,0x00000000,0x00088730,0x00001004,
-/* 0167 */ 0x00056f0a,0x000ea105,0x00000000,0x00000000,
-/* NULLALGORITHM */
-/* 0169 */ 0x0000473d,0x00001000,0x000f19b0,0x000bbc47,
-/* 016B */ 0x00080000,0x000bffc7,0x0000273d,0x00001000,
-/* HFGEXECCHILD */
-/* 016D */ 0x00000000,0x000eba44,
-/* HFGEXECCHILD_98 */
-/* 016E */ 0x00048f05,0x0000f440,0x00007401,0x0000f7c0,
-/* HFGEXECCHILD_PUSH1IND */
-/* 0170 */ 0x00000734,0x00001000,0x00010705,0x000a6880,
-/* 0172 */ 0x00006a88,0x000c75c4,
-/* HFGEXECSIBLING */
-/* 0173 */ 0x00000000,0x000e5084,0x00000000,0x000eba44,
-/* HFGEXECSIBLING_298 */
-/* 0175 */ 0x00087401,0x000e4782,
-/* HFGEXECSIBLING_2IND1 */
-/* 0176 */ 0x00000734,0x00001000,0x00010705,0x000a6880,
-/* 0178 */ 0x00006a88,0x000c75c4,
-/* S16_CODECOUTPUTTASK */
-/* 0179 */ 0x0007c108,0x000c0000,0x0007e721,0x000bed40,
-/* 017B */ 0x00005f25,0x000badc0,0x0003ba97,0x000beb80,
-/* 017D */ 0x00065590,0x000b2e00,0x00033217,0x00003ec0,
-/* 017F */ 0x00065590,0x000b8e40,0x0003ed80,0x000491c0,
-/* 0181 */ 0x00073fb0,0x00074c80,0x000583a0,0x0000100c,
-/* 0183 */ 0x000ee388,0x00042970,0x00008301,0x00021ef2,
-/* 0185 */ 0x000b8f14,0x0000000f,0x000c4d8d,0x0000001b,
-/* 0187 */ 0x000d6dc2,0x000e06c6,0x000032ac,0x000c3916,
-/* 0189 */ 0x0004edc2,0x00074c80,0x00078898,0x00001000,
-/* 018B */ 0x00038894,0x00000032,0x000c4d8d,0x00092e1b,
-/* 018D */ 0x000d6dc2,0x000e06c6,0x0004edc2,0x000c1956,
-/* 018F */ 0x0000722c,0x00034a00,0x00041705,0x0009ed40,
-/* 0191 */ 0x00058730,0x00001400,0x000d7488,0x000c3a00,
-/* 0193 */ 0x00048f05,0x00000000
-};
-/* #CODE_END */
-
-static u32 cwc4630_parameter[] = {
-/* 0000 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0004 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0008 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 000C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0010 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0014 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0018 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 001C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0020 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0024 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0028 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 002C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0030 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0034 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0038 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 003C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0040 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0044 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0048 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 004C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0050 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0054 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0058 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 005C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0060 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0064 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0068 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 006C */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0070 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0074 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 0078 */ 0x00000000,0x00000000,0x00000000,0x00000000,
-/* 007C */ 0x00000000,0x00000000,0x00000000,0x00000000
-}; /* #PARAMETER_END */
-
-
-static struct dsp_segment_desc cwc4630_segments[] = {
-  { SEGTYPE_SP_PROGRAM, 0x00000000, 0x00000328, cwc4630_code },
-  { SEGTYPE_SP_PARAMETER, 0x00000000, 0x00000080, cwc4630_parameter },
-};
-
-static struct dsp_module_desc cwc4630_module = {
-  "cwc4630",
-  {
-    38,
-    cwc4630_symbols
-  },
-  2,
-  cwc4630_segments,
-};
-
-#endif /* __HEADER_cwc4630_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcasync.h b/sound/pci/cs46xx/imgs/cwcasync.h
deleted file mode 100644
index 70e63e13c2b3..000000000000
--- a/sound/pci/cs46xx/imgs/cwcasync.h
+++ /dev/null
@@ -1,176 +0,0 @@
-/* generated from cwcasync.osp DO NOT MODIFY */
-
-#ifndef __HEADER_cwcasync_H__
-#define __HEADER_cwcasync_H__
-
-static struct dsp_symbol_entry cwcasync_symbols[] = {
-  { 0x8000, "EXECCHILD",0x03 },
-  { 0x8001, "EXECCHILD_98",0x03 },
-  { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
-  { 0x8008, "EXECSIBLING",0x03 },
-  { 0x800a, "EXECSIBLING_298",0x03 },
-  { 0x800b, "EXECSIBLING_2IND1",0x03 },
-  { 0x8010, "TIMINGMASTER",0x03 },
-  { 0x804f, "S16_CODECINPUTTASK",0x03 },
-  { 0x805e, "PCMSERIALINPUTTASK",0x03 },
-  { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
-  { 0x809a, "S16_MIX",0x03 },
-  { 0x80bb, "S16_UPSRC",0x03 },
-  { 0x813b, "MIX3_EXP",0x03 },
-  { 0x8164, "DECIMATEBYPOW2",0x03 },
-  { 0x8197, "VARIDECIMATE",0x03 },
-  { 0x81f2, "_3DINPUTTASK",0x03 },
-  { 0x820a, "_3DPRLGCINPTASK",0x03 },
-  { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
-  { 0x8242, "_3DOUTPUTTASK",0x03 },
-  { 0x82c4, "HRTF_MORPH_TASK",0x03 },
-  { 0x82c6, "WAIT4DATA",0x03 },
-  { 0x82fa, "PROLOGIC",0x03 },
-  { 0x8496, "DECORRELATOR",0x03 },
-  { 0x84a4, "STEREO2MONO",0x03 },
-  { 0x0000, "OVERLAYBEGINADDRESS",0x00 },
-  { 0x0000, "SPIOWRITE",0x03 },
-  { 0x000d, "S16_ASYNCCODECINPUTTASK",0x03 },
-  { 0x0043, "SPDIFITASK",0x03 },
-  { 0x007b, "SPDIFOTASK",0x03 },
-  { 0x0097, "ASYNCHFGTXCODE",0x03 },
-  { 0x00be, "ASYNCHFGRXCODE",0x03 },
-  { 0x00db, "#CODE_END",0x00 },
-}; /* cwcasync symbols */
-
-static u32 cwcasync_code[] = {
-/* OVERLAYBEGINADDRESS */
-/* 0000 */ 0x00002731,0x00001400,0x00003725,0x000a8440,
-/* 0002 */ 0x000000ae,0x00000000,0x00060630,0x00001000,
-/* 0004 */ 0x00000000,0x000c7560,0x00075282,0x0002d640,
-/* 0006 */ 0x00021705,0x00000000,0x00072ab8,0x0002d6c0,
-/* 0008 */ 0x00020630,0x00001000,0x000c74c2,0x000d4b82,
-/* 000A */ 0x000475c2,0x00000000,0x0003430a,0x000c0000,
-/* 000C */ 0x00042730,0x00001400,
-/* S16_ASYNCCODECINPUTTASK */
-/* 000D */ 0x0006a108,0x000cf2c4,0x0004f4c0,0x00000000,
-/* 000F */ 0x000fa418,0x0000101f,0x0005d402,0x0001c500,
-/* 0011 */ 0x000f0630,0x00001000,0x00004418,0x00001380,
-/* 0013 */ 0x000e243d,0x000d394a,0x00049705,0x00000000,
-/* 0015 */ 0x0007d530,0x000b4240,0x000e00f2,0x00001000,
-/* 0017 */ 0x00009134,0x000ca20a,0x00004c90,0x00001000,
-/* 0019 */ 0x0005d705,0x00000000,0x00004f25,0x00098240,
-/* 001B */ 0x00004725,0x00000000,0x0000e48a,0x00000000,
-/* 001D */ 0x00027295,0x0009c2c0,0x0003df25,0x00000000,
-/* 001F */ 0x000e8030,0x00001001,0x0005f718,0x000ac600,
-/* 0021 */ 0x0007cf30,0x000c2a01,0x00082630,0x00001001,
-/* 0023 */ 0x000504a0,0x00001001,0x00029314,0x000bcb80,
-/* 0025 */ 0x0003cf25,0x000b0e00,0x0004f5c0,0x00000000,
-/* 0027 */ 0x00049118,0x000d888a,0x0007dd02,0x000c6efa,
-/* 0029 */ 0x00000000,0x00000000,0x0004f5c0,0x00069c80,
-/* 002B */ 0x0000d402,0x00000000,0x000e8630,0x00001001,
-/* 002D */ 0x00079130,0x00000000,0x00049118,0x00090e00,
-/* 002F */ 0x0006c10a,0x00000000,0x00000000,0x000c0000,
-/* 0031 */ 0x0007cf30,0x00030580,0x00005725,0x00000000,
-/* 0033 */ 0x000d84a0,0x00001001,0x00029314,0x000b4780,
-/* 0035 */ 0x0003cf25,0x000b8600,0x00000000,0x00000000,
-/* 0037 */ 0x00000000,0x000c0000,0x00000000,0x00042c80,
-/* 0039 */ 0x0001dec1,0x000e488c,0x00031114,0x00000000,
-/* 003B */ 0x0004f5c2,0x00000000,0x0003640a,0x00000000,
-/* 003D */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
-/* 003F */ 0x00007001,0x00000000,0x00000734,0x00001000,
-/* 0041 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
-/* SPDIFITASK */
-/* 0043 */ 0x0006a108,0x000cf2c4,0x0004f4c0,0x000d5384,
-/* 0045 */ 0x0007e48a,0x00000000,0x00067718,0x00001000,
-/* 0047 */ 0x0007a418,0x00001000,0x0007221a,0x00000000,
-/* 0049 */ 0x0005d402,0x00014500,0x000b8630,0x00001002,
-/* 004B */ 0x00004418,0x00001780,0x000e243d,0x000d394a,
-/* 004D */ 0x00049705,0x00000000,0x0007d530,0x000b4240,
-/* 004F */ 0x000ac0f2,0x00001002,0x00014414,0x00000000,
-/* 0051 */ 0x00004c90,0x00001000,0x0005d705,0x00000000,
-/* 0053 */ 0x00004f25,0x00098240,0x00004725,0x00000000,
-/* 0055 */ 0x0000e48a,0x00000000,0x00027295,0x0009c2c0,
-/* 0057 */ 0x0007df25,0x00000000,0x000ac030,0x00001003,
-/* 0059 */ 0x0005f718,0x000fe798,0x00029314,0x000bcb80,
-/* 005B */ 0x00000930,0x000b0e00,0x0004f5c0,0x000de204,
-/* 005D */ 0x000884a0,0x00001003,0x0007cf25,0x000e3560,
-/* 005F */ 0x00049118,0x00000000,0x00049118,0x000d888a,
-/* 0061 */ 0x0007dd02,0x000c6efa,0x0000c434,0x00030040,
-/* 0063 */ 0x000fda82,0x000c2312,0x000fdc0e,0x00001001,
-/* 0065 */ 0x00083402,0x000c2b92,0x000706b0,0x00001003,
-/* 0067 */ 0x00075a82,0x00000000,0x0000d625,0x000b0940,
-/* 0069 */ 0x0000840e,0x00001002,0x0000aabc,0x000c511e,
-/* 006B */ 0x00078730,0x00001003,0x0000aaf4,0x000e910a,
-/* 006D */ 0x0004628a,0x00000000,0x00006aca,0x00000000,
-/* 006F */ 0x00000930,0x00000000,0x0004f5c0,0x00069c80,
-/* 0071 */ 0x00046ac0,0x00000000,0x0003c40a,0x000fc898,
-/* 0073 */ 0x00049118,0x00090e00,0x0006c10a,0x00000000,
-/* 0075 */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
-/* 0077 */ 0x00007001,0x00000000,0x00000734,0x00001000,
-/* 0079 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
-/* SPDIFOTASK */
-/* 007B */ 0x0006a108,0x000c0000,0x0004f4c0,0x000c3245,
-/* 007D */ 0x0000a418,0x00001000,0x0003a20a,0x00000000,
-/* 007F */ 0x00004418,0x00001380,0x000e243d,0x000d394a,
-/* 0081 */ 0x000c9705,0x000def92,0x0008c030,0x00001004,
-/* 0083 */ 0x0005f718,0x000fe798,0x00000000,0x000c0000,
-/* 0085 */ 0x00005725,0x00000000,0x000704a0,0x00001004,
-/* 0087 */ 0x00029314,0x000b4780,0x0003cf25,0x000b8600,
-/* 0089 */ 0x00000000,0x00000000,0x00000000,0x000c0000,
-/* 008B */ 0x00000000,0x00042c80,0x0001dec1,0x000e488c,
-/* 008D */ 0x00031114,0x00000000,0x0004f5c2,0x00000000,
-/* 008F */ 0x0004a918,0x00098600,0x0006c28a,0x00000000,
-/* 0091 */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
-/* 0093 */ 0x00007001,0x00000000,0x00000734,0x00001000,
-/* 0095 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
-/* ASYNCHFGTXCODE */
-/* 0097 */ 0x0002a880,0x000b4e40,0x00042214,0x000e5548,
-/* 0099 */ 0x000542bf,0x00000000,0x00000000,0x000481c0,
-/* 009B */ 0x00000000,0x00000000,0x00000000,0x00000030,
-/* 009D */ 0x0000072d,0x000fbf8a,0x00077f94,0x000ea7df,
-/* 009F */ 0x0002ac95,0x000d3145,0x00002731,0x00001400,
-/* 00A1 */ 0x00006288,0x000c71c4,0x00014108,0x000e6044,
-/* 00A3 */ 0x00035408,0x00000000,0x00025418,0x000a0ec0,
-/* 00A5 */ 0x0001443d,0x000ca21e,0x00046595,0x000d730c,
-/* 00A7 */ 0x0006538e,0x00000000,0x00064630,0x00001005,
-/* 00A9 */ 0x000e7b0e,0x000df782,0x000746b0,0x00001005,
-/* 00AB */ 0x00036f05,0x000c0000,0x00043695,0x000d598c,
-/* 00AD */ 0x0005331a,0x000f2185,0x00000000,0x00000000,
-/* 00AF */ 0x000007ae,0x000bdb00,0x00040630,0x00001400,
-/* 00B1 */ 0x0005e708,0x000c0000,0x0007ef30,0x000b1c00,
-/* 00B3 */ 0x000d86a0,0x00001005,0x00066408,0x000c0000,
-/* 00B5 */ 0x00000000,0x00000000,0x00021843,0x00000000,
-/* 00B7 */ 0x00000cac,0x00062c00,0x00001dac,0x00063400,
-/* 00B9 */ 0x00002cac,0x0006cc80,0x000db943,0x000e5ca1,
-/* 00BB */ 0x00000000,0x00000000,0x0006680a,0x000f3205,
-/* 00BD */ 0x00042730,0x00001400,
-/* ASYNCHFGRXCODE */
-/* 00BE */ 0x00014108,0x000f2204,0x00025418,0x000a2ec0,
-/* 00C0 */ 0x00015dbd,0x00038100,0x00015dbc,0x00000000,
-/* 00C2 */ 0x0005e415,0x00034880,0x0001258a,0x000d730c,
-/* 00C4 */ 0x0006538e,0x000baa40,0x00060630,0x00001006,
-/* 00C6 */ 0x00067b0e,0x000ac380,0x0003ef05,0x00000000,
-/* 00C8 */ 0x0000f734,0x0001c300,0x000586b0,0x00001400,
-/* 00CA */ 0x000b6f05,0x000c3a00,0x00048f05,0x00000000,
-/* 00CC */ 0x0005b695,0x0008c380,0x0002058e,0x00000000,
-/* 00CE */ 0x000500b0,0x00001400,0x0002b318,0x000e998d,
-/* 00D0 */ 0x0006430a,0x00000000,0x00000000,0x000ef384,
-/* 00D2 */ 0x00004725,0x000c0000,0x00000000,0x000f3204,
-/* 00D4 */ 0x00004f25,0x000c0000,0x00080000,0x000e5ca1,
-/* 00D6 */ 0x000cb943,0x000e5ca1,0x0004b943,0x00000000,
-/* 00D8 */ 0x00040730,0x00001400,0x000cb943,0x000e5ca1,
-/* 00DA */ 0x0004b943,0x00000000
-};
-/* #CODE_END */
-
-static struct dsp_segment_desc cwcasync_segments[] = {
-  { SEGTYPE_SP_PROGRAM, 0x00000000, 0x000001b6, cwcasync_code },
-};
-
-static struct dsp_module_desc cwcasync_module = {
-  "cwcasync",
-  {
-    32,
-    cwcasync_symbols
-  },
-  1,
-  cwcasync_segments,
-};
-
-#endif /* __HEADER_cwcasync_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcbinhack.h b/sound/pci/cs46xx/imgs/cwcbinhack.h
deleted file mode 100644
index f4d93689cd49..000000000000
--- a/sound/pci/cs46xx/imgs/cwcbinhack.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/* generated by Benny 
-   MODIFY ON YOUR OWN RISK */
-
-#ifndef __HEADER_cwcbinhack_H__
-#define __HEADER_cwcbinhack_H__
-
-static struct dsp_symbol_entry cwcbinhack_symbols[] = {
-  { 0x02c8, "OVERLAYBEGINADDRESS",0x00 },
-  { 0x02c8, "MAGICSNOOPTASK",0x03 },
-  { 0x0308, "#CODE_END",0x00 },
-}; /* cwcbinhack symbols */
-
-static u32 cwcbinhack_code[] = {
-  /* 0x02c8 */
-  0x0007bfb0,0x000bc240,0x00000c2e,0x000c6084, /* 1 */
-  0x000b8630,0x00001016,0x00006408,0x000efb84, /* 2 */
-  0x00016008,0x00000000,0x0001c088,0x000c0000, /* 3 */
-  0x000fc908,0x000e3392,0x0005f488,0x000efb84, /* 4 */
-  0x0001d402,0x000b2e00,0x0003d418,0x00001000, /* 5 */
-  0x0008d574,0x000c4293,0x00065625,0x000ea30e, /* 6 */
-  0x00096c01,0x000c6f92,0x0001a58a,0x000c6085, /* 7 */
-  0x00002f43,0x00000000,0x000e03a0,0x00001016, /* 8 */
-  0x0005e608,0x000c0000,0x00000000,0x00000000, /* 9 */
-  0x000ca108,0x000dcca1,0x00003bac,0x000c3205, /* 10 */
-  0x00073843,0x00000000,0x00010730,0x00001017, /* 11 */
-  0x0001600a,0x000c0000,0x00057488,0x00000000, /* 12 */
-  0x00000000,0x000e5084,0x00000000,0x000eba44, /* 13 */
-  0x00087401,0x000e4782,0x00000734,0x00001000, /* 14 */
-  0x00010705,0x000a6880,0x00006a88,0x000c75c4, /* 15 */
-  0x00000000,0x00000000,0x00000000,0x00000000, /* 16 */
-};
-/* #CODE_END */
-
-static struct dsp_segment_desc cwcbinhack_segments[] = {
-  { SEGTYPE_SP_PROGRAM, 0x00000000, 64, cwcbinhack_code },
-};
-
-static struct dsp_module_desc cwcbinhack_module = {
-  "cwcbinhack",
-  {
-    3,
-    cwcbinhack_symbols
-  },
-  1,
-  cwcbinhack_segments,
-};
-
-#endif /* __HEADER_cwcbinhack_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
deleted file mode 100644
index a65e1193c89a..000000000000
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ /dev/null
@@ -1,170 +0,0 @@
-// 
-//  Copyright(c) by Benny Sjostrand (benny@hostmobility.com)
-//
-//  This program is free software; you can redistribute it and/or modify
-//  it under the terms of the GNU General Public License as published by
-//  the Free Software Foundation; either version 2 of the License, or
-//  (at your option) any later version.
-//
-//  This program is distributed in the hope that it will be useful,
-//  but WITHOUT ANY WARRANTY; without even the implied warranty of
-//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-//  GNU General Public License for more details.
-//
-//  You should have received a copy of the GNU General Public License
-//  along with this program; if not, write to the Free Software
-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
-//
-
-
-//
-// This code runs inside the DSP (cs4610, cs4612, cs4624, or cs4630),
-// to compile it you need a tool named SPASM 3.0 and DSP code owned by 
-// Cirrus Logic(R). The SPASM program will generate a object file (cwcdma.osp),
-// the "ospparser"  tool will genereate the cwcdma.h file it's included from
-// the cs46xx_lib.c file.
-//
-//
-// The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader
-// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
-// SRC (source rate converters) task always alters the samples in somehow,
-// however it's from 48khz -> 48khz.
-// The alterations are not audible, but AC3 wont work. 
-//
-//        ...
-//         |
-// +---------------+
-// | AsynchFGTxSCB |
-// +---------------+
-//        |
-//    subListPtr
-//        |
-// +--------------+
-// |   DMAReader  |
-// +--------------+
-//        |
-//    subListPtr
-//        |
-// +-------------+
-// | PCMReader   |
-// +-------------+
-// (DMA from host)
-//
-
-struct dmaSCB
-  {
-    long  dma_reserved1[3];
-
-    short dma_reserved2:dma_outBufPtr;
-
-    short dma_unused1:dma_unused2;
-
-    long  dma_reserved3[4];
-
-    short dma_subListPtr:dma_nextSCB;
-    short dma_SPBptr:dma_entryPoint;
-
-    long  dma_strmRsConfig;
-    long  dma_strmBufPtr;
-
-    long  dma_reserved4;
-
-    VolumeControl s2m_volume;
-  };
-
-#export DMAReader
-void DMAReader()
-{
-  execChild();
-  r2 = r0->dma_subListPtr;
-  r1 = r0->nextSCB;
-	
-  rsConfig01 = r2->strmRsConfig;
-  // Load rsConfig for input buffer
-
-  rsDMA01 = r2->basicReq.daw,       ,                   tb = Z(0 - rf);
-  // Load rsDMA in case input buffer is a DMA buffer    Test to see if there is any data to transfer
-
-  if (tb) goto execSibling_2ind1 after {
-      r5 = rf + (-1);
-      r6 = r1->dma_entryPoint;           // r6 = entry point of sibling task
-      r1 = r1->dma_SPBptr,               // r1 = pointer to sibling task's SPB
-          ,   ind = r6;                  // Load entry point of sibling task
-  }
-
-  rsConfig23 = r0->dma_strmRsConfig;
-  // Load rsConfig for output buffer (never a DMA buffer)
-
-  r4 = r0->dma_outBufPtr;
-
-  rsa0 = r2->strmBufPtr;
-  // rsa0 = input buffer pointer                        
-
-  for (i = r5; i >= 0; --i)
-    after {
-      rsa2 = r4;
-      // rsa2 = output buffer pointer
-
-      nop;
-      nop;
-    }
-  //*****************************
-  // TODO: cycles to this point *
-  //*****************************
-    {
-      acc0 =  (rsd0 = *rsa0++1);
-      // get sample
-
-      nop;  // Those "nop"'s are really uggly, but there's
-      nop;  // something with DSP's pipelines which I don't
-      nop;  // understand, resulting this code to fail without
-            // having those "nop"'s (Benny)
-
-      rsa0?reqDMA = r2;
-      // Trigger DMA transfer on input stream, 
-      // if needed to replenish input buffer
-
-      nop;
-      // Yet another magic "nop" to make stuff work
-
-      ,,r98 = acc0 $+>> 0;
-      // store sample in ALU
-
-      nop;
-      // latency on load register.
-      // (this one is understandable)
-
-      *rsa2++1 = r98;
-      // store sample in output buffer
-
-      nop; // The same story
-      nop; // as above again ...
-      nop;
-    }
-  // TODO: cycles per loop iteration
-
-  r2->strmBufPtr = rsa0,,   ;
-  // Update the modified buffer pointers
-
-  r4 = rsa2;
-  // Load output pointer position into r4
-
-  r2 = r0->nextSCB;
-  // Sibling task
-
-  goto execSibling_2ind1 // takes 6 cycles
-    after {
-      r98 = r2->thisSPB:entryPoint;
-      // Load child routine entry and data address 
-
-      r1 = r9;
-      // r9 is r2->thisSPB
-
-      r0->dma_outBufPtr = r4,,
-      // Store updated output buffer pointer
-
-      ind = r8;
-      // r8 is r2->entryPoint
-    }
-}
diff --git a/sound/pci/cs46xx/imgs/cwcdma.h b/sound/pci/cs46xx/imgs/cwcdma.h
deleted file mode 100644
index 7ff0d4587161..000000000000
--- a/sound/pci/cs46xx/imgs/cwcdma.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/* generated from cwcdma.osp DO NOT MODIFY */
-
-#ifndef __HEADER_cwcdma_H__
-#define __HEADER_cwcdma_H__
-
-static struct dsp_symbol_entry cwcdma_symbols[] = {
-  { 0x8000, "EXECCHILD",0x03 },
-  { 0x8001, "EXECCHILD_98",0x03 },
-  { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
-  { 0x8008, "EXECSIBLING",0x03 },
-  { 0x800a, "EXECSIBLING_298",0x03 },
-  { 0x800b, "EXECSIBLING_2IND1",0x03 },
-  { 0x8010, "TIMINGMASTER",0x03 },
-  { 0x804f, "S16_CODECINPUTTASK",0x03 },
-  { 0x805e, "PCMSERIALINPUTTASK",0x03 },
-  { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
-  { 0x809a, "S16_MIX",0x03 },
-  { 0x80bb, "S16_UPSRC",0x03 },
-  { 0x813b, "MIX3_EXP",0x03 },
-  { 0x8164, "DECIMATEBYPOW2",0x03 },
-  { 0x8197, "VARIDECIMATE",0x03 },
-  { 0x81f2, "_3DINPUTTASK",0x03 },
-  { 0x820a, "_3DPRLGCINPTASK",0x03 },
-  { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
-  { 0x8242, "_3DOUTPUTTASK",0x03 },
-  { 0x82c4, "HRTF_MORPH_TASK",0x03 },
-  { 0x82c6, "WAIT4DATA",0x03 },
-  { 0x82fa, "PROLOGIC",0x03 },
-  { 0x8496, "DECORRELATOR",0x03 },
-  { 0x84a4, "STEREO2MONO",0x03 },
-  { 0x0000, "OVERLAYBEGINADDRESS",0x00 },
-  { 0x0000, "DMAREADER",0x03 },
-  { 0x0018, "#CODE_END",0x00 },
-}; /* cwcdma symbols */
-
-static u32 cwcdma_code[] = {
-/* OVERLAYBEGINADDRESS */
-/* 0000 */ 0x00002731,0x00001400,0x0004c108,0x000e5044,
-/* 0002 */ 0x0005f608,0x00000000,0x000007ae,0x000be300,
-/* 0004 */ 0x00058630,0x00001400,0x0007afb0,0x000e9584,
-/* 0006 */ 0x00007301,0x000a9840,0x0005e708,0x000cd104,
-/* 0008 */ 0x00067008,0x00000000,0x000902a0,0x00001000,
-/* 000A */ 0x00012a01,0x000c0000,0x00000000,0x00000000,
-/* 000C */ 0x00021843,0x000c0000,0x00000000,0x000c0000,
-/* 000E */ 0x0000e101,0x000c0000,0x00000cac,0x00000000,
-/* 0010 */ 0x00080000,0x000e5ca1,0x00000000,0x000c0000,
-/* 0012 */ 0x00000000,0x00000000,0x00000000,0x00092c00,
-/* 0014 */ 0x000122c1,0x000e5084,0x00058730,0x00001400,
-/* 0016 */ 0x000d7488,0x000e4782,0x00007401,0x0001c100
-};
-
-/* #CODE_END */
-
-static struct dsp_segment_desc cwcdma_segments[] = {
-  { SEGTYPE_SP_PROGRAM, 0x00000000, 0x00000030, cwcdma_code },
-};
-
-static struct dsp_module_desc cwcdma_module = {
-  "cwcdma",
-  {
-    27,
-    cwcdma_symbols
-  },
-  1,
-  cwcdma_segments,
-};
-
-#endif /* __HEADER_cwcdma_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcsnoop.h b/sound/pci/cs46xx/imgs/cwcsnoop.h
deleted file mode 100644
index 6929d0a5a3f3..000000000000
--- a/sound/pci/cs46xx/imgs/cwcsnoop.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/* generated from cwcsnoop.osp DO NOT MODIFY */
-
-#ifndef __HEADER_cwcsnoop_H__
-#define __HEADER_cwcsnoop_H__
-
-static struct dsp_symbol_entry cwcsnoop_symbols[] = {
-  { 0x0500, "OVERLAYBEGINADDRESS",0x00 },
-  { 0x0500, "OUTPUTSNOOP",0x03 },
-  { 0x051f, "#CODE_END",0x00 },
-}; /* cwcsnoop symbols */
-
-static u32 cwcsnoop_code[] = {
-/* 0000 */ 0x0007bfb0,0x000b4e40,0x0007c088,0x000c0617,
-/* 0002 */ 0x00049705,0x00000000,0x00080630,0x00001028,
-/* 0004 */ 0x00076408,0x000efb84,0x00066008,0x00000000,
-/* 0006 */ 0x0007c908,0x000c0000,0x00046725,0x000efa44,
-/* 0008 */ 0x0005f708,0x00000000,0x0001d402,0x000b2e00,
-/* 000A */ 0x0003d418,0x00001000,0x0008d574,0x000c4293,
-/* 000C */ 0x00065625,0x000ea30e,0x00096c01,0x000c6f92,
-/* 000E */ 0x0006a58a,0x000f6085,0x00002f43,0x00000000,
-/* 0010 */ 0x000a83a0,0x00001028,0x0005e608,0x000c0000,
-/* 0012 */ 0x00000000,0x00000000,0x000ca108,0x000dcca1,
-/* 0014 */ 0x00003bac,0x000fb205,0x00073843,0x00000000,
-/* 0016 */ 0x000d8730,0x00001028,0x0006600a,0x000c0000,
-/* 0018 */ 0x00057488,0x00000000,0x00000000,0x000e5084,
-/* 001A */ 0x00000000,0x000eba44,0x00087401,0x000e4782,
-/* 001C */ 0x00000734,0x00001000,0x00010705,0x000a6880,
-/* 001E */ 0x00006a88,0x000c75c4
-};
-/* #CODE_END */
-
-static struct dsp_segment_desc cwcsnoop_segments[] = {
-  { SEGTYPE_SP_PROGRAM, 0x00000000, 0x0000003e, cwcsnoop_code },
-};
-
-static struct dsp_module_desc cwcsnoop_module = {
-  "cwcsnoop",
-  {
-    3,
-    cwcsnoop_symbols
-  },
-  1,
-  cwcsnoop_segments,
-};
-
-#endif /* __HEADER_cwcsnoop_H__ */
-- 
cgit v1.2.2


From e64001e8efc107992fd835770f6383d0dc731594 Mon Sep 17 00:00:00 2001
From: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Date: Wed, 20 Nov 2013 13:17:07 +0000
Subject: ASoC: wm5110: Add extra AIF2 channels

Signed-off-by: D.J. Barrow <dbarrow@wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/arizona.c |  8 ++++++++
 sound/soc/codecs/arizona.h |  2 +-
 sound/soc/codecs/wm5110.c  | 48 ++++++++++++++++++++++++++++++++++++++++++++--
 3 files changed, 55 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 6f05b17d1965..6977bf9f19a8 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -292,6 +292,10 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"AIF1RX8",
 	"AIF2RX1",
 	"AIF2RX2",
+	"AIF2RX3",
+	"AIF2RX4",
+	"AIF2RX5",
+	"AIF2RX6",
 	"AIF3RX1",
 	"AIF3RX2",
 	"SLIMRX1",
@@ -395,6 +399,10 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
 	0x27,
 	0x28,  /* AIF2RX1 */
 	0x29,
+	0x2a,
+	0x2b,
+	0x2c,
+	0x2d,
 	0x30,  /* AIF3RX1 */
 	0x31,
 	0x38,  /* SLIMRX1 */
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 9e81b6392692..1f96672c7c1e 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -81,7 +81,7 @@ struct arizona_priv {
 	unsigned int spk_ena_pending:1;
 };
 
-#define ARIZONA_NUM_MIXER_INPUTS 99
+#define ARIZONA_NUM_MIXER_INPUTS 103
 
 extern const unsigned int arizona_mixer_tlv[];
 extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index bbd64384ca1c..181de7df2c87 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -302,6 +302,10 @@ ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
@@ -361,6 +365,10 @@ ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
@@ -561,11 +569,27 @@ SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
 		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
 		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0),
 
 SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
 		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
 SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
 		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0),
 
 SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0,
 		    ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE,
@@ -703,6 +727,10 @@ ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
 
 ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
 ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"),
+ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"),
+ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"),
+ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"),
 
 ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
 ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
@@ -764,6 +792,10 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"),
 	{ name, "AIF1RX8", "AIF1RX8" }, \
 	{ name, "AIF2RX1", "AIF2RX1" }, \
 	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF2RX3", "AIF2RX3" }, \
+	{ name, "AIF2RX4", "AIF2RX4" }, \
+	{ name, "AIF2RX5", "AIF2RX5" }, \
+	{ name, "AIF2RX6", "AIF2RX6" }, \
 	{ name, "AIF3RX1", "AIF3RX1" }, \
 	{ name, "AIF3RX2", "AIF3RX2" }, \
 	{ name, "SLIMRX1", "SLIMRX1" }, \
@@ -861,9 +893,17 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 
 	{ "AIF2 Capture", NULL, "AIF2TX1" },
 	{ "AIF2 Capture", NULL, "AIF2TX2" },
+	{ "AIF2 Capture", NULL, "AIF2TX3" },
+	{ "AIF2 Capture", NULL, "AIF2TX4" },
+	{ "AIF2 Capture", NULL, "AIF2TX5" },
+	{ "AIF2 Capture", NULL, "AIF2TX6" },
 
 	{ "AIF2RX1", NULL, "AIF2 Playback" },
 	{ "AIF2RX2", NULL, "AIF2 Playback" },
+	{ "AIF2RX3", NULL, "AIF2 Playback" },
+	{ "AIF2RX4", NULL, "AIF2 Playback" },
+	{ "AIF2RX5", NULL, "AIF2 Playback" },
+	{ "AIF2RX6", NULL, "AIF2 Playback" },
 
 	{ "AIF3 Capture", NULL, "AIF3TX1" },
 	{ "AIF3 Capture", NULL, "AIF3TX2" },
@@ -947,6 +987,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 
 	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
 	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+	ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"),
+	ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"),
+	ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"),
+	ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"),
 
 	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
 	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
@@ -1067,14 +1111,14 @@ static struct snd_soc_dai_driver wm5110_dai[] = {
 		.playback = {
 			.stream_name = "AIF2 Playback",
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 6,
 			.rates = WM5110_RATES,
 			.formats = WM5110_FORMATS,
 		},
 		.capture = {
 			 .stream_name = "AIF2 Capture",
 			 .channels_min = 1,
-			 .channels_max = 2,
+			 .channels_max = 6,
 			 .rates = WM5110_RATES,
 			 .formats = WM5110_FORMATS,
 		 },
-- 
cgit v1.2.2


From 254dc326dbfd23c2678fafad1b84fc0e42ac4374 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Tue, 19 Nov 2013 16:04:03 +0000
Subject: ASoC: wm5110: Expose input high pass filter controls

Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
---
 sound/soc/codecs/arizona.c | 10 ++++++++++
 sound/soc/codecs/arizona.h |  1 +
 sound/soc/codecs/wm5110.c  | 19 +++++++++++++++++++
 3 files changed, 30 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 657808ba1418..708326265a37 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -560,6 +560,16 @@ const struct soc_enum arizona_ng_hold =
 			4, arizona_ng_hold_text);
 EXPORT_SYMBOL_GPL(arizona_ng_hold);
 
+static const char * const arizona_in_hpf_cut_text[] = {
+	"2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz"
+};
+
+const struct soc_enum arizona_in_hpf_cut_enum =
+	SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT,
+			ARRAY_SIZE(arizona_in_hpf_cut_text),
+			arizona_in_hpf_cut_text);
+EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum);
+
 static const char * const arizona_in_dmic_osr_text[] = {
 	"1.536MHz", "3.072MHz", "6.144MHz",
 };
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 9e81b6392692..f8e63865a1c5 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -199,6 +199,7 @@ extern const struct soc_enum arizona_lhpf3_mode;
 extern const struct soc_enum arizona_lhpf4_mode;
 
 extern const struct soc_enum arizona_ng_hold;
+extern const struct soc_enum arizona_in_hpf_cut_enum;
 extern const struct soc_enum arizona_in_dmic_osr[];
 
 extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index bbd64384ca1c..ea18e88e0a07 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -76,6 +76,25 @@ SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
 SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
 		     ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
 
+SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
+
+SOC_SINGLE("IN1L HPF Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN1R HPF Switch", ARIZONA_IN1R_CONTROL,
+	   ARIZONA_IN1R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2L HPF Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN2R HPF Switch", ARIZONA_IN2R_CONTROL,
+	   ARIZONA_IN2R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN3L HPF Switch", ARIZONA_IN3L_CONTROL,
+	   ARIZONA_IN3L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN3R HPF Switch", ARIZONA_IN3R_CONTROL,
+	   ARIZONA_IN3R_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN4L HPF Switch", ARIZONA_IN4L_CONTROL,
+	   ARIZONA_IN4L_HPF_SHIFT, 1, 0),
+SOC_SINGLE("IN4R HPF Switch", ARIZONA_IN4R_CONTROL,
+	   ARIZONA_IN4R_HPF_SHIFT, 1, 0),
+
 SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
 	       ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv),
 SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R,
-- 
cgit v1.2.2


From 3635bf09a89cf92b80ac44198c5c8f0989624ea6 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Wed, 13 Nov 2013 18:56:24 +0800
Subject: ASoC: soc-pcm: add symmetry for channels and sample bits

Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.

In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.

Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.

Thus, this patch applys symmetry for channels and sample bits.

Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 130 ++++++++++++++++++++++++++++++++++++++++++----------
 1 file changed, 107 insertions(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..ed1e077114a2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -84,30 +84,97 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	int ret;
 
-	if (!soc_dai->driver->symmetric_rates &&
-	    !rtd->dai_link->symmetric_rates)
-		return 0;
+	if (soc_dai->rate && (soc_dai->driver->symmetric_rates ||
+				rtd->dai_link->symmetric_rates)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n",
+				soc_dai->rate);
+
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_RATE,
+						soc_dai->rate, soc_dai->rate);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply rate constraint: %d\n",
+				ret);
+			return ret;
+		}
+	}
 
-	/* This can happen if multiple streams are starting simultaneously -
-	 * the second can need to get its constraints before the first has
-	 * picked a rate.  Complain and allow the application to carry on.
-	 */
-	if (!soc_dai->rate) {
-		dev_warn(soc_dai->dev,
-			 "ASoC: Not enforcing symmetric_rates due to race\n");
-		return 0;
+	if (soc_dai->channels && (soc_dai->driver->symmetric_channels ||
+				rtd->dai_link->symmetric_channels)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d channel(s)\n",
+				soc_dai->channels);
+
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_CHANNELS,
+						soc_dai->channels,
+						soc_dai->channels);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply channel symmetry constraint: %d\n",
+				ret);
+			return ret;
+		}
 	}
 
-	dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %dHz rate\n", soc_dai->rate);
+	if (soc_dai->sample_bits && (soc_dai->driver->symmetric_samplebits ||
+				rtd->dai_link->symmetric_samplebits)) {
+		dev_dbg(soc_dai->dev, "ASoC: Symmetry forces %d sample bits\n",
+				soc_dai->sample_bits);
 
-	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-					   SNDRV_PCM_HW_PARAM_RATE,
-					   soc_dai->rate, soc_dai->rate);
-	if (ret < 0) {
-		dev_err(soc_dai->dev,
-			"ASoC: Unable to apply rate symmetry constraint: %d\n",
-			ret);
-		return ret;
+		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+						soc_dai->sample_bits,
+						soc_dai->sample_bits);
+		if (ret < 0) {
+			dev_err(soc_dai->dev,
+				"ASoC: Unable to apply sample bits symmetry constraint: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
+static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int rate, channels, sample_bits, symmetry;
+
+	rate = params_rate(params);
+	channels = params_channels(params);
+	sample_bits = snd_pcm_format_physical_width(params_format(params));
+
+	/* reject unmatched parameters when applying symmetry */
+	symmetry = cpu_dai->driver->symmetric_rates ||
+		codec_dai->driver->symmetric_rates ||
+		rtd->dai_link->symmetric_rates;
+	if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) {
+		dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n",
+				cpu_dai->rate, rate);
+		return -EINVAL;
+	}
+
+	symmetry = cpu_dai->driver->symmetric_channels ||
+		codec_dai->driver->symmetric_channels ||
+		rtd->dai_link->symmetric_channels;
+	if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) {
+		dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n",
+				cpu_dai->channels, channels);
+		return -EINVAL;
+	}
+
+	symmetry = cpu_dai->driver->symmetric_samplebits ||
+		codec_dai->driver->symmetric_samplebits ||
+		rtd->dai_link->symmetric_samplebits;
+	if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) {
+		dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n",
+				cpu_dai->sample_bits, sample_bits);
+		return -EINVAL;
 	}
 
 	return 0;
@@ -384,11 +451,17 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	codec->active--;
 
 	/* clear the corresponding DAIs rate when inactive */
-	if (!cpu_dai->active)
+	if (!cpu_dai->active) {
 		cpu_dai->rate = 0;
+		cpu_dai->channels = 0;
+		cpu_dai->sample_bits = 0;
+	}
 
-	if (!codec_dai->active)
+	if (!codec_dai->active) {
 		codec_dai->rate = 0;
+		codec_dai->channels = 0;
+		codec_dai->sample_bits = 0;
+	}
 
 	/* Muting the DAC suppresses artifacts caused during digital
 	 * shutdown, for example from stopping clocks.
@@ -525,6 +598,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
+	ret = soc_pcm_params_symmetry(substream, params);
+	if (ret)
+		goto out;
+
 	if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
 		ret = rtd->dai_link->ops->hw_params(substream, params);
 		if (ret < 0) {
@@ -561,9 +638,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	/* store the rate for each DAIs */
+	/* store the parameters for each DAIs */
 	cpu_dai->rate = params_rate(params);
+	cpu_dai->channels = params_channels(params);
+	cpu_dai->sample_bits =
+		snd_pcm_format_physical_width(params_format(params));
+
 	codec_dai->rate = params_rate(params);
+	codec_dai->channels = params_channels(params);
+	codec_dai->sample_bits =
+		snd_pcm_format_physical_width(params_format(params));
 
 out:
 	mutex_unlock(&rtd->pcm_mutex);
-- 
cgit v1.2.2


From d3383420c969c25deffd33270ebe321e8401191a Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Wed, 20 Nov 2013 18:37:09 +0800
Subject: ASoC: soc-pcm: move DAIs parameters cleaning into hw_free()

We're now applying soc_hw_params_symmetry() to reject unmatched parameters
while we clear parameters in soc_pcm_close(). So here's a use case might be
broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav

In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()
->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we
only clear parameters in soc_pcm_close(). The parameters would be remained
in the system even if the playback of 44100.wav is finished.

Thus, this patch is trying to move parameters cleaning into hw_free() so that
the system can continue to serve this kind of use case.

Also, since we set them in hw_params(), it should be better to clear them in
hw_free() for symmetry.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ed1e077114a2..170ff9695753 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -450,19 +450,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	codec_dai->active--;
 	codec->active--;
 
-	/* clear the corresponding DAIs rate when inactive */
-	if (!cpu_dai->active) {
-		cpu_dai->rate = 0;
-		cpu_dai->channels = 0;
-		cpu_dai->sample_bits = 0;
-	}
-
-	if (!codec_dai->active) {
-		codec_dai->rate = 0;
-		codec_dai->channels = 0;
-		codec_dai->sample_bits = 0;
-	}
-
 	/* Muting the DAC suppresses artifacts caused during digital
 	 * shutdown, for example from stopping clocks.
 	 */
@@ -682,6 +669,19 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
+	/* clear the corresponding DAIs parameters when going to be inactive */
+	if (cpu_dai->active == 1) {
+		cpu_dai->rate = 0;
+		cpu_dai->channels = 0;
+		cpu_dai->sample_bits = 0;
+	}
+
+	if (codec_dai->active == 1) {
+		codec_dai->rate = 0;
+		codec_dai->channels = 0;
+		codec_dai->sample_bits = 0;
+	}
+
 	/* apply codec digital mute */
 	if (!codec->active)
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
-- 
cgit v1.2.2


From 74c375cb85d7374734e6e53af41c724d9a937f8e Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:43 -0200
Subject: ASoC: ad193x: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ad193x.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index aea7e52cf714..12c27eb363dd 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -413,7 +413,7 @@ static struct spi_driver ad193x_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 
 static const struct regmap_config ad193x_i2c_regmap_config = {
 	.val_bits = 8,
@@ -470,7 +470,7 @@ static int __init ad193x_modinit(void)
 {
 	int ret;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret =  i2c_add_driver(&ad193x_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n",
@@ -495,7 +495,7 @@ static void __exit ad193x_modexit(void)
 	spi_unregister_driver(&ad193x_spi_driver);
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&ad193x_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From 04c3a852f51ff40f32a29e14078432038b5bcdbc Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:44 -0200
Subject: ASoC: adav80x: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/adav80x.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 14a7c169d004..f7bf45552749 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -939,7 +939,7 @@ static struct spi_driver adav80x_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static const struct regmap_config adav80x_i2c_regmap_config = {
 	.val_bits = 8,
 	.pad_bits = 1,
@@ -985,7 +985,7 @@ static int __init adav80x_init(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&adav80x_i2c_driver);
 	if (ret)
 		return ret;
@@ -1001,7 +1001,7 @@ module_init(adav80x_init);
 
 static void __exit adav80x_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&adav80x_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From b34d7cf355116f5107fad8d42fb91d067a3b15bc Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:46 -0200
Subject: ASoC: ak4642: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ak4642.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 090d499bb7eb..2f861c9b1d69 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -511,7 +511,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
 	.num_dapm_routes	= ARRAY_SIZE(ak4642_intercon),
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static struct of_device_id ak4642_of_match[];
 static int ak4642_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
@@ -576,7 +576,7 @@ static struct i2c_driver ak4642_i2c_driver = {
 static int __init ak4642_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&ak4642_i2c_driver);
 #endif
 	return ret;
@@ -586,7 +586,7 @@ module_init(ak4642_modinit);
 
 static void __exit ak4642_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&ak4642_i2c_driver);
 #endif
 
-- 
cgit v1.2.2


From d8764646e1cc0ad209af29396e59f836b7d8f164 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Wed, 20 Nov 2013 10:04:15 +0100
Subject: ASoC: fsl-ssi: Move ac97 specific setup to seperate function

This is a pure cleanup patch to increase code readability.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++++++++----------------------
 1 file changed, 32 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 35e277379b86..fb8f52a0e7b7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -321,6 +321,36 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	return ret;
 }
 
+static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
+{
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+	/*
+	 * Setup the clock control register
+	 */
+	write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+			&ssi->stccr);
+	write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
+			&ssi->srccr);
+
+	/*
+	 * Enable AC97 mode and startup the SSI
+	 */
+	write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV,
+			&ssi->sacnt);
+	write_ssi(0xff, &ssi->saccdis);
+	write_ssi(0x300, &ssi->saccen);
+
+	/*
+	 * Enable SSI, Transmit and Receive. AC97 has to communicate with the
+	 * codec before a stream is started.
+	 */
+	write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN |
+			CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE);
+
+	write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor);
+}
+
 static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 {
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
@@ -387,31 +417,8 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 	 * because it is also running without an active substream. Normally SSI
 	 * is only enabled when there is a substream.
 	 */
-	if (ssi_private->imx_ac97) {
-		/*
-		 * Setup the clock control register
-		 */
-		write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
-				&ssi->stccr);
-		write_ssi(CCSR_SSI_SxCCR_WL(17) | CCSR_SSI_SxCCR_DC(13),
-				&ssi->srccr);
-
-		/*
-		 * Enable AC97 mode and startup the SSI
-		 */
-		write_ssi(CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV,
-				&ssi->sacnt);
-		write_ssi(0xff, &ssi->saccdis);
-		write_ssi(0x300, &ssi->saccen);
-
-		/*
-		 * Enable SSI, Transmit and Receive
-		 */
-		write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN |
-				CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE);
-
-		write_ssi(CCSR_SSI_SOR_WAIT(3), &ssi->sor);
-	}
+	if (ssi_private->imx_ac97)
+		fsl_ssi_setup_ac97(ssi_private);
 
 	return 0;
 }
-- 
cgit v1.2.2


From c600e95360dac3a3b88f0a2106214dff8e5f56be Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Tue, 19 Nov 2013 14:12:25 +0200
Subject: ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format

The new playback format is needed for tda998x HDMI audio support. At
the moment the only other user of this codec is omap-hdmi-audio. This
change should not break anything because omap-hdmi-audio-dai, the
cpu-dai of omap-hdmi-audio, enforces sufficient constraints to
available sample formats.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
index 68342b121c96..32797a8e4ee9 100644
--- a/sound/soc/codecs/hdmi.c
+++ b/sound/soc/codecs/hdmi.c
@@ -44,7 +44,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = {
 			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
 			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S24_LE,
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
 	},
 	.capture = {
 		.stream_name = "Capture",
-- 
cgit v1.2.2


From 1b488a481c39d9cd36535b6c15fe474546e6460b Mon Sep 17 00:00:00 2001
From: Victor Kamensky <victor.kamensky@linaro.org>
Date: Sat, 16 Nov 2013 02:01:19 +0200
Subject: ASoC: omap: mcbsp, mcpdm, dmic: raw read and write endian fix

All OMAP IP blocks expect LE data, but CPU may operate in BE mode.
Need to use endian neutral functions to read/write h/w registers.
I.e instead of __raw_read[lw] and __raw_write[lw] functions code
need to use read[lw]_relaxed and write[lw]_relaxed functions.
If the first simply reads/writes register, the second will byteswap
it if host operates in BE mode.

Changes are trivial sed like replacement of __raw_xxx functions
with xxx_relaxed variant.

Signed-off-by: Victor Kamensky <victor.kamensky@linaro.org>
Signed-off-by: Taras Kondratiuk <taras.kondratiuk@linaro.org>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/omap/mcbsp.c      | 12 ++++++------
 sound/soc/omap/omap-dmic.c  |  4 ++--
 sound/soc/omap/omap-mcpdm.c |  4 ++--
 3 files changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 83433fdea32a..86c75384c3c8 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -36,10 +36,10 @@ static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
 
 	if (mcbsp->pdata->reg_size == 2) {
 		((u16 *)mcbsp->reg_cache)[reg] = (u16)val;
-		__raw_writew((u16)val, addr);
+		writew_relaxed((u16)val, addr);
 	} else {
 		((u32 *)mcbsp->reg_cache)[reg] = val;
-		__raw_writel(val, addr);
+		writel_relaxed(val, addr);
 	}
 }
 
@@ -48,22 +48,22 @@ static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache)
 	void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
 
 	if (mcbsp->pdata->reg_size == 2) {
-		return !from_cache ? __raw_readw(addr) :
+		return !from_cache ? readw_relaxed(addr) :
 				     ((u16 *)mcbsp->reg_cache)[reg];
 	} else {
-		return !from_cache ? __raw_readl(addr) :
+		return !from_cache ? readl_relaxed(addr) :
 				     ((u32 *)mcbsp->reg_cache)[reg];
 	}
 }
 
 static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
 {
-	__raw_writel(val, mcbsp->st_data->io_base_st + reg);
+	writel_relaxed(val, mcbsp->st_data->io_base_st + reg);
 }
 
 static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg)
 {
-	return __raw_readl(mcbsp->st_data->io_base_st + reg);
+	return readl_relaxed(mcbsp->st_data->io_base_st + reg);
 }
 
 #define MCBSP_READ(mcbsp, reg) \
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 12e566be3793..1bd531d718f9 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -61,12 +61,12 @@ struct omap_dmic {
 
 static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val)
 {
-	__raw_writel(val, dmic->io_base + reg);
+	writel_relaxed(val, dmic->io_base + reg);
 }
 
 static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg)
 {
-	return __raw_readl(dmic->io_base + reg);
+	return readl_relaxed(dmic->io_base + reg);
 }
 
 static inline void omap_dmic_start(struct omap_dmic *dmic)
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index cd9ee167959d..2f5b1536477e 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -74,12 +74,12 @@ struct omap_mcpdm {
 
 static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val)
 {
-	__raw_writel(val, mcpdm->io_base + reg);
+	writel_relaxed(val, mcpdm->io_base + reg);
 }
 
 static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg)
 {
-	return __raw_readl(mcpdm->io_base + reg);
+	return readl_relaxed(mcpdm->io_base + reg);
 }
 
 #ifdef DEBUG
-- 
cgit v1.2.2


From a37377314ff068c83f425979142263a17a6f18af Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Fri, 8 Nov 2013 12:46:53 +0300
Subject: ASoC: rcar: some dubious one-bit signed bitfields

Because these are signed they can either be 0 or -1 instead of 0 and 1
as intended.  It doesn't cause a problem from what I can see, but it's
dangerous and Sparse complains:

	sound/soc/sh/rcar/rsnd.h:177:25:
		error: dubious one-bit signed bitfield

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/rsnd.h | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 9e463e50e7e6..b5ac3a2afc58 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -174,11 +174,11 @@ struct rsnd_dai {
 	struct rsnd_dai_stream playback;
 	struct rsnd_dai_stream capture;
 
-	int clk_master:1;
-	int bit_clk_inv:1;
-	int frm_clk_inv:1;
-	int sys_delay:1;
-	int data_alignment:1;
+	unsigned int clk_master:1;
+	unsigned int bit_clk_inv:1;
+	unsigned int frm_clk_inv:1;
+	unsigned int sys_delay:1;
+	unsigned int data_alignment:1;
 };
 
 #define rsnd_dai_nr(priv) ((priv)->dai_nr)
-- 
cgit v1.2.2


From 06b2bd23057f9dad149f0dda436c7426c87b986f Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:52 -0200
Subject: ASoC: uda1380: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/uda1380.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index fd0a314bc209..726df6d43c2b 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -794,7 +794,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda1380 = {
 	.num_dapm_routes = ARRAY_SIZE(uda1380_dapm_routes),
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int uda1380_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
@@ -840,7 +840,7 @@ static struct i2c_driver uda1380_i2c_driver = {
 static int __init uda1380_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&uda1380_i2c_driver);
 	if (ret != 0)
 		pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
@@ -851,7 +851,7 @@ module_init(uda1380_modinit);
 
 static void __exit uda1380_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&uda1380_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From 784cbf8ab4641c874806a938b9c863c91d70fbe5 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Date: Thu, 21 Nov 2013 13:32:24 +0200
Subject: ASoC: Rename mid-x86 directory to intel

We have other Intel platforms coming having the Smart Sound Technology (SST)
so rename the mid-x86 directory to intel as originally directory name
reflected only Intel MID platform.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/Kconfig                |   2 +-
 sound/soc/Makefile               |   2 +-
 sound/soc/intel/Kconfig          |  13 +
 sound/soc/intel/Makefile         |   5 +
 sound/soc/intel/mfld_machine.c   | 427 +++++++++++++++++++++++
 sound/soc/intel/sst_dsp.h        | 134 +++++++
 sound/soc/intel/sst_platform.c   | 735 +++++++++++++++++++++++++++++++++++++++
 sound/soc/intel/sst_platform.h   | 157 +++++++++
 sound/soc/mid-x86/Kconfig        |  13 -
 sound/soc/mid-x86/Makefile       |   5 -
 sound/soc/mid-x86/mfld_machine.c | 427 -----------------------
 sound/soc/mid-x86/sst_dsp.h      | 134 -------
 sound/soc/mid-x86/sst_platform.c | 735 ---------------------------------------
 sound/soc/mid-x86/sst_platform.h | 157 ---------
 14 files changed, 1473 insertions(+), 1473 deletions(-)
 create mode 100644 sound/soc/intel/Kconfig
 create mode 100644 sound/soc/intel/Makefile
 create mode 100644 sound/soc/intel/mfld_machine.c
 create mode 100644 sound/soc/intel/sst_dsp.h
 create mode 100644 sound/soc/intel/sst_platform.c
 create mode 100644 sound/soc/intel/sst_platform.h
 delete mode 100644 sound/soc/mid-x86/Kconfig
 delete mode 100644 sound/soc/mid-x86/Makefile
 delete mode 100644 sound/soc/mid-x86/mfld_machine.c
 delete mode 100644 sound/soc/mid-x86/sst_dsp.h
 delete mode 100644 sound/soc/mid-x86/sst_platform.c
 delete mode 100644 sound/soc/mid-x86/sst_platform.h

(limited to 'sound')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 5138b8493051..463a9e25e04f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -42,7 +42,7 @@ source "sound/soc/jz4740/Kconfig"
 source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
-source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/intel/Kconfig"
 source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/samsung/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 8b9e70105dd2..ff291d3e60af 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -16,7 +16,7 @@ obj-$(CONFIG_SND_SOC)	+= davinci/
 obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
-obj-$(CONFIG_SND_SOC)	+= mid-x86/
+obj-$(CONFIG_SND_SOC)	+= intel/
 obj-$(CONFIG_SND_SOC)	+= mxs/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
new file mode 100644
index 000000000000..61c10bf503d2
--- /dev/null
+++ b/sound/soc/intel/Kconfig
@@ -0,0 +1,13 @@
+config SND_MFLD_MACHINE
+	tristate "SOC Machine Audio driver for Intel Medfield MID platform"
+	depends on INTEL_SCU_IPC
+	select SND_SOC_SN95031
+	select SND_SST_PLATFORM
+	help
+          This adds support for ASoC machine driver for Intel(R) MID Medfield platform
+          used as alsa device in audio substem in Intel(R) MID devices
+          Say Y if you have such a device
+          If unsure select "N".
+
+config SND_SST_PLATFORM
+	tristate
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
new file mode 100644
index 000000000000..639883339465
--- /dev/null
+++ b/sound/soc/intel/Makefile
@@ -0,0 +1,5 @@
+snd-soc-sst-platform-objs := sst_platform.o
+snd-soc-mfld-machine-objs := mfld_machine.o
+
+obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
+obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c
new file mode 100644
index 000000000000..d3d4c32434f7
--- /dev/null
+++ b/sound/soc/intel/mfld_machine.c
@@ -0,0 +1,427 @@
+/*
+ *  mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/sn95031.h"
+
+#define MID_MONO 1
+#define MID_STEREO 2
+#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+	MFLD_MV_START = 0,
+	/* mic bias volutage range for Headphones*/
+	MFLD_MV_HP = 400,
+	/* mic bias volutage range for American Headset*/
+	MFLD_MV_AM_HS = 650,
+	/* mic bias volutage range for Headset*/
+	MFLD_MV_HS = 2000,
+	MFLD_MV_UNDEFINED,
+};
+
+static unsigned int	hs_switch;
+static unsigned int	lo_dac;
+
+struct mfld_mc_private {
+	void __iomem *int_base;
+	u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin = "AMIC1",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
+/* sound card controls */
+static const char *headset_switch_text[] = {"Earpiece", "Headset"};
+
+static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
+
+static const struct soc_enum headset_enum =
+	SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
+
+static const struct soc_enum lo_enum =
+	SOC_ENUM_SINGLE_EXT(4, lo_text);
+
+static int headset_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = hs_switch;
+	return 0;
+}
+
+static int headset_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == hs_switch)
+		return 0;
+
+	if (ucontrol->value.integer.value[0]) {
+		pr_debug("hs_set HS path\n");
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		pr_debug("hs_set EP path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+	hs_switch = ucontrol->value.integer.value[0];
+
+	return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
+	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
+	snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
+	if (hs_switch) {
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+	} else {
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+	}
+}
+
+static int lo_get_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = lo_dac;
+	return 0;
+}
+
+static int lo_set_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] == lo_dac)
+		return 0;
+
+	/* we dont want to work with last state of lineout so just enable all
+	 * pins and then disable pins not required
+	 */
+	lo_enable_out_pins(codec);
+	switch (ucontrol->value.integer.value[0]) {
+	case 0:
+		pr_debug("set vibra path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
+		snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
+		break;
+
+	case 1:
+		pr_debug("set hs  path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
+		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
+		break;
+
+	case 2:
+		pr_debug("set spkr path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
+		break;
+
+	case 3:
+		pr_debug("set null path\n");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
+		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
+		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
+		break;
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+	lo_dac = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static const struct snd_kcontrol_new mfld_snd_controls[] = {
+	SOC_ENUM_EXT("Playback Switch", headset_enum,
+			headset_get_switch, headset_set_switch),
+	SOC_ENUM_EXT("Lineout Mux", lo_enum,
+			lo_get_switch, lo_set_switch),
+};
+
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+	{"Headphones", NULL, "HPOUTR"},
+	{"Headphones", NULL, "HPOUTL"},
+	{"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+	struct mfld_jack_data jack_data;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(&jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+	struct snd_soc_codec *codec = runtime->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret_val;
+
+	/* Add jack sense widgets */
+	snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
+
+	/* Set up the map */
+	snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Mic");
+
+	ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls,
+				ARRAY_SIZE(mfld_snd_controls));
+	if (ret_val) {
+		pr_err("soc_add_controls failed %d", ret_val);
+		return ret_val;
+	}
+	/* default is earpiece pin, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "Headphones");
+	/* default is lineout NC, userspace sets it explcitly */
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
+	lo_dac = 3;
+	hs_switch = 0;
+	/* we dont use linein in this so set to NC */
+	snd_soc_dapm_disable_pin(dapm, "LINEINL");
+	snd_soc_dapm_disable_pin(dapm, "LINEINR");
+
+	/* Headset and button jack detection */
+	ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 |
+			SND_JACK_BTN_1, &mfld_jack);
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_pins(&mfld_jack,
+			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
+	if (ret_val) {
+		pr_err("adding jack pins failed\n");
+		return ret_val;
+	}
+	ret_val = snd_soc_jack_add_zones(&mfld_jack,
+			ARRAY_SIZE(mfld_zones), mfld_zones);
+	if (ret_val) {
+		pr_err("adding jack zones failed\n");
+		return ret_val;
+	}
+
+	/* we want to check if anything is inserted at boot,
+	 * so send a fake event to codec and it will read adc
+	 * to find if anything is there or not */
+	mfld_jack_check(MFLD_JACK_INSERT);
+	return ret_val;
+}
+
+static struct snd_soc_dai_link mfld_msic_dailink[] = {
+	{
+		.name = "Medfield Headset",
+		.stream_name = "Headset",
+		.cpu_dai_name = "Headset-cpu-dai",
+		.codec_dai_name = "SN95031 Headset",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = mfld_init,
+	},
+	{
+		.name = "Medfield Speaker",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Speaker-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Vibra",
+		.stream_name = "Vibra1",
+		.cpu_dai_name = "Vibra1-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra1",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Haptics",
+		.stream_name = "Vibra2",
+		.cpu_dai_name = "Vibra2-cpu-dai",
+		.codec_dai_name = "SN95031 Vibra2",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+	{
+		.name = "Medfield Compress",
+		.stream_name = "Speaker",
+		.cpu_dai_name = "Compress-cpu-dai",
+		.codec_dai_name = "SN95031 Speaker",
+		.codec_name = "sn95031",
+		.platform_name = "sst-platform",
+		.init = NULL,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_mfld = {
+	.name = "medfield_audio",
+	.owner = THIS_MODULE,
+	.dai_link = mfld_msic_dailink,
+	.num_links = ARRAY_SIZE(mfld_msic_dailink),
+};
+
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+	memcpy_fromio(&mc_private->interrupt_status,
+			((void *)(mc_private->int_base)),
+			sizeof(u8));
+	return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+	if (mfld_jack.codec == NULL)
+		return IRQ_HANDLED;
+	mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+	return IRQ_HANDLED;
+}
+
+static int snd_mfld_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0, irq;
+	struct mfld_mc_private *mc_drv_ctx;
+	struct resource *irq_mem;
+
+	pr_debug("snd_mfld_mc_probe called\n");
+
+	/* retrive the irq number */
+	irq = platform_get_irq(pdev, 0);
+
+	/* audio interrupt base of SRAM location where
+	 * interrupts are stored by System FW */
+	mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
+	if (!mc_drv_ctx) {
+		pr_err("allocation failed\n");
+		return -ENOMEM;
+	}
+
+	irq_mem = platform_get_resource_byname(
+				pdev, IORESOURCE_MEM, "IRQ_BASE");
+	if (!irq_mem) {
+		pr_err("no mem resource given\n");
+		return -ENODEV;
+	}
+	mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
+						    resource_size(irq_mem));
+	if (!mc_drv_ctx->int_base) {
+		pr_err("Mapping of cache failed\n");
+		return -ENOMEM;
+	}
+	/* register for interrupt */
+	ret_val = devm_request_threaded_irq(&pdev->dev, irq,
+			snd_mfld_jack_intr_handler,
+			snd_mfld_jack_detection,
+			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+	if (ret_val) {
+		pr_err("cannot register IRQ\n");
+		return ret_val;
+	}
+	/* register the soc card */
+	snd_soc_card_mfld.dev = &pdev->dev;
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
+	if (ret_val) {
+		pr_debug("snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, mc_drv_ctx);
+	pr_debug("successfully exited probe\n");
+	return 0;
+}
+
+static struct platform_driver snd_mfld_mc_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "msic_audio",
+	},
+	.probe = snd_mfld_mc_probe,
+};
+
+module_platform_driver(snd_mfld_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:msic-audio");
diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst_dsp.h
new file mode 100644
index 000000000000..0fce1de284ff
--- /dev/null
+++ b/sound/soc/intel/sst_dsp.h
@@ -0,0 +1,134 @@
+#ifndef __SST_DSP_H__
+#define __SST_DSP_H__
+/*
+ *  sst_dsp.h - Intel SST Driver for audio engine
+ *
+ *  Copyright (C) 2008-12 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+enum sst_codec_types {
+	/*  AUDIO/MUSIC	CODEC Type Definitions */
+	SST_CODEC_TYPE_UNKNOWN = 0,
+	SST_CODEC_TYPE_PCM,	/* Pass through Audio codec */
+	SST_CODEC_TYPE_MP3,
+	SST_CODEC_TYPE_MP24,
+	SST_CODEC_TYPE_AAC,
+	SST_CODEC_TYPE_AACP,
+	SST_CODEC_TYPE_eAACP,
+};
+
+enum stream_type {
+	SST_STREAM_TYPE_NONE = 0,
+	SST_STREAM_TYPE_MUSIC = 1,
+};
+
+struct snd_pcm_params {
+	u16 codec;	/* codec type */
+	u8 num_chan;	/* 1=Mono, 2=Stereo */
+	u8 pcm_wd_sz;	/* 16/24 - bit*/
+	u32 reserved;	/* Bitrate in bits per second */
+	u32 sfreq;	/* Sampling rate in Hz */
+	u8 use_offload_path;
+	u8 reserved2;
+	u16 reserved3;
+	u8 channel_map[8];
+} __packed;
+
+/* MP3 Music Parameters Message */
+struct snd_mp3_params {
+	u16 codec;
+	u8  num_chan;	/* 1=Mono, 2=Stereo	*/
+	u8  pcm_wd_sz; /* 16/24 - bit*/
+	u8  crc_check; /* crc_check - disable (0) or enable (1) */
+	u8  reserved1; /* unused*/
+	u16 reserved2;	/* Unused */
+} __packed;
+
+#define AAC_BIT_STREAM_ADTS		0
+#define AAC_BIT_STREAM_ADIF		1
+#define AAC_BIT_STREAM_RAW		2
+
+/* AAC Music Parameters Message */
+struct snd_aac_params {
+	u16 codec;
+	u8 num_chan; /* 1=Mono, 2=Stereo*/
+	u8 pcm_wd_sz; /* 16/24 - bit*/
+	u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
+	u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
+	u16  reser2;
+	u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
+	u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
+	u8 reser1;
+	u16  reser3;
+} __packed;
+
+/* WMA Music Parameters Message */
+struct snd_wma_params {
+	u16 codec;
+	u8  num_chan;	/* 1=Mono, 2=Stereo */
+	u8  pcm_wd_sz;	/* 16/24 - bit*/
+	u32 brate;	/* Use the hard coded value. */
+	u32 sfreq;	/* Sampling freq eg. 8000, 441000, 48000 */
+	u32 channel_mask;  /* Channel Mask */
+	u16 format_tag;	/* Format Tag */
+	u16 block_align;	/* packet size */
+	u16 wma_encode_opt;/* Encoder option */
+	u8 op_align;	/* op align 0- 16 bit, 1- MSB, 2 LSB */
+	u8 reserved;	/* reserved */
+} __packed;
+
+/* Codec params struture */
+union  snd_sst_codec_params {
+	struct snd_pcm_params pcm_params;
+	struct snd_mp3_params mp3_params;
+	struct snd_aac_params aac_params;
+	struct snd_wma_params wma_params;
+} __packed;
+
+/* Address and size info of a frame buffer */
+struct sst_address_info {
+	u32 addr; /* Address at IA */
+	u32 size; /* Size of the buffer */
+};
+
+struct snd_sst_alloc_params_ext {
+	struct sst_address_info  ring_buf_info[8];
+	u8 sg_count;
+	u8 reserved;
+	u16 reserved2;
+	u32 frag_size;	/*Number of samples after which period elapsed
+				  message is sent valid only if path  = 0*/
+} __packed;
+
+struct snd_sst_stream_params {
+	union snd_sst_codec_params uc;
+} __packed;
+
+struct snd_sst_params {
+	u32 stream_id;
+	u8 codec;
+	u8 ops;
+	u8 stream_type;
+	u8 device_type;
+	struct snd_sst_stream_params sparams;
+	struct snd_sst_alloc_params_ext aparams;
+};
+
+#endif /* __SST_DSP_H__ */
diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c
new file mode 100644
index 000000000000..b6b5eb698d33
--- /dev/null
+++ b/sound/soc/intel/sst_platform.c
@@ -0,0 +1,735 @@
+/*
+ *  sst_platform.c - Intel MID Platform driver
+ *
+ *  Copyright (C) 2010-2013 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/compress_driver.h>
+#include "sst_platform.h"
+
+static struct sst_device *sst;
+static DEFINE_MUTEX(sst_lock);
+
+int sst_register_dsp(struct sst_device *dev)
+{
+	if (WARN_ON(!dev))
+		return -EINVAL;
+	if (!try_module_get(dev->dev->driver->owner))
+		return -ENODEV;
+	mutex_lock(&sst_lock);
+	if (sst) {
+		pr_err("we already have a device %s\n", sst->name);
+		module_put(dev->dev->driver->owner);
+		mutex_unlock(&sst_lock);
+		return -EEXIST;
+	}
+	pr_debug("registering device %s\n", dev->name);
+	sst = dev;
+	mutex_unlock(&sst_lock);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(sst_register_dsp);
+
+int sst_unregister_dsp(struct sst_device *dev)
+{
+	if (WARN_ON(!dev))
+		return -EINVAL;
+	if (dev != sst)
+		return -EINVAL;
+
+	mutex_lock(&sst_lock);
+
+	if (!sst) {
+		mutex_unlock(&sst_lock);
+		return -EIO;
+	}
+
+	module_put(sst->dev->driver->owner);
+	pr_debug("unreg %s\n", sst->name);
+	sst = NULL;
+	mutex_unlock(&sst_lock);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(sst_unregister_dsp);
+
+static struct snd_pcm_hardware sst_platform_pcm_hw = {
+	.info =	(SNDRV_PCM_INFO_INTERLEAVED |
+			SNDRV_PCM_INFO_DOUBLE |
+			SNDRV_PCM_INFO_PAUSE |
+			SNDRV_PCM_INFO_RESUME |
+			SNDRV_PCM_INFO_MMAP|
+			SNDRV_PCM_INFO_MMAP_VALID |
+			SNDRV_PCM_INFO_BLOCK_TRANSFER |
+			SNDRV_PCM_INFO_SYNC_START),
+	.formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |
+			SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |
+			SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32),
+	.rates = (SNDRV_PCM_RATE_8000|
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000),
+	.rate_min = SST_MIN_RATE,
+	.rate_max = SST_MAX_RATE,
+	.channels_min =	SST_MIN_CHANNEL,
+	.channels_max =	SST_MAX_CHANNEL,
+	.buffer_bytes_max = SST_MAX_BUFFER,
+	.period_bytes_min = SST_MIN_PERIOD_BYTES,
+	.period_bytes_max = SST_MAX_PERIOD_BYTES,
+	.periods_min = SST_MIN_PERIODS,
+	.periods_max = SST_MAX_PERIODS,
+	.fifo_size = SST_FIFO_SIZE,
+};
+
+/* MFLD - MSIC */
+static struct snd_soc_dai_driver sst_platform_dai[] = {
+{
+	.name = "Headset-cpu-dai",
+	.id = 0,
+	.playback = {
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 5,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Speaker-cpu-dai",
+	.id = 1,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra1-cpu-dai",
+	.id = 2,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_MONO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Vibra2-cpu-dai",
+	.id = 3,
+	.playback = {
+		.channels_min = SST_MONO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S24_LE,
+	},
+},
+{
+	.name = "Compress-cpu-dai",
+	.compress_dai = 1,
+	.playback = {
+		.channels_min = SST_STEREO,
+		.channels_max = SST_STEREO,
+		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+},
+};
+
+static const struct snd_soc_component_driver sst_component = {
+	.name		= "sst",
+};
+
+/* helper functions */
+static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
+					int state)
+{
+	unsigned long flags;
+	spin_lock_irqsave(&stream->status_lock, flags);
+	stream->stream_status = state;
+	spin_unlock_irqrestore(&stream->status_lock, flags);
+}
+
+static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
+{
+	int state;
+	unsigned long flags;
+
+	spin_lock_irqsave(&stream->status_lock, flags);
+	state = stream->stream_status;
+	spin_unlock_irqrestore(&stream->status_lock, flags);
+	return state;
+}
+
+static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
+				struct sst_pcm_params *param)
+{
+
+	param->codec = SST_CODEC_TYPE_PCM;
+	param->num_chan = (u8) substream->runtime->channels;
+	param->pcm_wd_sz = substream->runtime->sample_bits;
+	param->reserved = 0;
+	param->sfreq = substream->runtime->rate;
+	param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	param->period_count = substream->runtime->period_size;
+	param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
+	pr_debug("period_cnt = %d\n", param->period_count);
+	pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
+}
+
+static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	struct sst_pcm_params param = {0};
+	struct sst_stream_params str_params = {0};
+	int ret_val;
+
+	/* set codec params and inform SST driver the same */
+	sst_fill_pcm_params(substream, &param);
+	substream->runtime->dma_area = substream->dma_buffer.area;
+	str_params.sparams = param;
+	str_params.codec =  param.codec;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		str_params.ops = STREAM_OPS_PLAYBACK;
+		str_params.device_type = substream->pcm->device + 1;
+		pr_debug("Playbck stream,Device %d\n",
+					substream->pcm->device);
+	} else {
+		str_params.ops = STREAM_OPS_CAPTURE;
+		str_params.device_type = SND_SST_DEVICE_CAPTURE;
+		pr_debug("Capture stream,Device %d\n",
+					substream->pcm->device);
+	}
+	ret_val = stream->ops->open(&str_params);
+	pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
+	if (ret_val < 0)
+		return ret_val;
+
+	stream->stream_info.str_id = ret_val;
+	pr_debug("str id :  %d\n", stream->stream_info.str_id);
+	return ret_val;
+}
+
+static void sst_period_elapsed(void *mad_substream)
+{
+	struct snd_pcm_substream *substream = mad_substream;
+	struct sst_runtime_stream *stream;
+	int status;
+
+	if (!substream || !substream->runtime)
+		return;
+	stream = substream->runtime->private_data;
+	if (!stream)
+		return;
+	status = sst_get_stream_status(stream);
+	if (status != SST_PLATFORM_RUNNING)
+		return;
+	snd_pcm_period_elapsed(substream);
+}
+
+static int sst_platform_init_stream(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream =
+			substream->runtime->private_data;
+	int ret_val;
+
+	pr_debug("setting buffer ptr param\n");
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	stream->stream_info.period_elapsed = sst_period_elapsed;
+	stream->stream_info.mad_substream = substream;
+	stream->stream_info.buffer_ptr = 0;
+	stream->stream_info.sfreq = substream->runtime->rate;
+	ret_val = stream->ops->device_control(
+			SST_SND_STREAM_INIT, &stream->stream_info);
+	if (ret_val)
+		pr_err("control_set ret error %d\n", ret_val);
+	return ret_val;
+
+}
+/* end -- helper functions */
+
+static int sst_platform_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct sst_runtime_stream *stream;
+	int ret_val;
+
+	pr_debug("sst_platform_open called\n");
+
+	snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
+	ret_val = snd_pcm_hw_constraint_integer(runtime,
+						SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret_val < 0)
+		return ret_val;
+
+	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+	if (!stream)
+		return -ENOMEM;
+	spin_lock_init(&stream->status_lock);
+
+	/* get the sst ops */
+	mutex_lock(&sst_lock);
+	if (!sst) {
+		pr_err("no device available to run\n");
+		mutex_unlock(&sst_lock);
+		kfree(stream);
+		return -ENODEV;
+	}
+	if (!try_module_get(sst->dev->driver->owner)) {
+		mutex_unlock(&sst_lock);
+		kfree(stream);
+		return -ENODEV;
+	}
+	stream->ops = sst->ops;
+	mutex_unlock(&sst_lock);
+
+	stream->stream_info.str_id = 0;
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	stream->stream_info.mad_substream = substream;
+	/* allocate memory for SST API set */
+	runtime->private_data = stream;
+
+	return 0;
+}
+
+static int sst_platform_close(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	pr_debug("sst_platform_close called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	if (str_id)
+		ret_val = stream->ops->close(str_id);
+	module_put(sst->dev->driver->owner);
+	kfree(stream);
+	return ret_val;
+}
+
+static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	pr_debug("sst_platform_pcm_prepare called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	if (stream->stream_info.str_id) {
+		ret_val = stream->ops->device_control(
+				SST_SND_DROP, &str_id);
+		return ret_val;
+	}
+
+	ret_val = sst_platform_alloc_stream(substream);
+	if (ret_val < 0)
+		return ret_val;
+	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
+			"%d", stream->stream_info.str_id);
+
+	ret_val = sst_platform_init_stream(substream);
+	if (ret_val)
+		return ret_val;
+	substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER;
+	return ret_val;
+}
+
+static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
+					int cmd)
+{
+	int ret_val = 0, str_id;
+	struct sst_runtime_stream *stream;
+	int str_cmd, status;
+
+	pr_debug("sst_platform_pcm_trigger called\n");
+	stream = substream->runtime->private_data;
+	str_id = stream->stream_info.str_id;
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		pr_debug("sst: Trigger Start\n");
+		str_cmd = SST_SND_START;
+		status = SST_PLATFORM_RUNNING;
+		stream->stream_info.mad_substream = substream;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		pr_debug("sst: in stop\n");
+		str_cmd = SST_SND_DROP;
+		status = SST_PLATFORM_DROPPED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		pr_debug("sst: in pause\n");
+		str_cmd = SST_SND_PAUSE;
+		status = SST_PLATFORM_PAUSED;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		pr_debug("sst: in pause release\n");
+		str_cmd = SST_SND_RESUME;
+		status = SST_PLATFORM_RUNNING;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ret_val = stream->ops->device_control(str_cmd, &str_id);
+	if (!ret_val)
+		sst_set_stream_status(stream, status);
+
+	return ret_val;
+}
+
+
+static snd_pcm_uframes_t sst_platform_pcm_pointer
+			(struct snd_pcm_substream *substream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val, status;
+	struct pcm_stream_info *str_info;
+
+	stream = substream->runtime->private_data;
+	status = sst_get_stream_status(stream);
+	if (status == SST_PLATFORM_INIT)
+		return 0;
+	str_info = &stream->stream_info;
+	ret_val = stream->ops->device_control(
+				SST_SND_BUFFER_POINTER, str_info);
+	if (ret_val) {
+		pr_err("sst: error code = %d\n", ret_val);
+		return ret_val;
+	}
+	return stream->stream_info.buffer_ptr;
+}
+
+static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+
+	return 0;
+}
+
+static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_ops sst_platform_ops = {
+	.open = sst_platform_open,
+	.close = sst_platform_close,
+	.ioctl = snd_pcm_lib_ioctl,
+	.prepare = sst_platform_pcm_prepare,
+	.trigger = sst_platform_pcm_trigger,
+	.pointer = sst_platform_pcm_pointer,
+	.hw_params = sst_platform_pcm_hw_params,
+	.hw_free = sst_platform_pcm_hw_free,
+};
+
+static void sst_pcm_free(struct snd_pcm *pcm)
+{
+	pr_debug("sst_pcm_free called\n");
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm *pcm = rtd->pcm;
+	int retval = 0;
+
+	pr_debug("sst_pcm_new called\n");
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
+			SNDRV_DMA_TYPE_CONTINUOUS,
+			snd_dma_continuous_data(GFP_KERNEL),
+			SST_MIN_BUFFER, SST_MAX_BUFFER);
+		if (retval) {
+			pr_err("dma buffer allocationf fail\n");
+			return retval;
+		}
+	}
+	return retval;
+}
+
+/* compress stream operations */
+static void sst_compr_fragment_elapsed(void *arg)
+{
+	struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
+
+	pr_debug("fragment elapsed by driver\n");
+	if (cstream)
+		snd_compr_fragment_elapsed(cstream);
+}
+
+static int sst_platform_compr_open(struct snd_compr_stream *cstream)
+{
+
+	int ret_val = 0;
+	struct snd_compr_runtime *runtime = cstream->runtime;
+	struct sst_runtime_stream *stream;
+
+	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+	if (!stream)
+		return -ENOMEM;
+
+	spin_lock_init(&stream->status_lock);
+
+	/* get the sst ops */
+	if (!sst || !try_module_get(sst->dev->driver->owner)) {
+		pr_err("no device available to run\n");
+		ret_val = -ENODEV;
+		goto out_ops;
+	}
+	stream->compr_ops = sst->compr_ops;
+
+	stream->id = 0;
+	sst_set_stream_status(stream, SST_PLATFORM_INIT);
+	runtime->private_data = stream;
+	return 0;
+out_ops:
+	kfree(stream);
+	return ret_val;
+}
+
+static int sst_platform_compr_free(struct snd_compr_stream *cstream)
+{
+	struct sst_runtime_stream *stream;
+	int ret_val = 0, str_id;
+
+	stream = cstream->runtime->private_data;
+	/*need to check*/
+	str_id = stream->id;
+	if (str_id)
+		ret_val = stream->compr_ops->close(str_id);
+	module_put(sst->dev->driver->owner);
+	kfree(stream);
+	pr_debug("%s: %d\n", __func__, ret_val);
+	return 0;
+}
+
+static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
+					struct snd_compr_params *params)
+{
+	struct sst_runtime_stream *stream;
+	int retval;
+	struct snd_sst_params str_params;
+	struct sst_compress_cb cb;
+
+	stream = cstream->runtime->private_data;
+	/* construct fw structure for this*/
+	memset(&str_params, 0, sizeof(str_params));
+
+	str_params.ops = STREAM_OPS_PLAYBACK;
+	str_params.stream_type = SST_STREAM_TYPE_MUSIC;
+	str_params.device_type = SND_SST_DEVICE_COMPRESS;
+
+	switch (params->codec.id) {
+	case SND_AUDIOCODEC_MP3: {
+		str_params.codec = SST_CODEC_TYPE_MP3;
+		str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
+		str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
+		str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
+		break;
+	}
+
+	case SND_AUDIOCODEC_AAC: {
+		str_params.codec = SST_CODEC_TYPE_AAC;
+		str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
+		str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
+		str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
+		if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
+			str_params.sparams.uc.aac_params.bs_format =
+							AAC_BIT_STREAM_ADTS;
+		else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
+			str_params.sparams.uc.aac_params.bs_format =
+							AAC_BIT_STREAM_RAW;
+		else {
+			pr_err("Undefined format%d\n", params->codec.format);
+			return -EINVAL;
+		}
+		str_params.sparams.uc.aac_params.externalsr =
+						params->codec.sample_rate;
+		break;
+	}
+
+	default:
+		pr_err("codec not supported, id =%d\n", params->codec.id);
+		return -EINVAL;
+	}
+
+	str_params.aparams.ring_buf_info[0].addr  =
+					virt_to_phys(cstream->runtime->buffer);
+	str_params.aparams.ring_buf_info[0].size =
+					cstream->runtime->buffer_size;
+	str_params.aparams.sg_count = 1;
+	str_params.aparams.frag_size = cstream->runtime->fragment_size;
+
+	cb.param = cstream;
+	cb.compr_cb = sst_compr_fragment_elapsed;
+
+	retval = stream->compr_ops->open(&str_params, &cb);
+	if (retval < 0) {
+		pr_err("stream allocation failed %d\n", retval);
+		return retval;
+	}
+
+	stream->id = retval;
+	return 0;
+}
+
+static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->control(cmd, stream->id);
+}
+
+static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
+					struct snd_compr_tstamp *tstamp)
+{
+	struct sst_runtime_stream *stream;
+
+	stream  = cstream->runtime->private_data;
+	stream->compr_ops->tstamp(stream->id, tstamp);
+	tstamp->byte_offset = tstamp->copied_total %
+				 (u32)cstream->runtime->buffer_size;
+	pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
+	return 0;
+}
+
+static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
+					size_t bytes)
+{
+	struct sst_runtime_stream *stream;
+
+	stream  = cstream->runtime->private_data;
+	stream->compr_ops->ack(stream->id, (unsigned long)bytes);
+	stream->bytes_written += bytes;
+
+	return 0;
+}
+
+static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
+					struct snd_compr_caps *caps)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->get_caps(caps);
+}
+
+static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
+					struct snd_compr_codec_caps *codec)
+{
+	struct sst_runtime_stream *stream =
+		cstream->runtime->private_data;
+
+	return stream->compr_ops->get_codec_caps(codec);
+}
+
+static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream,
+					struct snd_compr_metadata *metadata)
+{
+	struct sst_runtime_stream *stream  =
+		 cstream->runtime->private_data;
+
+	return stream->compr_ops->set_metadata(stream->id, metadata);
+}
+
+static struct snd_compr_ops sst_platform_compr_ops = {
+
+	.open = sst_platform_compr_open,
+	.free = sst_platform_compr_free,
+	.set_params = sst_platform_compr_set_params,
+	.set_metadata = sst_platform_compr_set_metadata,
+	.trigger = sst_platform_compr_trigger,
+	.pointer = sst_platform_compr_pointer,
+	.ack = sst_platform_compr_ack,
+	.get_caps = sst_platform_compr_get_caps,
+	.get_codec_caps = sst_platform_compr_get_codec_caps,
+};
+
+static struct snd_soc_platform_driver sst_soc_platform_drv = {
+	.ops		= &sst_platform_ops,
+	.compr_ops	= &sst_platform_compr_ops,
+	.pcm_new	= sst_pcm_new,
+	.pcm_free	= sst_pcm_free,
+};
+
+static int sst_platform_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	pr_debug("sst_platform_probe called\n");
+	sst = NULL;
+	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
+	if (ret) {
+		pr_err("registering soc platform failed\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_component(&pdev->dev, &sst_component,
+				sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
+	if (ret) {
+		pr_err("registering cpu dais failed\n");
+		snd_soc_unregister_platform(&pdev->dev);
+	}
+	return ret;
+}
+
+static int sst_platform_remove(struct platform_device *pdev)
+{
+
+	snd_soc_unregister_component(&pdev->dev);
+	snd_soc_unregister_platform(&pdev->dev);
+	pr_debug("sst_platform_remove success\n");
+	return 0;
+}
+
+static struct platform_driver sst_platform_driver = {
+	.driver		= {
+		.name		= "sst-platform",
+		.owner		= THIS_MODULE,
+	},
+	.probe		= sst_platform_probe,
+	.remove		= sst_platform_remove,
+};
+
+module_platform_driver(sst_platform_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sst-platform");
diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h
new file mode 100644
index 000000000000..cacc9066ec52
--- /dev/null
+++ b/sound/soc/intel/sst_platform.h
@@ -0,0 +1,157 @@
+/*
+ *  sst_platform.h - Intel MID Platform driver header file
+ *
+ *  Copyright (C) 2010 Intel Corp
+ *  Author: Vinod Koul <vinod.koul@intel.com>
+ *  Author: Harsha Priya <priya.harsha@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *
+ */
+
+#ifndef __SST_PLATFORMDRV_H__
+#define __SST_PLATFORMDRV_H__
+
+#include "sst_dsp.h"
+
+#define SST_MONO		1
+#define SST_STEREO		2
+#define SST_MAX_CAP		5
+
+#define SST_MIN_RATE		8000
+#define SST_MAX_RATE		48000
+#define SST_MIN_CHANNEL		1
+#define SST_MAX_CHANNEL		5
+#define SST_MAX_BUFFER		(800*1024)
+#define SST_MIN_BUFFER		(800*1024)
+#define SST_MIN_PERIOD_BYTES	32
+#define SST_MAX_PERIOD_BYTES	SST_MAX_BUFFER
+#define SST_MIN_PERIODS		2
+#define SST_MAX_PERIODS		(1024*2)
+#define SST_FIFO_SIZE		0
+
+struct pcm_stream_info {
+	int str_id;
+	void *mad_substream;
+	void (*period_elapsed) (void *mad_substream);
+	unsigned long long buffer_ptr;
+	int sfreq;
+};
+
+enum sst_drv_status {
+	SST_PLATFORM_INIT = 1,
+	SST_PLATFORM_STARTED,
+	SST_PLATFORM_RUNNING,
+	SST_PLATFORM_PAUSED,
+	SST_PLATFORM_DROPPED,
+};
+
+enum sst_controls {
+	SST_SND_ALLOC =			0x00,
+	SST_SND_PAUSE =			0x01,
+	SST_SND_RESUME =		0x02,
+	SST_SND_DROP =			0x03,
+	SST_SND_FREE =			0x04,
+	SST_SND_BUFFER_POINTER =	0x05,
+	SST_SND_STREAM_INIT =		0x06,
+	SST_SND_START	 =		0x07,
+	SST_MAX_CONTROLS =		0x07,
+};
+
+enum sst_stream_ops {
+	STREAM_OPS_PLAYBACK = 0,
+	STREAM_OPS_CAPTURE,
+};
+
+enum sst_audio_device_type {
+	SND_SST_DEVICE_HEADSET = 1,
+	SND_SST_DEVICE_IHF,
+	SND_SST_DEVICE_VIBRA,
+	SND_SST_DEVICE_HAPTIC,
+	SND_SST_DEVICE_CAPTURE,
+	SND_SST_DEVICE_COMPRESS,
+};
+
+/* PCM Parameters */
+struct sst_pcm_params {
+	u16 codec;	/* codec type */
+	u8 num_chan;	/* 1=Mono, 2=Stereo */
+	u8 pcm_wd_sz;	/* 16/24 - bit*/
+	u32 reserved;	/* Bitrate in bits per second */
+	u32 sfreq;	/* Sampling rate in Hz */
+	u32 ring_buffer_size;
+	u32 period_count;	/* period elapsed in samples*/
+	u32 ring_buffer_addr;
+};
+
+struct sst_stream_params {
+	u32 result;
+	u32 stream_id;
+	u8 codec;
+	u8 ops;
+	u8 stream_type;
+	u8 device_type;
+	struct sst_pcm_params sparams;
+};
+
+struct sst_compress_cb {
+	void *param;
+	void (*compr_cb)(void *param);
+};
+
+struct compress_sst_ops {
+	const char *name;
+	int (*open) (struct snd_sst_params *str_params,
+			struct sst_compress_cb *cb);
+	int (*control) (unsigned int cmd, unsigned int str_id);
+	int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
+	int (*ack) (unsigned int str_id, unsigned long bytes);
+	int (*close) (unsigned int str_id);
+	int (*get_caps) (struct snd_compr_caps *caps);
+	int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
+	int (*set_metadata) (unsigned int str_id,
+			struct snd_compr_metadata *mdata);
+
+};
+
+struct sst_ops {
+	int (*open) (struct sst_stream_params *str_param);
+	int (*device_control) (int cmd, void *arg);
+	int (*close) (unsigned int str_id);
+};
+
+struct sst_runtime_stream {
+	int     stream_status;
+	unsigned int id;
+	size_t bytes_written;
+	struct pcm_stream_info stream_info;
+	struct sst_ops *ops;
+	struct compress_sst_ops *compr_ops;
+	spinlock_t	status_lock;
+};
+
+struct sst_device {
+	char *name;
+	struct device *dev;
+	struct sst_ops *ops;
+	struct compress_sst_ops *compr_ops;
+};
+
+int sst_register_dsp(struct sst_device *sst);
+int sst_unregister_dsp(struct sst_device *sst);
+#endif
diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig
deleted file mode 100644
index 61c10bf503d2..000000000000
--- a/sound/soc/mid-x86/Kconfig
+++ /dev/null
@@ -1,13 +0,0 @@
-config SND_MFLD_MACHINE
-	tristate "SOC Machine Audio driver for Intel Medfield MID platform"
-	depends on INTEL_SCU_IPC
-	select SND_SOC_SN95031
-	select SND_SST_PLATFORM
-	help
-          This adds support for ASoC machine driver for Intel(R) MID Medfield platform
-          used as alsa device in audio substem in Intel(R) MID devices
-          Say Y if you have such a device
-          If unsure select "N".
-
-config SND_SST_PLATFORM
-	tristate
diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile
deleted file mode 100644
index 639883339465..000000000000
--- a/sound/soc/mid-x86/Makefile
+++ /dev/null
@@ -1,5 +0,0 @@
-snd-soc-sst-platform-objs := sst_platform.o
-snd-soc-mfld-machine-objs := mfld_machine.o
-
-obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
-obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
deleted file mode 100644
index d3d4c32434f7..000000000000
--- a/sound/soc/mid-x86/mfld_machine.c
+++ /dev/null
@@ -1,427 +0,0 @@
-/*
- *  mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
- *
- *  Copyright (C) 2010 Intel Corp
- *  Author: Vinod Koul <vinod.koul@intel.com>
- *  Author: Harsha Priya <priya.harsha@intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- */
-
-#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
-
-#include <linux/init.h>
-#include <linux/device.h>
-#include <linux/slab.h>
-#include <linux/io.h>
-#include <linux/module.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-#include "../codecs/sn95031.h"
-
-#define MID_MONO 1
-#define MID_STEREO 2
-#define MID_MAX_CAP 5
-#define MFLD_JACK_INSERT 0x04
-
-enum soc_mic_bias_zones {
-	MFLD_MV_START = 0,
-	/* mic bias volutage range for Headphones*/
-	MFLD_MV_HP = 400,
-	/* mic bias volutage range for American Headset*/
-	MFLD_MV_AM_HS = 650,
-	/* mic bias volutage range for Headset*/
-	MFLD_MV_HS = 2000,
-	MFLD_MV_UNDEFINED,
-};
-
-static unsigned int	hs_switch;
-static unsigned int	lo_dac;
-
-struct mfld_mc_private {
-	void __iomem *int_base;
-	u8 interrupt_status;
-};
-
-struct snd_soc_jack mfld_jack;
-
-/*Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin mfld_jack_pins[] = {
-	{
-		.pin = "Headphones",
-		.mask = SND_JACK_HEADPHONE,
-	},
-	{
-		.pin = "AMIC1",
-		.mask = SND_JACK_MICROPHONE,
-	},
-};
-
-/* jack detection voltage zones */
-static struct snd_soc_jack_zone mfld_zones[] = {
-	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
-	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
-};
-
-/* sound card controls */
-static const char *headset_switch_text[] = {"Earpiece", "Headset"};
-
-static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
-
-static const struct soc_enum headset_enum =
-	SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
-
-static const struct soc_enum lo_enum =
-	SOC_ENUM_SINGLE_EXT(4, lo_text);
-
-static int headset_get_switch(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = hs_switch;
-	return 0;
-}
-
-static int headset_set_switch(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-
-	if (ucontrol->value.integer.value[0] == hs_switch)
-		return 0;
-
-	if (ucontrol->value.integer.value[0]) {
-		pr_debug("hs_set HS path\n");
-		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
-		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
-	} else {
-		pr_debug("hs_set EP path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
-		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
-	}
-	snd_soc_dapm_sync(&codec->dapm);
-	hs_switch = ucontrol->value.integer.value[0];
-
-	return 0;
-}
-
-static void lo_enable_out_pins(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
-	snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
-	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
-	snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
-	snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
-	snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
-	if (hs_switch) {
-		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
-		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
-	} else {
-		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
-		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
-	}
-}
-
-static int lo_get_switch(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = lo_dac;
-	return 0;
-}
-
-static int lo_set_switch(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-
-	if (ucontrol->value.integer.value[0] == lo_dac)
-		return 0;
-
-	/* we dont want to work with last state of lineout so just enable all
-	 * pins and then disable pins not required
-	 */
-	lo_enable_out_pins(codec);
-	switch (ucontrol->value.integer.value[0]) {
-	case 0:
-		pr_debug("set vibra path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
-		snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
-		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
-		break;
-
-	case 1:
-		pr_debug("set hs  path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
-		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
-		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
-		break;
-
-	case 2:
-		pr_debug("set spkr path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
-		snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
-		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
-		break;
-
-	case 3:
-		pr_debug("set null path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
-		snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
-		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
-		break;
-	}
-	snd_soc_dapm_sync(&codec->dapm);
-	lo_dac = ucontrol->value.integer.value[0];
-	return 0;
-}
-
-static const struct snd_kcontrol_new mfld_snd_controls[] = {
-	SOC_ENUM_EXT("Playback Switch", headset_enum,
-			headset_get_switch, headset_set_switch),
-	SOC_ENUM_EXT("Lineout Mux", lo_enum,
-			lo_get_switch, lo_set_switch),
-};
-
-static const struct snd_soc_dapm_widget mfld_widgets[] = {
-	SND_SOC_DAPM_HP("Headphones", NULL),
-	SND_SOC_DAPM_MIC("Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route mfld_map[] = {
-	{"Headphones", NULL, "HPOUTR"},
-	{"Headphones", NULL, "HPOUTL"},
-	{"Mic", NULL, "AMIC1"},
-};
-
-static void mfld_jack_check(unsigned int intr_status)
-{
-	struct mfld_jack_data jack_data;
-
-	jack_data.mfld_jack = &mfld_jack;
-	jack_data.intr_id = intr_status;
-
-	sn95031_jack_detection(&jack_data);
-	/* TODO: add american headset detection post gpiolib support */
-}
-
-static int mfld_init(struct snd_soc_pcm_runtime *runtime)
-{
-	struct snd_soc_codec *codec = runtime->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret_val;
-
-	/* Add jack sense widgets */
-	snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
-
-	/* Set up the map */
-	snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
-
-	/* always connected */
-	snd_soc_dapm_enable_pin(dapm, "Headphones");
-	snd_soc_dapm_enable_pin(dapm, "Mic");
-
-	ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls,
-				ARRAY_SIZE(mfld_snd_controls));
-	if (ret_val) {
-		pr_err("soc_add_controls failed %d", ret_val);
-		return ret_val;
-	}
-	/* default is earpiece pin, userspace sets it explcitly */
-	snd_soc_dapm_disable_pin(dapm, "Headphones");
-	/* default is lineout NC, userspace sets it explcitly */
-	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
-	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
-	lo_dac = 3;
-	hs_switch = 0;
-	/* we dont use linein in this so set to NC */
-	snd_soc_dapm_disable_pin(dapm, "LINEINL");
-	snd_soc_dapm_disable_pin(dapm, "LINEINR");
-
-	/* Headset and button jack detection */
-	ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
-			SND_JACK_HEADSET | SND_JACK_BTN_0 |
-			SND_JACK_BTN_1, &mfld_jack);
-	if (ret_val) {
-		pr_err("jack creation failed\n");
-		return ret_val;
-	}
-
-	ret_val = snd_soc_jack_add_pins(&mfld_jack,
-			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
-	if (ret_val) {
-		pr_err("adding jack pins failed\n");
-		return ret_val;
-	}
-	ret_val = snd_soc_jack_add_zones(&mfld_jack,
-			ARRAY_SIZE(mfld_zones), mfld_zones);
-	if (ret_val) {
-		pr_err("adding jack zones failed\n");
-		return ret_val;
-	}
-
-	/* we want to check if anything is inserted at boot,
-	 * so send a fake event to codec and it will read adc
-	 * to find if anything is there or not */
-	mfld_jack_check(MFLD_JACK_INSERT);
-	return ret_val;
-}
-
-static struct snd_soc_dai_link mfld_msic_dailink[] = {
-	{
-		.name = "Medfield Headset",
-		.stream_name = "Headset",
-		.cpu_dai_name = "Headset-cpu-dai",
-		.codec_dai_name = "SN95031 Headset",
-		.codec_name = "sn95031",
-		.platform_name = "sst-platform",
-		.init = mfld_init,
-	},
-	{
-		.name = "Medfield Speaker",
-		.stream_name = "Speaker",
-		.cpu_dai_name = "Speaker-cpu-dai",
-		.codec_dai_name = "SN95031 Speaker",
-		.codec_name = "sn95031",
-		.platform_name = "sst-platform",
-		.init = NULL,
-	},
-	{
-		.name = "Medfield Vibra",
-		.stream_name = "Vibra1",
-		.cpu_dai_name = "Vibra1-cpu-dai",
-		.codec_dai_name = "SN95031 Vibra1",
-		.codec_name = "sn95031",
-		.platform_name = "sst-platform",
-		.init = NULL,
-	},
-	{
-		.name = "Medfield Haptics",
-		.stream_name = "Vibra2",
-		.cpu_dai_name = "Vibra2-cpu-dai",
-		.codec_dai_name = "SN95031 Vibra2",
-		.codec_name = "sn95031",
-		.platform_name = "sst-platform",
-		.init = NULL,
-	},
-	{
-		.name = "Medfield Compress",
-		.stream_name = "Speaker",
-		.cpu_dai_name = "Compress-cpu-dai",
-		.codec_dai_name = "SN95031 Speaker",
-		.codec_name = "sn95031",
-		.platform_name = "sst-platform",
-		.init = NULL,
-	},
-};
-
-/* SoC card */
-static struct snd_soc_card snd_soc_card_mfld = {
-	.name = "medfield_audio",
-	.owner = THIS_MODULE,
-	.dai_link = mfld_msic_dailink,
-	.num_links = ARRAY_SIZE(mfld_msic_dailink),
-};
-
-static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
-{
-	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
-
-	memcpy_fromio(&mc_private->interrupt_status,
-			((void *)(mc_private->int_base)),
-			sizeof(u8));
-	return IRQ_WAKE_THREAD;
-}
-
-static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
-{
-	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
-
-	if (mfld_jack.codec == NULL)
-		return IRQ_HANDLED;
-	mfld_jack_check(mc_drv_ctx->interrupt_status);
-
-	return IRQ_HANDLED;
-}
-
-static int snd_mfld_mc_probe(struct platform_device *pdev)
-{
-	int ret_val = 0, irq;
-	struct mfld_mc_private *mc_drv_ctx;
-	struct resource *irq_mem;
-
-	pr_debug("snd_mfld_mc_probe called\n");
-
-	/* retrive the irq number */
-	irq = platform_get_irq(pdev, 0);
-
-	/* audio interrupt base of SRAM location where
-	 * interrupts are stored by System FW */
-	mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
-	if (!mc_drv_ctx) {
-		pr_err("allocation failed\n");
-		return -ENOMEM;
-	}
-
-	irq_mem = platform_get_resource_byname(
-				pdev, IORESOURCE_MEM, "IRQ_BASE");
-	if (!irq_mem) {
-		pr_err("no mem resource given\n");
-		return -ENODEV;
-	}
-	mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
-						    resource_size(irq_mem));
-	if (!mc_drv_ctx->int_base) {
-		pr_err("Mapping of cache failed\n");
-		return -ENOMEM;
-	}
-	/* register for interrupt */
-	ret_val = devm_request_threaded_irq(&pdev->dev, irq,
-			snd_mfld_jack_intr_handler,
-			snd_mfld_jack_detection,
-			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
-	if (ret_val) {
-		pr_err("cannot register IRQ\n");
-		return ret_val;
-	}
-	/* register the soc card */
-	snd_soc_card_mfld.dev = &pdev->dev;
-	ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
-	if (ret_val) {
-		pr_debug("snd_soc_register_card failed %d\n", ret_val);
-		return ret_val;
-	}
-	platform_set_drvdata(pdev, mc_drv_ctx);
-	pr_debug("successfully exited probe\n");
-	return 0;
-}
-
-static struct platform_driver snd_mfld_mc_driver = {
-	.driver = {
-		.owner = THIS_MODULE,
-		.name = "msic_audio",
-	},
-	.probe = snd_mfld_mc_probe,
-};
-
-module_platform_driver(snd_mfld_mc_driver);
-
-MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
-MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
-MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:msic-audio");
diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h
deleted file mode 100644
index 0fce1de284ff..000000000000
--- a/sound/soc/mid-x86/sst_dsp.h
+++ /dev/null
@@ -1,134 +0,0 @@
-#ifndef __SST_DSP_H__
-#define __SST_DSP_H__
-/*
- *  sst_dsp.h - Intel SST Driver for audio engine
- *
- *  Copyright (C) 2008-12 Intel Corporation
- *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- */
-
-enum sst_codec_types {
-	/*  AUDIO/MUSIC	CODEC Type Definitions */
-	SST_CODEC_TYPE_UNKNOWN = 0,
-	SST_CODEC_TYPE_PCM,	/* Pass through Audio codec */
-	SST_CODEC_TYPE_MP3,
-	SST_CODEC_TYPE_MP24,
-	SST_CODEC_TYPE_AAC,
-	SST_CODEC_TYPE_AACP,
-	SST_CODEC_TYPE_eAACP,
-};
-
-enum stream_type {
-	SST_STREAM_TYPE_NONE = 0,
-	SST_STREAM_TYPE_MUSIC = 1,
-};
-
-struct snd_pcm_params {
-	u16 codec;	/* codec type */
-	u8 num_chan;	/* 1=Mono, 2=Stereo */
-	u8 pcm_wd_sz;	/* 16/24 - bit*/
-	u32 reserved;	/* Bitrate in bits per second */
-	u32 sfreq;	/* Sampling rate in Hz */
-	u8 use_offload_path;
-	u8 reserved2;
-	u16 reserved3;
-	u8 channel_map[8];
-} __packed;
-
-/* MP3 Music Parameters Message */
-struct snd_mp3_params {
-	u16 codec;
-	u8  num_chan;	/* 1=Mono, 2=Stereo	*/
-	u8  pcm_wd_sz; /* 16/24 - bit*/
-	u8  crc_check; /* crc_check - disable (0) or enable (1) */
-	u8  reserved1; /* unused*/
-	u16 reserved2;	/* Unused */
-} __packed;
-
-#define AAC_BIT_STREAM_ADTS		0
-#define AAC_BIT_STREAM_ADIF		1
-#define AAC_BIT_STREAM_RAW		2
-
-/* AAC Music Parameters Message */
-struct snd_aac_params {
-	u16 codec;
-	u8 num_chan; /* 1=Mono, 2=Stereo*/
-	u8 pcm_wd_sz; /* 16/24 - bit*/
-	u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
-	u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
-	u16  reser2;
-	u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
-	u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
-	u8 reser1;
-	u16  reser3;
-} __packed;
-
-/* WMA Music Parameters Message */
-struct snd_wma_params {
-	u16 codec;
-	u8  num_chan;	/* 1=Mono, 2=Stereo */
-	u8  pcm_wd_sz;	/* 16/24 - bit*/
-	u32 brate;	/* Use the hard coded value. */
-	u32 sfreq;	/* Sampling freq eg. 8000, 441000, 48000 */
-	u32 channel_mask;  /* Channel Mask */
-	u16 format_tag;	/* Format Tag */
-	u16 block_align;	/* packet size */
-	u16 wma_encode_opt;/* Encoder option */
-	u8 op_align;	/* op align 0- 16 bit, 1- MSB, 2 LSB */
-	u8 reserved;	/* reserved */
-} __packed;
-
-/* Codec params struture */
-union  snd_sst_codec_params {
-	struct snd_pcm_params pcm_params;
-	struct snd_mp3_params mp3_params;
-	struct snd_aac_params aac_params;
-	struct snd_wma_params wma_params;
-} __packed;
-
-/* Address and size info of a frame buffer */
-struct sst_address_info {
-	u32 addr; /* Address at IA */
-	u32 size; /* Size of the buffer */
-};
-
-struct snd_sst_alloc_params_ext {
-	struct sst_address_info  ring_buf_info[8];
-	u8 sg_count;
-	u8 reserved;
-	u16 reserved2;
-	u32 frag_size;	/*Number of samples after which period elapsed
-				  message is sent valid only if path  = 0*/
-} __packed;
-
-struct snd_sst_stream_params {
-	union snd_sst_codec_params uc;
-} __packed;
-
-struct snd_sst_params {
-	u32 stream_id;
-	u8 codec;
-	u8 ops;
-	u8 stream_type;
-	u8 device_type;
-	struct snd_sst_stream_params sparams;
-	struct snd_sst_alloc_params_ext aparams;
-};
-
-#endif /* __SST_DSP_H__ */
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
deleted file mode 100644
index b6b5eb698d33..000000000000
--- a/sound/soc/mid-x86/sst_platform.c
+++ /dev/null
@@ -1,735 +0,0 @@
-/*
- *  sst_platform.c - Intel MID Platform driver
- *
- *  Copyright (C) 2010-2013 Intel Corp
- *  Author: Vinod Koul <vinod.koul@intel.com>
- *  Author: Harsha Priya <priya.harsha@intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *
- */
-#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
-
-#include <linux/slab.h>
-#include <linux/io.h>
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/compress_driver.h>
-#include "sst_platform.h"
-
-static struct sst_device *sst;
-static DEFINE_MUTEX(sst_lock);
-
-int sst_register_dsp(struct sst_device *dev)
-{
-	if (WARN_ON(!dev))
-		return -EINVAL;
-	if (!try_module_get(dev->dev->driver->owner))
-		return -ENODEV;
-	mutex_lock(&sst_lock);
-	if (sst) {
-		pr_err("we already have a device %s\n", sst->name);
-		module_put(dev->dev->driver->owner);
-		mutex_unlock(&sst_lock);
-		return -EEXIST;
-	}
-	pr_debug("registering device %s\n", dev->name);
-	sst = dev;
-	mutex_unlock(&sst_lock);
-	return 0;
-}
-EXPORT_SYMBOL_GPL(sst_register_dsp);
-
-int sst_unregister_dsp(struct sst_device *dev)
-{
-	if (WARN_ON(!dev))
-		return -EINVAL;
-	if (dev != sst)
-		return -EINVAL;
-
-	mutex_lock(&sst_lock);
-
-	if (!sst) {
-		mutex_unlock(&sst_lock);
-		return -EIO;
-	}
-
-	module_put(sst->dev->driver->owner);
-	pr_debug("unreg %s\n", sst->name);
-	sst = NULL;
-	mutex_unlock(&sst_lock);
-	return 0;
-}
-EXPORT_SYMBOL_GPL(sst_unregister_dsp);
-
-static struct snd_pcm_hardware sst_platform_pcm_hw = {
-	.info =	(SNDRV_PCM_INFO_INTERLEAVED |
-			SNDRV_PCM_INFO_DOUBLE |
-			SNDRV_PCM_INFO_PAUSE |
-			SNDRV_PCM_INFO_RESUME |
-			SNDRV_PCM_INFO_MMAP|
-			SNDRV_PCM_INFO_MMAP_VALID |
-			SNDRV_PCM_INFO_BLOCK_TRANSFER |
-			SNDRV_PCM_INFO_SYNC_START),
-	.formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |
-			SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |
-			SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32),
-	.rates = (SNDRV_PCM_RATE_8000|
-			SNDRV_PCM_RATE_44100 |
-			SNDRV_PCM_RATE_48000),
-	.rate_min = SST_MIN_RATE,
-	.rate_max = SST_MAX_RATE,
-	.channels_min =	SST_MIN_CHANNEL,
-	.channels_max =	SST_MAX_CHANNEL,
-	.buffer_bytes_max = SST_MAX_BUFFER,
-	.period_bytes_min = SST_MIN_PERIOD_BYTES,
-	.period_bytes_max = SST_MAX_PERIOD_BYTES,
-	.periods_min = SST_MIN_PERIODS,
-	.periods_max = SST_MAX_PERIODS,
-	.fifo_size = SST_FIFO_SIZE,
-};
-
-/* MFLD - MSIC */
-static struct snd_soc_dai_driver sst_platform_dai[] = {
-{
-	.name = "Headset-cpu-dai",
-	.id = 0,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 5,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Speaker-cpu-dai",
-	.id = 1,
-	.playback = {
-		.channels_min = SST_MONO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Vibra1-cpu-dai",
-	.id = 2,
-	.playback = {
-		.channels_min = SST_MONO,
-		.channels_max = SST_MONO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Vibra2-cpu-dai",
-	.id = 3,
-	.playback = {
-		.channels_min = SST_MONO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S24_LE,
-	},
-},
-{
-	.name = "Compress-cpu-dai",
-	.compress_dai = 1,
-	.playback = {
-		.channels_min = SST_STEREO,
-		.channels_max = SST_STEREO,
-		.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-},
-};
-
-static const struct snd_soc_component_driver sst_component = {
-	.name		= "sst",
-};
-
-/* helper functions */
-static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
-					int state)
-{
-	unsigned long flags;
-	spin_lock_irqsave(&stream->status_lock, flags);
-	stream->stream_status = state;
-	spin_unlock_irqrestore(&stream->status_lock, flags);
-}
-
-static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
-{
-	int state;
-	unsigned long flags;
-
-	spin_lock_irqsave(&stream->status_lock, flags);
-	state = stream->stream_status;
-	spin_unlock_irqrestore(&stream->status_lock, flags);
-	return state;
-}
-
-static void sst_fill_pcm_params(struct snd_pcm_substream *substream,
-				struct sst_pcm_params *param)
-{
-
-	param->codec = SST_CODEC_TYPE_PCM;
-	param->num_chan = (u8) substream->runtime->channels;
-	param->pcm_wd_sz = substream->runtime->sample_bits;
-	param->reserved = 0;
-	param->sfreq = substream->runtime->rate;
-	param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream);
-	param->period_count = substream->runtime->period_size;
-	param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area);
-	pr_debug("period_cnt = %d\n", param->period_count);
-	pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz);
-}
-
-static int sst_platform_alloc_stream(struct snd_pcm_substream *substream)
-{
-	struct sst_runtime_stream *stream =
-			substream->runtime->private_data;
-	struct sst_pcm_params param = {0};
-	struct sst_stream_params str_params = {0};
-	int ret_val;
-
-	/* set codec params and inform SST driver the same */
-	sst_fill_pcm_params(substream, &param);
-	substream->runtime->dma_area = substream->dma_buffer.area;
-	str_params.sparams = param;
-	str_params.codec =  param.codec;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		str_params.ops = STREAM_OPS_PLAYBACK;
-		str_params.device_type = substream->pcm->device + 1;
-		pr_debug("Playbck stream,Device %d\n",
-					substream->pcm->device);
-	} else {
-		str_params.ops = STREAM_OPS_CAPTURE;
-		str_params.device_type = SND_SST_DEVICE_CAPTURE;
-		pr_debug("Capture stream,Device %d\n",
-					substream->pcm->device);
-	}
-	ret_val = stream->ops->open(&str_params);
-	pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val);
-	if (ret_val < 0)
-		return ret_val;
-
-	stream->stream_info.str_id = ret_val;
-	pr_debug("str id :  %d\n", stream->stream_info.str_id);
-	return ret_val;
-}
-
-static void sst_period_elapsed(void *mad_substream)
-{
-	struct snd_pcm_substream *substream = mad_substream;
-	struct sst_runtime_stream *stream;
-	int status;
-
-	if (!substream || !substream->runtime)
-		return;
-	stream = substream->runtime->private_data;
-	if (!stream)
-		return;
-	status = sst_get_stream_status(stream);
-	if (status != SST_PLATFORM_RUNNING)
-		return;
-	snd_pcm_period_elapsed(substream);
-}
-
-static int sst_platform_init_stream(struct snd_pcm_substream *substream)
-{
-	struct sst_runtime_stream *stream =
-			substream->runtime->private_data;
-	int ret_val;
-
-	pr_debug("setting buffer ptr param\n");
-	sst_set_stream_status(stream, SST_PLATFORM_INIT);
-	stream->stream_info.period_elapsed = sst_period_elapsed;
-	stream->stream_info.mad_substream = substream;
-	stream->stream_info.buffer_ptr = 0;
-	stream->stream_info.sfreq = substream->runtime->rate;
-	ret_val = stream->ops->device_control(
-			SST_SND_STREAM_INIT, &stream->stream_info);
-	if (ret_val)
-		pr_err("control_set ret error %d\n", ret_val);
-	return ret_val;
-
-}
-/* end -- helper functions */
-
-static int sst_platform_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct sst_runtime_stream *stream;
-	int ret_val;
-
-	pr_debug("sst_platform_open called\n");
-
-	snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
-	ret_val = snd_pcm_hw_constraint_integer(runtime,
-						SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret_val < 0)
-		return ret_val;
-
-	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
-	if (!stream)
-		return -ENOMEM;
-	spin_lock_init(&stream->status_lock);
-
-	/* get the sst ops */
-	mutex_lock(&sst_lock);
-	if (!sst) {
-		pr_err("no device available to run\n");
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
-	}
-	if (!try_module_get(sst->dev->driver->owner)) {
-		mutex_unlock(&sst_lock);
-		kfree(stream);
-		return -ENODEV;
-	}
-	stream->ops = sst->ops;
-	mutex_unlock(&sst_lock);
-
-	stream->stream_info.str_id = 0;
-	sst_set_stream_status(stream, SST_PLATFORM_INIT);
-	stream->stream_info.mad_substream = substream;
-	/* allocate memory for SST API set */
-	runtime->private_data = stream;
-
-	return 0;
-}
-
-static int sst_platform_close(struct snd_pcm_substream *substream)
-{
-	struct sst_runtime_stream *stream;
-	int ret_val = 0, str_id;
-
-	pr_debug("sst_platform_close called\n");
-	stream = substream->runtime->private_data;
-	str_id = stream->stream_info.str_id;
-	if (str_id)
-		ret_val = stream->ops->close(str_id);
-	module_put(sst->dev->driver->owner);
-	kfree(stream);
-	return ret_val;
-}
-
-static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct sst_runtime_stream *stream;
-	int ret_val = 0, str_id;
-
-	pr_debug("sst_platform_pcm_prepare called\n");
-	stream = substream->runtime->private_data;
-	str_id = stream->stream_info.str_id;
-	if (stream->stream_info.str_id) {
-		ret_val = stream->ops->device_control(
-				SST_SND_DROP, &str_id);
-		return ret_val;
-	}
-
-	ret_val = sst_platform_alloc_stream(substream);
-	if (ret_val < 0)
-		return ret_val;
-	snprintf(substream->pcm->id, sizeof(substream->pcm->id),
-			"%d", stream->stream_info.str_id);
-
-	ret_val = sst_platform_init_stream(substream);
-	if (ret_val)
-		return ret_val;
-	substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER;
-	return ret_val;
-}
-
-static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream,
-					int cmd)
-{
-	int ret_val = 0, str_id;
-	struct sst_runtime_stream *stream;
-	int str_cmd, status;
-
-	pr_debug("sst_platform_pcm_trigger called\n");
-	stream = substream->runtime->private_data;
-	str_id = stream->stream_info.str_id;
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		pr_debug("sst: Trigger Start\n");
-		str_cmd = SST_SND_START;
-		status = SST_PLATFORM_RUNNING;
-		stream->stream_info.mad_substream = substream;
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-		pr_debug("sst: in stop\n");
-		str_cmd = SST_SND_DROP;
-		status = SST_PLATFORM_DROPPED;
-		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		pr_debug("sst: in pause\n");
-		str_cmd = SST_SND_PAUSE;
-		status = SST_PLATFORM_PAUSED;
-		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		pr_debug("sst: in pause release\n");
-		str_cmd = SST_SND_RESUME;
-		status = SST_PLATFORM_RUNNING;
-		break;
-	default:
-		return -EINVAL;
-	}
-	ret_val = stream->ops->device_control(str_cmd, &str_id);
-	if (!ret_val)
-		sst_set_stream_status(stream, status);
-
-	return ret_val;
-}
-
-
-static snd_pcm_uframes_t sst_platform_pcm_pointer
-			(struct snd_pcm_substream *substream)
-{
-	struct sst_runtime_stream *stream;
-	int ret_val, status;
-	struct pcm_stream_info *str_info;
-
-	stream = substream->runtime->private_data;
-	status = sst_get_stream_status(stream);
-	if (status == SST_PLATFORM_INIT)
-		return 0;
-	str_info = &stream->stream_info;
-	ret_val = stream->ops->device_control(
-				SST_SND_BUFFER_POINTER, str_info);
-	if (ret_val) {
-		pr_err("sst: error code = %d\n", ret_val);
-		return ret_val;
-	}
-	return stream->stream_info.buffer_ptr;
-}
-
-static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params)
-{
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
-	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-
-	return 0;
-}
-
-static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
-}
-
-static struct snd_pcm_ops sst_platform_ops = {
-	.open = sst_platform_open,
-	.close = sst_platform_close,
-	.ioctl = snd_pcm_lib_ioctl,
-	.prepare = sst_platform_pcm_prepare,
-	.trigger = sst_platform_pcm_trigger,
-	.pointer = sst_platform_pcm_pointer,
-	.hw_params = sst_platform_pcm_hw_params,
-	.hw_free = sst_platform_pcm_hw_free,
-};
-
-static void sst_pcm_free(struct snd_pcm *pcm)
-{
-	pr_debug("sst_pcm_free called\n");
-	snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
-static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_pcm *pcm = rtd->pcm;
-	int retval = 0;
-
-	pr_debug("sst_pcm_new called\n");
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
-			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		retval =  snd_pcm_lib_preallocate_pages_for_all(pcm,
-			SNDRV_DMA_TYPE_CONTINUOUS,
-			snd_dma_continuous_data(GFP_KERNEL),
-			SST_MIN_BUFFER, SST_MAX_BUFFER);
-		if (retval) {
-			pr_err("dma buffer allocationf fail\n");
-			return retval;
-		}
-	}
-	return retval;
-}
-
-/* compress stream operations */
-static void sst_compr_fragment_elapsed(void *arg)
-{
-	struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
-
-	pr_debug("fragment elapsed by driver\n");
-	if (cstream)
-		snd_compr_fragment_elapsed(cstream);
-}
-
-static int sst_platform_compr_open(struct snd_compr_stream *cstream)
-{
-
-	int ret_val = 0;
-	struct snd_compr_runtime *runtime = cstream->runtime;
-	struct sst_runtime_stream *stream;
-
-	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
-	if (!stream)
-		return -ENOMEM;
-
-	spin_lock_init(&stream->status_lock);
-
-	/* get the sst ops */
-	if (!sst || !try_module_get(sst->dev->driver->owner)) {
-		pr_err("no device available to run\n");
-		ret_val = -ENODEV;
-		goto out_ops;
-	}
-	stream->compr_ops = sst->compr_ops;
-
-	stream->id = 0;
-	sst_set_stream_status(stream, SST_PLATFORM_INIT);
-	runtime->private_data = stream;
-	return 0;
-out_ops:
-	kfree(stream);
-	return ret_val;
-}
-
-static int sst_platform_compr_free(struct snd_compr_stream *cstream)
-{
-	struct sst_runtime_stream *stream;
-	int ret_val = 0, str_id;
-
-	stream = cstream->runtime->private_data;
-	/*need to check*/
-	str_id = stream->id;
-	if (str_id)
-		ret_val = stream->compr_ops->close(str_id);
-	module_put(sst->dev->driver->owner);
-	kfree(stream);
-	pr_debug("%s: %d\n", __func__, ret_val);
-	return 0;
-}
-
-static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
-					struct snd_compr_params *params)
-{
-	struct sst_runtime_stream *stream;
-	int retval;
-	struct snd_sst_params str_params;
-	struct sst_compress_cb cb;
-
-	stream = cstream->runtime->private_data;
-	/* construct fw structure for this*/
-	memset(&str_params, 0, sizeof(str_params));
-
-	str_params.ops = STREAM_OPS_PLAYBACK;
-	str_params.stream_type = SST_STREAM_TYPE_MUSIC;
-	str_params.device_type = SND_SST_DEVICE_COMPRESS;
-
-	switch (params->codec.id) {
-	case SND_AUDIOCODEC_MP3: {
-		str_params.codec = SST_CODEC_TYPE_MP3;
-		str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
-		str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
-		str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
-		break;
-	}
-
-	case SND_AUDIOCODEC_AAC: {
-		str_params.codec = SST_CODEC_TYPE_AAC;
-		str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
-		str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
-		str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
-		if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
-			str_params.sparams.uc.aac_params.bs_format =
-							AAC_BIT_STREAM_ADTS;
-		else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
-			str_params.sparams.uc.aac_params.bs_format =
-							AAC_BIT_STREAM_RAW;
-		else {
-			pr_err("Undefined format%d\n", params->codec.format);
-			return -EINVAL;
-		}
-		str_params.sparams.uc.aac_params.externalsr =
-						params->codec.sample_rate;
-		break;
-	}
-
-	default:
-		pr_err("codec not supported, id =%d\n", params->codec.id);
-		return -EINVAL;
-	}
-
-	str_params.aparams.ring_buf_info[0].addr  =
-					virt_to_phys(cstream->runtime->buffer);
-	str_params.aparams.ring_buf_info[0].size =
-					cstream->runtime->buffer_size;
-	str_params.aparams.sg_count = 1;
-	str_params.aparams.frag_size = cstream->runtime->fragment_size;
-
-	cb.param = cstream;
-	cb.compr_cb = sst_compr_fragment_elapsed;
-
-	retval = stream->compr_ops->open(&str_params, &cb);
-	if (retval < 0) {
-		pr_err("stream allocation failed %d\n", retval);
-		return retval;
-	}
-
-	stream->id = retval;
-	return 0;
-}
-
-static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
-{
-	struct sst_runtime_stream *stream =
-		cstream->runtime->private_data;
-
-	return stream->compr_ops->control(cmd, stream->id);
-}
-
-static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
-					struct snd_compr_tstamp *tstamp)
-{
-	struct sst_runtime_stream *stream;
-
-	stream  = cstream->runtime->private_data;
-	stream->compr_ops->tstamp(stream->id, tstamp);
-	tstamp->byte_offset = tstamp->copied_total %
-				 (u32)cstream->runtime->buffer_size;
-	pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
-	return 0;
-}
-
-static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
-					size_t bytes)
-{
-	struct sst_runtime_stream *stream;
-
-	stream  = cstream->runtime->private_data;
-	stream->compr_ops->ack(stream->id, (unsigned long)bytes);
-	stream->bytes_written += bytes;
-
-	return 0;
-}
-
-static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
-					struct snd_compr_caps *caps)
-{
-	struct sst_runtime_stream *stream =
-		cstream->runtime->private_data;
-
-	return stream->compr_ops->get_caps(caps);
-}
-
-static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
-					struct snd_compr_codec_caps *codec)
-{
-	struct sst_runtime_stream *stream =
-		cstream->runtime->private_data;
-
-	return stream->compr_ops->get_codec_caps(codec);
-}
-
-static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream,
-					struct snd_compr_metadata *metadata)
-{
-	struct sst_runtime_stream *stream  =
-		 cstream->runtime->private_data;
-
-	return stream->compr_ops->set_metadata(stream->id, metadata);
-}
-
-static struct snd_compr_ops sst_platform_compr_ops = {
-
-	.open = sst_platform_compr_open,
-	.free = sst_platform_compr_free,
-	.set_params = sst_platform_compr_set_params,
-	.set_metadata = sst_platform_compr_set_metadata,
-	.trigger = sst_platform_compr_trigger,
-	.pointer = sst_platform_compr_pointer,
-	.ack = sst_platform_compr_ack,
-	.get_caps = sst_platform_compr_get_caps,
-	.get_codec_caps = sst_platform_compr_get_codec_caps,
-};
-
-static struct snd_soc_platform_driver sst_soc_platform_drv = {
-	.ops		= &sst_platform_ops,
-	.compr_ops	= &sst_platform_compr_ops,
-	.pcm_new	= sst_pcm_new,
-	.pcm_free	= sst_pcm_free,
-};
-
-static int sst_platform_probe(struct platform_device *pdev)
-{
-	int ret;
-
-	pr_debug("sst_platform_probe called\n");
-	sst = NULL;
-	ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv);
-	if (ret) {
-		pr_err("registering soc platform failed\n");
-		return ret;
-	}
-
-	ret = snd_soc_register_component(&pdev->dev, &sst_component,
-				sst_platform_dai, ARRAY_SIZE(sst_platform_dai));
-	if (ret) {
-		pr_err("registering cpu dais failed\n");
-		snd_soc_unregister_platform(&pdev->dev);
-	}
-	return ret;
-}
-
-static int sst_platform_remove(struct platform_device *pdev)
-{
-
-	snd_soc_unregister_component(&pdev->dev);
-	snd_soc_unregister_platform(&pdev->dev);
-	pr_debug("sst_platform_remove success\n");
-	return 0;
-}
-
-static struct platform_driver sst_platform_driver = {
-	.driver		= {
-		.name		= "sst-platform",
-		.owner		= THIS_MODULE,
-	},
-	.probe		= sst_platform_probe,
-	.remove		= sst_platform_remove,
-};
-
-module_platform_driver(sst_platform_driver);
-
-MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver");
-MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
-MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:sst-platform");
diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h
deleted file mode 100644
index cacc9066ec52..000000000000
--- a/sound/soc/mid-x86/sst_platform.h
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- *  sst_platform.h - Intel MID Platform driver header file
- *
- *  Copyright (C) 2010 Intel Corp
- *  Author: Vinod Koul <vinod.koul@intel.com>
- *  Author: Harsha Priya <priya.harsha@intel.com>
- *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms of the GNU General Public License as published by
- *  the Free Software Foundation; version 2 of the License.
- *
- *  This program is distributed in the hope that it will be useful, but
- *  WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  General Public License for more details.
- *
- *  You should have received a copy of the GNU General Public License along
- *  with this program; if not, write to the Free Software Foundation, Inc.,
- *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
- *
- * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
- *
- *
- */
-
-#ifndef __SST_PLATFORMDRV_H__
-#define __SST_PLATFORMDRV_H__
-
-#include "sst_dsp.h"
-
-#define SST_MONO		1
-#define SST_STEREO		2
-#define SST_MAX_CAP		5
-
-#define SST_MIN_RATE		8000
-#define SST_MAX_RATE		48000
-#define SST_MIN_CHANNEL		1
-#define SST_MAX_CHANNEL		5
-#define SST_MAX_BUFFER		(800*1024)
-#define SST_MIN_BUFFER		(800*1024)
-#define SST_MIN_PERIOD_BYTES	32
-#define SST_MAX_PERIOD_BYTES	SST_MAX_BUFFER
-#define SST_MIN_PERIODS		2
-#define SST_MAX_PERIODS		(1024*2)
-#define SST_FIFO_SIZE		0
-
-struct pcm_stream_info {
-	int str_id;
-	void *mad_substream;
-	void (*period_elapsed) (void *mad_substream);
-	unsigned long long buffer_ptr;
-	int sfreq;
-};
-
-enum sst_drv_status {
-	SST_PLATFORM_INIT = 1,
-	SST_PLATFORM_STARTED,
-	SST_PLATFORM_RUNNING,
-	SST_PLATFORM_PAUSED,
-	SST_PLATFORM_DROPPED,
-};
-
-enum sst_controls {
-	SST_SND_ALLOC =			0x00,
-	SST_SND_PAUSE =			0x01,
-	SST_SND_RESUME =		0x02,
-	SST_SND_DROP =			0x03,
-	SST_SND_FREE =			0x04,
-	SST_SND_BUFFER_POINTER =	0x05,
-	SST_SND_STREAM_INIT =		0x06,
-	SST_SND_START	 =		0x07,
-	SST_MAX_CONTROLS =		0x07,
-};
-
-enum sst_stream_ops {
-	STREAM_OPS_PLAYBACK = 0,
-	STREAM_OPS_CAPTURE,
-};
-
-enum sst_audio_device_type {
-	SND_SST_DEVICE_HEADSET = 1,
-	SND_SST_DEVICE_IHF,
-	SND_SST_DEVICE_VIBRA,
-	SND_SST_DEVICE_HAPTIC,
-	SND_SST_DEVICE_CAPTURE,
-	SND_SST_DEVICE_COMPRESS,
-};
-
-/* PCM Parameters */
-struct sst_pcm_params {
-	u16 codec;	/* codec type */
-	u8 num_chan;	/* 1=Mono, 2=Stereo */
-	u8 pcm_wd_sz;	/* 16/24 - bit*/
-	u32 reserved;	/* Bitrate in bits per second */
-	u32 sfreq;	/* Sampling rate in Hz */
-	u32 ring_buffer_size;
-	u32 period_count;	/* period elapsed in samples*/
-	u32 ring_buffer_addr;
-};
-
-struct sst_stream_params {
-	u32 result;
-	u32 stream_id;
-	u8 codec;
-	u8 ops;
-	u8 stream_type;
-	u8 device_type;
-	struct sst_pcm_params sparams;
-};
-
-struct sst_compress_cb {
-	void *param;
-	void (*compr_cb)(void *param);
-};
-
-struct compress_sst_ops {
-	const char *name;
-	int (*open) (struct snd_sst_params *str_params,
-			struct sst_compress_cb *cb);
-	int (*control) (unsigned int cmd, unsigned int str_id);
-	int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
-	int (*ack) (unsigned int str_id, unsigned long bytes);
-	int (*close) (unsigned int str_id);
-	int (*get_caps) (struct snd_compr_caps *caps);
-	int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
-	int (*set_metadata) (unsigned int str_id,
-			struct snd_compr_metadata *mdata);
-
-};
-
-struct sst_ops {
-	int (*open) (struct sst_stream_params *str_param);
-	int (*device_control) (int cmd, void *arg);
-	int (*close) (unsigned int str_id);
-};
-
-struct sst_runtime_stream {
-	int     stream_status;
-	unsigned int id;
-	size_t bytes_written;
-	struct pcm_stream_info stream_info;
-	struct sst_ops *ops;
-	struct compress_sst_ops *compr_ops;
-	spinlock_t	status_lock;
-};
-
-struct sst_device {
-	char *name;
-	struct device *dev;
-	struct sst_ops *ops;
-	struct compress_sst_ops *compr_ops;
-};
-
-int sst_register_dsp(struct sst_device *sst);
-int sst_unregister_dsp(struct sst_device *sst);
-#endif
-- 
cgit v1.2.2


From 8e6ad35a31e7ebc59543df875fc970200df2cf68 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:51 -0200
Subject: ASoC: wm8510: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8510.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 6ed5433943ea..7df7d4572755 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -684,7 +684,7 @@ static struct spi_driver wm8510_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8510_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -735,7 +735,7 @@ static struct i2c_driver wm8510_i2c_driver = {
 static int __init wm8510_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8510_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register WM8510 I2C driver: %d\n",
@@ -755,7 +755,7 @@ module_init(wm8510_modinit);
 
 static void __exit wm8510_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8510_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 008ef947d0c5d14442256a37f6bf6b14015efe26 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:50 -0200
Subject: ASoC: wm8523: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8523.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 139bf9ac9407..74d106dc7667 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -452,7 +452,7 @@ static const struct regmap_config wm8523_regmap = {
 	.volatile_reg = wm8523_volatile_register,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8523_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -555,7 +555,7 @@ static struct i2c_driver wm8523_i2c_driver = {
 static int __init wm8523_modinit(void)
 {
 	int ret;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8523_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n",
@@ -568,7 +568,7 @@ module_init(wm8523_modinit);
 
 static void __exit wm8523_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8523_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From f58c4fc4a3bf9eb699a638634d20ef4d16069366 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:49 -0200
Subject: ASoC: wm8580: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8580.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 08a414b57b1e..318989acbbe5 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -941,7 +941,7 @@ static const struct regmap_config wm8580_regmap = {
 	.volatile_reg = wm8580_volatile,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1003,7 +1003,7 @@ static int __init wm8580_modinit(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8580_i2c_driver);
 	if (ret != 0) {
 		pr_err("Failed to register WM8580 I2C driver: %d\n", ret);
@@ -1016,7 +1016,7 @@ module_init(wm8580_modinit);
 
 static void __exit wm8580_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8580_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From 2309d6757900c4a6909fa673724976935b408a25 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:48 -0200
Subject: ASoC: wm8711: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8711.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 5b428b060d41..d99f948c513c 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -469,7 +469,7 @@ static struct spi_driver wm8711_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8711_i2c_probe(struct i2c_client *client,
 			    const struct i2c_device_id *id)
 {
@@ -520,7 +520,7 @@ static struct i2c_driver wm8711_i2c_driver = {
 static int __init wm8711_modinit(void)
 {
 	int ret;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8711_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n",
@@ -540,7 +540,7 @@ module_init(wm8711_modinit);
 
 static void __exit wm8711_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8711_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 5c1537163ce716e317776565b8210ea06fa2b5de Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:47 -0200
Subject: ASoC: wm8728: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8728.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index c6a292dcded0..cd89033e84c0 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -320,7 +320,7 @@ static struct spi_driver wm8728_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8728_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -371,7 +371,7 @@ static struct i2c_driver wm8728_i2c_driver = {
 static int __init wm8728_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8728_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register wm8728 I2C driver: %d\n",
@@ -391,7 +391,7 @@ module_init(wm8728_modinit);
 
 static void __exit wm8728_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8728_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 6435e5be652633b87f73460f9c17be361404ee01 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 8 Nov 2013 17:19:55 +0000
Subject: ASoC: wm8940: Convert to table based control and DAPM init

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Jonathan Cameron <jic23@kernel.org>
---
 sound/soc/codecs/wm8940.c | 28 +++++++---------------------
 1 file changed, 7 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b1591c61c254..4858b5c039fa 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -264,7 +264,7 @@ static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("AUX"),
 };
 
-static const struct snd_soc_dapm_route audio_map[] = {
+static const struct snd_soc_dapm_route wm8940_dapm_routes[] = {
 	/* Mono output mixer */
 	{"Mono Mixer", "PCM Playback Switch", "DAC"},
 	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
@@ -296,21 +296,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"ADC", NULL, "Boost Mixer"},
 };
 
-static int wm8940_add_widgets(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret;
-
-	ret = snd_soc_dapm_new_controls(dapm, wm8940_dapm_widgets,
-					ARRAY_SIZE(wm8940_dapm_widgets));
-	if (ret)
-		goto error_ret;
-	ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-error_ret:
-	return ret;
-}
-
 #define wm8940_reset(c) snd_soc_write(c, WM8940_SOFTRESET, 0);
 
 static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
@@ -716,11 +701,6 @@ static int wm8940_probe(struct snd_soc_codec *codec)
 			return ret;
 	}
 
-	ret = snd_soc_add_codec_controls(codec, wm8940_snd_controls,
-			     ARRAY_SIZE(wm8940_snd_controls));
-	if (ret)
-		return ret;
-	ret = wm8940_add_widgets(codec);
 	return ret;
 }
 
@@ -736,6 +716,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = {
 	.suspend =	wm8940_suspend,
 	.resume =	wm8940_resume,
 	.set_bias_level = wm8940_set_bias_level,
+	.controls =     wm8940_snd_controls,
+	.num_controls = ARRAY_SIZE(wm8940_snd_controls),
+	.dapm_widgets = wm8940_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets),
+	.dapm_routes =  wm8940_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes),
 	.reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8940_reg_defaults,
-- 
cgit v1.2.2


From fbbf7fea8e806ccc3ce0e059ad1d9671d57b4309 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 8 Nov 2013 17:22:23 +0000
Subject: ASoC: wm8940: Convert to direct regmap API usage

This helps move us towards being able to remove the ASoC level I/O code.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Jonathan Cameron <jic23@kernel.org>
---
 sound/soc/codecs/wm8940.c | 193 +++++++++++++++++++++++++++++-----------------
 1 file changed, 121 insertions(+), 72 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 4858b5c039fa..b404c26c1753 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -28,7 +28,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -41,78 +41,116 @@
 
 struct wm8940_priv {
 	unsigned int sysclk;
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 };
 
-static int wm8940_volatile_register(struct snd_soc_codec *codec,
-				    unsigned int reg)
+static bool wm8940_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case WM8940_SOFTRESET:
-		return 1;
+		return true;
 	default:
-		return 0;
+		return false;
+	}
+}
+
+static bool wm8940_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case WM8940_SOFTRESET:
+	case WM8940_POWER1:
+	case WM8940_POWER2:
+	case WM8940_POWER3:
+	case WM8940_IFACE:
+	case WM8940_COMPANDINGCTL:
+	case WM8940_CLOCK:
+	case WM8940_ADDCNTRL:
+	case WM8940_GPIO:
+	case WM8940_CTLINT:
+	case WM8940_DAC:
+	case WM8940_DACVOL:
+	case WM8940_ADC:
+	case WM8940_ADCVOL:
+	case WM8940_NOTCH1:
+	case WM8940_NOTCH2:
+	case WM8940_NOTCH3:
+	case WM8940_NOTCH4:
+	case WM8940_NOTCH5:
+	case WM8940_NOTCH6:
+	case WM8940_NOTCH7:
+	case WM8940_NOTCH8:
+	case WM8940_DACLIM1:
+	case WM8940_DACLIM2:
+	case WM8940_ALC1:
+	case WM8940_ALC2:
+	case WM8940_ALC3:
+	case WM8940_NOISEGATE:
+	case WM8940_PLLN:
+	case WM8940_PLLK1:
+	case WM8940_PLLK2:
+	case WM8940_PLLK3:
+	case WM8940_ALC4:
+	case WM8940_INPUTCTL:
+	case WM8940_PGAGAIN:
+	case WM8940_ADCBOOST:
+	case WM8940_OUTPUTCTL:
+	case WM8940_SPKMIX:
+	case WM8940_SPKVOL:
+	case WM8940_MONOMIX:
+		return true;
+	default:
+		return false;
 	}
 }
 
-static u16 wm8940_reg_defaults[] = {
-	0x8940, /* Soft Reset */
-	0x0000, /* Power 1 */
-	0x0000, /* Power 2 */
-	0x0000, /* Power 3 */
-	0x0010, /* Interface Control */
-	0x0000, /* Companding Control */
-	0x0140, /* Clock Control */
-	0x0000, /* Additional Controls */
-	0x0000, /* GPIO Control */
-	0x0002, /* Auto Increment Control */
-	0x0000, /* DAC Control */
-	0x00FF, /* DAC Volume */
-	0,
-	0,
-	0x0100, /* ADC Control */
-	0x00FF, /* ADC Volume */
-	0x0000, /* Notch Filter 1 Control 1 */
-	0x0000, /* Notch Filter 1 Control 2 */
-	0x0000, /* Notch Filter 2 Control 1 */
-	0x0000, /* Notch Filter 2 Control 2 */
-	0x0000, /* Notch Filter 3 Control 1 */
-	0x0000, /* Notch Filter 3 Control 2 */
-	0x0000, /* Notch Filter 4 Control 1 */
-	0x0000, /* Notch Filter 4 Control 2 */
-	0x0032, /* DAC Limit Control 1 */
-	0x0000, /* DAC Limit Control 2 */
-	0,
-	0,
-	0,
-	0,
-	0,
-	0,
-	0x0038, /* ALC Control 1 */
-	0x000B, /* ALC Control 2 */
-	0x0032, /* ALC Control 3 */
-	0x0000, /* Noise Gate */
-	0x0041, /* PLLN */
-	0x000C, /* PLLK1 */
-	0x0093, /* PLLK2 */
-	0x00E9, /* PLLK3 */
-	0,
-	0,
-	0x0030, /* ALC Control 4 */
-	0,
-	0x0002, /* Input Control */
-	0x0050, /* PGA Gain */
-	0,
-	0x0002, /* ADC Boost Control */
-	0,
-	0x0002, /* Output Control */
-	0x0000, /* Speaker Mixer Control */
-	0,
-	0,
-	0,
-	0x0079, /* Speaker Volume */
-	0,
-	0x0000, /* Mono Mixer Control */
+static const struct reg_default wm8940_reg_defaults[] = {
+	{  0x1, 0x0000 }, /* Power 1 */
+	{  0x2, 0x0000 }, /* Power 2 */
+	{  0x3, 0x0000 }, /* Power 3 */
+	{  0x4, 0x0010 }, /* Interface Control */
+	{  0x5, 0x0000 }, /* Companding Control */
+	{  0x6, 0x0140 }, /* Clock Control */
+	{  0x7, 0x0000 }, /* Additional Controls */
+	{  0x8, 0x0000 }, /* GPIO Control */
+	{  0x9, 0x0002 }, /* Auto Increment Control */
+	{  0xa, 0x0000 }, /* DAC Control */
+	{  0xb, 0x00FF }, /* DAC Volume */
+
+	{  0xe, 0x0100 }, /* ADC Control */
+	{  0xf, 0x00FF }, /* ADC Volume */
+	{ 0x10, 0x0000 }, /* Notch Filter 1 Control 1 */
+	{ 0x11, 0x0000 }, /* Notch Filter 1 Control 2 */
+	{ 0x12, 0x0000 }, /* Notch Filter 2 Control 1 */
+	{ 0x13, 0x0000 }, /* Notch Filter 2 Control 2 */
+	{ 0x14, 0x0000 }, /* Notch Filter 3 Control 1 */
+	{ 0x15, 0x0000 }, /* Notch Filter 3 Control 2 */
+	{ 0x16, 0x0000 }, /* Notch Filter 4 Control 1 */
+	{ 0x17, 0x0000 }, /* Notch Filter 4 Control 2 */
+	{ 0x18, 0x0032 }, /* DAC Limit Control 1 */
+	{ 0x19, 0x0000 }, /* DAC Limit Control 2 */
+
+	{ 0x20, 0x0038 }, /* ALC Control 1 */
+	{ 0x21, 0x000B }, /* ALC Control 2 */
+	{ 0x22, 0x0032 }, /* ALC Control 3 */
+	{ 0x23, 0x0000 }, /* Noise Gate */
+	{ 0x24, 0x0041 }, /* PLLN */
+	{ 0x25, 0x000C }, /* PLLK1 */
+	{ 0x26, 0x0093 }, /* PLLK2 */
+	{ 0x27, 0x00E9 }, /* PLLK3 */
+
+	{ 0x2a, 0x0030 }, /* ALC Control 4 */
+
+	{ 0x2c, 0x0002 }, /* Input Control */
+	{ 0x2d, 0x0050 }, /* PGA Gain */
+
+	{ 0x2f, 0x0002 }, /* ADC Boost Control */
+
+	{ 0x31, 0x0002 }, /* Output Control */
+	{ 0x32, 0x0000 }, /* Speaker Mixer Control */
+
+	{ 0x36, 0x0079 }, /* Speaker Volume */
+
+	{ 0x38, 0x0000 }, /* Mono Mixer Control */
 };
 
 static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
@@ -431,6 +469,7 @@ static int wm8940_mute(struct snd_soc_dai *dai, int mute)
 static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
+	struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec);
 	u16 val;
 	u16 pwr_reg = snd_soc_read(codec, WM8940_POWER1) & 0x1F0;
 	int ret = 0;
@@ -454,7 +493,7 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_cache_sync(codec);
+			ret = regcache_sync(wm8940->regmap);
 			if (ret < 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
 				return ret;
@@ -669,12 +708,11 @@ static int wm8940_resume(struct snd_soc_codec *codec)
 
 static int wm8940_probe(struct snd_soc_codec *codec)
 {
-	struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec);
 	struct wm8940_setup_data *pdata = codec->dev->platform_data;
 	int ret;
 	u16 reg;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -722,10 +760,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = {
 	.num_dapm_widgets = ARRAY_SIZE(wm8940_dapm_widgets),
 	.dapm_routes =  wm8940_dapm_routes,
 	.num_dapm_routes = ARRAY_SIZE(wm8940_dapm_routes),
-	.reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults),
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = wm8940_reg_defaults,
-	.volatile_register = wm8940_volatile_register,
+};
+
+static const struct regmap_config wm8940_regmap = {
+	.reg_bits = 8,
+	.val_bits = 16,
+
+	.max_register = WM8940_MONOMIX,
+	.reg_defaults = wm8940_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults),
+
+	.readable_reg = wm8940_readable_register,
+	.volatile_reg = wm8940_volatile_register,
 };
 
 static int wm8940_i2c_probe(struct i2c_client *i2c,
@@ -739,8 +785,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c,
 	if (wm8940 == NULL)
 		return -ENOMEM;
 
+	wm8940->regmap = devm_regmap_init_i2c(i2c, &wm8940_regmap);
+	if (IS_ERR(wm8940->regmap))
+		return PTR_ERR(wm8940->regmap);
+
 	i2c_set_clientdata(i2c, wm8940);
-	wm8940->control_type = SND_SOC_I2C;
 
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8940, &wm8940_dai, 1);
-- 
cgit v1.2.2


From c3e8494c001ce0bec0ebaa49c6f5eeb2aa5ab36a Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:41 -0200
Subject: ASoC: wm8962: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8962.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..07da601f8343 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -74,7 +74,7 @@ struct wm8962_priv {
 	struct regulator_bulk_data supplies[WM8962_NUM_SUPPLIES];
 	struct notifier_block disable_nb[WM8962_NUM_SUPPLIES];
 
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
 	struct input_dev *beep;
 	struct work_struct beep_work;
 	int beep_rate;
@@ -3108,7 +3108,7 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 }
 EXPORT_SYMBOL_GPL(wm8962_mic_detect);
 
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
 static int beep_rates[] = {
 	500, 1000, 2000, 4000,
 };
-- 
cgit v1.2.2


From e40e0b5da87bb4256a6dc62db7663b8a0c204f1f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 8 Nov 2013 14:01:39 +0000
Subject: ASoC: wm8974: Convert to direct regmap API usage

Moves us towards removing the ASoC level I/O functions.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8974.c | 54 +++++++++++++++++++++++++++++------------------
 1 file changed, 33 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index a2d01d10a5dd..15f45c7bd833 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -27,22 +28,22 @@
 
 #include "wm8974.h"
 
-static const u16 wm8974_reg[WM8974_CACHEREGNUM] = {
-	0x0000, 0x0000, 0x0000, 0x0000,
-	0x0050, 0x0000, 0x0140, 0x0000,
-	0x0000, 0x0000, 0x0000, 0x00ff,
-	0x0000, 0x0000, 0x0100, 0x00ff,
-	0x0000, 0x0000, 0x012c, 0x002c,
-	0x002c, 0x002c, 0x002c, 0x0000,
-	0x0032, 0x0000, 0x0000, 0x0000,
-	0x0000, 0x0000, 0x0000, 0x0000,
-	0x0038, 0x000b, 0x0032, 0x0000,
-	0x0008, 0x000c, 0x0093, 0x00e9,
-	0x0000, 0x0000, 0x0000, 0x0000,
-	0x0003, 0x0010, 0x0000, 0x0000,
-	0x0000, 0x0002, 0x0000, 0x0000,
-	0x0000, 0x0000, 0x0039, 0x0000,
-	0x0000,
+static const struct reg_default wm8974_reg_defaults[] = {
+	{  0, 0x0000 }, {  1, 0x0000 }, {  2, 0x0000 }, {  3, 0x0000 },
+	{  4, 0x0050 }, {  5, 0x0000 }, {  6, 0x0140 }, {  7, 0x0000 },
+	{  8, 0x0000 }, {  9, 0x0000 }, { 10, 0x0000 }, { 11, 0x00ff },
+	{ 12, 0x0000 }, { 13, 0x0000 }, { 14, 0x0100 }, { 15, 0x00ff },
+	{ 16, 0x0000 }, { 17, 0x0000 }, { 18, 0x012c }, { 19, 0x002c },
+	{ 20, 0x002c }, { 21, 0x002c }, { 22, 0x002c }, { 23, 0x0000 },
+	{ 24, 0x0032 }, { 25, 0x0000 }, { 26, 0x0000 }, { 27, 0x0000 },
+	{ 28, 0x0000 }, { 29, 0x0000 }, { 30, 0x0000 }, { 31, 0x0000 },
+	{ 32, 0x0038 }, { 33, 0x000b }, { 34, 0x0032 }, { 35, 0x0000 },
+	{ 36, 0x0008 }, { 37, 0x000c }, { 38, 0x0093 }, { 39, 0x00e9 },
+	{ 40, 0x0000 }, { 41, 0x0000 }, { 42, 0x0000 }, { 43, 0x0000 },
+	{ 44, 0x0003 }, { 45, 0x0010 }, { 46, 0x0000 }, { 47, 0x0000 },
+	{ 48, 0x0000 }, { 49, 0x0002 }, { 50, 0x0000 }, { 51, 0x0000 },
+	{ 52, 0x0000 }, { 53, 0x0000 }, { 54, 0x0039 }, { 55, 0x0000 },
+	{ 56, 0x0000 },
 };
 
 #define WM8974_POWER1_BIASEN  0x08
@@ -514,7 +515,7 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
 		power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
 
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			snd_soc_cache_sync(codec);
+			regcache_sync(dev_get_regmap(codec->dev, NULL));
 
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8974_POWER1, power1 | 0x3);
@@ -579,11 +580,20 @@ static int wm8974_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static const struct regmap_config wm8974_regmap = {
+	.reg_bits = 7,
+	.val_bits = 9,
+
+	.max_register = WM8974_MONOMIX,
+	.reg_defaults = wm8974_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults),
+};
+
 static int wm8974_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -613,9 +623,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = {
 	.suspend = 	wm8974_suspend,
 	.resume =	wm8974_resume,
 	.set_bias_level = wm8974_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(wm8974_reg),
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = wm8974_reg,
 
 	.controls = wm8974_snd_controls,
 	.num_controls = ARRAY_SIZE(wm8974_snd_controls),
@@ -628,8 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = {
 static int wm8974_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
+	struct regmap *regmap;
 	int ret;
 
+	regmap = devm_regmap_init_i2c(i2c, &wm8974_regmap);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8974, &wm8974_dai, 1);
 
-- 
cgit v1.2.2


From 6a0773368619644432fbe46f64ded6aa204baab9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 11:16:18 +0000
Subject: ASoC: wm8991: Use a supply to manage input power

Instead of using a fake register and events to manage input power use a
supply to do the job, saving code and preparing for regmap conversion of
the driver.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8991.c | 53 +++++++++++++++--------------------------------
 sound/soc/codecs/wm8991.h |  9 --------
 2 files changed, 17 insertions(+), 45 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3a39df7a3829..5078fc8e10f7 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -374,30 +374,6 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = {
 /*
  * _DAPM_ Controls
  */
-static int inmixer_event(struct snd_soc_dapm_widget *w,
-			 struct snd_kcontrol *kcontrol, int event)
-{
-	u16 reg, fakepower;
-
-	reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2);
-	fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS);
-
-	if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) |
-			 (1 << WM8991_AINLMUX_PWR_BIT)))
-		reg |= WM8991_AINL_ENA;
-	else
-		reg &= ~WM8991_AINL_ENA;
-
-	if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) |
-			 (1 << WM8991_AINRMUX_PWR_BIT)))
-		reg |= WM8991_AINR_ENA;
-	else
-		reg &= ~WM8991_AINR_ENA;
-
-	snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg);
-	return 0;
-}
-
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 			  struct snd_kcontrol *kcontrol, int event)
 {
@@ -655,6 +631,11 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("RIN2"),
 	SND_SOC_DAPM_INPUT("Internal ADC Source"),
 
+	SND_SOC_DAPM_SUPPLY("INL", WM8991_POWER_MANAGEMENT_2,
+			    WM8991_AINL_ENA_BIT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("INR", WM8991_POWER_MANAGEMENT_2,
+			    WM8991_AINR_ENA_BIT, 0, NULL, 0),
+
 	/* DACs */
 	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2,
 		WM8991_ADCL_ENA_BIT, 0),
@@ -676,26 +657,22 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = {
 		ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)),
 
 	/* INMIXL */
-	SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0,
+	SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0,
 		&wm8991_dapm_inmixl_controls[0],
-		ARRAY_SIZE(wm8991_dapm_inmixl_controls),
-		inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+		ARRAY_SIZE(wm8991_dapm_inmixl_controls)),
 
 	/* AINLMUX */
-	SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0,
-		&wm8991_dapm_ainlmux_controls, inmixer_event,
-		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0,
+		&wm8991_dapm_ainlmux_controls),
 
 	/* INMIXR */
-	SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0,
+	SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0,
 		&wm8991_dapm_inmixr_controls[0],
-		ARRAY_SIZE(wm8991_dapm_inmixr_controls),
-		inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+		ARRAY_SIZE(wm8991_dapm_inmixr_controls)),
 
 	/* AINRMUX */
-	SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0,
-		&wm8991_dapm_ainrmux_controls, inmixer_event,
-		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0,
+		&wm8991_dapm_ainrmux_controls),
 
 	/* Output Side */
 	/* DACs */
@@ -797,6 +774,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Right ADC", NULL, "Internal ADC Source"},
 
 	/* Input Side */
+	{"INMIXL", NULL, "INL"},
+	{"AINLMUX", NULL, "INL"},
+	{"INMIXR", NULL, "INR"},
+	{"AINRMUX", NULL, "INR"},
 	/* LIN12 PGA */
 	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
 	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h
index 07707d8d7e20..08ed383303c0 100644
--- a/sound/soc/codecs/wm8991.h
+++ b/sound/soc/codecs/wm8991.h
@@ -76,7 +76,6 @@
 #define WM8991_PLL1                             0x3C
 #define WM8991_PLL2                             0x3D
 #define WM8991_PLL3                             0x3E
-#define WM8991_INTDRIVBITS			0x3F
 
 #define WM8991_REGISTER_COUNT                   60
 #define WM8991_MAX_REGISTER                     0x3F
@@ -807,14 +806,6 @@
  */
 #define WM8991_PLLK2_MASK                       0x00FF  /* PLLK2 - [7:0] */
 
-/*
- * R63 (0x3F) - Internal Driver Bits
- */
-#define WM8991_INMIXL_PWR_BIT			0
-#define WM8991_AINLMUX_PWR_BIT			1
-#define WM8991_INMIXR_PWR_BIT			2
-#define WM8991_AINRMUX_PWR_BIT			3
-
 #define WM8991_MCLK_DIV 0
 #define WM8991_DACCLK_DIV 1
 #define WM8991_ADCCLK_DIV 2
-- 
cgit v1.2.2


From 898249958281ebf52288eb0d3ed6797e1fc4b0bd Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 12:28:06 +0000
Subject: ASoC: wm8991: Convert to table based control and widget init

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8991.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 5078fc8e10f7..86aa33fd11de 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -764,7 +764,7 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
 };
 
-static const struct snd_soc_dapm_route audio_map[] = {
+static const struct snd_soc_dapm_route wm8991_dapm_routes[] = {
 	/* Make DACs turn on when playing even if not mixed into any outputs */
 	{"Internal DAC Sink", NULL, "Left DAC"},
 	{"Internal DAC Sink", NULL, "Right DAC"},
@@ -1278,13 +1278,6 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
-	snd_soc_add_codec_controls(codec, wm8991_snd_controls,
-			     ARRAY_SIZE(wm8991_snd_controls));
-
-	snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets,
-				  ARRAY_SIZE(wm8991_dapm_widgets));
-	snd_soc_dapm_add_routes(&codec->dapm, audio_map,
-				ARRAY_SIZE(audio_map));
 	return 0;
 }
 
@@ -1333,6 +1326,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = {
 	.suspend = wm8991_suspend,
 	.resume = wm8991_resume,
 	.set_bias_level = wm8991_set_bias_level,
+	.controls = wm8991_snd_controls,
+	.num_controls = ARRAY_SIZE(wm8991_snd_controls),
+	.dapm_widgets = wm8991_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets),
+	.dapm_routes = wm8991_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes),
 	.reg_cache_size = WM8991_MAX_REGISTER + 1,
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8991_reg_defs
-- 
cgit v1.2.2


From a86652e51a8776bc0fe811e32ec3118f03c7e3bb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 13:28:55 +0000
Subject: ASoC: wm8991: Convert to direct regmap API usage

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8991.c | 166 ++++++++++++++++++++++++++--------------------
 1 file changed, 94 insertions(+), 72 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 86aa33fd11de..7006f9732af6 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -18,6 +18,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -31,76 +32,85 @@
 #include "wm8991.h"
 
 struct wm8991_priv {
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 	unsigned int pcmclk;
 };
 
-static const u16 wm8991_reg_defs[] = {
-	0x8991,     /* R0  - Reset */
-	0x0000,     /* R1  - Power Management (1) */
-	0x6000,     /* R2  - Power Management (2) */
-	0x0000,     /* R3  - Power Management (3) */
-	0x4050,     /* R4  - Audio Interface (1) */
-	0x4000,     /* R5  - Audio Interface (2) */
-	0x01C8,     /* R6  - Clocking (1) */
-	0x0000,     /* R7  - Clocking (2) */
-	0x0040,     /* R8  - Audio Interface (3) */
-	0x0040,     /* R9  - Audio Interface (4) */
-	0x0004,     /* R10 - DAC CTRL */
-	0x00C0,     /* R11 - Left DAC Digital Volume */
-	0x00C0,     /* R12 - Right DAC Digital Volume */
-	0x0000,     /* R13 - Digital Side Tone */
-	0x0100,     /* R14 - ADC CTRL */
-	0x00C0,     /* R15 - Left ADC Digital Volume */
-	0x00C0,     /* R16 - Right ADC Digital Volume */
-	0x0000,     /* R17 */
-	0x0000,     /* R18 - GPIO CTRL 1 */
-	0x1000,     /* R19 - GPIO1 & GPIO2 */
-	0x1010,     /* R20 - GPIO3 & GPIO4 */
-	0x1010,     /* R21 - GPIO5 & GPIO6 */
-	0x8000,     /* R22 - GPIOCTRL 2 */
-	0x0800,     /* R23 - GPIO_POL */
-	0x008B,     /* R24 - Left Line Input 1&2 Volume */
-	0x008B,     /* R25 - Left Line Input 3&4 Volume */
-	0x008B,     /* R26 - Right Line Input 1&2 Volume */
-	0x008B,     /* R27 - Right Line Input 3&4 Volume */
-	0x0000,     /* R28 - Left Output Volume */
-	0x0000,     /* R29 - Right Output Volume */
-	0x0066,     /* R30 - Line Outputs Volume */
-	0x0022,     /* R31 - Out3/4 Volume */
-	0x0079,     /* R32 - Left OPGA Volume */
-	0x0079,     /* R33 - Right OPGA Volume */
-	0x0003,     /* R34 - Speaker Volume */
-	0x0003,     /* R35 - ClassD1 */
-	0x0000,     /* R36 */
-	0x0100,     /* R37 - ClassD3 */
-	0x0000,     /* R38 */
-	0x0000,     /* R39 - Input Mixer1 */
-	0x0000,     /* R40 - Input Mixer2 */
-	0x0000,     /* R41 - Input Mixer3 */
-	0x0000,     /* R42 - Input Mixer4 */
-	0x0000,     /* R43 - Input Mixer5 */
-	0x0000,     /* R44 - Input Mixer6 */
-	0x0000,     /* R45 - Output Mixer1 */
-	0x0000,     /* R46 - Output Mixer2 */
-	0x0000,     /* R47 - Output Mixer3 */
-	0x0000,     /* R48 - Output Mixer4 */
-	0x0000,     /* R49 - Output Mixer5 */
-	0x0000,     /* R50 - Output Mixer6 */
-	0x0180,     /* R51 - Out3/4 Mixer */
-	0x0000,     /* R52 - Line Mixer1 */
-	0x0000,     /* R53 - Line Mixer2 */
-	0x0000,     /* R54 - Speaker Mixer */
-	0x0000,     /* R55 - Additional Control */
-	0x0000,     /* R56 - AntiPOP1 */
-	0x0000,     /* R57 - AntiPOP2 */
-	0x0000,     /* R58 - MICBIAS */
-	0x0000,     /* R59 */
-	0x0008,     /* R60 - PLL1 */
-	0x0031,     /* R61 - PLL2 */
-	0x0026,     /* R62 - PLL3 */
+static const struct reg_default wm8991_reg_defaults[] = {
+	{  1, 0x0000 },     /* R1  - Power Management (1) */
+	{  2, 0x6000 },     /* R2  - Power Management (2) */
+	{  3, 0x0000 },     /* R3  - Power Management (3) */
+	{  4, 0x4050 },     /* R4  - Audio Interface (1) */
+	{  5, 0x4000 },     /* R5  - Audio Interface (2) */
+	{  6, 0x01C8 },     /* R6  - Clocking (1) */
+	{  7, 0x0000 },     /* R7  - Clocking (2) */
+	{  8, 0x0040 },     /* R8  - Audio Interface (3) */
+	{  9, 0x0040 },     /* R9  - Audio Interface (4) */
+	{ 10, 0x0004 },     /* R10 - DAC CTRL */
+	{ 11, 0x00C0 },     /* R11 - Left DAC Digital Volume */
+	{ 12, 0x00C0 },     /* R12 - Right DAC Digital Volume */
+	{ 13, 0x0000 },     /* R13 - Digital Side Tone */
+	{ 14, 0x0100 },     /* R14 - ADC CTRL */
+	{ 15, 0x00C0 },     /* R15 - Left ADC Digital Volume */
+	{ 16, 0x00C0 },     /* R16 - Right ADC Digital Volume */
+
+	{ 18, 0x0000 },     /* R18 - GPIO CTRL 1 */
+	{ 19, 0x1000 },     /* R19 - GPIO1 & GPIO2 */
+	{ 20, 0x1010 },     /* R20 - GPIO3 & GPIO4 */
+	{ 21, 0x1010 },     /* R21 - GPIO5 & GPIO6 */
+	{ 22, 0x8000 },     /* R22 - GPIOCTRL 2 */
+	{ 23, 0x0800 },     /* R23 - GPIO_POL */
+	{ 24, 0x008B },     /* R24 - Left Line Input 1&2 Volume */
+	{ 25, 0x008B },     /* R25 - Left Line Input 3&4 Volume */
+	{ 26, 0x008B },     /* R26 - Right Line Input 1&2 Volume */
+	{ 27, 0x008B },     /* R27 - Right Line Input 3&4 Volume */
+	{ 28, 0x0000 },     /* R28 - Left Output Volume */
+	{ 29, 0x0000 },     /* R29 - Right Output Volume */
+	{ 30, 0x0066 },     /* R30 - Line Outputs Volume */
+	{ 31, 0x0022 },     /* R31 - Out3/4 Volume */
+	{ 32, 0x0079 },     /* R32 - Left OPGA Volume */
+	{ 33, 0x0079 },     /* R33 - Right OPGA Volume */
+	{ 34, 0x0003 },     /* R34 - Speaker Volume */
+	{ 35, 0x0003 },     /* R35 - ClassD1 */
+
+	{ 37, 0x0100 },     /* R37 - ClassD3 */
+
+	{ 39, 0x0000 },     /* R39 - Input Mixer1 */
+	{ 40, 0x0000 },     /* R40 - Input Mixer2 */
+	{ 41, 0x0000 },     /* R41 - Input Mixer3 */
+	{ 42, 0x0000 },     /* R42 - Input Mixer4 */
+	{ 43, 0x0000 },     /* R43 - Input Mixer5 */
+	{ 44, 0x0000 },     /* R44 - Input Mixer6 */
+	{ 45, 0x0000 },     /* R45 - Output Mixer1 */
+	{ 46, 0x0000 },     /* R46 - Output Mixer2 */
+	{ 47, 0x0000 },     /* R47 - Output Mixer3 */
+	{ 48, 0x0000 },     /* R48 - Output Mixer4 */
+	{ 49, 0x0000 },     /* R49 - Output Mixer5 */
+	{ 50, 0x0000 },     /* R50 - Output Mixer6 */
+	{ 51, 0x0180 },     /* R51 - Out3/4 Mixer */
+	{ 52, 0x0000 },     /* R52 - Line Mixer1 */
+	{ 53, 0x0000 },     /* R53 - Line Mixer2 */
+	{ 54, 0x0000 },     /* R54 - Speaker Mixer */
+	{ 55, 0x0000 },     /* R55 - Additional Control */
+	{ 56, 0x0000 },     /* R56 - AntiPOP1 */
+	{ 57, 0x0000 },     /* R57 - AntiPOP2 */
+	{ 58, 0x0000 },     /* R58 - MICBIAS */
+
+	{ 60, 0x0008 },     /* R60 - PLL1 */
+	{ 61, 0x0031 },     /* R61 - PLL2 */
+	{ 62, 0x0026 },     /* R62 - PLL3 */
 };
 
+static bool wm8991_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case WM8991_RESET:
+		return true;
+	default:
+		return false;
+	}
+}
+
 #define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0)
 
 static const unsigned int rec_mix_tlv[] = {
@@ -1110,6 +1120,7 @@ static int wm8991_mute(struct snd_soc_dai *dai, int mute)
 static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
+	struct wm8991_priv *wm8991 = snd_soc_codec_get_drvdata(codec);
 	u16 val;
 
 	switch (level) {
@@ -1125,7 +1136,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			snd_soc_cache_sync(codec);
+			regcache_sync(wm8991->regmap);
 			/* Enable all output discharge bits */
 			snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE |
 				      WM8991_DIS_RLINE | WM8991_DIS_OUT3 |
@@ -1213,7 +1224,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 
 		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 		snd_soc_write(codec, WM8991_ANTIPOP2, 0x0);
-		codec->cache_sync = 1;
+		regcache_mark_dirty(wm8991->regmap);
 		break;
 	}
 
@@ -1247,7 +1258,7 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 
 	wm8991 = snd_soc_codec_get_drvdata(codec);
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
 		return ret;
@@ -1332,9 +1343,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = {
 	.num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets),
 	.dapm_routes = wm8991_dapm_routes,
 	.num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes),
-	.reg_cache_size = WM8991_MAX_REGISTER + 1,
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = wm8991_reg_defs
+};
+
+static const struct regmap_config wm8991_regmap = {
+	.reg_bits = 8,
+	.val_bits = 16,
+
+	.max_register = WM8991_PLL3,
+	.volatile_reg = wm8991_volatile,
+	.reg_defaults = wm8991_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(wm8991_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
 };
 
 static int wm8991_i2c_probe(struct i2c_client *i2c,
@@ -1347,7 +1366,10 @@ static int wm8991_i2c_probe(struct i2c_client *i2c,
 	if (!wm8991)
 		return -ENOMEM;
 
-	wm8991->control_type = SND_SOC_I2C;
+	wm8991->regmap = devm_regmap_init_i2c(i2c, &wm8991_regmap);
+	if (IS_ERR(wm8991->regmap))
+		return PTR_ERR(wm8991->regmap);
+
 	i2c_set_clientdata(i2c, wm8991);
 
 	ret = snd_soc_register_codec(&i2c->dev,
-- 
cgit v1.2.2


From e4634804cacce6fe1ec34d92786f764fcb75cb97 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 13:32:36 +0000
Subject: ASoC: wm8991: Move basic initialisation to I2C level probe

This is better practice, though some of this stuff ought not to be here at
all.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8991.c | 50 +++++++++++++++++++++++------------------------
 1 file changed, 25 insertions(+), 25 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 7006f9732af6..5fdcf788e990 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -111,8 +111,6 @@ static bool wm8991_volatile(struct device *dev, unsigned int reg)
 	}
 }
 
-#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0)
-
 static const unsigned int rec_mix_tlv[] = {
 	TLV_DB_RANGE_HEAD(1),
 	0, 7, TLV_DB_LINEAR_ITEM(-1500, 600),
@@ -1264,31 +1262,8 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	ret = wm8991_reset(codec);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to issue reset\n");
-		return ret;
-	}
-
 	wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4,
-			    WM8991_ALRCGPIO1, WM8991_ALRCGPIO1);
-
-	snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2,
-			    WM8991_GPIO1_SEL_MASK, 1);
-
-	snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1,
-			    WM8991_VREF_ENA | WM8991_VMID_MODE_MASK,
-			    WM8991_VREF_ENA | WM8991_VMID_MODE_MASK);
-
-	snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2,
-			    WM8991_OPCLK_ENA, WM8991_OPCLK_ENA);
-
-	snd_soc_write(codec, WM8991_DAC_CTRL, 0);
-	snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
-	snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
-
 	return 0;
 }
 
@@ -1372,6 +1347,31 @@ static int wm8991_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, wm8991);
 
+	ret = regmap_write(wm8991->regmap, WM8991_RESET, 0);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+		return ret;
+	}
+
+	regmap_update_bits(wm8991->regmap, WM8991_AUDIO_INTERFACE_4,
+			   WM8991_ALRCGPIO1, WM8991_ALRCGPIO1);
+
+	regmap_update_bits(wm8991->regmap, WM8991_GPIO1_GPIO2,
+			   WM8991_GPIO1_SEL_MASK, 1);
+
+	regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_1,
+			   WM8991_VREF_ENA | WM8991_VMID_MODE_MASK,
+			   WM8991_VREF_ENA | WM8991_VMID_MODE_MASK);
+
+	regmap_update_bits(wm8991->regmap, WM8991_POWER_MANAGEMENT_2,
+			   WM8991_OPCLK_ENA, WM8991_OPCLK_ENA);
+
+	regmap_write(wm8991->regmap, WM8991_DAC_CTRL, 0);
+	regmap_write(wm8991->regmap, WM8991_LEFT_OUTPUT_VOLUME,
+		     0x50 | (1<<8));
+	regmap_write(wm8991->regmap, WM8991_RIGHT_OUTPUT_VOLUME,
+		     0x50 | (1<<8));
+
 	ret = snd_soc_register_codec(&i2c->dev,
 				     &soc_codec_dev_wm8991, &wm8991_dai, 1);
 
-- 
cgit v1.2.2


From a0a05916cf67a007f4ee0071fd0fa04e45137a38 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 13:34:48 +0000
Subject: ASoC: wm8991: Verify device ID during probe()

Just in case.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8991.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 5fdcf788e990..dba0306c42a5 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1335,6 +1335,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct wm8991_priv *wm8991;
+	unsigned int val;
 	int ret;
 
 	wm8991 = devm_kzalloc(&i2c->dev, sizeof(*wm8991), GFP_KERNEL);
@@ -1347,6 +1348,16 @@ static int wm8991_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, wm8991);
 
+	ret = regmap_read(wm8991->regmap, WM8991_RESET, &val);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to read device ID: %d\n", ret);
+		return ret;
+	}
+	if (val != 0x8991) {
+		dev_err(&i2c->dev, "Device with ID %x is not a WM8991\n", val);
+		return -EINVAL;
+	}
+
 	ret = regmap_write(wm8991->regmap, WM8991_RESET, 0);
 	if (ret < 0) {
 		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
-- 
cgit v1.2.2


From 4f534777c130180f9338f0fb96090d43464b7ddf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 15:14:48 +0000
Subject: ASoC: ak4641: Convert to direct regmap API usage

We're trying to remove the ASoC level I/O functions.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ak4641.c | 38 +++++++++++++++++++++++++-------------
 1 file changed, 25 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 49cc5f6d6dba..94cbe508dd37 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -17,6 +17,7 @@
 #include <linux/gpio.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -30,6 +31,7 @@
 
 /* codec private data */
 struct ak4641_priv {
+	struct regmap *regmap;
 	unsigned int sysclk;
 	int deemph;
 	int playback_fs;
@@ -38,12 +40,12 @@ struct ak4641_priv {
 /*
  * ak4641 register cache
  */
-static const u8 ak4641_reg[AK4641_CACHEREGNUM] = {
-	0x00, 0x80, 0x00, 0x80,
-	0x02, 0x00, 0x11, 0x05,
-	0x00, 0x00, 0x36, 0x10,
-	0x00, 0x00, 0x57, 0x00,
-	0x88, 0x88, 0x08, 0x08
+static const struct reg_default ak4641_reg_defaults[] = {
+	{  0, 0x00 }, {  1, 0x80 }, {  2, 0x00 }, {  3, 0x80 },
+	{  4, 0x02 }, {  5, 0x00 }, {  6, 0x11 }, {  7, 0x05 },
+	{  8, 0x00 }, {  9, 0x00 }, { 10, 0x36 }, { 11, 0x10 },
+	{ 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 },
+	{ 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 }
 };
 
 static const int deemph_settings[] = {44100, 0, 48000, 32000};
@@ -396,6 +398,7 @@ static int ak4641_mute(struct snd_soc_dai *dai, int mute)
 static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
 	struct ak4641_platform_data *pdata = codec->dev->platform_data;
 	int ret;
 
@@ -417,7 +420,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 				gpio_set_value(pdata->gpio_npdn, 1);
 			mdelay(1);
 
-			ret = snd_soc_cache_sync(codec);
+			ret = regcache_sync(ak4641->regmap);
 			if (ret) {
 				dev_err(codec->dev,
 					"Failed to sync cache: %d\n", ret);
@@ -433,7 +436,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 			gpio_set_value(pdata->gpio_npdn, 0);
 		if (pdata && gpio_is_valid(pdata->gpio_power))
 			gpio_set_value(pdata->gpio_power, 0);
-		codec->cache_sync = 1;
+		regcache_mark_dirty(ak4641->regmap);
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -518,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -550,12 +553,17 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
 	.dapm_routes		= ak4641_audio_map,
 	.num_dapm_routes	= ARRAY_SIZE(ak4641_audio_map),
 	.set_bias_level		= ak4641_set_bias_level,
-	.reg_cache_size		= ARRAY_SIZE(ak4641_reg),
-	.reg_word_size		= sizeof(u8),
-	.reg_cache_default	= ak4641_reg,
-	.reg_cache_step		= 1,
 };
 
+static const struct regmap_config ak4641_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = AK4641_BTIF,
+	.reg_defaults = ak4641_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
 
 static int ak4641_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
@@ -569,6 +577,10 @@ static int ak4641_i2c_probe(struct i2c_client *i2c,
 	if (!ak4641)
 		return -ENOMEM;
 
+	ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap);
+	if (IS_ERR(ak4641->regmap))
+		return PTR_ERR(ak4641->regmap);
+
 	if (pdata) {
 		if (gpio_is_valid(pdata->gpio_power)) {
 			ret = gpio_request_one(pdata->gpio_power,
-- 
cgit v1.2.2


From 9a3e1b8c503590a2a89164586c0e65415bb90979 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 19 Nov 2013 16:04:01 +0000
Subject: ASoC: wm5110: Hook up ADSP2 cores

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/Kconfig  |   2 +
 sound/soc/codecs/wm5110.c | 119 +++++++++++++++++++++++++++++++++++++++++++++-
 2 files changed, 119 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b33b45dfceec..983d087aa92a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -163,8 +163,10 @@ config SND_SOC_WM_HUBS
 config SND_SOC_WM_ADSP
 	tristate
 	default y if SND_SOC_WM5102=y
+	default y if SND_SOC_WM5110=y
 	default y if SND_SOC_WM2200=y
 	default m if SND_SOC_WM5102=m
+	default m if SND_SOC_WM5110=m
 	default m if SND_SOC_WM2200=m
 
 config SND_SOC_AB8500_CODEC
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c6dbc1ddaf7f..e5f2804086a1 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -30,13 +30,51 @@
 #include <linux/mfd/arizona/registers.h>
 
 #include "arizona.h"
+#include "wm_adsp.h"
 #include "wm5110.h"
 
+#define WM5110_NUM_ADSP 4
+
 struct wm5110_priv {
 	struct arizona_priv core;
 	struct arizona_fll fll[2];
 };
 
+static const struct wm_adsp_region wm5110_dsp1_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x100000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x180000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x190000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x1a8000 },
+};
+
+static const struct wm_adsp_region wm5110_dsp2_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x200000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x280000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x290000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x2a8000 },
+};
+
+static const struct wm_adsp_region wm5110_dsp3_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x300000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x380000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x390000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x3a8000 },
+};
+
+static const struct wm_adsp_region wm5110_dsp4_regions[] = {
+	{ .type = WMFW_ADSP2_PM, .base = 0x400000 },
+	{ .type = WMFW_ADSP2_ZM, .base = 0x480000 },
+	{ .type = WMFW_ADSP2_XM, .base = 0x490000 },
+	{ .type = WMFW_ADSP2_YM, .base = 0x4a8000 },
+};
+
+static const struct wm_adsp_region *wm5110_dsp_regions[] = {
+	wm5110_dsp1_regions,
+	wm5110_dsp2_regions,
+	wm5110_dsp3_regions,
+	wm5110_dsp4_regions,
+};
+
 static const struct reg_default wm5110_sysclk_revd_patch[] = {
 	{ 0x3093, 0x1001 },
 	{ 0x30E3, 0x1301 },
@@ -395,6 +433,22 @@ ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
 
+ARIZONA_MIXER_ENUMS(DSP1L, ARIZONA_DSP1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DSP1R, ARIZONA_DSP1RMIX_INPUT_1_SOURCE);
+ARIZONA_DSP_AUX_ENUMS(DSP1, ARIZONA_DSP1AUX1MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DSP2L, ARIZONA_DSP2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DSP2R, ARIZONA_DSP2RMIX_INPUT_1_SOURCE);
+ARIZONA_DSP_AUX_ENUMS(DSP2, ARIZONA_DSP2AUX1MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DSP3L, ARIZONA_DSP3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DSP3R, ARIZONA_DSP3RMIX_INPUT_1_SOURCE);
+ARIZONA_DSP_AUX_ENUMS(DSP3, ARIZONA_DSP3AUX1MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DSP4L, ARIZONA_DSP4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DSP4R, ARIZONA_DSP4RMIX_INPUT_1_SOURCE);
+ARIZONA_DSP_AUX_ENUMS(DSP4, ARIZONA_DSP4AUX1MIX_INPUT_1_SOURCE);
+
 ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
 
@@ -587,6 +641,11 @@ SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
 SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
 		 NULL, 0),
 
+WM_ADSP2("DSP1", 0),
+WM_ADSP2("DSP2", 1),
+WM_ADSP2("DSP3", 2),
+WM_ADSP2("DSP4", 3),
+
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
 		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
 		       &wm5110_aec_loopback_mux),
@@ -809,6 +868,11 @@ ARIZONA_MUX_WIDGETS(ASRC1R, "ASRC1R"),
 ARIZONA_MUX_WIDGETS(ASRC2L, "ASRC2L"),
 ARIZONA_MUX_WIDGETS(ASRC2R, "ASRC2R"),
 
+ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
+ARIZONA_DSP_WIDGETS(DSP2, "DSP2"),
+ARIZONA_DSP_WIDGETS(DSP3, "DSP3"),
+ARIZONA_DSP_WIDGETS(DSP4, "DSP4"),
+
 SND_SOC_DAPM_OUTPUT("HPOUT1L"),
 SND_SOC_DAPM_OUTPUT("HPOUT1R"),
 SND_SOC_DAPM_OUTPUT("HPOUT2L"),
@@ -881,7 +945,31 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"),
 	{ name, "ASRC1L", "ASRC1L" }, \
 	{ name, "ASRC1R", "ASRC1R" }, \
 	{ name, "ASRC2L", "ASRC2L" }, \
-	{ name, "ASRC2R", "ASRC2R" }
+	{ name, "ASRC2R", "ASRC2R" }, \
+	{ name, "DSP1.1", "DSP1" }, \
+	{ name, "DSP1.2", "DSP1" }, \
+	{ name, "DSP1.3", "DSP1" }, \
+	{ name, "DSP1.4", "DSP1" }, \
+	{ name, "DSP1.5", "DSP1" }, \
+	{ name, "DSP1.6", "DSP1" }, \
+	{ name, "DSP2.1", "DSP2" }, \
+	{ name, "DSP2.2", "DSP2" }, \
+	{ name, "DSP2.3", "DSP2" }, \
+	{ name, "DSP2.4", "DSP2" }, \
+	{ name, "DSP2.5", "DSP2" }, \
+	{ name, "DSP2.6", "DSP2" }, \
+	{ name, "DSP3.1", "DSP3" }, \
+	{ name, "DSP3.2", "DSP3" }, \
+	{ name, "DSP3.3", "DSP3" }, \
+	{ name, "DSP3.4", "DSP3" }, \
+	{ name, "DSP3.5", "DSP3" }, \
+	{ name, "DSP3.6", "DSP3" }, \
+	{ name, "DSP4.1", "DSP4" }, \
+	{ name, "DSP4.2", "DSP4" }, \
+	{ name, "DSP4.3", "DSP4" }, \
+	{ name, "DSP4.4", "DSP4" }, \
+	{ name, "DSP4.5", "DSP4" }, \
+	{ name, "DSP4.6", "DSP4" }
 
 static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "DBVDD2" },
@@ -1087,6 +1175,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"),
 	ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"),
 
+	ARIZONA_DSP_ROUTES("DSP1"),
+	ARIZONA_DSP_ROUTES("DSP2"),
+	ARIZONA_DSP_ROUTES("DSP3"),
+	ARIZONA_DSP_ROUTES("DSP4"),
+
 	{ "AEC Loopback", "HPOUT1L", "OUT1L" },
 	{ "AEC Loopback", "HPOUT1R", "OUT1R" },
 	{ "HPOUT1L", NULL, "OUT1L" },
@@ -1292,6 +1385,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	arizona_init_spk(codec);
 	arizona_init_gpio(codec);
 
+	ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8);
+	if (ret != 0)
+		return ret;
+
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
 	priv->core.arizona->dapm = &codec->dapm;
@@ -1346,7 +1443,7 @@ static int wm5110_probe(struct platform_device *pdev)
 {
 	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
 	struct wm5110_priv *wm5110;
-	int i;
+	int i, ret;
 
 	wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv),
 			      GFP_KERNEL);
@@ -1357,6 +1454,24 @@ static int wm5110_probe(struct platform_device *pdev)
 	wm5110->core.arizona = arizona;
 	wm5110->core.num_inputs = 8;
 
+	for (i = 0; i < WM5110_NUM_ADSP; i++) {
+		wm5110->core.adsp[i].part = "wm5110";
+		wm5110->core.adsp[i].num = i + 1;
+		wm5110->core.adsp[i].type = WMFW_ADSP2;
+		wm5110->core.adsp[i].dev = arizona->dev;
+		wm5110->core.adsp[i].regmap = arizona->regmap;
+
+		wm5110->core.adsp[i].base = ARIZONA_DSP1_CONTROL_1
+			+ (0x100 * i);
+		wm5110->core.adsp[i].mem = wm5110_dsp_regions[i];
+		wm5110->core.adsp[i].num_mems
+			= ARRAY_SIZE(wm5110_dsp1_regions);
+
+		ret = wm_adsp2_init(&wm5110->core.adsp[i], false);
+		if (ret != 0)
+			return ret;
+	}
+
 	for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
 		wm5110->fll[i].vco_mult = 3;
 
-- 
cgit v1.2.2


From 0da2e5baf4233e2744a7fc691932638c39d9b245 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Tue, 19 Nov 2013 16:04:02 +0000
Subject: ASoC: wm5110: Add basic support for ISRCs

Add support for the ISRCs that matches the current support on the w5102.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm5110.c | 178 ++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 178 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index e5f2804086a1..0e63d8ce533f 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -277,6 +277,10 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
 SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
 SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
 
+SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
+SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
+SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]),
+
 ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP2L", ARIZONA_DSP2LMIX_INPUT_1_SOURCE),
@@ -501,6 +505,36 @@ ARIZONA_MUX_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
 ARIZONA_MUX_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
 ARIZONA_MUX_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
 
+ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT3, ARIZONA_ISRC1INT3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1INT4, ARIZONA_ISRC1INT4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC3, ARIZONA_ISRC1DEC3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC1DEC4, ARIZONA_ISRC1DEC4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT3, ARIZONA_ISRC2INT3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2INT4, ARIZONA_ISRC2INT4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC3, ARIZONA_ISRC2DEC3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC2DEC4, ARIZONA_ISRC2DEC4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC3INT1, ARIZONA_ISRC3INT1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3INT2, ARIZONA_ISRC3INT2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3INT3, ARIZONA_ISRC3INT3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3INT4, ARIZONA_ISRC3INT4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MUX_ENUMS(ISRC3DEC1, ARIZONA_ISRC3DEC1MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3DEC2, ARIZONA_ISRC3DEC2MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3DEC3, ARIZONA_ISRC3DEC3MIX_INPUT_1_SOURCE);
+ARIZONA_MUX_ENUMS(ISRC3DEC4, ARIZONA_ISRC3DEC4MIX_INPUT_1_SOURCE);
+
 static const char *wm5110_aec_loopback_texts[] = {
 	"HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "HPOUT3L", "HPOUT3R",
 	"SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", "SPKDAT2L", "SPKDAT2R",
@@ -646,6 +680,60 @@ WM_ADSP2("DSP2", 1),
 WM_ADSP2("DSP3", 2),
 WM_ADSP2("DSP4", 3),
 
+SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT3", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1INT4", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC3", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC1DEC4", ARIZONA_ISRC_1_CTRL_3,
+		 ARIZONA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT3", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2INT4", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC3", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC2DEC4", ARIZONA_ISRC_2_CTRL_3,
+		 ARIZONA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC3INT1", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_INT0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3INT2", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_INT1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3INT3", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_INT2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3INT4", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_INT3_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ISRC3DEC1", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_DEC0_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3DEC2", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_DEC1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3DEC3", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_DEC2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ISRC3DEC4", ARIZONA_ISRC_3_CTRL_3,
+		 ARIZONA_ISRC3_DEC3_ENA_SHIFT, 0, NULL, 0),
+
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
 		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
 		       &wm5110_aec_loopback_mux),
@@ -873,6 +961,36 @@ ARIZONA_DSP_WIDGETS(DSP2, "DSP2"),
 ARIZONA_DSP_WIDGETS(DSP3, "DSP3"),
 ARIZONA_DSP_WIDGETS(DSP4, "DSP4"),
 
+ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"),
+ARIZONA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"),
+
+ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"),
+ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"),
+ARIZONA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"),
+ARIZONA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"),
+
+ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"),
+ARIZONA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"),
+
+ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"),
+ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"),
+ARIZONA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"),
+ARIZONA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"),
+
+ARIZONA_MUX_WIDGETS(ISRC3DEC1, "ISRC3DEC1"),
+ARIZONA_MUX_WIDGETS(ISRC3DEC2, "ISRC3DEC2"),
+ARIZONA_MUX_WIDGETS(ISRC3DEC3, "ISRC3DEC3"),
+ARIZONA_MUX_WIDGETS(ISRC3DEC4, "ISRC3DEC4"),
+
+ARIZONA_MUX_WIDGETS(ISRC3INT1, "ISRC3INT1"),
+ARIZONA_MUX_WIDGETS(ISRC3INT2, "ISRC3INT2"),
+ARIZONA_MUX_WIDGETS(ISRC3INT3, "ISRC3INT3"),
+ARIZONA_MUX_WIDGETS(ISRC3INT4, "ISRC3INT4"),
+
 SND_SOC_DAPM_OUTPUT("HPOUT1L"),
 SND_SOC_DAPM_OUTPUT("HPOUT1R"),
 SND_SOC_DAPM_OUTPUT("HPOUT2L"),
@@ -946,6 +1064,30 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"),
 	{ name, "ASRC1R", "ASRC1R" }, \
 	{ name, "ASRC2L", "ASRC2L" }, \
 	{ name, "ASRC2R", "ASRC2R" }, \
+	{ name, "ISRC1DEC1", "ISRC1DEC1" }, \
+	{ name, "ISRC1DEC2", "ISRC1DEC2" }, \
+	{ name, "ISRC1DEC3", "ISRC1DEC3" }, \
+	{ name, "ISRC1DEC4", "ISRC1DEC4" }, \
+	{ name, "ISRC1INT1", "ISRC1INT1" }, \
+	{ name, "ISRC1INT2", "ISRC1INT2" }, \
+	{ name, "ISRC1INT3", "ISRC1INT3" }, \
+	{ name, "ISRC1INT4", "ISRC1INT4" }, \
+	{ name, "ISRC2DEC1", "ISRC2DEC1" }, \
+	{ name, "ISRC2DEC2", "ISRC2DEC2" }, \
+	{ name, "ISRC2DEC3", "ISRC2DEC3" }, \
+	{ name, "ISRC2DEC4", "ISRC2DEC4" }, \
+	{ name, "ISRC2INT1", "ISRC2INT1" }, \
+	{ name, "ISRC2INT2", "ISRC2INT2" }, \
+	{ name, "ISRC2INT3", "ISRC2INT3" }, \
+	{ name, "ISRC2INT4", "ISRC2INT4" }, \
+	{ name, "ISRC3DEC1", "ISRC3DEC1" }, \
+	{ name, "ISRC3DEC2", "ISRC3DEC2" }, \
+	{ name, "ISRC3DEC3", "ISRC3DEC3" }, \
+	{ name, "ISRC3DEC4", "ISRC3DEC4" }, \
+	{ name, "ISRC3INT1", "ISRC3INT1" }, \
+	{ name, "ISRC3INT2", "ISRC3INT2" }, \
+	{ name, "ISRC3INT3", "ISRC3INT3" }, \
+	{ name, "ISRC3INT4", "ISRC3INT4" }, \
 	{ name, "DSP1.1", "DSP1" }, \
 	{ name, "DSP1.2", "DSP1" }, \
 	{ name, "DSP1.3", "DSP1" }, \
@@ -1180,6 +1322,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	ARIZONA_DSP_ROUTES("DSP3"),
 	ARIZONA_DSP_ROUTES("DSP4"),
 
+	ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"),
+	ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"),
+	ARIZONA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"),
+	ARIZONA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"),
+
+	ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"),
+	ARIZONA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"),
+
+	ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"),
+	ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"),
+	ARIZONA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"),
+	ARIZONA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"),
+
+	ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"),
+	ARIZONA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"),
+
+	ARIZONA_MUX_ROUTES("ISRC3INT1", "ISRC3INT1"),
+	ARIZONA_MUX_ROUTES("ISRC3INT2", "ISRC3INT2"),
+	ARIZONA_MUX_ROUTES("ISRC3INT3", "ISRC3INT3"),
+	ARIZONA_MUX_ROUTES("ISRC3INT4", "ISRC3INT4"),
+
+	ARIZONA_MUX_ROUTES("ISRC3DEC1", "ISRC3DEC1"),
+	ARIZONA_MUX_ROUTES("ISRC3DEC2", "ISRC3DEC2"),
+	ARIZONA_MUX_ROUTES("ISRC3DEC3", "ISRC3DEC3"),
+	ARIZONA_MUX_ROUTES("ISRC3DEC4", "ISRC3DEC4"),
+
 	{ "AEC Loopback", "HPOUT1L", "OUT1L" },
 	{ "AEC Loopback", "HPOUT1R", "OUT1R" },
 	{ "HPOUT1L", NULL, "OUT1L" },
@@ -1482,6 +1654,12 @@ static int wm5110_probe(struct platform_device *pdev)
 			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
 			 &wm5110->fll[1]);
 
+	/* SR2 fixed at 8kHz, SR3 fixed at 16kHz */
+	regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2,
+			   ARIZONA_SAMPLE_RATE_2_MASK, 0x11);
+	regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3,
+			   ARIZONA_SAMPLE_RATE_3_MASK, 0x12);
+
 	for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++)
 		arizona_init_dai(&wm5110->core, i);
 
-- 
cgit v1.2.2


From 93818c9a12dd38f2b32f960f979815ac2e15a176 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 13:42:51 +0000
Subject: ASoC: wm8990: Convet to module_i2c_driver()

The device is I2C only (or at least current support is).

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8990.c | 24 +-----------------------
 1 file changed, 1 insertion(+), 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 4f05fb88bddf..33bec5672418 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1378,7 +1378,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
 	.volatile_register = wm8990_volatile_register,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8990_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1420,29 +1419,8 @@ static struct i2c_driver wm8990_i2c_driver = {
 	.remove =   wm8990_i2c_remove,
 	.id_table = wm8990_i2c_id,
 };
-#endif
 
-static int __init wm8990_modinit(void)
-{
-	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	ret = i2c_add_driver(&wm8990_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register wm8990 I2C driver: %d\n",
-		       ret);
-	}
-#endif
-	return ret;
-}
-module_init(wm8990_modinit);
-
-static void __exit wm8990_exit(void)
-{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_del_driver(&wm8990_i2c_driver);
-#endif
-}
-module_exit(wm8990_exit);
+module_i2c_driver(wm8990_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC WM8990 driver");
 MODULE_AUTHOR("Liam Girdwood");
-- 
cgit v1.2.2


From f6b415b6065041c0970426cc8ac81a980b2998f2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 13:44:56 +0000
Subject: ASoC: wm8990: Convert to table based control and DAPM init

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8990.c | 24 +++++++-----------------
 1 file changed, 7 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 33bec5672418..6ee1cf1d353d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -787,7 +787,7 @@ SND_SOC_DAPM_OUTPUT("RON"),
 SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
 };
 
-static const struct snd_soc_dapm_route audio_map[] = {
+static const struct snd_soc_dapm_route wm8990_dapm_routes[] = {
 	/* Make DACs turn on when playing even if not mixed into any outputs */
 	{"Internal DAC Sink", NULL, "Left DAC"},
 	{"Internal DAC Sink", NULL, "Right DAC"},
@@ -912,18 +912,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"RON", NULL, "RONMIX"},
 };
 
-static int wm8990_add_widgets(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, wm8990_dapm_widgets,
-				  ARRAY_SIZE(wm8990_dapm_widgets));
-	/* set up the WM8990 audio map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-	return 0;
-}
-
 /* PLL divisors */
 struct _pll_div {
 	u32 div2;
@@ -1352,10 +1340,6 @@ static int wm8990_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
-	snd_soc_add_codec_controls(codec, wm8990_snd_controls,
-				ARRAY_SIZE(wm8990_snd_controls));
-	wm8990_add_widgets(codec);
-
 	return 0;
 }
 
@@ -1376,6 +1360,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8990_reg,
 	.volatile_register = wm8990_volatile_register,
+	.controls =	wm8990_snd_controls,
+	.num_controls = ARRAY_SIZE(wm8990_snd_controls),
+	.dapm_widgets = wm8990_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8990_dapm_widgets),
+	.dapm_routes =	wm8990_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes),
 };
 
 static int wm8990_i2c_probe(struct i2c_client *i2c,
-- 
cgit v1.2.2


From d2fd5fe7ee3bc231e21aeb9ee120e0e61a79f8be Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 14:25:04 +0000
Subject: ASoC: wm8990: Use supplies to manage input power

Instead of using a fake register use a supply widget to manage the power
for the inputs, this is more idiomatic and supports regmap conversion.

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8990.c | 54 ++++++++++++++---------------------------------
 sound/soc/codecs/wm8990.h |  9 --------
 2 files changed, 16 insertions(+), 47 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 6ee1cf1d353d..2261fe1b68ce 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -376,32 +376,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
  * _DAPM_ Controls
  */
 
-static int inmixer_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	u16 reg, fakepower;
-
-	reg = snd_soc_read(w->codec, WM8990_POWER_MANAGEMENT_2);
-	fakepower = snd_soc_read(w->codec, WM8990_INTDRIVBITS);
-
-	if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) |
-		(1 << WM8990_AINLMUX_PWR_BIT))) {
-		reg |= WM8990_AINL_ENA;
-	} else {
-		reg &= ~WM8990_AINL_ENA;
-	}
-
-	if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) |
-		(1 << WM8990_AINRMUX_PWR_BIT))) {
-		reg |= WM8990_AINR_ENA;
-	} else {
-		reg &= ~WM8990_AINR_ENA;
-	}
-	snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
-
-	return 0;
-}
-
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -656,6 +630,11 @@ SND_SOC_DAPM_INPUT("RIN1"),
 SND_SOC_DAPM_INPUT("RIN2"),
 SND_SOC_DAPM_INPUT("Internal ADC Source"),
 
+SND_SOC_DAPM_SUPPLY("INL", WM8990_POWER_MANAGEMENT_2, WM8990_AINL_ENA_BIT, 0,
+		    NULL, 0),
+SND_SOC_DAPM_SUPPLY("INR", WM8990_POWER_MANAGEMENT_2, WM8990_AINR_ENA_BIT, 0,
+		    NULL, 0),
+
 /* DACs */
 SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2,
 	WM8990_ADCL_ENA_BIT, 0),
@@ -677,26 +656,20 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT,
 	ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)),
 
 /* INMIXL */
-SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
+SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0,
 	&wm8990_dapm_inmixl_controls[0],
-	ARRAY_SIZE(wm8990_dapm_inmixl_controls),
-	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	ARRAY_SIZE(wm8990_dapm_inmixl_controls)),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
-	&wm8990_dapm_ainlmux_controls, inmixer_event,
-	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainlmux_controls),
 
 /* INMIXR */
-SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
+SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0,
 	&wm8990_dapm_inmixr_controls[0],
-	ARRAY_SIZE(wm8990_dapm_inmixr_controls),
-	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	ARRAY_SIZE(wm8990_dapm_inmixr_controls)),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
-	&wm8990_dapm_ainrmux_controls, inmixer_event,
-	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, &wm8990_dapm_ainrmux_controls),
 
 /* Output Side */
 /* DACs */
@@ -796,6 +769,11 @@ static const struct snd_soc_dapm_route wm8990_dapm_routes[] = {
 	{"Left ADC", NULL, "Internal ADC Source"},
 	{"Right ADC", NULL, "Internal ADC Source"},
 
+	{"AINLMUX", NULL, "INL"},
+	{"INMIXL", NULL, "INL"},
+	{"AINRMUX", NULL, "INR"},
+	{"INMIXR", NULL, "INR"},
+
 	/* Input Side */
 	/* LIN12 PGA */
 	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 77c98a4bfe9c..0e9c78040c4c 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -78,7 +78,6 @@
 #define WM8990_PLL1                             0x3C
 #define WM8990_PLL2                             0x3D
 #define WM8990_PLL3                             0x3E
-#define WM8990_INTDRIVBITS			0x3F
 
 #define WM8990_EXT_ACCESS_ENA			0x75
 #define WM8990_EXT_CTL1				0x7a
@@ -818,14 +817,6 @@
  */
 #define WM8990_PLLK2_MASK                       0x00FF  /* PLLK2 - [7:0] */
 
-/*
- * R63 (0x3F) - Internal Driver Bits
- */
-#define WM8990_INMIXL_PWR_BIT			0
-#define WM8990_AINLMUX_PWR_BIT			1
-#define WM8990_INMIXR_PWR_BIT			2
-#define WM8990_AINRMUX_PWR_BIT			3
-
 #define WM8990_MCLK_DIV 0
 #define WM8990_DACCLK_DIV 1
 #define WM8990_ADCCLK_DIV 2
-- 
cgit v1.2.2


From 0112b62b12e18b883e1027689acab8eaa8830bac Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 22 Nov 2013 14:36:23 +0000
Subject: ASoC: wm8990: Convert to direct regmap API usage

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8990.c | 156 ++++++++++++++++++++++++----------------------
 1 file changed, 81 insertions(+), 75 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 2261fe1b68ce..0ccd4d8d043b 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regmap.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -30,13 +31,12 @@
 
 /* codec private data */
 struct wm8990_priv {
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 	unsigned int sysclk;
 	unsigned int pcmclk;
 };
 
-static int wm8990_volatile_register(struct snd_soc_codec *codec,
-				    unsigned int reg)
+static bool wm8990_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
 	case WM8990_RESET:
@@ -46,71 +46,69 @@ static int wm8990_volatile_register(struct snd_soc_codec *codec,
 	}
 }
 
-static const u16 wm8990_reg[] = {
-	0x8990,     /* R0  - Reset */
-	0x0000,     /* R1  - Power Management (1) */
-	0x6000,     /* R2  - Power Management (2) */
-	0x0000,     /* R3  - Power Management (3) */
-	0x4050,     /* R4  - Audio Interface (1) */
-	0x4000,     /* R5  - Audio Interface (2) */
-	0x01C8,     /* R6  - Clocking (1) */
-	0x0000,     /* R7  - Clocking (2) */
-	0x0040,     /* R8  - Audio Interface (3) */
-	0x0040,     /* R9  - Audio Interface (4) */
-	0x0004,     /* R10 - DAC CTRL */
-	0x00C0,     /* R11 - Left DAC Digital Volume */
-	0x00C0,     /* R12 - Right DAC Digital Volume */
-	0x0000,     /* R13 - Digital Side Tone */
-	0x0100,     /* R14 - ADC CTRL */
-	0x00C0,     /* R15 - Left ADC Digital Volume */
-	0x00C0,     /* R16 - Right ADC Digital Volume */
-	0x0000,     /* R17 */
-	0x0000,     /* R18 - GPIO CTRL 1 */
-	0x1000,     /* R19 - GPIO1 & GPIO2 */
-	0x1010,     /* R20 - GPIO3 & GPIO4 */
-	0x1010,     /* R21 - GPIO5 & GPIO6 */
-	0x8000,     /* R22 - GPIOCTRL 2 */
-	0x0800,     /* R23 - GPIO_POL */
-	0x008B,     /* R24 - Left Line Input 1&2 Volume */
-	0x008B,     /* R25 - Left Line Input 3&4 Volume */
-	0x008B,     /* R26 - Right Line Input 1&2 Volume */
-	0x008B,     /* R27 - Right Line Input 3&4 Volume */
-	0x0000,     /* R28 - Left Output Volume */
-	0x0000,     /* R29 - Right Output Volume */
-	0x0066,     /* R30 - Line Outputs Volume */
-	0x0022,     /* R31 - Out3/4 Volume */
-	0x0079,     /* R32 - Left OPGA Volume */
-	0x0079,     /* R33 - Right OPGA Volume */
-	0x0003,     /* R34 - Speaker Volume */
-	0x0003,     /* R35 - ClassD1 */
-	0x0000,     /* R36 */
-	0x0100,     /* R37 - ClassD3 */
-	0x0079,     /* R38 - ClassD4 */
-	0x0000,     /* R39 - Input Mixer1 */
-	0x0000,     /* R40 - Input Mixer2 */
-	0x0000,     /* R41 - Input Mixer3 */
-	0x0000,     /* R42 - Input Mixer4 */
-	0x0000,     /* R43 - Input Mixer5 */
-	0x0000,     /* R44 - Input Mixer6 */
-	0x0000,     /* R45 - Output Mixer1 */
-	0x0000,     /* R46 - Output Mixer2 */
-	0x0000,     /* R47 - Output Mixer3 */
-	0x0000,     /* R48 - Output Mixer4 */
-	0x0000,     /* R49 - Output Mixer5 */
-	0x0000,     /* R50 - Output Mixer6 */
-	0x0180,     /* R51 - Out3/4 Mixer */
-	0x0000,     /* R52 - Line Mixer1 */
-	0x0000,     /* R53 - Line Mixer2 */
-	0x0000,     /* R54 - Speaker Mixer */
-	0x0000,     /* R55 - Additional Control */
-	0x0000,     /* R56 - AntiPOP1 */
-	0x0000,     /* R57 - AntiPOP2 */
-	0x0000,     /* R58 - MICBIAS */
-	0x0000,     /* R59 */
-	0x0008,     /* R60 - PLL1 */
-	0x0031,     /* R61 - PLL2 */
-	0x0026,     /* R62 - PLL3 */
-	0x0000,	    /* R63 - Driver internal */
+static const struct reg_default wm8990_reg_defaults[] = {
+	{  1, 0x0000 },     /* R1  - Power Management (1) */
+	{  2, 0x6000 },     /* R2  - Power Management (2) */
+	{  3, 0x0000 },     /* R3  - Power Management (3) */
+	{  4, 0x4050 },     /* R4  - Audio Interface (1) */
+	{  5, 0x4000 },     /* R5  - Audio Interface (2) */
+	{  6, 0x01C8 },     /* R6  - Clocking (1) */
+	{  7, 0x0000 },     /* R7  - Clocking (2) */
+	{  8, 0x0040 },     /* R8  - Audio Interface (3) */
+	{  9, 0x0040 },     /* R9  - Audio Interface (4) */
+	{ 10, 0x0004 },     /* R10 - DAC CTRL */
+	{ 11, 0x00C0 },     /* R11 - Left DAC Digital Volume */
+	{ 12, 0x00C0 },     /* R12 - Right DAC Digital Volume */
+	{ 13, 0x0000 },     /* R13 - Digital Side Tone */
+	{ 14, 0x0100 },     /* R14 - ADC CTRL */
+	{ 15, 0x00C0 },     /* R15 - Left ADC Digital Volume */
+	{ 16, 0x00C0 },     /* R16 - Right ADC Digital Volume */
+
+	{ 18, 0x0000 },     /* R18 - GPIO CTRL 1 */
+	{ 19, 0x1000 },     /* R19 - GPIO1 & GPIO2 */
+	{ 20, 0x1010 },     /* R20 - GPIO3 & GPIO4 */
+	{ 21, 0x1010 },     /* R21 - GPIO5 & GPIO6 */
+	{ 22, 0x8000 },     /* R22 - GPIOCTRL 2 */
+	{ 23, 0x0800 },     /* R23 - GPIO_POL */
+	{ 24, 0x008B },     /* R24 - Left Line Input 1&2 Volume */
+	{ 25, 0x008B },     /* R25 - Left Line Input 3&4 Volume */
+	{ 26, 0x008B },     /* R26 - Right Line Input 1&2 Volume */
+	{ 27, 0x008B },     /* R27 - Right Line Input 3&4 Volume */
+	{ 28, 0x0000 },     /* R28 - Left Output Volume */
+	{ 29, 0x0000 },     /* R29 - Right Output Volume */
+	{ 30, 0x0066 },     /* R30 - Line Outputs Volume */
+	{ 31, 0x0022 },     /* R31 - Out3/4 Volume */
+	{ 32, 0x0079 },     /* R32 - Left OPGA Volume */
+	{ 33, 0x0079 },     /* R33 - Right OPGA Volume */
+	{ 34, 0x0003 },     /* R34 - Speaker Volume */
+	{ 35, 0x0003 },     /* R35 - ClassD1 */
+
+	{ 37, 0x0100 },     /* R37 - ClassD3 */
+	{ 38, 0x0079 },     /* R38 - ClassD4 */
+	{ 39, 0x0000 },     /* R39 - Input Mixer1 */
+	{ 40, 0x0000 },     /* R40 - Input Mixer2 */
+	{ 41, 0x0000 },     /* R41 - Input Mixer3 */
+	{ 42, 0x0000 },     /* R42 - Input Mixer4 */
+	{ 43, 0x0000 },     /* R43 - Input Mixer5 */
+	{ 44, 0x0000 },     /* R44 - Input Mixer6 */
+	{ 45, 0x0000 },     /* R45 - Output Mixer1 */
+	{ 46, 0x0000 },     /* R46 - Output Mixer2 */
+	{ 47, 0x0000 },     /* R47 - Output Mixer3 */
+	{ 48, 0x0000 },     /* R48 - Output Mixer4 */
+	{ 49, 0x0000 },     /* R49 - Output Mixer5 */
+	{ 50, 0x0000 },     /* R50 - Output Mixer6 */
+	{ 51, 0x0180 },     /* R51 - Out3/4 Mixer */
+	{ 52, 0x0000 },     /* R52 - Line Mixer1 */
+	{ 53, 0x0000 },     /* R53 - Line Mixer2 */
+	{ 54, 0x0000 },     /* R54 - Speaker Mixer */
+	{ 55, 0x0000 },     /* R55 - Additional Control */
+	{ 56, 0x0000 },     /* R56 - AntiPOP1 */
+	{ 57, 0x0000 },     /* R57 - AntiPOP2 */
+	{ 58, 0x0000 },     /* R58 - MICBIAS */
+
+	{ 60, 0x0008 },     /* R60 - PLL1 */
+	{ 61, 0x0031 },     /* R61 - PLL2 */
+	{ 62, 0x0026 },     /* R62 - PLL3 */
 };
 
 #define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0)
@@ -1114,6 +1112,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute)
 static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
+	struct wm8990_priv *wm8990 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (level) {
@@ -1128,7 +1127,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_cache_sync(codec);
+			ret = regcache_sync(wm8990->regmap);
 			if (ret < 0) {
 				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
 				return ret;
@@ -1226,7 +1225,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 		snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
 
-		codec->cache_sync = 1;
+		regcache_mark_dirty(wm8990->regmap);
 		break;
 	}
 
@@ -1295,7 +1294,7 @@ static int wm8990_probe(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -1334,10 +1333,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
 	.suspend =	wm8990_suspend,
 	.resume =	wm8990_resume,
 	.set_bias_level = wm8990_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(wm8990_reg),
-	.reg_word_size = sizeof(u16),
-	.reg_cache_default = wm8990_reg,
-	.volatile_register = wm8990_volatile_register,
 	.controls =	wm8990_snd_controls,
 	.num_controls = ARRAY_SIZE(wm8990_snd_controls),
 	.dapm_widgets = wm8990_dapm_widgets,
@@ -1346,6 +1341,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
 	.num_dapm_routes = ARRAY_SIZE(wm8990_dapm_routes),
 };
 
+static const struct regmap_config wm8990_regmap = {
+	.reg_bits = 8,
+	.val_bits = 16,
+
+	.max_register = WM8990_PLL3,
+	.volatile_reg = wm8990_volatile_register,
+	.reg_defaults = wm8990_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(wm8990_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+};
+
 static int wm8990_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-- 
cgit v1.2.2


From d44008b358588cf6fcc74716b50584a8e59cbe65 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:48:31 -0200
Subject: ASoC: wm8995: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8995.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index da2899e6c401..4300caff1783 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2293,7 +2293,7 @@ static struct spi_driver wm8995_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8995_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -2350,7 +2350,7 @@ static int __init wm8995_modinit(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8995_i2c_driver);
 	if (ret) {
 		printk(KERN_ERR "Failed to register wm8995 I2C driver: %d\n",
@@ -2371,7 +2371,7 @@ module_init(wm8995_modinit);
 
 static void __exit wm8995_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8995_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From c6aeb7de226dd08ad9b343fc6cbaf2ff36f69c84 Mon Sep 17 00:00:00 2001
From: Florian Meier <florian.meier@koalo.de>
Date: Fri, 22 Nov 2013 16:24:08 +0100
Subject: ASoC: Add support for BCM2835

This driver adds support for digital audio (I2S)
for the BCM2835 SoC that is used by the
Raspberry Pi. External audio codecs can be
connected to the Raspberry Pi via P5 header.

It relies on cyclic DMA engine support for BCM2835.

Signed-off-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/Kconfig           |   1 +
 sound/soc/Makefile          |   1 +
 sound/soc/bcm/Kconfig       |  10 +
 sound/soc/bcm/Makefile      |   5 +
 sound/soc/bcm/bcm2835-i2s.c | 886 ++++++++++++++++++++++++++++++++++++++++++++
 5 files changed, 903 insertions(+)
 create mode 100644 sound/soc/bcm/Kconfig
 create mode 100644 sound/soc/bcm/Makefile
 create mode 100644 sound/soc/bcm/bcm2835-i2s.c

(limited to 'sound')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 5138b8493051..a5e3a70c0d3d 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -33,6 +33,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM
 # All the supported SoCs
 source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
+source "sound/soc/bcm/Kconfig"
 source "sound/soc/blackfin/Kconfig"
 source "sound/soc/cirrus/Kconfig"
 source "sound/soc/davinci/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 8b9e70105dd2..b52d4aad0716 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC)	+= codecs/
 obj-$(CONFIG_SND_SOC)	+= generic/
 obj-$(CONFIG_SND_SOC)	+= atmel/
 obj-$(CONFIG_SND_SOC)	+= au1x/
+obj-$(CONFIG_SND_SOC)	+= bcm/
 obj-$(CONFIG_SND_SOC)	+= blackfin/
 obj-$(CONFIG_SND_SOC)	+= cirrus/
 obj-$(CONFIG_SND_SOC)	+= davinci/
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
new file mode 100644
index 000000000000..3d82a29ce3a8
--- /dev/null
+++ b/sound/soc/bcm/Kconfig
@@ -0,0 +1,10 @@
+config SND_BCM2835_SOC_I2S
+	tristate "SoC Audio support for the Broadcom BCM2835 I2S module"
+	depends on ARCH_BCM2835 || COMPILE_TEST
+	select SND_SOC_DMAENGINE_PCM
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the BCM2835 I2S interface. You will also need
+	  to select the audio interfaces to support below.
diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile
new file mode 100644
index 000000000000..bc816b71e5a4
--- /dev/null
+++ b/sound/soc/bcm/Makefile
@@ -0,0 +1,5 @@
+# BCM2835 Platform Support
+snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
+
+obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+
diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
new file mode 100644
index 000000000000..f49b007c8b43
--- /dev/null
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -0,0 +1,886 @@
+/*
+ * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC
+ *
+ * Author:	Florian Meier <florian.meier@koalo.de>
+ *		Copyright 2013
+ *
+ * Based on
+ *	Raspberry Pi PCM I2S ALSA Driver
+ *	Copyright (c) by Phil Poole 2013
+ *
+ *	ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *      Vladimir Barinov, <vbarinov@embeddedalley.com>
+ *	Copyright (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ *	OMAP ALSA SoC DAI driver using McBSP port
+ *	Copyright (C) 2008 Nokia Corporation
+ *	Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *		 Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ *	Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
+ *	Author: Timur Tabi <timur@freescale.com>
+ *	Copyright 2007-2010 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+/* Clock registers */
+#define BCM2835_CLK_PCMCTL_REG  0x00
+#define BCM2835_CLK_PCMDIV_REG  0x04
+
+/* Clock register settings */
+#define BCM2835_CLK_PASSWD		(0x5a000000)
+#define BCM2835_CLK_PASSWD_MASK	(0xff000000)
+#define BCM2835_CLK_MASH(v)		((v) << 9)
+#define BCM2835_CLK_FLIP		BIT(8)
+#define BCM2835_CLK_BUSY		BIT(7)
+#define BCM2835_CLK_KILL		BIT(5)
+#define BCM2835_CLK_ENAB		BIT(4)
+#define BCM2835_CLK_SRC(v)		(v)
+
+#define BCM2835_CLK_SHIFT		(12)
+#define BCM2835_CLK_DIVI(v)		((v) << BCM2835_CLK_SHIFT)
+#define BCM2835_CLK_DIVF(v)		(v)
+#define BCM2835_CLK_DIVF_MASK		(0xFFF)
+
+enum {
+	BCM2835_CLK_MASH_0 = 0,
+	BCM2835_CLK_MASH_1,
+	BCM2835_CLK_MASH_2,
+	BCM2835_CLK_MASH_3,
+};
+
+enum {
+	BCM2835_CLK_SRC_GND = 0,
+	BCM2835_CLK_SRC_OSC,
+	BCM2835_CLK_SRC_DBG0,
+	BCM2835_CLK_SRC_DBG1,
+	BCM2835_CLK_SRC_PLLA,
+	BCM2835_CLK_SRC_PLLC,
+	BCM2835_CLK_SRC_PLLD,
+	BCM2835_CLK_SRC_HDMI,
+};
+
+/* Most clocks are not useable (freq = 0) */
+static const unsigned int bcm2835_clk_freq[BCM2835_CLK_SRC_HDMI+1] = {
+	[BCM2835_CLK_SRC_GND]		= 0,
+	[BCM2835_CLK_SRC_OSC]		= 19200000,
+	[BCM2835_CLK_SRC_DBG0]		= 0,
+	[BCM2835_CLK_SRC_DBG1]		= 0,
+	[BCM2835_CLK_SRC_PLLA]		= 0,
+	[BCM2835_CLK_SRC_PLLC]		= 0,
+	[BCM2835_CLK_SRC_PLLD]		= 500000000,
+	[BCM2835_CLK_SRC_HDMI]		= 0,
+};
+
+/* I2S registers */
+#define BCM2835_I2S_CS_A_REG		0x00
+#define BCM2835_I2S_FIFO_A_REG		0x04
+#define BCM2835_I2S_MODE_A_REG		0x08
+#define BCM2835_I2S_RXC_A_REG		0x0c
+#define BCM2835_I2S_TXC_A_REG		0x10
+#define BCM2835_I2S_DREQ_A_REG		0x14
+#define BCM2835_I2S_INTEN_A_REG	0x18
+#define BCM2835_I2S_INTSTC_A_REG	0x1c
+#define BCM2835_I2S_GRAY_REG		0x20
+
+/* I2S register settings */
+#define BCM2835_I2S_STBY		BIT(25)
+#define BCM2835_I2S_SYNC		BIT(24)
+#define BCM2835_I2S_RXSEX		BIT(23)
+#define BCM2835_I2S_RXF		BIT(22)
+#define BCM2835_I2S_TXE		BIT(21)
+#define BCM2835_I2S_RXD		BIT(20)
+#define BCM2835_I2S_TXD		BIT(19)
+#define BCM2835_I2S_RXR		BIT(18)
+#define BCM2835_I2S_TXW		BIT(17)
+#define BCM2835_I2S_CS_RXERR		BIT(16)
+#define BCM2835_I2S_CS_TXERR		BIT(15)
+#define BCM2835_I2S_RXSYNC		BIT(14)
+#define BCM2835_I2S_TXSYNC		BIT(13)
+#define BCM2835_I2S_DMAEN		BIT(9)
+#define BCM2835_I2S_RXTHR(v)		((v) << 7)
+#define BCM2835_I2S_TXTHR(v)		((v) << 5)
+#define BCM2835_I2S_RXCLR		BIT(4)
+#define BCM2835_I2S_TXCLR		BIT(3)
+#define BCM2835_I2S_TXON		BIT(2)
+#define BCM2835_I2S_RXON		BIT(1)
+#define BCM2835_I2S_EN			(1)
+
+#define BCM2835_I2S_CLKDIS		BIT(28)
+#define BCM2835_I2S_PDMN		BIT(27)
+#define BCM2835_I2S_PDME		BIT(26)
+#define BCM2835_I2S_FRXP		BIT(25)
+#define BCM2835_I2S_FTXP		BIT(24)
+#define BCM2835_I2S_CLKM		BIT(23)
+#define BCM2835_I2S_CLKI		BIT(22)
+#define BCM2835_I2S_FSM		BIT(21)
+#define BCM2835_I2S_FSI		BIT(20)
+#define BCM2835_I2S_FLEN(v)		((v) << 10)
+#define BCM2835_I2S_FSLEN(v)		(v)
+
+#define BCM2835_I2S_CHWEX		BIT(15)
+#define BCM2835_I2S_CHEN		BIT(14)
+#define BCM2835_I2S_CHPOS(v)		((v) << 4)
+#define BCM2835_I2S_CHWID(v)		(v)
+#define BCM2835_I2S_CH1(v)		((v) << 16)
+#define BCM2835_I2S_CH2(v)		(v)
+
+#define BCM2835_I2S_TX_PANIC(v)	((v) << 24)
+#define BCM2835_I2S_RX_PANIC(v)	((v) << 16)
+#define BCM2835_I2S_TX(v)		((v) << 8)
+#define BCM2835_I2S_RX(v)		(v)
+
+#define BCM2835_I2S_INT_RXERR		BIT(3)
+#define BCM2835_I2S_INT_TXERR		BIT(2)
+#define BCM2835_I2S_INT_RXR		BIT(1)
+#define BCM2835_I2S_INT_TXW		BIT(0)
+
+/* I2S DMA interface */
+/* FIXME: Needs IOMMU support */
+#define BCM2835_VCMMU_SHIFT		(0x7E000000 - 0x20000000)
+
+/* General device struct */
+struct bcm2835_i2s_dev {
+	struct device				*dev;
+	struct snd_dmaengine_dai_dma_data	dma_data[2];
+	unsigned int				fmt;
+	unsigned int				bclk_ratio;
+
+	struct regmap *i2s_regmap;
+	struct regmap *clk_regmap;
+};
+
+static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev)
+{
+	/* Start the clock if in master mode */
+	unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+
+	switch (master) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG,
+			BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB,
+			BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB);
+		break;
+	default:
+		break;
+	}
+}
+
+static void bcm2835_i2s_stop_clock(struct bcm2835_i2s_dev *dev)
+{
+	uint32_t clkreg;
+	int timeout = 1000;
+
+	/* Stop clock */
+	regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG,
+			BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB,
+			BCM2835_CLK_PASSWD);
+
+	/* Wait for the BUSY flag going down */
+	while (--timeout) {
+		regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg);
+		if (!(clkreg & BCM2835_CLK_BUSY))
+			break;
+	}
+
+	if (!timeout) {
+		/* KILL the clock */
+		dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n");
+		regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG,
+			BCM2835_CLK_KILL | BCM2835_CLK_PASSWD_MASK,
+			BCM2835_CLK_KILL | BCM2835_CLK_PASSWD);
+	}
+}
+
+static void bcm2835_i2s_clear_fifos(struct bcm2835_i2s_dev *dev,
+				    bool tx, bool rx)
+{
+	int timeout = 1000;
+	uint32_t syncval;
+	uint32_t csreg;
+	uint32_t i2s_active_state;
+	uint32_t clkreg;
+	uint32_t clk_active_state;
+	uint32_t off;
+	uint32_t clr;
+
+	off =  tx ? BCM2835_I2S_TXON : 0;
+	off |= rx ? BCM2835_I2S_RXON : 0;
+
+	clr =  tx ? BCM2835_I2S_TXCLR : 0;
+	clr |= rx ? BCM2835_I2S_RXCLR : 0;
+
+	/* Backup the current state */
+	regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg);
+	i2s_active_state = csreg & (BCM2835_I2S_RXON | BCM2835_I2S_TXON);
+
+	regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg);
+	clk_active_state = clkreg & BCM2835_CLK_ENAB;
+
+	/* Start clock if not running */
+	if (!clk_active_state) {
+		regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG,
+			BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB,
+			BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB);
+	}
+
+	/* Stop I2S module */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, off, 0);
+
+	/*
+	 * Clear the FIFOs
+	 * Requires at least 2 PCM clock cycles to take effect
+	 */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, clr, clr);
+
+	/* Wait for 2 PCM clock cycles */
+
+	/*
+	 * Toggle the SYNC flag. After 2 PCM clock cycles it can be read back
+	 * FIXME: This does not seem to work for slave mode!
+	 */
+	regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &syncval);
+	syncval &= BCM2835_I2S_SYNC;
+
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_SYNC, ~syncval);
+
+	/* Wait for the SYNC flag changing it's state */
+	while (--timeout) {
+		regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg);
+		if ((csreg & BCM2835_I2S_SYNC) != syncval)
+			break;
+	}
+
+	if (!timeout)
+		dev_err(dev->dev, "I2S SYNC error!\n");
+
+	/* Stop clock if it was not running before */
+	if (!clk_active_state)
+		bcm2835_i2s_stop_clock(dev);
+
+	/* Restore I2S state */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_RXON | BCM2835_I2S_TXON, i2s_active_state);
+}
+
+static int bcm2835_i2s_set_dai_fmt(struct snd_soc_dai *dai,
+				      unsigned int fmt)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	dev->fmt = fmt;
+	return 0;
+}
+
+static int bcm2835_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai,
+				      unsigned int ratio)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	dev->bclk_ratio = ratio;
+	return 0;
+}
+
+static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	unsigned int sampling_rate = params_rate(params);
+	unsigned int data_length, data_delay, bclk_ratio;
+	unsigned int ch1pos, ch2pos, mode, format;
+	unsigned int mash = BCM2835_CLK_MASH_1;
+	unsigned int divi, divf, target_frequency;
+	int clk_src = -1;
+	unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+	bool bit_master =	(master == SND_SOC_DAIFMT_CBS_CFS
+					|| master == SND_SOC_DAIFMT_CBS_CFM);
+
+	bool frame_master =	(master == SND_SOC_DAIFMT_CBS_CFS
+					|| master == SND_SOC_DAIFMT_CBM_CFS);
+	uint32_t csreg;
+
+	/*
+	 * If a stream is already enabled,
+	 * the registers are already set properly.
+	 */
+	regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg);
+
+	if (csreg & (BCM2835_I2S_TXON | BCM2835_I2S_RXON))
+		return 0;
+
+	/*
+	 * Adjust the data length according to the format.
+	 * We prefill the half frame length with an integer
+	 * divider of 2400 as explained at the clock settings.
+	 * Maybe it is overwritten there, if the Integer mode
+	 * does not apply.
+	 */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		data_length = 16;
+		bclk_ratio = 40;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		data_length = 32;
+		bclk_ratio = 80;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* If bclk_ratio already set, use that one. */
+	if (dev->bclk_ratio)
+		bclk_ratio = dev->bclk_ratio;
+
+	/*
+	 * Clock Settings
+	 *
+	 * The target frequency of the bit clock is
+	 *	sampling rate * frame length
+	 *
+	 * Integer mode:
+	 * Sampling rates that are multiples of 8000 kHz
+	 * can be driven by the oscillator of 19.2 MHz
+	 * with an integer divider as long as the frame length
+	 * is an integer divider of 19200000/8000=2400 as set up above.
+	 * This is no longer possible if the sampling rate
+	 * is too high (e.g. 192 kHz), because the oscillator is too slow.
+	 *
+	 * MASH mode:
+	 * For all other sampling rates, it is not possible to
+	 * have an integer divider. Approximate the clock
+	 * with the MASH module that induces a slight frequency
+	 * variance. To minimize that it is best to have the fastest
+	 * clock here. That is PLLD with 500 MHz.
+	 */
+	target_frequency = sampling_rate * bclk_ratio;
+	clk_src = BCM2835_CLK_SRC_OSC;
+	mash = BCM2835_CLK_MASH_0;
+
+	if (bcm2835_clk_freq[clk_src] % target_frequency == 0
+			&& bit_master && frame_master) {
+		divi = bcm2835_clk_freq[clk_src] / target_frequency;
+		divf = 0;
+	} else {
+		uint64_t dividend;
+
+		if (!dev->bclk_ratio) {
+			/*
+			 * Overwrite bclk_ratio, because the
+			 * above trick is not needed or can
+			 * not be used.
+			 */
+			bclk_ratio = 2 * data_length;
+		}
+
+		target_frequency = sampling_rate * bclk_ratio;
+
+		clk_src = BCM2835_CLK_SRC_PLLD;
+		mash = BCM2835_CLK_MASH_1;
+
+		dividend = bcm2835_clk_freq[clk_src];
+		dividend <<= BCM2835_CLK_SHIFT;
+		do_div(dividend, target_frequency);
+		divi = dividend >> BCM2835_CLK_SHIFT;
+		divf = dividend & BCM2835_CLK_DIVF_MASK;
+	}
+
+	/* Set clock divider */
+	regmap_write(dev->clk_regmap, BCM2835_CLK_PCMDIV_REG, BCM2835_CLK_PASSWD
+			| BCM2835_CLK_DIVI(divi)
+			| BCM2835_CLK_DIVF(divf));
+
+	/* Setup clock, but don't start it yet */
+	regmap_write(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, BCM2835_CLK_PASSWD
+			| BCM2835_CLK_MASH(mash)
+			| BCM2835_CLK_SRC(clk_src));
+
+	/* Setup the frame format */
+	format = BCM2835_I2S_CHEN;
+
+	if (data_length > 24)
+		format |= BCM2835_I2S_CHWEX;
+
+	format |= BCM2835_I2S_CHWID((data_length-8)&0xf);
+
+	switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		data_delay = 1;
+		break;
+	default:
+		/*
+		 * TODO
+		 * Others are possible but are not implemented at the moment.
+		 */
+		dev_err(dev->dev, "%s:bad format\n", __func__);
+		return -EINVAL;
+	}
+
+	ch1pos = data_delay;
+	ch2pos = bclk_ratio / 2 + data_delay;
+
+	switch (params_channels(params)) {
+	case 2:
+		format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format);
+		format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos));
+		format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos));
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/*
+	 * Set format for both streams.
+	 * We cannot set another frame length
+	 * (and therefore word length) anyway,
+	 * so the format will be the same.
+	 */
+	regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, format);
+	regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, format);
+
+	/* Setup the I2S mode */
+	mode = 0;
+
+	if (data_length <= 16) {
+		/*
+		 * Use frame packed mode (2 channels per 32 bit word)
+		 * We cannot set another frame length in the second stream
+		 * (and therefore word length) anyway,
+		 * so the format will be the same.
+		 */
+		mode |= BCM2835_I2S_FTXP | BCM2835_I2S_FRXP;
+	}
+
+	mode |= BCM2835_I2S_FLEN(bclk_ratio - 1);
+	mode |= BCM2835_I2S_FSLEN(bclk_ratio / 2);
+
+	/* Master or slave? */
+	switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* CPU is master */
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/*
+		 * CODEC is bit clock master
+		 * CPU is frame master
+		 */
+		mode |= BCM2835_I2S_CLKM;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		/*
+		 * CODEC is frame master
+		 * CPU is bit clock master
+		 */
+		mode |= BCM2835_I2S_FSM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* CODEC is master */
+		mode |= BCM2835_I2S_CLKM;
+		mode |= BCM2835_I2S_FSM;
+		break;
+	default:
+		dev_err(dev->dev, "%s:bad master\n", __func__);
+		return -EINVAL;
+	}
+
+	/*
+	 * Invert clocks?
+	 *
+	 * The BCM approach seems to be inverted to the classical I2S approach.
+	 */
+	switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* None. Therefore, both for BCM */
+		mode |= BCM2835_I2S_CLKI;
+		mode |= BCM2835_I2S_FSI;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		/* Both. Therefore, none for BCM */
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/*
+		 * Invert only frame sync. Therefore,
+		 * invert only bit clock for BCM
+		 */
+		mode |= BCM2835_I2S_CLKI;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/*
+		 * Invert only bit clock. Therefore,
+		 * invert only frame sync for BCM
+		 */
+		mode |= BCM2835_I2S_FSI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_write(dev->i2s_regmap, BCM2835_I2S_MODE_A_REG, mode);
+
+	/* Setup the DMA parameters */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_RXTHR(1)
+			| BCM2835_I2S_TXTHR(1)
+			| BCM2835_I2S_DMAEN, 0xffffffff);
+
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_DREQ_A_REG,
+			  BCM2835_I2S_TX_PANIC(0x10)
+			| BCM2835_I2S_RX_PANIC(0x30)
+			| BCM2835_I2S_TX(0x30)
+			| BCM2835_I2S_RX(0x20), 0xffffffff);
+
+	/* Clear FIFOs */
+	bcm2835_i2s_clear_fifos(dev, true, true);
+
+	return 0;
+}
+
+static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	uint32_t cs_reg;
+
+	bcm2835_i2s_start_clock(dev);
+
+	/*
+	 * Clear both FIFOs if the one that should be started
+	 * is not empty at the moment. This should only happen
+	 * after overrun. Otherwise, hw_params would have cleared
+	 * the FIFO.
+	 */
+	regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+			&& !(cs_reg & BCM2835_I2S_TXE))
+		bcm2835_i2s_clear_fifos(dev, true, false);
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+			&& (cs_reg & BCM2835_I2S_RXD))
+		bcm2835_i2s_clear_fifos(dev, false, true);
+
+	return 0;
+}
+
+static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev,
+		struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	uint32_t mask;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		mask = BCM2835_I2S_RXON;
+	else
+		mask = BCM2835_I2S_TXON;
+
+	regmap_update_bits(dev->i2s_regmap,
+			BCM2835_I2S_CS_A_REG, mask, 0);
+
+	/* Stop also the clock when not SND_SOC_DAIFMT_CONT */
+	if (!dai->active && !(dev->fmt & SND_SOC_DAIFMT_CONT))
+		bcm2835_i2s_stop_clock(dev);
+}
+
+static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	uint32_t mask;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		bcm2835_i2s_start_clock(dev);
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+			mask = BCM2835_I2S_RXON;
+		else
+			mask = BCM2835_I2S_TXON;
+
+		regmap_update_bits(dev->i2s_regmap,
+				BCM2835_I2S_CS_A_REG, mask, mask);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		bcm2835_i2s_stop(dev, substream, dai);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int bcm2835_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	if (dai->active)
+		return 0;
+
+	/* Should this still be running stop it */
+	bcm2835_i2s_stop_clock(dev);
+
+	/* Enable PCM block */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_EN, BCM2835_I2S_EN);
+
+	/*
+	 * Disable STBY.
+	 * Requires at least 4 PCM clock cycles to take effect.
+	 */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_STBY, BCM2835_I2S_STBY);
+
+	return 0;
+}
+
+static void bcm2835_i2s_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	bcm2835_i2s_stop(dev, substream, dai);
+
+	/* If both streams are stopped, disable module and clock */
+	if (dai->active)
+		return;
+
+	/* Disable the module */
+	regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG,
+			BCM2835_I2S_EN, 0);
+
+	/*
+	 * Stopping clock is necessary, because stop does
+	 * not stop the clock when SND_SOC_DAIFMT_CONT
+	 */
+	bcm2835_i2s_stop_clock(dev);
+}
+
+static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = {
+	.startup	= bcm2835_i2s_startup,
+	.shutdown	= bcm2835_i2s_shutdown,
+	.prepare	= bcm2835_i2s_prepare,
+	.trigger	= bcm2835_i2s_trigger,
+	.hw_params	= bcm2835_i2s_hw_params,
+	.set_fmt	= bcm2835_i2s_set_dai_fmt,
+	.set_bclk_ratio	= bcm2835_i2s_set_dai_bclk_ratio
+};
+
+static int bcm2835_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai,
+			&dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+			&dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver bcm2835_i2s_dai = {
+	.name	= "bcm2835-i2s",
+	.probe	= bcm2835_i2s_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates =	SNDRV_PCM_RATE_8000_192000,
+		.formats =	SNDRV_PCM_FMTBIT_S16_LE
+				| SNDRV_PCM_FMTBIT_S32_LE
+		},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates =	SNDRV_PCM_RATE_8000_192000,
+		.formats =	SNDRV_PCM_FMTBIT_S16_LE
+				| SNDRV_PCM_FMTBIT_S32_LE
+		},
+	.ops = &bcm2835_i2s_dai_ops,
+	.symmetric_rates = 1
+};
+
+static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case BCM2835_I2S_CS_A_REG:
+	case BCM2835_I2S_FIFO_A_REG:
+	case BCM2835_I2S_INTSTC_A_REG:
+	case BCM2835_I2S_GRAY_REG:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case BCM2835_I2S_FIFO_A_REG:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static bool bcm2835_clk_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case BCM2835_CLK_PCMCTL_REG:
+		return true;
+	default:
+		return false;
+	};
+}
+
+static const struct regmap_config bcm2835_regmap_config[] = {
+	{
+		.reg_bits = 32,
+		.reg_stride = 4,
+		.val_bits = 32,
+		.max_register = BCM2835_I2S_GRAY_REG,
+		.precious_reg = bcm2835_i2s_precious_reg,
+		.volatile_reg = bcm2835_i2s_volatile_reg,
+		.cache_type = REGCACHE_RBTREE,
+	},
+	{
+		.reg_bits = 32,
+		.reg_stride = 4,
+		.val_bits = 32,
+		.max_register = BCM2835_CLK_PCMDIV_REG,
+		.volatile_reg = bcm2835_clk_volatile_reg,
+		.cache_type = REGCACHE_RBTREE,
+	},
+};
+
+static const struct snd_soc_component_driver bcm2835_i2s_component = {
+	.name		= "bcm2835-i2s-comp",
+};
+
+static int bcm2835_i2s_probe(struct platform_device *pdev)
+{
+	struct bcm2835_i2s_dev *dev;
+	int i;
+	int ret;
+	struct regmap *regmap[2];
+	struct resource *mem[2];
+
+	/* Request both ioareas */
+	for (i = 0; i <= 1; i++) {
+		void __iomem *base;
+
+		mem[i] = platform_get_resource(pdev, IORESOURCE_MEM, i);
+		base = devm_ioremap_resource(&pdev->dev, mem[i]);
+		if (IS_ERR(base))
+			return PTR_ERR(base);
+
+		regmap[i] = devm_regmap_init_mmio(&pdev->dev, base,
+					    &bcm2835_regmap_config[i]);
+		if (IS_ERR(regmap[i]))
+			return PTR_ERR(regmap[i]);
+	}
+
+	dev = devm_kzalloc(&pdev->dev, sizeof(*dev),
+			   GFP_KERNEL);
+	if (!dev)
+		return -ENOMEM;
+
+	dev->i2s_regmap = regmap[0];
+	dev->clk_regmap = regmap[1];
+
+	/* Set the DMA address */
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
+		(dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG
+					  + BCM2835_VCMMU_SHIFT;
+
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
+		(dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG
+					  + BCM2835_VCMMU_SHIFT;
+
+	/* Set the bus width */
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr_width =
+		DMA_SLAVE_BUSWIDTH_4_BYTES;
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr_width =
+		DMA_SLAVE_BUSWIDTH_4_BYTES;
+
+	/* Set burst */
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2;
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2;
+
+	/* BCLK ratio - use default */
+	dev->bclk_ratio = 0;
+
+	/* Store the pdev */
+	dev->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+			&bcm2835_i2s_component, &bcm2835_i2s_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	if (ret) {
+		dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct of_device_id bcm2835_i2s_of_match[] = {
+	{ .compatible = "brcm,bcm2835-i2s", },
+	{},
+};
+
+static int bcm2835_i2s_remove(struct platform_device *pdev)
+{
+	snd_dmaengine_pcm_unregister(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver bcm2835_i2s_driver = {
+	.probe		= bcm2835_i2s_probe,
+	.remove		= bcm2835_i2s_remove,
+	.driver		= {
+		.name	= "bcm2835-i2s",
+		.owner	= THIS_MODULE,
+		.of_match_table = bcm2835_i2s_of_match,
+	},
+};
+
+module_platform_driver(bcm2835_i2s_driver);
+
+MODULE_ALIAS("platform:bcm2835-i2s");
+MODULE_DESCRIPTION("BCM2835 I2S interface");
+MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.2


From b21bdd0d343c8b2496690283211e27c9af598ed3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 18 Nov 2013 12:03:56 +0100
Subject: ALSA: hda - Split the generic parser as an individual module

Drop the hard dependency on the generic parser code and load / unload
the generic parser code dynamically if built as a module.  This allows
us to avoid the generic parser if only HDMI/DP codecs are found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Makefile      |  5 +++-
 sound/pci/hda/hda_codec.c   | 67 +++++++++++++++++++++++++++++++++------------
 sound/pci/hda/hda_codec.h   |  1 +
 sound/pci/hda/hda_generic.c |  4 +++
 sound/pci/hda/hda_local.h   |  7 -----
 5 files changed, 58 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index c091438286a3..5a40c652df41 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -3,7 +3,6 @@ snd-hda-intel-objs := hda_intel.o
 snd-hda-intel-$(CONFIG_SND_HDA_I915) +=	hda_i915.o
 
 snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o
-snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
 snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
 snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
 snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
@@ -12,6 +11,7 @@ snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
 CFLAGS_hda_codec.o := -I$(src)
 CFLAGS_hda_intel.o := -I$(src)
 
+snd-hda-codec-generic-objs :=	hda_generic.o
 snd-hda-codec-realtek-objs :=	patch_realtek.o
 snd-hda-codec-cmedia-objs :=	patch_cmedia.o
 snd-hda-codec-analog-objs :=	patch_analog.o
@@ -28,6 +28,9 @@ snd-hda-codec-hdmi-objs :=	patch_hdmi.o hda_eld.o
 obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o
 
 # codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans)
+ifdef CONFIG_SND_HDA_GENERIC
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-generic.o
+endif
 ifdef CONFIG_SND_HDA_CODEC_REALTEK
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o
 endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 69178c4f4113..4d3f46329eb5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -945,9 +945,6 @@ find_codec_preset(struct hda_codec *codec)
 	const struct hda_codec_preset *preset;
 	unsigned int mod_requested = 0;
 
-	if (is_generic_config(codec))
-		return NULL; /* use the generic parser */
-
  again:
 	mutex_lock(&preset_mutex);
 	list_for_each_entry(tbl, &hda_preset_tables, list) {
@@ -1329,6 +1326,28 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid)
 	return p;
 }
 
+/*
+ * Dynamic symbol binding for the codec parsers
+ */
+#ifdef MODULE
+#define load_parser_sym(sym)		((int (*)(struct hda_codec *))symbol_request(sym))
+#define unload_parser_addr(addr)	symbol_put_addr(addr)
+#else
+#define load_parser_sym(sym)		(sym)
+#define unload_parser_addr(addr)	do {} while (0)
+#endif
+
+#define load_parser(codec, sym) \
+	((codec)->parser = load_parser_sym(sym))
+
+static void unload_parser(struct hda_codec *codec)
+{
+	if (codec->parser) {
+		unload_parser_addr(codec->parser);
+		codec->parser = NULL;
+	}
+}
+
 /*
  * codec destructor
  */
@@ -1356,6 +1375,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	if (!codec->pm_down_notified) /* cancel leftover refcounts */
 		hda_call_pm_notify(codec->bus, false);
 #endif
+	unload_parser(codec);
 	module_put(codec->owner);
 	free_hda_cache(&codec->amp_cache);
 	free_hda_cache(&codec->cmd_cache);
@@ -1548,6 +1568,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_update_widgets);
  */
 int snd_hda_codec_configure(struct hda_codec *codec)
 {
+	int (*patch)(struct hda_codec *) = NULL;
 	int err;
 
 	codec->preset = find_codec_preset(codec);
@@ -1557,29 +1578,38 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 			return err;
 	}
 
-	if (is_generic_config(codec)) {
-		err = snd_hda_parse_generic_codec(codec);
-		goto patched;
-	}
-	if (codec->preset && codec->preset->patch) {
-		err = codec->preset->patch(codec);
-		goto patched;
+	if (!is_generic_config(codec) && codec->preset)
+		patch = codec->preset->patch;
+	if (!patch) {
+		unload_parser(codec); /* to be sure */
+#ifdef CONFIG_SND_HDA_GENERIC
+		if (!patch)
+			patch = load_parser(codec, snd_hda_parse_generic_codec);
+#endif
+		if (!patch) {
+			printk(KERN_ERR "hda-codec: No codec parser is available\n");
+			return -ENODEV;
+		}
 	}
 
-	/* call the default parser */
-	err = snd_hda_parse_generic_codec(codec);
-	if (err < 0)
-		printk(KERN_ERR "hda-codec: No codec parser is available\n");
+	err = patch(codec);
+	if (err < 0) {
+		unload_parser(codec);
+		return err;
+	}
 
- patched:
-	if (!err && codec->patch_ops.unsol_event)
+	if (codec->patch_ops.unsol_event) {
 		err = init_unsol_queue(codec->bus);
+		if (err < 0)
+			return err;
+	}
+
 	/* audio codec should override the mixer name */
-	if (!err && (codec->afg || !*codec->bus->card->mixername))
+	if (codec->afg || !*codec->bus->card->mixername)
 		snprintf(codec->bus->card->mixername,
 			 sizeof(codec->bus->card->mixername),
 			 "%s %s", codec->vendor_name, codec->chip_name);
-	return err;
+	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
 
@@ -2610,6 +2640,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	codec->preset = NULL;
 	codec->slave_dig_outs = NULL;
 	codec->spdif_status_reset = 0;
+	unload_parser(codec);
 	module_put(codec->owner);
 	codec->owner = NULL;
 
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 7aa9870040c1..14e4b5a9c43c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -834,6 +834,7 @@ struct hda_codec {
 	/* detected preset */
 	const struct hda_codec_preset *preset;
 	struct module *owner;
+	int (*parser)(struct hda_codec *codec);
 	const char *vendor_name;	/* codec vendor name */
 	const char *chip_name;		/* codec chip name */
 	const char *modelname;	/* model name for preset */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 3067ed4fe3b2..9b251456f2a0 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -28,6 +28,7 @@
 #include <linux/ctype.h>
 #include <linux/string.h>
 #include <linux/bitops.h>
+#include <linux/module.h>
 #include <sound/core.h>
 #include <sound/jack.h>
 #include "hda_codec.h"
@@ -5291,3 +5292,6 @@ error:
 	return err;
 }
 EXPORT_SYMBOL_HDA(snd_hda_parse_generic_codec);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Generic HD-audio codec parser");
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index d398b648bb5d..4d571a6d9230 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -352,14 +352,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 /*
  * generic codec parser
  */
-#ifdef CONFIG_SND_HDA_GENERIC
 int snd_hda_parse_generic_codec(struct hda_codec *codec);
-#else
-static inline int snd_hda_parse_generic_codec(struct hda_codec *codec)
-{
-	return -ENODEV;
-}
-#endif
 
 /*
  * generic proc interface
-- 
cgit v1.2.2


From f0639272d6c8e4706ea6d1b0dbc6eb83eaaff412 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 18 Nov 2013 12:07:29 +0100
Subject: ALSA: hda - Bind with HDMI codec parser automatically

If a codec contains only the digital outputs, it's very likely a
HDMI/DP codec, which isn't supported by the generic parser but via
HDMI codec parser code.  Detect such a case and bind with the proper
parser object if available.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c  | 27 +++++++++++++++++++++++++++
 sound/pci/hda/hda_local.h  |  1 +
 sound/pci/hda/patch_hdmi.c |  9 +++++++++
 3 files changed, 37 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4d3f46329eb5..bada677df8a7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1557,6 +1557,31 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec)
 EXPORT_SYMBOL_HDA(snd_hda_codec_update_widgets);
 
 
+#ifdef CONFIG_SND_HDA_CODEC_HDMI
+/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */
+static bool is_likely_hdmi_codec(struct hda_codec *codec)
+{
+	hda_nid_t nid = codec->start_nid;
+	int i;
+
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		unsigned int wcaps = get_wcaps(codec, nid);
+		switch (get_wcaps_type(wcaps)) {
+		case AC_WID_AUD_IN:
+			return false; /* HDMI parser supports only HDMI out */
+		case AC_WID_AUD_OUT:
+			if (!(wcaps & AC_WCAP_DIGITAL))
+				return false;
+			break;
+		}
+	}
+	return true;
+}
+#else
+/* no HDMI codec parser support */
+#define is_likely_hdmi_codec(codec)	false
+#endif /* CONFIG_SND_HDA_CODEC_HDMI */
+
 /**
  * snd_hda_codec_configure - (Re-)configure the HD-audio codec
  * @codec: the HDA codec
@@ -1582,6 +1607,8 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 		patch = codec->preset->patch;
 	if (!patch) {
 		unload_parser(codec); /* to be sure */
+		if (is_likely_hdmi_codec(codec))
+			patch = load_parser(codec, snd_hda_parse_hdmi_codec);
 #ifdef CONFIG_SND_HDA_GENERIC
 		if (!patch)
 			patch = load_parser(codec, snd_hda_parse_generic_codec);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4d571a6d9230..da80c5bd7fd4 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -353,6 +353,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
  * generic codec parser
  */
 int snd_hda_parse_generic_codec(struct hda_codec *codec);
+int snd_hda_parse_hdmi_codec(struct hda_codec *codec);
 
 /*
  * generic proc interface
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 08407bed093e..a6081ed34b88 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3213,6 +3213,15 @@ static int patch_via_hdmi(struct hda_codec *codec)
 	return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID);
 }
 
+/*
+ * called from hda_codec.c for generic HDMI support
+ */
+int snd_hda_parse_hdmi_codec(struct hda_codec *codec)
+{
+	return patch_generic_hdmi(codec);
+}
+EXPORT_SYMBOL_HDA(snd_hda_parse_hdmi_codec);
+
 /*
  * patch entries
  */
-- 
cgit v1.2.2


From a40e0a880b34d95f583552be7fd3905b6b4e6891 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Nov 2013 12:41:20 +0100
Subject: ALSA: hda - Refactor pm notification in hda_codec.c

No functional change.

- codec->pm_down_notified flag is renamed to codec->pm_up_notified
  flag (and takes the reversed meaning), which is managed solely in
  hda_call_pm_notify() now.

- The explicit call of hda_call_pm_notify() is moved into
  hda_keep_power_on().

- Removed a redundant call of hda_call_pm_notify() in
  snd_hda_power_up(), as it's called in hda_call_codec_resume()
  anyway.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 37 ++++++++++++++-----------------------
 sound/pci/hda/hda_codec.h |  2 +-
 2 files changed, 15 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index bada677df8a7..8f7295bdccbd 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -99,16 +99,24 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
 #define hda_codec_is_power_on(codec)	((codec)->power_on)
-static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up)
+
+static void hda_call_pm_notify(struct hda_codec *codec, bool power_up)
 {
+	struct hda_bus *bus = codec->bus;
+
+	if ((power_up && codec->pm_up_notified) ||
+	    (!power_up && !codec->pm_up_notified))
+		return;
 	if (bus->ops.pm_notify)
 		bus->ops.pm_notify(bus, power_up);
+	codec->pm_up_notified = power_up;
 }
+
 #else
 #define codec_in_pm(codec)	0
 static inline void hda_keep_power_on(struct hda_codec *codec) {}
 #define hda_codec_is_power_on(codec)	1
-#define hda_call_pm_notify(bus, state) {}
+#define hda_call_pm_notify(codec, state) {}
 #endif
 
 /**
@@ -1371,10 +1379,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	codec->bus->caddr_tbl[codec->addr] = NULL;
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
-#ifdef CONFIG_PM
-	if (!codec->pm_down_notified) /* cancel leftover refcounts */
-		hda_call_pm_notify(codec->bus, false);
-#endif
+	hda_call_pm_notify(codec, false); /* cancel leftover refcounts */
 	unload_parser(codec);
 	module_put(codec->owner);
 	free_hda_cache(&codec->amp_cache);
@@ -1453,7 +1458,6 @@ int snd_hda_codec_new(struct hda_bus *bus,
 	 * phase.
 	 */
 	hda_keep_power_on(codec);
-	hda_call_pm_notify(bus, true);
 #endif
 
 	if (codec->bus->modelname) {
@@ -4058,10 +4062,6 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	 * in the resume / power-save sequence
 	 */
 	hda_keep_power_on(codec);
-	if (codec->pm_down_notified) {
-		codec->pm_down_notified = 0;
-		hda_call_pm_notify(codec->bus, true);
-	}
 	hda_set_power_state(codec, AC_PWRST_D0);
 	restore_shutup_pins(codec);
 	hda_exec_init_verbs(codec);
@@ -4935,11 +4935,8 @@ static void hda_power_work(struct work_struct *work)
 	spin_unlock(&codec->power_lock);
 
 	state = hda_call_codec_suspend(codec, true);
-	if (!codec->pm_down_notified &&
-	    !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) {
-		codec->pm_down_notified = 1;
-		hda_call_pm_notify(bus, false);
-	}
+	if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK))
+		hda_call_pm_notify(codec, false);
 }
 
 static void hda_keep_power_on(struct hda_codec *codec)
@@ -4949,6 +4946,7 @@ static void hda_keep_power_on(struct hda_codec *codec)
 	codec->power_on = 1;
 	codec->power_jiffies = jiffies;
 	spin_unlock(&codec->power_lock);
+	hda_call_pm_notify(codec, true);
 }
 
 /* update the power on/off account with the current jiffies */
@@ -4968,8 +4966,6 @@ void snd_hda_update_power_acct(struct hda_codec *codec)
 /* call this with codec->power_lock held! */
 static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 {
-	struct hda_bus *bus = codec->bus;
-
 	/* Return if power_on or transitioning to power_on, unless currently
 	 * powering down. */
 	if ((codec->power_on || codec->power_transition > 0) &&
@@ -4996,11 +4992,6 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 	codec->power_transition = 1; /* avoid reentrance */
 	spin_unlock(&codec->power_lock);
 
-	if (codec->pm_down_notified) {
-		codec->pm_down_notified = 0;
-		hda_call_pm_notify(bus, true);
-	}
-
 	hda_call_codec_resume(codec);
 
 	spin_lock(&codec->power_lock);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 14e4b5a9c43c..a3e5f45d74c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -908,7 +908,7 @@ struct hda_codec {
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
-	unsigned int pm_down_notified:1; /* PM notified to controller */
+	unsigned int pm_up_notified:1;	/* PM notified to controller */
 	unsigned int in_pm:1;		/* suspend/resume being performed */
 	int power_transition;	/* power-state in transition */
 	int power_count;	/* current (global) power refcount */
-- 
cgit v1.2.2


From b1920c21102af1d22f27ce36378aa48d97a62bc0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 22 Nov 2013 12:43:25 +0100
Subject: ALSA: hda - Enable runtime PM on Panther Point

Now we fixed the long-standing bugs of runtime PM, let's enable
Panther Point again.  The runtime PM was disabled in the HDMI codec
driver due to the S3 issue, and this should have been fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c  | 2 +-
 sound/pci/hda/patch_hdmi.c | 6 +-----
 2 files changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c6d230193da6..dfdb96603636 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3978,7 +3978,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
 	/* Lynx Point */
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index a6081ed34b88..1773a640e72e 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1693,16 +1693,12 @@ static int hdmi_parse_codec(struct hda_codec *codec)
 	}
 
 #ifdef CONFIG_PM
-	/* We're seeing some problems with unsolicited hot plug events on
-	 * PantherPoint after S3, if this is not enabled */
-	if (codec->vendor_id == 0x80862806)
-		codec->bus->power_keep_link_on = 1;
 	/*
 	 * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event
 	 * can be lost and presence sense verb will become inaccurate if the
 	 * HDA link is powered off at hot plug or hw initialization time.
 	 */
-	else if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) &
+	if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) &
 	      AC_PWRST_EPSS))
 		codec->bus->power_keep_link_on = 1;
 #endif
-- 
cgit v1.2.2


From 3e9bc58fef3b7a1ee95b36c26671198a985293c7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 26 Nov 2013 09:58:46 +0100
Subject: ALSA: hda - Disable runtime PM when EPSS is unavailable

According to Mengdong, we shouldn't enable runtime PM when a codec
doesn't support EPSS, based on the experiences on Windows.
We have already this check in HDMI codec drivers, but now apply it in
general in hda_codec.c.

Credit goes to Mengdong Lin <mengdong.lin@intel.com>

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c  |  7 +++++--
 sound/pci/hda/patch_hdmi.c | 11 -----------
 2 files changed, 5 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8f7295bdccbd..6deb8d1852cc 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1510,11 +1510,14 @@ int snd_hda_codec_new(struct hda_bus *bus,
 #ifdef CONFIG_PM
 	codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, fg,
 					AC_PWRST_CLKSTOP);
-	if (!codec->d3_stop_clk)
-		bus->power_keep_link_on = 1;
 #endif
 	codec->epss = snd_hda_codec_get_supported_ps(codec, fg,
 					AC_PWRST_EPSS);
+#ifdef CONFIG_PM
+	if (!codec->d3_stop_clk || !codec->epss)
+		bus->power_keep_link_on = 1;
+#endif
+
 
 	/* power-up all before initialization */
 	hda_set_power_state(codec, AC_PWRST_D0);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 1773a640e72e..4098196b2300 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1692,17 +1692,6 @@ static int hdmi_parse_codec(struct hda_codec *codec)
 		}
 	}
 
-#ifdef CONFIG_PM
-	/*
-	 * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event
-	 * can be lost and presence sense verb will become inaccurate if the
-	 * HDA link is powered off at hot plug or hw initialization time.
-	 */
-	if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) &
-	      AC_PWRST_EPSS))
-		codec->bus->power_keep_link_on = 1;
-#endif
-
 	return 0;
 }
 
-- 
cgit v1.2.2


From 0e24dbb7f44877a3565ee461d2209b0942b1a753 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Tue, 26 Nov 2013 23:00:51 -0500
Subject: ALSA: hda - suspend codecs in parallel

The time to suspend a single codec may be several hundreds of ms. When runtime
power saving is disabled, driver suspend time can be long especially when there
are more than one codec on the bus.

To reduce driver suspend time, this patch creates a work queue for the bus, and
suspends the codecs in parallel if there are multiple codecs on the bus.

[fixed cosmetic issues by tiwai]

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 47 +++++++++++++++++++++++++++++++++++++++++++++--
 sound/pci/hda/hda_codec.h |  5 +++++
 2 files changed, 50 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 6deb8d1852cc..85886220ee74 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -96,6 +96,7 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 
 #ifdef CONFIG_PM
 #define codec_in_pm(codec)	((codec)->in_pm)
+static void hda_pm_work(struct work_struct *work);
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
 #define hda_codec_is_power_on(codec)	((codec)->power_on)
@@ -839,6 +840,12 @@ static int snd_hda_bus_free(struct hda_bus *bus)
 		bus->ops.private_free(bus);
 	if (bus->workq)
 		destroy_workqueue(bus->workq);
+
+#ifdef CONFIG_PM
+	if (bus->pm_wq)
+		destroy_workqueue(bus->pm_wq);
+#endif
+
 	kfree(bus);
 	return 0;
 }
@@ -883,6 +890,9 @@ int snd_hda_bus_new(struct snd_card *card,
 		.dev_register = snd_hda_bus_dev_register,
 		.dev_free = snd_hda_bus_dev_free,
 	};
+#ifdef CONFIG_PM
+	char wqname[16];
+#endif
 
 	if (snd_BUG_ON(!temp))
 		return -EINVAL;
@@ -919,6 +929,16 @@ int snd_hda_bus_new(struct snd_card *card,
 		return -ENOMEM;
 	}
 
+#ifdef CONFIG_PM
+	sprintf(wqname, "hda-pm-wq-%d", card->number);
+	bus->pm_wq = create_workqueue(wqname);
+	if (!bus->pm_wq) {
+		snd_printk(KERN_ERR "cannot create PM workqueue\n");
+		snd_hda_bus_free(bus);
+		return -ENOMEM;
+	}
+#endif
+
 	err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops);
 	if (err < 0) {
 		snd_hda_bus_free(bus);
@@ -1388,6 +1408,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	kfree(codec->chip_name);
 	kfree(codec->modelname);
 	kfree(codec->wcaps);
+	codec->bus->num_codecs--;
 	kfree(codec);
 }
 
@@ -1453,6 +1474,7 @@ int snd_hda_codec_new(struct hda_bus *bus,
 #ifdef CONFIG_PM
 	spin_lock_init(&codec->power_lock);
 	INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
+	INIT_WORK(&codec->pm_work, hda_pm_work);
 	/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
 	 * the caller has to power down appropriatley after initialization
 	 * phase.
@@ -1469,6 +1491,11 @@ int snd_hda_codec_new(struct hda_bus *bus,
 	}
 
 	list_add_tail(&codec->list, &bus->codec_list);
+	bus->num_codecs++;
+#ifdef CONFIG_PM
+	workqueue_set_max_active(bus->pm_wq, bus->num_codecs);
+#endif
+
 	bus->caddr_tbl[codec_addr] = codec;
 
 	codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT,
@@ -5088,6 +5115,14 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
+
+static void hda_pm_work(struct work_struct *work)
+{
+	struct hda_codec *codec =
+		container_of(work, struct hda_codec, pm_work);
+
+	hda_call_codec_suspend(codec, false);
+}
 #endif
 
 /*
@@ -5663,9 +5698,17 @@ int snd_hda_suspend(struct hda_bus *bus)
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		cancel_delayed_work_sync(&codec->jackpoll_work);
-		if (hda_codec_is_power_on(codec))
-			hda_call_codec_suspend(codec, false);
+		if (hda_codec_is_power_on(codec)) {
+			if (bus->num_codecs > 1)
+				queue_work(bus->pm_wq, &codec->pm_work);
+			else
+				hda_call_codec_suspend(codec, false);
+		}
 	}
+
+	if (bus->num_codecs > 1)
+		flush_workqueue(bus->pm_wq);
+
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_suspend);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index a3e5f45d74c1..9a2798a0313c 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -673,6 +673,7 @@ struct hda_bus {
 
 	/* codec linked list */
 	struct list_head codec_list;
+	unsigned int num_codecs;
 	/* link caddr -> codec */
 	struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
 
@@ -683,6 +684,9 @@ struct hda_bus {
 	struct hda_bus_unsolicited *unsol;
 	char workq_name[16];
 	struct workqueue_struct *workq;	/* common workqueue for codecs */
+#ifdef CONFIG_PM
+	struct workqueue_struct *pm_wq;	/* workqueue to parallel codec PM */
+#endif
 
 	/* assigned PCMs */
 	DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES);
@@ -917,6 +921,7 @@ struct hda_codec {
 	unsigned long power_off_acct;
 	unsigned long power_jiffies;
 	spinlock_t power_lock;
+	struct work_struct pm_work; /* task to parallel multi-codec PM */
 #endif
 
 	/* filter the requested power state per nid */
-- 
cgit v1.2.2


From 12edb893082b89f4d3af85ab9f4750aa8d3c34f3 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Tue, 26 Nov 2013 23:32:23 -0500
Subject: ALSA: hda - resume codecs in parallel

To reduce driver resume time, this patch resumes the codecs in parallel
if there are multiple codecs on the bus.

- The PM workqueue of bus is also used to parallel resuming multiple codecs.
- The work item 'pm_work' is renamed to 'suspend_work' to parallel suspending
  codecs.
- Add a work item 'resume_work' to parallel resuming codecs.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 29 +++++++++++++++++++++++------
 sound/pci/hda/hda_codec.h |  4 +++-
 2 files changed, 26 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 85886220ee74..472e5d04701e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -96,7 +96,8 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 
 #ifdef CONFIG_PM
 #define codec_in_pm(codec)	((codec)->in_pm)
-static void hda_pm_work(struct work_struct *work);
+static void hda_suspend_work(struct work_struct *work);
+static void hda_resume_work(struct work_struct *work);
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
 #define hda_codec_is_power_on(codec)	((codec)->power_on)
@@ -1474,7 +1475,8 @@ int snd_hda_codec_new(struct hda_bus *bus,
 #ifdef CONFIG_PM
 	spin_lock_init(&codec->power_lock);
 	INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
-	INIT_WORK(&codec->pm_work, hda_pm_work);
+	INIT_WORK(&codec->suspend_work, hda_suspend_work);
+	INIT_WORK(&codec->resume_work, hda_resume_work);
 	/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
 	 * the caller has to power down appropriatley after initialization
 	 * phase.
@@ -5116,13 +5118,21 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
 
-static void hda_pm_work(struct work_struct *work)
+static void hda_suspend_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
-		container_of(work, struct hda_codec, pm_work);
+		container_of(work, struct hda_codec, suspend_work);
 
 	hda_call_codec_suspend(codec, false);
 }
+
+static void hda_resume_work(struct work_struct *work)
+{
+	struct hda_codec *codec =
+		container_of(work, struct hda_codec, resume_work);
+
+	hda_call_codec_resume(codec);
+}
 #endif
 
 /*
@@ -5700,7 +5710,7 @@ int snd_hda_suspend(struct hda_bus *bus)
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		if (hda_codec_is_power_on(codec)) {
 			if (bus->num_codecs > 1)
-				queue_work(bus->pm_wq, &codec->pm_work);
+				queue_work(bus->pm_wq, &codec->suspend_work);
 			else
 				hda_call_codec_suspend(codec, false);
 		}
@@ -5724,8 +5734,15 @@ int snd_hda_resume(struct hda_bus *bus)
 	struct hda_codec *codec;
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
-		hda_call_codec_resume(codec);
+		if (bus->num_codecs > 1)
+			queue_work(bus->pm_wq, &codec->resume_work);
+		else
+			hda_call_codec_resume(codec);
 	}
+
+	if (bus->num_codecs > 1)
+		flush_workqueue(bus->pm_wq);
+
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_resume);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9a2798a0313c..01b248d3f7b2 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -921,7 +921,9 @@ struct hda_codec {
 	unsigned long power_off_acct;
 	unsigned long power_jiffies;
 	spinlock_t power_lock;
-	struct work_struct pm_work; /* task to parallel multi-codec PM */
+	/* tasks to parallel multi-codec suspend/resume */
+	struct work_struct suspend_work;
+	struct work_struct resume_work;
 #endif
 
 	/* filter the requested power state per nid */
-- 
cgit v1.2.2


From 4d9127faa864e7068d7e06527dfdf099ad06f64a Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 20 Nov 2013 15:37:42 -0200
Subject: ASoC: soc-io: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-io.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 4f11d23f2062..aa886cca3ecf 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -99,14 +99,14 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	config.val_bits = data_bits;
 
 	switch (control) {
-#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE)
+#if IS_ENABLED(CONFIG_REGMAP_I2C)
 	case SND_SOC_I2C:
 		codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
 						      &config);
 		break;
 #endif
 
-#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE)
+#if IS_ENABLED(CONFIG_REGMAP_SPI)
 	case SND_SOC_SPI:
 		codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
 						      &config);
-- 
cgit v1.2.2


From 516ea4b584332f511d3bf1b98ceabd974b1a2313 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:38 -0200
Subject: ASoC: cs4271: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs4271.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index f6e953454bc0..ce05fd93dc74 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -675,7 +675,7 @@ static struct spi_driver cs4271_spi_driver = {
 };
 #endif /* defined(CONFIG_SPI_MASTER) */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static const struct i2c_device_id cs4271_i2c_id[] = {
 	{"cs4271", 0},
 	{}
@@ -728,7 +728,7 @@ static struct i2c_driver cs4271_i2c_driver = {
 	.probe		= cs4271_i2c_probe,
 	.remove		= cs4271_i2c_remove,
 };
-#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */
+#endif /* IS_ENABLED(CONFIG_I2C) */
 
 /*
  * We only register our serial bus driver here without
@@ -741,7 +741,7 @@ static int __init cs4271_modinit(void)
 {
 	int ret;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&cs4271_i2c_driver);
 	if (ret) {
 		pr_err("Failed to register CS4271 I2C driver: %d\n", ret);
@@ -767,7 +767,7 @@ static void __exit cs4271_modexit(void)
 	spi_unregister_driver(&cs4271_spi_driver);
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&cs4271_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From 25c1a63f43ca40f1581c076b7f7618297ef1cbba Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:40 -0200
Subject: ASoC: da7210: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/da7210.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 9c1231456502..8166dcb2e4a3 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1188,7 +1188,7 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
 	.num_dapm_routes	= ARRAY_SIZE(da7210_audio_map),
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 
 static struct reg_default da7210_regmap_i2c_patch[] = {
 
@@ -1362,7 +1362,7 @@ static struct spi_driver da7210_spi_driver = {
 static int __init da7210_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&da7210_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
@@ -1378,7 +1378,7 @@ module_init(da7210_modinit);
 
 static void __exit da7210_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&da7210_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From e42644c64c0de145ab2041ceb904e44421dd6794 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <festevam@gmail.com>
Date: Thu, 21 Nov 2013 12:38:42 -0200
Subject: ASoC: ssm2602: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ssm2602.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 492644e67ace..480074d86476 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -730,7 +730,7 @@ static struct spi_driver ssm2602_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 /*
  * ssm2602 2 wire address is determined by GPIO5
  * state during powerup.
@@ -797,7 +797,7 @@ static int __init ssm2602_modinit(void)
 		return ret;
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&ssm2602_i2c_driver);
 	if (ret)
 		return ret;
@@ -813,7 +813,7 @@ static void __exit ssm2602_exit(void)
 	spi_unregister_driver(&ssm2602_spi_driver);
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&ssm2602_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From b65ab73e5d624eb4a88bc6094a3627007cb92500 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:43 -0200
Subject: ASoC: wm8731: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8731.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..6117107ea560 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -732,7 +732,7 @@ static struct spi_driver wm8731_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8731_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -791,7 +791,7 @@ static struct i2c_driver wm8731_i2c_driver = {
 static int __init wm8731_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8731_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n",
@@ -811,7 +811,7 @@ module_init(wm8731_modinit);
 
 static void __exit wm8731_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8731_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 26090a834b49673945458b185be0afa03c2737fe Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:45 -0200
Subject: ASoC: wm8741: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8741.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index b18813cc7ba9..2895c8d3b5e4 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -500,7 +500,7 @@ static const struct regmap_config wm8741_regmap = {
 	.readable_reg = wm8741_readable,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8741_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -617,7 +617,7 @@ static int __init wm8741_modinit(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8741_i2c_driver);
 	if (ret != 0)
 		pr_err("Failed to register WM8741 I2C driver: %d\n", ret);
@@ -639,7 +639,7 @@ static void __exit wm8741_exit(void)
 #if defined(CONFIG_SPI_MASTER)
 	spi_unregister_driver(&wm8741_spi_driver);
 #endif
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8741_i2c_driver);
 #endif
 }
-- 
cgit v1.2.2


From 9ea6fbc66d15c83089e177b445872a9ba40f125d Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:46 -0200
Subject: ASoC: wm8750: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8750.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 50d5ff616232..78616a638a55 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -816,7 +816,7 @@ static struct spi_driver wm8750_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8750_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -868,7 +868,7 @@ static struct i2c_driver wm8750_i2c_driver = {
 static int __init wm8750_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8750_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register wm8750 I2C driver: %d\n",
@@ -888,7 +888,7 @@ module_init(wm8750_modinit);
 
 static void __exit wm8750_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8750_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 2c4864334c4d9a23fa810638ad27e80ea0ceb9a4 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:47 -0200
Subject: ASoC: wm8753: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8753.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d96ebf52d953..be85da93a268 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1596,7 +1596,7 @@ static struct spi_driver wm8753_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8753_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1653,7 +1653,7 @@ static struct i2c_driver wm8753_i2c_driver = {
 static int __init wm8753_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8753_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register wm8753 I2C driver: %d\n",
@@ -1673,7 +1673,7 @@ module_init(wm8753_modinit);
 
 static void __exit wm8753_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8753_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From b888edbc68fbace3101cb092c6910476e85ae922 Mon Sep 17 00:00:00 2001
From: wangbiao <biao.wang@intel.com>
Date: Fri, 22 Nov 2013 10:44:30 +0800
Subject: ASoC: wm8994: Move DCS done IRQ request later

once code return from request_threaded_irq, irq was setup enabled by
default, but completion var dcs_done not got initialized yet, if then a
dcs done irq was raised, system will got hung as the sync mechanism is
invalid now.

so this patch move dcs done irq request to the end of initialization
of completion.

Signed-off-by: wang, biao <biao.wang@intel.com>
Signed-off-by: Zhang, Di <di.zhang@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8994.c | 11 +++++------
 1 file changed, 5 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 86426a117b07..b9be9cbc4603 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4077,12 +4077,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT,
 			   wm8994_temp_shut, "Thermal shutdown", codec);
 
-	ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE,
-				 wm_hubs_dcs_done, "DC servo done",
-				 &wm8994->hubs);
-	if (ret == 0)
-		wm8994->hubs.dcs_done_irq = true;
-
 	switch (control->type) {
 	case WM8994:
 		if (wm8994->micdet_irq) {
@@ -4313,6 +4307,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	}
 
 	wm_hubs_add_analogue_routes(codec, 0, 0);
+	ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE,
+				 wm_hubs_dcs_done, "DC servo done",
+				 &wm8994->hubs);
+	if (ret == 0)
+		wm8994->hubs.dcs_done_irq = true;
 	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	switch (control->type) {
-- 
cgit v1.2.2


From 1769267bb01303ac73b48535454461819ef1dcc2 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:48 -0200
Subject: ASoC: wm8776: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8776.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 942d58e455f3..ef8246725232 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -532,7 +532,7 @@ static struct spi_driver wm8776_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8776_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -584,7 +584,7 @@ static struct i2c_driver wm8776_i2c_driver = {
 static int __init wm8776_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8776_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register wm8776 I2C driver: %d\n",
@@ -604,7 +604,7 @@ module_init(wm8776_modinit);
 
 static void __exit wm8776_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8776_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From f3f9a60f7947b6bd2f970d5680dd3df624405027 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:49 -0200
Subject: ASoC: wm8804: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8804.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 1704b1e119cb..9bc8206a6807 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -739,7 +739,7 @@ static struct spi_driver wm8804_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8804_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -791,7 +791,7 @@ static int __init wm8804_modinit(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8804_i2c_driver);
 	if (ret) {
 		printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n",
@@ -811,7 +811,7 @@ module_init(wm8804_modinit);
 
 static void __exit wm8804_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8804_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From f25cf34969823ab7197ce9ff2521c33f0141075b Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:50 -0200
Subject: ASoC: wm8900: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8900.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 734209e252c3..e98bc7038a08 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1288,7 +1288,7 @@ static struct spi_driver wm8900_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8900_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1338,7 +1338,7 @@ static struct i2c_driver wm8900_i2c_driver = {
 static int __init wm8900_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8900_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register wm8900 I2C driver: %d\n",
@@ -1358,7 +1358,7 @@ module_init(wm8900_modinit);
 
 static void __exit wm8900_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8900_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 50c9697320434d1489d087cbf38f7907a9894609 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:52 -0200
Subject: ASoC: wm8985: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8985.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 18f2babe1090..271b517911a4 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -1148,7 +1148,7 @@ static struct spi_driver wm8985_spi_driver = {
 };
 #endif
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8985_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1201,7 +1201,7 @@ static int __init wm8985_modinit(void)
 {
 	int ret = 0;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8985_i2c_driver);
 	if (ret) {
 		printk(KERN_ERR "Failed to register wm8985 I2C driver: %d\n",
@@ -1221,7 +1221,7 @@ module_init(wm8985_modinit);
 
 static void __exit wm8985_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8985_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 63587116811bd23d22693b50447a2a356602e70b Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:53 -0200
Subject: ASoC: wm8988: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm8988.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 39b9acceb595..a55e1c2c382e 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -912,7 +912,7 @@ static struct spi_driver wm8988_spi_driver = {
 };
 #endif /* CONFIG_SPI_MASTER */
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm8988_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -964,7 +964,7 @@ static struct i2c_driver wm8988_i2c_driver = {
 static int __init wm8988_modinit(void)
 {
 	int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	ret = i2c_add_driver(&wm8988_i2c_driver);
 	if (ret != 0) {
 		printk(KERN_ERR "Failed to register WM8988 I2C driver: %d\n",
@@ -984,7 +984,7 @@ module_init(wm8988_modinit);
 
 static void __exit wm8988_exit(void)
 {
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 	i2c_del_driver(&wm8988_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-- 
cgit v1.2.2


From 3f3002692ce8fa1e9b257183ea1a36baacfdcfcf Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:48:32 -0200
Subject: AsoC: wm9081: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm9081.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 630b3d776ec2..0982c1d38ec4 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = {
 	.cache_type = REGCACHE_RBTREE,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm9081_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-- 
cgit v1.2.2


From 9a199b8e9933edf83585bac2c9030870e014381b Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:48:32 -0200
Subject: ASoC: wm9081: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm9081.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 630b3d776ec2..0982c1d38ec4 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1326,7 +1326,7 @@ static const struct regmap_config wm9081_regmap = {
 	.cache_type = REGCACHE_RBTREE,
 };
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
 static int wm9081_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-- 
cgit v1.2.2


From 7ae10ed2ee757f2ce19188e540eaa44f337c7cd2 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Thu, 21 Nov 2013 12:38:39 -0200
Subject: ASoC: cs42l52: Use IS_ENABLED() macro

Using the IS_ENABLED() macro can make the code shorter and simpler.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l52.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 8b427c977083..19ee10b6d6ca 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -50,7 +50,7 @@ struct  cs42l52_private {
 	u8 mclksel;
 	u32 mclk;
 	u8 flags;
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
 	struct input_dev *beep;
 	struct work_struct beep_work;
 	int beep_rate;
@@ -953,7 +953,7 @@ static int cs42l52_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE)
+#if IS_ENABLED(CONFIG_INPUT)
 static int beep_rates[] = {
 	261, 522, 585, 667, 706, 774, 889, 1000,
 	1043, 1200, 1333, 1412, 1600, 1714, 2000, 2182
-- 
cgit v1.2.2


From a3d36bc2aba531328f7311ef57dec7687283ec57 Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Wed, 13 Nov 2013 16:05:40 -0600
Subject: ASoC: cs42l52: Reorganize MICA/B Config and Select

This patch reworks the MICA an MICB config for single-ended or
differential and the selection of which MIC for the single config

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l52.c | 25 ++++---------------------
 1 file changed, 4 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 19ee10b6d6ca..18010639d0c5 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -233,7 +233,7 @@ static const struct soc_enum mic_bias_level_enum =
 	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
 			ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
-static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
+static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" };
 
 static const struct soc_enum mica_enum =
 	SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
@@ -243,12 +243,6 @@ static const struct soc_enum micb_enum =
 	SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
 			ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
 
-static const struct snd_kcontrol_new mica_mux =
-	SOC_DAPM_ENUM("Left Mic Input Capture Mux", mica_enum);
-
-static const struct snd_kcontrol_new micb_mux =
-	SOC_DAPM_ENUM("Right Mic Input Capture Mux", micb_enum);
-
 static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
 
 static const struct soc_enum digital_output_mux_enum =
@@ -425,6 +419,9 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
+	SOC_ENUM("MICA Select", mica_enum),
+	SOC_ENUM("MICB Select", micb_enum),
+
 	SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
 			      CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
 
@@ -550,9 +547,6 @@ static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_OUT("AIFOUTR", NULL,  0,
 			SND_SOC_NOPM, 0, 0),
 
-	SND_SOC_DAPM_MUX("MICA Mux", SND_SOC_NOPM, 0, 0, &mica_mux),
-	SND_SOC_DAPM_MUX("MICB Mux", SND_SOC_NOPM, 0, 0, &micb_mux),
-
 	SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L52_PWRCTL1, 1, 1),
 	SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L52_PWRCTL1, 2, 1),
 	SND_SOC_DAPM_PGA("PGA Left", CS42L52_PWRCTL1, 3, 1, NULL, 0),
@@ -1239,17 +1233,6 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 				cs42l52->pdata.micb_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
-	if (cs42l52->pdata.mica_sel)
-		regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
-				   CS42L52_MIC_CTL_MIC_SEL_MASK,
-				cs42l52->pdata.mica_sel <<
-				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
-	if (cs42l52->pdata.micb_sel)
-		regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
-				   CS42L52_MIC_CTL_MIC_SEL_MASK,
-				cs42l52->pdata.micb_sel <<
-				CS42L52_MIC_CTL_MIC_SEL_SHIFT);
-
 	if (cs42l52->pdata.chgfreq)
 		regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP,
 				   CS42L52_CHARGE_PUMP_MASK,
-- 
cgit v1.2.2


From 44b2ed54036ecec36ad27adf356f0274a72e5f05 Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Thu, 14 Nov 2013 11:46:11 -0600
Subject: ASoC: cs42l52: Make MICA/B mixer dependent on mic config

MICA/B Single-Ended input selection depends on mica/b config so lets
make the mixer controls for them only show for selected mic's

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l52.c | 37 ++++++++++++++++++++++++++++++-------
 1 file changed, 30 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 18010639d0c5..78d2dd669e89 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -419,9 +419,6 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
 
-	SOC_ENUM("MICA Select", mica_enum),
-	SOC_ENUM("MICB Select", micb_enum),
-
 	SOC_DOUBLE_R_TLV("MIC Gain Volume", CS42L52_MICA_CTL,
 			      CS42L52_MICB_CTL, 0, 0x10, 0, mic_tlv),
 
@@ -528,6 +525,30 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 };
 
+static const struct snd_kcontrol_new cs42l52_mica_controls[] = {
+	SOC_ENUM("MICA Select", mica_enum),
+};
+
+static const struct snd_kcontrol_new cs42l52_micb_controls[] = {
+	SOC_ENUM("MICB Select", micb_enum),
+};
+
+static int cs42l52_add_mic_controls(struct snd_soc_codec *codec)
+{
+	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
+	struct cs42l52_platform_data *pdata = &cs42l52->pdata;
+
+	if (!pdata->mica_diff_cfg)
+		snd_soc_add_codec_controls(codec, cs42l52_mica_controls,
+				     ARRAY_SIZE(cs42l52_mica_controls));
+
+	if (!pdata->micb_diff_cfg)
+		snd_soc_add_codec_controls(codec, cs42l52_micb_controls,
+				     ARRAY_SIZE(cs42l52_micb_controls));
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget cs42l52_dapm_widgets[] = {
 
 	SND_SOC_DAPM_INPUT("AIN1L"),
@@ -1104,6 +1125,8 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
 	}
 	regcache_cache_only(cs42l52->regmap, true);
 
+	cs42l52_add_mic_controls(codec);
+
 	cs42l52_init_beep(codec);
 
 	cs42l52_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1221,16 +1244,16 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 			reg & 0xFF);
 
 	/* Set Platform Data */
-	if (cs42l52->pdata.mica_cfg)
+	if (cs42l52->pdata.mica_diff_cfg)
 		regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL,
 				   CS42L52_MIC_CTL_TYPE_MASK,
-				cs42l52->pdata.mica_cfg <<
+				cs42l52->pdata.mica_diff_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
-	if (cs42l52->pdata.micb_cfg)
+	if (cs42l52->pdata.micb_diff_cfg)
 		regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL,
 				   CS42L52_MIC_CTL_TYPE_MASK,
-				cs42l52->pdata.micb_cfg <<
+				cs42l52->pdata.micb_diff_cfg <<
 				CS42L52_MIC_CTL_TYPE_SHIFT);
 
 	if (cs42l52->pdata.chgfreq)
-- 
cgit v1.2.2


From 391fc59db87615e07e8a6ab5fbffe3cc04f2b19c Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Fri, 15 Nov 2013 09:35:33 -0600
Subject: ASoC: cs42l52: Add devicetree support for CS42L52

This patch adds device tree support for the CS42L52 Codec

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l52.c | 45 ++++++++++++++++++++++++++++++++++++++++++---
 1 file changed, 42 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 78d2dd669e89..4a47a634f376 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -17,7 +17,7 @@
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
-#include <linux/gpio.h>
+#include <linux/of_gpio.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/input.h>
@@ -1193,6 +1193,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 	int ret;
 	unsigned int devid = 0;
 	unsigned int reg;
+	u32 val32;
 
 	cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private),
 			       GFP_KERNEL);
@@ -1206,9 +1207,39 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
 		return ret;
 	}
-
-	if (pdata)
+	if (pdata) {
+		cs42l52->pdata = *pdata;
+	} else {
+		pdata = devm_kzalloc(&i2c_client->dev,
+				     sizeof(struct cs42l52_platform_data),
+				GFP_KERNEL);
+		if (!pdata) {
+			dev_err(&i2c_client->dev, "could not allocate pdata\n");
+			return -ENOMEM;
+		}
+		if (i2c_client->dev.of_node) {
+			if (of_property_read_bool(i2c_client->dev.of_node,
+				"cirrus,mica-differential-cfg"))
+				pdata->mica_diff_cfg = true;
+
+			if (of_property_read_bool(i2c_client->dev.of_node,
+				"cirrus,micb-differential-cfg"))
+				pdata->micb_diff_cfg = true;
+
+			if (of_property_read_u32(i2c_client->dev.of_node,
+				"cirrus,micbias-lvl", &val32) >= 0)
+				pdata->micbias_lvl = val32;
+
+			if (of_property_read_u32(i2c_client->dev.of_node,
+				"cirrus,chgfreq-divisor", &val32) >= 0)
+				pdata->chgfreq_divisor = val32;
+
+			pdata->reset_gpio =
+				of_get_named_gpio(i2c_client->dev.of_node,
+						"cirrus,reset-gpio", 0);
+		}
 		cs42l52->pdata = *pdata;
+	}
 
 	if (cs42l52->pdata.reset_gpio) {
 		ret = gpio_request_one(cs42l52->pdata.reset_gpio,
@@ -1280,6 +1311,13 @@ static int cs42l52_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+static const struct of_device_id cs42l52_of_match[] = {
+	{ .compatible = "cirrus,cs42l52", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, cs42l52_of_match);
+
+
 static const struct i2c_device_id cs42l52_id[] = {
 	{ "cs42l52", 0 },
 	{ }
@@ -1290,6 +1328,7 @@ static struct i2c_driver cs42l52_i2c_driver = {
 	.driver = {
 		.name = "cs42l52",
 		.owner = THIS_MODULE,
+		.of_match_table = cs42l52_of_match,
 	},
 	.id_table = cs42l52_id,
 	.probe =    cs42l52_i2c_probe,
-- 
cgit v1.2.2


From 21585ee848078b12d0d1a513e93936bf96b444a0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:32 +0100
Subject: ASoC: Add resource managed snd_dmaengine_pcm_register()

For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-devres.c | 41 +++++++++++++++++++++++++++++++++++++++++
 1 file changed, 41 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..999861942d28 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -12,6 +12,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
 static void devm_component_release(struct device *dev, void *res)
 {
@@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 	return ret;
 }
 EXPORT_SYMBOL_GPL(devm_snd_soc_register_card);
+
+#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM
+
+static void devm_dmaengine_pcm_release(struct device *dev, void *res)
+{
+	snd_dmaengine_pcm_unregister(*(struct device **)res);
+}
+
+/**
+ * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration
+ * @dev: The parent device for the PCM device
+ * @config: Platform specific PCM configuration
+ * @flags: Platform specific quirks
+ *
+ * Register a dmaengine based PCM device with automatic unregistration when the
+ * device is unregistered.
+ */
+int devm_snd_dmaengine_pcm_register(struct device *dev,
+	const struct snd_dmaengine_pcm_config *config, unsigned int flags)
+{
+	struct device **ptr;
+	int ret;
+
+	ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL);
+	if (!ptr)
+		return -ENOMEM;
+
+	ret = snd_dmaengine_pcm_register(dev, config, flags);
+	if (ret == 0) {
+		*ptr = dev;
+		devres_add(dev, ptr);
+	} else {
+		devres_free(ptr);
+	}
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register);
+
+#endif
-- 
cgit v1.2.2


From d71b3ef44f9e5cfda2499768f6420b784845af06 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:36 +0100
Subject: ASoC: spear: Use devm_snd_dmaengine_pcm_register

Makes the code slightly shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/spear/spear_pcm.c | 9 +--------
 1 file changed, 1 insertion(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 4707f2b862c3..9a02141666ea 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -49,18 +49,12 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = {
 
 static int spear_soc_platform_probe(struct platform_device *pdev)
 {
-	return snd_dmaengine_pcm_register(&pdev->dev,
+	return devm_snd_dmaengine_pcm_register(&pdev->dev,
 		&spear_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 
-static int spear_soc_platform_remove(struct platform_device *pdev)
-{
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-	return 0;
-}
-
 static struct platform_driver spear_pcm_driver = {
 	.driver = {
 		.name = "spear-pcm-audio",
@@ -68,7 +62,6 @@ static struct platform_driver spear_pcm_driver = {
 	},
 
 	.probe = spear_soc_platform_probe,
-	.remove = spear_soc_platform_remove,
 };
 
 module_platform_driver(spear_pcm_driver);
-- 
cgit v1.2.2


From d733dc0828cfb230171ae7420a6e8c344ec8473a Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 28 Nov 2013 16:37:51 +0000
Subject: ASoC: wm_adsp: Stop region iteration when the desired region is found

When locating the memory region relating to a coefficient block written
through a bin file we keep processing the list of regions even after we
have found the region we require. This patch adds a break, so we don't
process redundant list items.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm_adsp.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..b42f9af163c8 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1286,6 +1286,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 					reg = wm_adsp_region_to_reg(mem,
 								    reg);
 					reg += offset;
+					break;
 				}
 			}
 
-- 
cgit v1.2.2


From 17b9a2b78586c42916a2bfc55ea6c0ef962b2c1e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:42:49 -0800
Subject: ASoC: rcar: remove unused register settings

AUDIO_CLK_SEL4/5 are not used

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c  | 3 ---
 sound/soc/sh/rcar/rsnd.h | 3 ---
 2 files changed, 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 61212ee97c28..b94d4cefa8b8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -208,9 +208,6 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 		RSND_GEN1_S_REG(gen, ADG,	SSICKR,		0x08),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL0,	0x0c),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL1,	0x10),
-		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL3,	0x18),
-		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL4,	0x1c),
-		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL5,	0x20),
 
 		RSND_GEN1_M_REG(gen, SSI,	SSICR,		0x00,	0x40),
 		RSND_GEN1_M_REG(gen, SSI,	SSISR,		0x04,	0x40),
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index b5ac3a2afc58..63a9d7081c60 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -49,9 +49,6 @@ enum rsnd_reg {
 	RSND_REG_AUDIO_CLK_SEL0,
 	RSND_REG_AUDIO_CLK_SEL1,
 	RSND_REG_AUDIO_CLK_SEL2,
-	RSND_REG_AUDIO_CLK_SEL3,
-	RSND_REG_AUDIO_CLK_SEL4,
-	RSND_REG_AUDIO_CLK_SEL5,
 
 	/* SSI */
 	RSND_REG_SSICR,
-- 
cgit v1.2.2


From c1e6cc5e577d1d446a645aea02d28a924f20b834 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:43:01 -0800
Subject: ASoC: rcar: separate regmap init common field

The repmap initialization difference between Gen1/Gen2 is
only register offset.
This patch separates rsnd_gen1_regmap_init()
into common part and Gen1 specific part.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c | 49 ++++++++++++++++++++++++++++---------------------
 1 file changed, 28 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index b94d4cefa8b8..0ebea44e890f 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -115,6 +115,33 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
 				  mask, data);
 }
 
+static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
+				struct rsnd_gen  *gen,
+				struct reg_field *regf)
+{
+	int i;
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct regmap_config regc;
+
+	memset(&regc, 0, sizeof(regc));
+	regc.reg_bits = 32;
+	regc.val_bits = 32;
+
+	gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, &regc);
+	if (IS_ERR(gen->regmap)) {
+		dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap));
+		return PTR_ERR(gen->regmap);
+	}
+
+	for (i = 0; i < RSND_REG_MAX; i++) {
+		gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]);
+		if (IS_ERR(gen->regs[i]))
+			return PTR_ERR(gen->regs[i]);
+
+	}
+
+	return 0;
+}
 /*
  *		Gen2
  *		will be filled in the future
@@ -189,9 +216,6 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv,
 
 static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 {
-	int i;
-	struct device *dev = rsnd_priv_to_dev(priv);
-	struct regmap_config regc;
 	struct reg_field regf[RSND_REG_MAX] = {
 		RSND_GEN1_S_REG(gen, SRU,	SRC_ROUTE_SEL,	0x00),
 		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL0,	0x08),
@@ -216,24 +240,7 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 		RSND_GEN1_M_REG(gen, SSI,	SSIWSR,		0x20,	0x40),
 	};
 
-	memset(&regc, 0, sizeof(regc));
-	regc.reg_bits = 32;
-	regc.val_bits = 32;
-
-	gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, &regc);
-	if (IS_ERR(gen->regmap)) {
-		dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap));
-		return PTR_ERR(gen->regmap);
-	}
-
-	for (i = 0; i < RSND_REG_MAX; i++) {
-		gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]);
-		if (IS_ERR(gen->regs[i]))
-			return PTR_ERR(gen->regs[i]);
-
-	}
-
-	return 0;
+	return rsnd_gen_regmap_init(priv, gen, regf);
 }
 
 static int rsnd_gen1_probe(struct platform_device *pdev,
-- 
cgit v1.2.2


From 42ee5d22e3d2550a49bc5d3e6f19c92da9a19446 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:43:13 -0800
Subject: ASoC: rcar: add rsnd_is_accessible_reg()

Current rcar driver is supporting Gen1,
and Gen2 will be supported soon.
Then, some registers are used from Gen1 only,
or from Gen2 only.
To avoid NULL pointer access, this patch adds
register accessible check function.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c | 26 ++++++++++++++++++++++++++
 1 file changed, 26 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 0ebea44e890f..970439d5ec0e 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -86,12 +86,28 @@ static struct regmap_bus rsnd_regmap_bus = {
 	.val_format_endian_default	= REGMAP_ENDIAN_NATIVE,
 };
 
+static int rsnd_is_accessible_reg(struct rsnd_priv *priv,
+				  struct rsnd_gen *gen, enum rsnd_reg reg)
+{
+	if (!gen->regs[reg]) {
+		struct device *dev = rsnd_priv_to_dev(priv);
+
+		dev_err(dev, "unsupported register access %x\n", reg);
+		return 0;
+	}
+
+	return 1;
+}
+
 u32 rsnd_read(struct rsnd_priv *priv,
 	      struct rsnd_mod *mod, enum rsnd_reg reg)
 {
 	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
 	u32 val;
 
+	if (!rsnd_is_accessible_reg(priv, gen, reg))
+		return 0;
+
 	regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val);
 
 	return val;
@@ -103,6 +119,9 @@ void rsnd_write(struct rsnd_priv *priv,
 {
 	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
 
+	if (!rsnd_is_accessible_reg(priv, gen, reg))
+		return;
+
 	regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data);
 }
 
@@ -111,6 +130,9 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod,
 {
 	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
 
+	if (!rsnd_is_accessible_reg(priv, gen, reg))
+		return;
+
 	regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod),
 				  mask, data);
 }
@@ -134,6 +156,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
 	}
 
 	for (i = 0; i < RSND_REG_MAX; i++) {
+		gen->regs[i] = NULL;
+		if (!regf[i].reg)
+			continue;
+
 		gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]);
 		if (IS_ERR(gen->regs[i]))
 			return PTR_ERR(gen->regs[i]);
-- 
cgit v1.2.2


From 994a9df1e3ea3244e94309e4f893e9a5121116c9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:43:23 -0800
Subject: ASoC: rcar: remove .path_init/exit from rsnd_gen_ops

rsnd_gen_ops has .path_init/exit callback function
which cares SRU/SSI (if Gen1) SCU/SSIU/SSI (if Gen2)
path settings.
But, the differences between Gen1/Gen2 are cared
in ssi.c/scu.c, and the path itself is same in Gen1/Gen2.
This patch removes .path_init/exit callback.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c | 54 ++++++++++++++-----------------------------------
 1 file changed, 15 insertions(+), 39 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 970439d5ec0e..4f2eaa3262d7 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -16,12 +16,6 @@ struct rsnd_gen_ops {
 		     struct rsnd_priv *priv);
 	void (*remove)(struct platform_device *pdev,
 		      struct rsnd_priv *priv);
-	int (*path_init)(struct rsnd_priv *priv,
-			 struct rsnd_dai *rdai,
-			 struct rsnd_dai_stream *io);
-	int (*path_exit)(struct rsnd_priv *priv,
-			 struct rsnd_dai *rdai,
-			 struct rsnd_dai_stream *io);
 };
 
 struct rsnd_gen {
@@ -168,17 +162,10 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
 
 	return 0;
 }
-/*
- *		Gen2
- *		will be filled in the future
- */
 
-/*
- *		Gen1
- */
-static int rsnd_gen1_path_init(struct rsnd_priv *priv,
-			       struct rsnd_dai *rdai,
-			       struct rsnd_dai_stream *io)
+int rsnd_gen_path_init(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io)
 {
 	struct rsnd_mod *mod;
 	int ret;
@@ -216,9 +203,9 @@ static int rsnd_gen1_path_init(struct rsnd_priv *priv,
 	return ret;
 }
 
-static int rsnd_gen1_path_exit(struct rsnd_priv *priv,
-			       struct rsnd_dai *rdai,
-			       struct rsnd_dai_stream *io)
+int rsnd_gen_path_exit(struct rsnd_priv *priv,
+		       struct rsnd_dai *rdai,
+		       struct rsnd_dai_stream *io)
 {
 	struct rsnd_mod *mod, *n;
 	int ret = 0;
@@ -232,6 +219,15 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv,
 	return ret;
 }
 
+/*
+ *		Gen2
+ *		will be filled in the future
+ */
+
+/*
+ *		Gen1
+ */
+
 /* single address mapping */
 #define RSND_GEN1_S_REG(gen, reg, id, offset)	\
 	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9)
@@ -319,31 +315,11 @@ static void rsnd_gen1_remove(struct platform_device *pdev,
 static struct rsnd_gen_ops rsnd_gen1_ops = {
 	.probe		= rsnd_gen1_probe,
 	.remove		= rsnd_gen1_remove,
-	.path_init	= rsnd_gen1_path_init,
-	.path_exit	= rsnd_gen1_path_exit,
 };
 
 /*
  *		Gen
  */
-int rsnd_gen_path_init(struct rsnd_priv *priv,
-		       struct rsnd_dai *rdai,
-		       struct rsnd_dai_stream *io)
-{
-	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
-
-	return gen->ops->path_init(priv, rdai, io);
-}
-
-int rsnd_gen_path_exit(struct rsnd_priv *priv,
-		       struct rsnd_dai *rdai,
-		       struct rsnd_dai_stream *io)
-{
-	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
-
-	return gen->ops->path_exit(priv, rdai, io);
-}
-
 int rsnd_gen_probe(struct platform_device *pdev,
 		   struct rcar_snd_info *info,
 		   struct rsnd_priv *priv)
-- 
cgit v1.2.2


From 531eaf491e25ce215bbcf434e23e77f53fc98171 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:43:34 -0800
Subject: ASoC: rcar: remove rcar_gen_ops

Current rcar driver gen.c is using rcar_gen_ops
which was made with the assumption that
Gen1 and Gen2 need different behavior.
but it was not needed.
This patch removes unnecessary complex method.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c | 33 ++++++---------------------------
 1 file changed, 6 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 4f2eaa3262d7..a29e36eb1a30 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -10,14 +10,6 @@
  */
 #include "rsnd.h"
 
-struct rsnd_gen_ops {
-	int (*probe)(struct platform_device *pdev,
-		     struct rcar_snd_info *info,
-		     struct rsnd_priv *priv);
-	void (*remove)(struct platform_device *pdev,
-		      struct rsnd_priv *priv);
-};
-
 struct rsnd_gen {
 	void __iomem *base[RSND_BASE_MAX];
 
@@ -307,16 +299,6 @@ static int rsnd_gen1_probe(struct platform_device *pdev,
 
 }
 
-static void rsnd_gen1_remove(struct platform_device *pdev,
-			     struct rsnd_priv *priv)
-{
-}
-
-static struct rsnd_gen_ops rsnd_gen1_ops = {
-	.probe		= rsnd_gen1_probe,
-	.remove		= rsnd_gen1_remove,
-};
-
 /*
  *		Gen
  */
@@ -326,6 +308,7 @@ int rsnd_gen_probe(struct platform_device *pdev,
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_gen *gen;
+	int ret;
 
 	gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL);
 	if (!gen) {
@@ -333,23 +316,19 @@ int rsnd_gen_probe(struct platform_device *pdev,
 		return -ENOMEM;
 	}
 
-	if (rsnd_is_gen1(priv))
-		gen->ops = &rsnd_gen1_ops;
+	priv->gen = gen;
 
-	if (!gen->ops) {
+	if (rsnd_is_gen1(priv)) {
+		ret = rsnd_gen1_probe(pdev, info, priv);
+	} else {
 		dev_err(dev, "unknown generation R-Car sound device\n");
 		return -ENODEV;
 	}
 
-	priv->gen = gen;
-
-	return gen->ops->probe(pdev, info, priv);
+	return ret;
 }
 
 void rsnd_gen_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv)
 {
-	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
-
-	gen->ops->remove(pdev, priv);
 }
-- 
cgit v1.2.2


From 507d466c733e28d132a7be87040a3da126df7947 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 28 Nov 2013 18:43:45 -0800
Subject: ASoC: rcar: add Gen2 sound support

This patch adds Gen2 sound support for Renesas R-Car.
But, it is supporting PIO transfer only at this point

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c  | 91 +++++++++++++++++++++++++++++++++++++++++++++---
 sound/soc/sh/rcar/rsnd.h |  3 +-
 sound/soc/sh/rcar/ssi.c  |  3 ++
 3 files changed, 91 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index a29e36eb1a30..bf066f73ef05 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -213,9 +213,88 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv,
 
 /*
  *		Gen2
- *		will be filled in the future
  */
 
+/* single address mapping */
+#define RSND_GEN2_S_REG(gen, reg, id, offset)				\
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9)
+
+/* multi address mapping */
+#define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset)		\
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9)
+
+static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
+{
+	struct reg_field regf[RSND_REG_MAX] = {
+		RSND_GEN2_S_REG(gen, SSIU,	SSI_MODE0,	0x800),
+		RSND_GEN2_S_REG(gen, SSIU,	SSI_MODE1,	0x804),
+		/* FIXME: it needs SSI_MODE2/3 in the future */
+		RSND_GEN2_M_REG(gen, SSIU,	INT_ENABLE,	0x18,	0x80),
+
+		RSND_GEN2_S_REG(gen, ADG,	BRRA,		0x00),
+		RSND_GEN2_S_REG(gen, ADG,	BRRB,		0x04),
+		RSND_GEN2_S_REG(gen, ADG,	SSICKR,		0x08),
+		RSND_GEN2_S_REG(gen, ADG,	AUDIO_CLK_SEL0,	0x0c),
+		RSND_GEN2_S_REG(gen, ADG,	AUDIO_CLK_SEL1,	0x10),
+		RSND_GEN2_S_REG(gen, ADG,	AUDIO_CLK_SEL2,	0x14),
+
+		RSND_GEN2_M_REG(gen, SSI,	SSICR,		0x00,	0x40),
+		RSND_GEN2_M_REG(gen, SSI,	SSISR,		0x04,	0x40),
+		RSND_GEN2_M_REG(gen, SSI,	SSITDR,		0x08,	0x40),
+		RSND_GEN2_M_REG(gen, SSI,	SSIRDR,		0x0c,	0x40),
+		RSND_GEN2_M_REG(gen, SSI,	SSIWSR,		0x20,	0x40),
+	};
+
+	return rsnd_gen_regmap_init(priv, gen, regf);
+}
+
+static int rsnd_gen2_probe(struct platform_device *pdev,
+			   struct rcar_snd_info *info,
+			   struct rsnd_priv *priv)
+{
+	struct device *dev = rsnd_priv_to_dev(priv);
+	struct rsnd_gen *gen = rsnd_priv_to_gen(priv);
+	struct resource *scu_res;
+	struct resource *adg_res;
+	struct resource *ssiu_res;
+	struct resource *ssi_res;
+	int ret;
+
+	/*
+	 * map address
+	 */
+	scu_res  = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SCU);
+	adg_res  = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_ADG);
+	ssiu_res = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSIU);
+	ssi_res  = platform_get_resource(pdev, IORESOURCE_MEM, RSND_GEN2_SSI);
+
+	gen->base[RSND_GEN2_SCU]  = devm_ioremap_resource(dev, scu_res);
+	gen->base[RSND_GEN2_ADG]  = devm_ioremap_resource(dev, adg_res);
+	gen->base[RSND_GEN2_SSIU] = devm_ioremap_resource(dev, ssiu_res);
+	gen->base[RSND_GEN2_SSI]  = devm_ioremap_resource(dev, ssi_res);
+	if (IS_ERR(gen->base[RSND_GEN2_SCU])  ||
+	    IS_ERR(gen->base[RSND_GEN2_ADG])  ||
+	    IS_ERR(gen->base[RSND_GEN2_SSIU]) ||
+	    IS_ERR(gen->base[RSND_GEN2_SSI]))
+		return -ENODEV;
+
+	ret = rsnd_gen2_regmap_init(priv, gen);
+	if (ret < 0)
+		return ret;
+
+	dev_dbg(dev, "Gen2 device probed\n");
+	dev_dbg(dev, "SRU  : %08x => %p\n", scu_res->start,
+		gen->base[RSND_GEN2_SCU]);
+	dev_dbg(dev, "ADG  : %08x => %p\n", adg_res->start,
+		gen->base[RSND_GEN2_ADG]);
+	dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start,
+		gen->base[RSND_GEN2_SSIU]);
+	dev_dbg(dev, "SSI  : %08x => %p\n", ssi_res->start,
+		gen->base[RSND_GEN2_SSI]);
+
+	return 0;
+}
+
 /*
  *		Gen1
  */
@@ -318,12 +397,14 @@ int rsnd_gen_probe(struct platform_device *pdev,
 
 	priv->gen = gen;
 
-	if (rsnd_is_gen1(priv)) {
+	ret = -ENODEV;
+	if (rsnd_is_gen1(priv))
 		ret = rsnd_gen1_probe(pdev, info, priv);
-	} else {
+	else if (rsnd_is_gen2(priv))
+		ret = rsnd_gen2_probe(pdev, info, priv);
+
+	if (ret < 0)
 		dev_err(dev, "unknown generation R-Car sound device\n");
-		return -ENODEV;
-	}
 
 	return ret;
 }
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 63a9d7081c60..bff7b9e53066 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -31,7 +31,7 @@
  * see gen1/gen2 for detail
  */
 enum rsnd_reg {
-	/* SRU/SCU */
+	/* SRU/SCU/SSIU */
 	RSND_REG_SRC_ROUTE_SEL,
 	RSND_REG_SRC_TMG_SEL0,
 	RSND_REG_SRC_TMG_SEL1,
@@ -41,6 +41,7 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
 	RSND_REG_BUSIF_ADINR,
+	RSND_REG_INT_ENABLE,
 
 	/* ADG */
 	RSND_REG_BRRA,
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index b71cf9d7dd3f..477465f9507b 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -457,6 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
 	/* enable PIO IRQ */
 	ssi->cr_etc = UIEN | OIEN | DIEN;
 
+	/* enable PIO interrupt */
+	rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000);
+
 	rsnd_ssi_hw_start(ssi, rdai, io);
 
 	dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-- 
cgit v1.2.2


From 69ae8489076fa0fa98609155434f3c286c7364a0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Fri, 29 Nov 2013 11:42:18 +0000
Subject: ASoC: cs42l52: Fix build

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l52.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 4a47a634f376..0bac6d5a4ac8 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1232,7 +1232,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 
 			if (of_property_read_u32(i2c_client->dev.of_node,
 				"cirrus,chgfreq-divisor", &val32) >= 0)
-				pdata->chgfreq_divisor = val32;
+				pdata->chgfreq = val32;
 
 			pdata->reset_gpio =
 				of_get_named_gpio(i2c_client->dev.of_node,
-- 
cgit v1.2.2


From 4b016931a997be59a2be0da3398985a19fdfa5be Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Thu, 28 Nov 2013 11:55:09 +0100
Subject: ALSA: hda/realtek - Add more codecs alias name for Dell

Dell assigned alias name for more codecs.
ALC3220 ALC3221 ALC3223 ALC3226 ALC3234 ALC3661.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++++++++-
 1 file changed, 30 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a9a1b0c1dbe..01fadf0e0b3c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -903,7 +903,7 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name)
 }
 
 /*
- * Rename codecs appropriately from COEF value
+ * Rename codecs appropriately from COEF value or subvendor id
  */
 struct alc_codec_rename_table {
 	unsigned int vendor_id;
@@ -912,6 +912,13 @@ struct alc_codec_rename_table {
 	const char *name;
 };
 
+struct alc_codec_rename_pci_table {
+	unsigned int codec_vendor_id;
+	unsigned short pci_subvendor;
+	unsigned short pci_subdevice;
+	const char *name;
+};
+
 static struct alc_codec_rename_table rename_tbl[] = {
 	{ 0x10ec0269, 0xfff0, 0x3010, "ALC277" },
 	{ 0x10ec0269, 0xf0f0, 0x2010, "ALC259" },
@@ -931,9 +938,20 @@ static struct alc_codec_rename_table rename_tbl[] = {
 	{ } /* terminator */
 };
 
+static struct alc_codec_rename_pci_table rename_pci_tbl[] = {
+	{ 0x10ec0280, 0x1028, 0, "ALC3220" },
+	{ 0x10ec0282, 0x1028, 0, "ALC3221" },
+	{ 0x10ec0283, 0x1028, 0, "ALC3223" },
+	{ 0x10ec0292, 0x1028, 0, "ALC3226" },
+	{ 0x10ec0255, 0x1028, 0, "ALC3234" },
+	{ 0x10ec0668, 0x1028, 0, "ALC3661" },
+	{ } /* terminator */
+};
+
 static int alc_codec_rename_from_preset(struct hda_codec *codec)
 {
 	const struct alc_codec_rename_table *p;
+	const struct alc_codec_rename_pci_table *q;
 
 	for (p = rename_tbl; p->vendor_id; p++) {
 		if (p->vendor_id != codec->vendor_id)
@@ -941,6 +959,17 @@ static int alc_codec_rename_from_preset(struct hda_codec *codec)
 		if ((alc_get_coef0(codec) & p->coef_mask) == p->coef_bits)
 			return alc_codec_rename(codec, p->name);
 	}
+
+	for (q = rename_pci_tbl; q->codec_vendor_id; q++) {
+		if (q->codec_vendor_id != codec->vendor_id)
+			continue;
+		if (q->pci_subvendor != codec->bus->pci->subsystem_vendor)
+			continue;
+		if (!q->pci_subdevice ||
+		    q->pci_subdevice == codec->bus->pci->subsystem_device)
+			return alc_codec_rename(codec, q->name);
+	}
+
 	return 0;
 }
 
-- 
cgit v1.2.2


From 97a2657076f463ef06b6ef4ab6d54bb903ea9ff3 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Fri, 29 Nov 2013 00:35:26 -0500
Subject: ALSA: hda/realtek - Remove depop delay for suspend and resume if
 applicable

This patch defines a flag "no_depop_delay" in alc_spec. If this flag is set,
delay in alc_eapd_shutup and alc_resume will be skipped.

Machine-specific fixup can set this flag to reduce suspend/resume time, if
the codec and hardware analog design can avoid pop noise without this delay.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 15 +++++++++++----
 1 file changed, 11 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 01fadf0e0b3c..521a9d281a5f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -118,7 +118,8 @@ struct alc_spec {
 
 	int init_amp;
 	int codec_variant;	/* flag for other variants */
-	bool has_alc5505_dsp;
+	unsigned int has_alc5505_dsp:1;
+	unsigned int no_depop_delay:1;
 
 	/* for PLL fix */
 	hda_nid_t pll_nid;
@@ -280,8 +281,11 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on)
  */
 static void alc_eapd_shutup(struct hda_codec *codec)
 {
+	struct alc_spec *spec = codec->spec;
+
 	alc_auto_setup_eapd(codec, false);
-	msleep(200);
+	if (!spec->no_depop_delay)
+		msleep(200);
 	snd_hda_shutup_pins(codec);
 }
 
@@ -863,7 +867,10 @@ static int alc_suspend(struct hda_codec *codec)
 #ifdef CONFIG_PM
 static int alc_resume(struct hda_codec *codec)
 {
-	msleep(150); /* to avoid pop noise */
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->no_depop_delay)
+		msleep(150); /* to avoid pop noise */
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
@@ -4457,7 +4464,7 @@ static int patch_alc269(struct hda_codec *codec)
 	}
 
 	if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) {
-		spec->has_alc5505_dsp = true;
+		spec->has_alc5505_dsp = 1;
 		spec->init_hook = alc5505_dsp_init;
 	}
 
-- 
cgit v1.2.2


From b5c6611fb4ef631cfb484a7cfe916154e52bc7d9 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Fri, 29 Nov 2013 00:35:35 -0500
Subject: ALSA: hda - fixup ALC262 to remove depop delay on Intel BayleyBay
 board

This patch defines a fixup 'alc_fixup_no_depop_delay', which sets alc_spec flag
'no_depop_delay' to indicate skipping delay in Realtek codec suspend/resume.

And Intel Baytrail BayleyBay board applies this fixup to reduce driver
suspend/resume time.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 521a9d281a5f..b092dfd2670e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -369,6 +369,15 @@ static void alc_fixup_sku_ignore(struct hda_codec *codec,
 	}
 }
 
+static void alc_fixup_no_depop_delay(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PROBE)
+		spec->no_depop_delay = 1;
+}
+
 static int alc_auto_parse_customize_define(struct hda_codec *codec)
 {
 	unsigned int ass, tmp, i;
@@ -2308,6 +2317,7 @@ enum {
 	ALC262_FIXUP_BENQ,
 	ALC262_FIXUP_BENQ_T31,
 	ALC262_FIXUP_INV_DMIC,
+	ALC262_FIXUP_INTEL_BAYLEYBAY,
 };
 
 static const struct hda_fixup alc262_fixups[] = {
@@ -2372,6 +2382,10 @@ static const struct hda_fixup alc262_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic_0x12,
 	},
+	[ALC262_FIXUP_INTEL_BAYLEYBAY] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_no_depop_delay,
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
@@ -2383,6 +2397,7 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000),
 	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ),
 	SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_FIXUP_BENQ_T31),
+	SND_PCI_QUIRK(0x8086, 0x7270, "BayleyBay", ALC262_FIXUP_INTEL_BAYLEYBAY),
 	{}
 };
 
-- 
cgit v1.2.2


From 7f132927435cf09d26ceb81cd0043b542ac7206d Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Fri, 29 Nov 2013 01:48:45 -0500
Subject: ALSA: hda - allow a codec to define its own depop delay time

This patch adds 'depop_delay' to struct hda_codec, to indicate a depop delay
time in ms when power-down, in function set_power_state() to D3.

Default value is -1, for a default delay time.
Machine fixup can set a suitable value according to the codec chip and HW audio
design.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 7 +++++--
 sound/pci/hda/hda_codec.h | 2 ++
 2 files changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 472e5d04701e..5b7622034eee 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1471,6 +1471,7 @@ int snd_hda_codec_new(struct hda_bus *bus,
 	INIT_LIST_HEAD(&codec->conn_list);
 
 	INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work);
+	codec->depop_delay = -1;
 
 #ifdef CONFIG_PM
 	spin_lock_init(&codec->power_lock);
@@ -3975,8 +3976,10 @@ static unsigned int hda_set_power_state(struct hda_codec *codec,
 
 	/* this delay seems necessary to avoid click noise at power-down */
 	if (power_state == AC_PWRST_D3) {
-		/* transition time less than 10ms for power down */
-		msleep(codec->epss ? 10 : 100);
+		if (codec->depop_delay < 0)
+			msleep(codec->epss ? 10 : 100);
+		else if (codec->depop_delay > 0)
+			msleep(codec->depop_delay);
 		flags = HDA_RW_NO_RESPONSE_FALLBACK;
 	}
 
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 01b248d3f7b2..3ab4834761a8 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -944,6 +944,8 @@ struct hda_codec {
 	struct snd_array jacks;
 #endif
 
+	int depop_delay; /* depop delay in ms, -1 for default delay time */
+
 	/* fix-up list */
 	int fixup_id;
 	const struct hda_fixup *fixup_list;
-- 
cgit v1.2.2


From 4ded61eb3ea87e9c563e09662be3ed5e942ff2a2 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Mon, 2 Dec 2013 00:41:24 -0200
Subject: ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure

No need to have a specific OOM message, since there is generic MM out of memory
message in place.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/imx-spdif.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 8499d5292f08..980dd1fc9113 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -35,7 +35,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 
 	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
 	if (!data) {
-		dev_err(&pdev->dev, "failed to allocate memory\n");
 		ret = -ENOMEM;
 		goto end;
 	}
-- 
cgit v1.2.2


From a22f33b00346f26d29483cdacdbf26df7947ef23 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 30 Nov 2013 18:00:45 +0100
Subject: ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is
 not supported

For dmaengine drivers which do not support transfer residue reporting we update
the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in
this case to let userspace know about this.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96ce1f4..87e86357124e 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -137,6 +137,9 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 	hw.buffer_bytes_max = SIZE_MAX;
 	hw.fifo_size = dma_data->fifo_size;
 
+	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
+		hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	ret = dma_get_slave_caps(chan, &dma_caps);
 	if (ret == 0) {
 		if (dma_caps.cmd_pause)
-- 
cgit v1.2.2


From 62e5f676f6a063e1ab0d6b8fcaef2feb026ee00e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 30 Nov 2013 17:38:58 +0100
Subject: ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry
 constraints

If there are symmetry constraints between the playback and the capture channel
set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 18 ++++++++++++++++++
 1 file changed, 18 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 170ff9695753..f3592f142832 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -180,6 +180,21 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver;
+	struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver;
+	struct snd_soc_dai_link *link = rtd->dai_link;
+
+	return cpu_driver->symmetric_rates || codec_driver->symmetric_rates ||
+		link->symmetric_rates || cpu_driver->symmetric_channels ||
+		codec_driver->symmetric_channels || link->symmetric_channels ||
+		cpu_driver->symmetric_samplebits ||
+		codec_driver->symmetric_samplebits ||
+		link->symmetric_samplebits;
+}
+
 /*
  * List of sample sizes that might go over the bus for parameter
  * application.  There ought to be a wildcard sample size for things
@@ -309,6 +324,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			&cpu_dai_drv->capture);
 	}
 
+	if (soc_pcm_has_symmetry(substream))
+		runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
+
 	ret = -EINVAL;
 	snd_pcm_limit_hw_rates(runtime);
 	if (!runtime->hw.rates) {
-- 
cgit v1.2.2


From 3990c516de66af940c5c366a81531787aefe81ae Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:33 +0100
Subject: ASoC: bcm2835-i2s: Use devm_snd_dmaengine_pcm_register()

Makes the code slightly shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/bcm/bcm2835-i2s.c | 9 +--------
 1 file changed, 1 insertion(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c
index f49b007c8b43..2685fe4f8427 100644
--- a/sound/soc/bcm/bcm2835-i2s.c
+++ b/sound/soc/bcm/bcm2835-i2s.c
@@ -848,7 +848,7 @@ static int bcm2835_i2s_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register PCM: %d\n", ret);
 		return ret;
@@ -862,15 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = {
 	{},
 };
 
-static int bcm2835_i2s_remove(struct platform_device *pdev)
-{
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-	return 0;
-}
-
 static struct platform_driver bcm2835_i2s_driver = {
 	.probe		= bcm2835_i2s_probe,
-	.remove		= bcm2835_i2s_remove,
 	.driver		= {
 		.name	= "bcm2835-i2s",
 		.owner	= THIS_MODULE,
-- 
cgit v1.2.2


From 237eeb1c044fdd0f406a8484ee31884e34b9dfc5 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:34 +0100
Subject: ASoC: ep93xx: Use devm_snd_dmaengine_pcm_register()

Makes the code slightly shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/cirrus/ep93xx-pcm.c | 9 +--------
 1 file changed, 1 insertion(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index cfe517e68009..fdb8b8feb4ed 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -78,19 +78,13 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = {
 
 static int ep93xx_soc_platform_probe(struct platform_device *pdev)
 {
-	return snd_dmaengine_pcm_register(&pdev->dev,
+	return devm_snd_dmaengine_pcm_register(&pdev->dev,
 		&ep93xx_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 
-static int ep93xx_soc_platform_remove(struct platform_device *pdev)
-{
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-	return 0;
-}
-
 static struct platform_driver ep93xx_pcm_driver = {
 	.driver = {
 			.name = "ep93xx-pcm-audio",
@@ -98,7 +92,6 @@ static struct platform_driver ep93xx_pcm_driver = {
 	},
 
 	.probe = ep93xx_soc_platform_probe,
-	.remove = ep93xx_soc_platform_remove,
 };
 
 module_platform_driver(ep93xx_pcm_driver);
-- 
cgit v1.2.2


From 7e6d18ac7ea1372b462778ff7c416ceaabe71b66 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:35 +0100
Subject: ASoC: fsl: Use devm_snd_dmaengine_pcm_register()

Makes the code shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_spdif.c   | 8 --------
 sound/soc/fsl/fsl_ssi.c     | 4 ----
 sound/soc/fsl/imx-pcm-dma.c | 9 ++-------
 sound/soc/fsl/imx-pcm.h     | 5 -----
 sound/soc/fsl/imx-ssi.c     | 3 ---
 5 files changed, 2 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 55193a5596ca..4d075f1abe78 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1181,13 +1181,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	return ret;
 }
 
-static int fsl_spdif_remove(struct platform_device *pdev)
-{
-	imx_pcm_dma_exit(pdev);
-
-	return 0;
-}
-
 static const struct of_device_id fsl_spdif_dt_ids[] = {
 	{ .compatible = "fsl,imx35-spdif", },
 	{}
@@ -1201,7 +1194,6 @@ static struct platform_driver fsl_spdif_driver = {
 		.of_match_table = fsl_spdif_dt_ids,
 	},
 	.probe = fsl_spdif_probe,
-	.remove = fsl_spdif_remove,
 };
 
 module_platform_driver(fsl_spdif_driver);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index fb8f52a0e7b7..3df0318e71df 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1109,8 +1109,6 @@ done:
 	return 0;
 
 error_dai:
-	if (ssi_private->ssi_on_imx)
-		imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
@@ -1132,8 +1130,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 
 	if (!ssi_private->new_binding)
 		platform_device_unregister(ssi_private->pdev);
-	if (ssi_private->ssi_on_imx)
-		imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 	if (ssi_private->ssi_on_imx)
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index aee23077080a..c5e47f866b4b 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -61,16 +61,11 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = {
 
 int imx_pcm_dma_init(struct platform_device *pdev)
 {
-	return snd_dmaengine_pcm_register(&pdev->dev, &imx_dmaengine_pcm_config,
+	return devm_snd_dmaengine_pcm_register(&pdev->dev,
+		&imx_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
 
-void imx_pcm_dma_exit(struct platform_device *pdev)
-{
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-}
-EXPORT_SYMBOL_GPL(imx_pcm_dma_exit);
-
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h
index 5d5b73303e11..c79cb27473be 100644
--- a/sound/soc/fsl/imx-pcm.h
+++ b/sound/soc/fsl/imx-pcm.h
@@ -40,16 +40,11 @@ struct imx_pcm_fiq_params {
 
 #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA)
 int imx_pcm_dma_init(struct platform_device *pdev);
-void imx_pcm_dma_exit(struct platform_device *pdev);
 #else
 static inline int imx_pcm_dma_init(struct platform_device *pdev)
 {
 	return -ENODEV;
 }
-
-static inline void imx_pcm_dma_exit(struct platform_device *pdev)
-{
-}
 #endif
 
 #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_FIQ)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index f5f248c91c16..cc7376f87f88 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -624,9 +624,6 @@ static int imx_ssi_remove(struct platform_device *pdev)
 {
 	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 
-	if (!ssi->dma_init)
-		imx_pcm_dma_exit(pdev);
-
 	if (!ssi->fiq_init)
 		imx_pcm_fiq_exit(pdev);
 
-- 
cgit v1.2.2


From 2650bc4f6d0c36f1219d2070485cc2980a88fab3 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 28 Nov 2013 08:50:37 +0100
Subject: ASoC: mxs: Use devm_snd_dmaengine_pcm_register()

Makes the code shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/mxs/mxs-pcm.c  | 8 +-------
 sound/soc/mxs/mxs-pcm.h  | 1 -
 sound/soc/mxs/mxs-saif.c | 8 --------
 3 files changed, 1 insertion(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index b16abbbf7764..04a6b0d60944 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -56,16 +56,10 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = {
 
 int mxs_pcm_platform_register(struct device *dev)
 {
-	return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config,
+	return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX);
 }
 EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
 
-void mxs_pcm_platform_unregister(struct device *dev)
-{
-	snd_dmaengine_pcm_unregister(dev);
-}
-EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister);
-
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
index bc685b67cac7..035ea0436ca5 100644
--- a/sound/soc/mxs/mxs-pcm.h
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -20,6 +20,5 @@
 #define _MXS_PCM_H
 
 int mxs_pcm_platform_register(struct device *dev);
-void mxs_pcm_platform_unregister(struct device *dev);
 
 #endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 54e622acac33..92db74dc3177 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -804,13 +804,6 @@ static int mxs_saif_probe(struct platform_device *pdev)
 	return 0;
 }
 
-static int mxs_saif_remove(struct platform_device *pdev)
-{
-	mxs_pcm_platform_unregister(&pdev->dev);
-
-	return 0;
-}
-
 static const struct of_device_id mxs_saif_dt_ids[] = {
 	{ .compatible = "fsl,imx28-saif", },
 	{ /* sentinel */ }
@@ -819,7 +812,6 @@ MODULE_DEVICE_TABLE(of, mxs_saif_dt_ids);
 
 static struct platform_driver mxs_saif_driver = {
 	.probe = mxs_saif_probe,
-	.remove = mxs_saif_remove,
 
 	.driver = {
 		.name = "mxs-saif",
-- 
cgit v1.2.2


From a8ca52b7911378864e6defb42be9166c248a3749 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 28 Nov 2013 17:27:02 +0000
Subject: ASoC: ak4642: Convert to table based control init

Improves error handling and saves code.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ak4642.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 2f861c9b1d69..7fe1e9030c4f 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -469,9 +469,6 @@ static int ak4642_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	snd_soc_add_codec_controls(codec, ak4642_snd_controls,
-			     ARRAY_SIZE(ak4642_snd_controls));
-
 	ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
@@ -491,6 +488,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.reg_cache_default	= ak4642_reg,			/* ak4642 reg */
 	.reg_cache_size		= ARRAY_SIZE(ak4642_reg),	/* ak4642 reg */
 	.reg_word_size		= sizeof(u8),
+	.controls		= ak4642_snd_controls,
+	.num_controls		= ARRAY_SIZE(ak4642_snd_controls),
 	.dapm_widgets		= ak4642_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(ak4642_dapm_widgets),
 	.dapm_routes		= ak4642_intercon,
@@ -505,6 +504,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
 	.reg_cache_default	= ak4648_reg,			/* ak4648 reg */
 	.reg_cache_size		= ARRAY_SIZE(ak4648_reg),	/* ak4648 reg */
 	.reg_word_size		= sizeof(u8),
+	.controls		= ak4642_snd_controls,
+	.num_controls		= ARRAY_SIZE(ak4642_snd_controls),
 	.dapm_widgets		= ak4642_dapm_widgets,
 	.num_dapm_widgets	= ARRAY_SIZE(ak4642_dapm_widgets),
 	.dapm_routes		= ak4642_intercon,
-- 
cgit v1.2.2


From 4574cd94a717eff3021a3e187dd48846adbd21ea Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 28 Nov 2013 18:03:49 +0000
Subject: ASoC: ak4642: Convert to direct regmap API usage

This moves us towards being able to remove the ASoC level I/O code which
duplicates regmap functionality. Currently the only difference between
the supported devices in the driver is the regmap so we can replace the
CODEC driver selections with regmap selection instead.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ak4642.c | 113 ++++++++++++++++++++++++----------------------
 1 file changed, 59 insertions(+), 54 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 7fe1e9030c4f..5af237468618 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -28,6 +28,7 @@
 #include <linux/slab.h>
 #include <linux/of_device.h>
 #include <linux/module.h>
+#include <linux/regmap.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
@@ -198,30 +199,30 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = {
 /*
  * ak4642 register cache
  */
-static const u8 ak4642_reg[] = {
-	0x00, 0x00, 0x01, 0x00,
-	0x02, 0x00, 0x00, 0x00,
-	0xe1, 0xe1, 0x18, 0x00,
-	0xe1, 0x18, 0x11, 0x08,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00,
+static const struct reg_default ak4642_reg[] = {
+	{  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
+	{  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
+	{  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
+	{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0x08 },
+	{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
+	{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
+	{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
+	{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
+	{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
+	{ 36, 0x00 },
 };
 
-static const u8 ak4648_reg[] = {
-	0x00, 0x00, 0x01, 0x00,
-	0x02, 0x00, 0x00, 0x00,
-	0xe1, 0xe1, 0x18, 0x00,
-	0xe1, 0x18, 0x11, 0xb8,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x00, 0x00, 0x00,
-	0x00, 0x88, 0x88, 0x08,
+static const struct reg_default ak4648_reg[] = {
+	{  0, 0x00 }, {  1, 0x00 }, {  2, 0x01 }, {  3, 0x00 },
+	{  4, 0x02 }, {  5, 0x00 }, {  6, 0x00 }, {  7, 0x00 },
+	{  8, 0xe1 }, {  9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 },
+	{ 12, 0xe1 }, { 13, 0x18 }, { 14, 0x11 }, { 15, 0xb8 },
+	{ 16, 0x00 }, { 17, 0x00 }, { 18, 0x00 }, { 19, 0x00 },
+	{ 20, 0x00 }, { 21, 0x00 }, { 22, 0x00 }, { 23, 0x00 },
+	{ 24, 0x00 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0x00 },
+	{ 28, 0x00 }, { 29, 0x00 }, { 30, 0x00 }, { 31, 0x00 },
+	{ 32, 0x00 }, { 33, 0x00 }, { 34, 0x00 }, { 35, 0x00 },
+	{ 36, 0x00 }, { 37, 0x88 }, { 38, 0x88 }, { 39, 0x08 },
 };
 
 static int ak4642_dai_startup(struct snd_pcm_substream *substream,
@@ -454,7 +455,10 @@ static struct snd_soc_dai_driver ak4642_dai = {
 
 static int ak4642_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+	regcache_mark_dirty(regmap);
+	regcache_sync(regmap);
 	return 0;
 }
 
@@ -463,7 +467,7 @@ static int ak4642_probe(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -485,9 +489,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.remove			= ak4642_remove,
 	.resume			= ak4642_resume,
 	.set_bias_level		= ak4642_set_bias_level,
-	.reg_cache_default	= ak4642_reg,			/* ak4642 reg */
-	.reg_cache_size		= ARRAY_SIZE(ak4642_reg),	/* ak4642 reg */
-	.reg_word_size		= sizeof(u8),
 	.controls		= ak4642_snd_controls,
 	.num_controls		= ARRAY_SIZE(ak4642_snd_controls),
 	.dapm_widgets		= ak4642_dapm_widgets,
@@ -496,20 +497,20 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.num_dapm_routes	= ARRAY_SIZE(ak4642_intercon),
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
-	.probe			= ak4642_probe,
-	.remove			= ak4642_remove,
-	.resume			= ak4642_resume,
-	.set_bias_level		= ak4642_set_bias_level,
-	.reg_cache_default	= ak4648_reg,			/* ak4648 reg */
-	.reg_cache_size		= ARRAY_SIZE(ak4648_reg),	/* ak4648 reg */
-	.reg_word_size		= sizeof(u8),
-	.controls		= ak4642_snd_controls,
-	.num_controls		= ARRAY_SIZE(ak4642_snd_controls),
-	.dapm_widgets		= ak4642_dapm_widgets,
-	.num_dapm_widgets	= ARRAY_SIZE(ak4642_dapm_widgets),
-	.dapm_routes		= ak4642_intercon,
-	.num_dapm_routes	= ARRAY_SIZE(ak4642_intercon),
+static const struct regmap_config ak4642_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= ARRAY_SIZE(ak4642_reg) + 1,
+	.reg_defaults		= ak4642_reg,
+	.num_reg_defaults	= ARRAY_SIZE(ak4642_reg),
+};
+
+static const struct regmap_config ak4648_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= ARRAY_SIZE(ak4648_reg) + 1,
+	.reg_defaults		= ak4648_reg,
+	.num_reg_defaults	= ARRAY_SIZE(ak4648_reg),
 };
 
 #if IS_ENABLED(CONFIG_I2C)
@@ -518,26 +519,30 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct device_node *np = i2c->dev.of_node;
-	const struct snd_soc_codec_driver *driver;
+	const struct regmap_config *regmap_config = NULL;
+	struct regmap *regmap;
 
-	driver = NULL;
 	if (np) {
 		const struct of_device_id *of_id;
 
 		of_id = of_match_device(ak4642_of_match, &i2c->dev);
 		if (of_id)
-			driver = of_id->data;
+			regmap_config = of_id->data;
 	} else {
-		driver = (struct snd_soc_codec_driver *)id->driver_data;
+		regmap_config = (const struct regmap_config *)id->driver_data;
 	}
 
-	if (!driver) {
-		dev_err(&i2c->dev, "no driver\n");
+	if (!regmap_config) {
+		dev_err(&i2c->dev, "Unknown device type\n");
 		return -EINVAL;
 	}
 
+	regmap = devm_regmap_init_i2c(i2c, regmap_config);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
 	return snd_soc_register_codec(&i2c->dev,
-				      driver, &ak4642_dai, 1);
+				      &soc_codec_dev_ak4642, &ak4642_dai, 1);
 }
 
 static int ak4642_i2c_remove(struct i2c_client *client)
@@ -547,17 +552,17 @@ static int ak4642_i2c_remove(struct i2c_client *client)
 }
 
 static struct of_device_id ak4642_of_match[] = {
-	{ .compatible = "asahi-kasei,ak4642",	.data = &soc_codec_dev_ak4642},
-	{ .compatible = "asahi-kasei,ak4643",	.data = &soc_codec_dev_ak4642},
-	{ .compatible = "asahi-kasei,ak4648",	.data = &soc_codec_dev_ak4648},
+	{ .compatible = "asahi-kasei,ak4642",	.data = &ak4642_regmap},
+	{ .compatible = "asahi-kasei,ak4643",	.data = &ak4642_regmap},
+	{ .compatible = "asahi-kasei,ak4648",	.data = &ak4648_regmap},
 	{},
 };
 MODULE_DEVICE_TABLE(of, ak4642_of_match);
 
 static const struct i2c_device_id ak4642_i2c_id[] = {
-	{ "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 },
-	{ "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 },
-	{ "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 },
+	{ "ak4642", (kernel_ulong_t)&ak4642_regmap },
+	{ "ak4643", (kernel_ulong_t)&ak4642_regmap },
+	{ "ak4648", (kernel_ulong_t)&ak4648_regmap },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
-- 
cgit v1.2.2


From 3621dbbc27ff347f2e4476013054bab18ebd906c Mon Sep 17 00:00:00 2001
From: Oskar Schirmer <oskar@scara.com>
Date: Sat, 16 Nov 2013 07:52:25 +0000
Subject: ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced
 situation

Unbalanced calls to imx_ssi_trigger() may result in endless
SSI activity and thus provoke eternal sound. While on the first glance,
the switch statement looks pretty symmetric, the SUSPEND/RESUME
pair is not: the suspend case comes along snd_pcm_suspend_all(),
which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(),
but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME.
This way userland may provoke an unbalanced resume, which might cause
the ssi->enabled counter to increase and never return to zero again,
so eventually SSI_SCR_SSIEN is never disabled.

As the information on whether to enable the SSI or not is contained
in the two bits for TE/RE, we save all the software mirroring of
hardware state here and simply use the hardware register itself
to keep the state of whether someone is currently playing or capturing.

This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c
which I send a patch for three days ago. Astonishing enough this
highly fragile scheme is used twice in parallel to serve the very
same control function, synchronously: Once out of sync you are lost
until reboot.

Note, that these fixes wont prevent state machine distortion on alsa
level to cut sound or the like. It just makes sure we have a chance
to synchronise again later on.

Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/imx-ssi.c | 5 ++---
 sound/soc/fsl/imx-ssi.h | 1 -
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index cc7376f87f88..6336757e967a 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -304,8 +304,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 			scr |= SSI_SCR_RE;
 		sier |= sier_bits;
 
-		if (++ssi->enabled == 1)
-			scr |= SSI_SCR_SSIEN;
+		scr |= SSI_SCR_SSIEN;
 
 		break;
 
@@ -318,7 +317,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 			scr &= ~SSI_SCR_RE;
 		sier &= ~sier_bits;
 
-		if (--ssi->enabled == 0)
+		if (!(scr & (SSI_SCR_TE | SSI_SCR_RE)))
 			scr &= ~SSI_SCR_SSIEN;
 
 		break;
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index 560c40fc9ebb..be6562365b6a 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -213,7 +213,6 @@ struct imx_ssi {
 
 	int fiq_init;
 	int dma_init;
-	int enabled;
 };
 
 #endif /* _IMX_SSI_H */
-- 
cgit v1.2.2


From fa558c2801fc173758c742d836f3cc4621851557 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 20 Nov 2013 15:25:02 +0900
Subject: ASoC: simple-card: add Device Tree support

Support for loading the simple-card module via DeviceTree.
It requests CPU/CODEC information.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 156 ++++++++++++++++++++++++++++++++++++++--
 1 file changed, 150 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index b2fbb7075a6c..7a9b6b4898c0 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -8,7 +8,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
-
+#include <linux/clk.h>
+#include <linux/of.h>
 #include <linux/platform_device.h>
 #include <linux/module.h>
 #include <sound/simple_card.h>
@@ -57,11 +58,144 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int
+asoc_simple_card_sub_parse_of(struct device_node *np,
+			      struct asoc_simple_dai *dai,
+			      struct device_node **node)
+{
+	struct clk *clk;
+	int ret;
+
+	/*
+	 * get node via "sound-dai = <&phandle port>"
+	 * it will be used as xxx_of_node on soc_bind_dai_link()
+	 */
+	*node = of_parse_phandle(np, "sound-dai", 0);
+	if (!*node)
+		return -ENODEV;
+
+	/* get dai->name */
+	ret = snd_soc_of_get_dai_name(np, &dai->name);
+	if (ret < 0)
+		goto parse_error;
+
+	/*
+	 * bitclock-inversion, frame-inversion
+	 * bitclock-master,    frame-master
+	 * and specific "format" if it has
+	 */
+	dai->fmt = snd_soc_of_parse_daifmt(np, NULL);
+
+	/*
+	 * dai->sysclk come from
+	 *  "clocks = <&xxx>" (if system has common clock)
+	 *  or "system-clock-frequency = <xxx>"
+	 */
+	clk = of_clk_get(np, 0);
+	if (IS_ERR(clk))
+		of_property_read_u32(np,
+				     "system-clock-frequency",
+				     &dai->sysclk);
+	else
+		dai->sysclk = clk_get_rate(clk);
+
+	ret = 0;
+
+parse_error:
+	of_node_put(*node);
+
+	return ret;
+}
+
+static int asoc_simple_card_parse_of(struct device_node *node,
+				     struct asoc_simple_card_info *info,
+				     struct device *dev,
+				     struct device_node **of_cpu,
+				     struct device_node **of_codec,
+				     struct device_node **of_platform)
+{
+	struct device_node *np;
+	char *name;
+	int ret = 0;
+
+	/* get CPU/CODEC common format via simple-audio-card,format */
+	info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
+		(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
+
+	/* CPU sub-node */
+	ret = -EINVAL;
+	np = of_get_child_by_name(node, "simple-audio-card,cpu");
+	if (np)
+		ret = asoc_simple_card_sub_parse_of(np,
+						  &info->cpu_dai,
+						  of_cpu);
+	if (ret < 0)
+		return ret;
+
+	/* CODEC sub-node */
+	ret = -EINVAL;
+	np = of_get_child_by_name(node, "simple-audio-card,codec");
+	if (np)
+		ret = asoc_simple_card_sub_parse_of(np,
+						  &info->codec_dai,
+						  of_codec);
+	if (ret < 0)
+		return ret;
+
+	/* card name is created from CPU/CODEC dai name */
+	name = devm_kzalloc(dev,
+			    strlen(info->cpu_dai.name)   +
+			    strlen(info->codec_dai.name) + 2,
+			    GFP_KERNEL);
+	sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name);
+	info->name = info->card = name;
+
+	/* simple-card assumes platform == cpu */
+	*of_platform = *of_cpu;
+
+	dev_dbg(dev, "card-name : %s\n", info->card);
+	dev_dbg(dev, "platform : %04x\n", info->daifmt);
+	dev_dbg(dev, "cpu : %s / %04x / %d\n",
+		info->cpu_dai.name,
+		info->cpu_dai.fmt,
+		info->cpu_dai.sysclk);
+	dev_dbg(dev, "codec : %s / %04x / %d\n",
+		info->codec_dai.name,
+		info->codec_dai.fmt,
+		info->codec_dai.sysclk);
+
+	return 0;
+}
+
 static int asoc_simple_card_probe(struct platform_device *pdev)
 {
-	struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
+	struct asoc_simple_card_info *cinfo;
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *of_cpu, *of_codec, *of_platform;
 	struct device *dev = &pdev->dev;
 
+	cinfo		= NULL;
+	of_cpu		= NULL;
+	of_codec	= NULL;
+	of_platform	= NULL;
+	if (np && of_device_is_available(np)) {
+		cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
+		if (cinfo) {
+			int ret;
+			ret = asoc_simple_card_parse_of(np, cinfo, dev,
+							&of_cpu,
+							&of_codec,
+							&of_platform);
+			if (ret < 0) {
+				if (ret != -EPROBE_DEFER)
+					dev_err(dev, "parse error %d\n", ret);
+				return ret;
+			}
+		}
+	} else {
+		cinfo = pdev->dev.platform_data;
+	}
+
 	if (!cinfo) {
 		dev_err(dev, "no info for asoc-simple-card\n");
 		return -EINVAL;
@@ -69,10 +203,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 
 	if (!cinfo->name	||
 	    !cinfo->card	||
-	    !cinfo->codec	||
-	    !cinfo->platform	||
-	    !cinfo->cpu_dai.name ||
-	    !cinfo->codec_dai.name) {
+	    !cinfo->codec_dai.name	||
+	    !(cinfo->codec		|| of_codec)	||
+	    !(cinfo->platform		|| of_platform)	||
+	    !(cinfo->cpu_dai.name	|| of_cpu)) {
 		dev_err(dev, "insufficient asoc_simple_card_info settings\n");
 		return -EINVAL;
 	}
@@ -86,6 +220,9 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	cinfo->snd_link.platform_name	= cinfo->platform;
 	cinfo->snd_link.codec_name	= cinfo->codec;
 	cinfo->snd_link.codec_dai_name	= cinfo->codec_dai.name;
+	cinfo->snd_link.cpu_of_node	= of_cpu;
+	cinfo->snd_link.codec_of_node	= of_codec;
+	cinfo->snd_link.platform_of_node = of_platform;
 	cinfo->snd_link.init		= asoc_simple_card_dai_init;
 
 	/*
@@ -107,10 +244,17 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 	return snd_soc_unregister_card(&cinfo->snd_card);
 }
 
+static const struct of_device_id asoc_simple_of_match[] = {
+	{ .compatible = "simple-audio-card", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, asoc_simple_of_match);
+
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name	= "asoc-simple-card",
 		.owner = THIS_MODULE,
+		.of_match_table = asoc_simple_of_match,
 	},
 	.probe		= asoc_simple_card_probe,
 	.remove		= asoc_simple_card_remove,
-- 
cgit v1.2.2


From 2f54d2a1cf7e62f56b1b0bcf44bd704f65359f38 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 28 Nov 2013 17:24:55 +0000
Subject: ASoC: ak4642: Convert to module_i2c_driver()

The device does not support anything other than I2C (at least with the
current driver) so save code.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ak4642.c | 22 +---------------------
 1 file changed, 1 insertion(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5af237468618..1f646c6e90c6 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -513,7 +513,6 @@ static const struct regmap_config ak4648_regmap = {
 	.num_reg_defaults	= ARRAY_SIZE(ak4648_reg),
 };
 
-#if IS_ENABLED(CONFIG_I2C)
 static struct of_device_id ak4642_of_match[];
 static int ak4642_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
@@ -577,27 +576,8 @@ static struct i2c_driver ak4642_i2c_driver = {
 	.remove		= ak4642_i2c_remove,
 	.id_table	= ak4642_i2c_id,
 };
-#endif
 
-static int __init ak4642_modinit(void)
-{
-	int ret = 0;
-#if IS_ENABLED(CONFIG_I2C)
-	ret = i2c_add_driver(&ak4642_i2c_driver);
-#endif
-	return ret;
-
-}
-module_init(ak4642_modinit);
-
-static void __exit ak4642_exit(void)
-{
-#if IS_ENABLED(CONFIG_I2C)
-	i2c_del_driver(&ak4642_i2c_driver);
-#endif
-
-}
-module_exit(ak4642_exit);
+module_i2c_driver(ak4642_i2c_driver);
 
 MODULE_DESCRIPTION("Soc AK4642 driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-- 
cgit v1.2.2


From aad730d0704545ad97654bd929b0aba05adb1436 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Dec 2013 13:33:57 +0100
Subject: ALSA: hda - Always do delayed probes for HD-audio devices

HD-audio devices tend to take long time for finishing the whole
probing procedure.  In this patch, the time-consuming part of the
probing procedure, the codec probe and the rest initializations, are
moved in the work, so that they can be done asynchronously in parallel
with probes of other devices.

Since we already have this mechanism in the driver code for the
firmware and i915 request_symbol() stuff, we just need to enable it
always; the resultant patch even reduces more lines, which is an
additional bonus.

Credit goes to David Henningsson, who suggested this workaround.

Reported-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 28 +++++++---------------------
 1 file changed, 7 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 486d25fa7e97..80c5f14e8ecd 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -543,9 +543,7 @@ struct azx {
 	/* for pending irqs */
 	struct work_struct irq_pending_work;
 
-#ifdef CONFIG_SND_HDA_I915
 	struct work_struct probe_work;
-#endif
 
 	/* reboot notifier (for mysterious hangup problem at power-down) */
 	struct notifier_block reboot_notifier;
@@ -3500,12 +3498,10 @@ static void azx_check_snoop_available(struct azx *chip)
 	}
 }
 
-#ifdef CONFIG_SND_HDA_I915
 static void azx_probe_work(struct work_struct *work)
 {
 	azx_probe_continue(container_of(work, struct azx, probe_work));
 }
-#endif
 
 /*
  * constructor
@@ -3582,10 +3578,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 		return err;
 	}
 
-#ifdef CONFIG_SND_HDA_I915
 	/* continue probing in work context as may trigger request module */
 	INIT_WORK(&chip->probe_work, azx_probe_work);
-#endif
 
 	*rchip = chip;
 
@@ -3805,7 +3799,7 @@ static int azx_probe(struct pci_dev *pci,
 	static int dev;
 	struct snd_card *card;
 	struct azx *chip;
-	bool probe_now;
+	bool schedule_probe;
 	int err;
 
 	if (dev >= SNDRV_CARDS)
@@ -3844,7 +3838,7 @@ static int azx_probe(struct pci_dev *pci,
 		chip->disabled = true;
 	}
 
-	probe_now = !chip->disabled;
+	schedule_probe = !chip->disabled;
 
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
 	if (patch[dev] && *patch[dev]) {
@@ -3855,25 +3849,17 @@ static int azx_probe(struct pci_dev *pci,
 					      azx_firmware_cb);
 		if (err < 0)
 			goto out_free;
-		probe_now = false; /* continued in azx_firmware_cb() */
+		schedule_probe = false; /* continued in azx_firmware_cb() */
 	}
 #endif /* CONFIG_SND_HDA_PATCH_LOADER */
 
-	/* continue probing in work context, avoid request_module deadlock */
-	if (probe_now && (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)) {
-#ifdef CONFIG_SND_HDA_I915
-		probe_now = false;
-		schedule_work(&chip->probe_work);
-#else
+#ifndef CONFIG_SND_HDA_I915
+	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		snd_printk(KERN_ERR SFX "Haswell must build in CONFIG_SND_HDA_I915\n");
 #endif
-	}
 
-	if (probe_now) {
-		err = azx_probe_continue(chip);
-		if (err < 0)
-			goto out_free;
-	}
+	if (schedule_probe)
+		schedule_work(&chip->probe_work);
 
 	dev++;
 	if (chip->disabled)
-- 
cgit v1.2.2


From 2924a9981006ad01efb46c754689fa7d03e3eb4f Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Mon, 2 Dec 2013 23:29:03 +0800
Subject: ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface

The normal mode of SSI allows it to send/receive data to/from the first
slot of each period. So we can use this normal mode to trick I2S signal
by puting/getting data to/from the first slot only (the left channel)
so as to support monaural audio playback and recording.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++-------
 1 file changed, 12 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 3df0318e71df..90ff1071e29c 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -143,6 +143,7 @@ struct fsl_ssi_private {
 	bool ssi_on_imx;
 	bool imx_ac97;
 	bool use_dma;
+	u8 i2s_mode;
 	struct clk *clk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
 static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 {
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-	u8 i2s_mode;
 	u8 wm;
 	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
 
 	if (ssi_private->imx_ac97)
-		i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET;
+		ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET;
 	else
-		i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE;
+		ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE;
 
 	/*
 	 * Section 16.5 of the MPC8610 reference manual says that the SSI needs
@@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 	write_ssi_mask(&ssi->scr,
 		CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
 		CCSR_SSI_SCR_TFR_CLK_DIS |
-		i2s_mode |
+		ssi_private->i2s_mode |
 		(synchronous ? CCSR_SSI_SCR_SYN : 0));
 
 	write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
@@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 {
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	unsigned int channels = params_channels(hw_params);
 	unsigned int sample_size =
 		snd_pcm_format_width(params_format(hw_params));
 	u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
@@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	else
 		write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
 
+	if (!ssi_private->imx_ac97)
+		write_ssi_mask(&ssi->scr,
+				CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
+				channels == 1 ? 0 : ssi_private->i2s_mode);
+
 	return 0;
 }
 
@@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 	.probe = fsl_ssi_dai_probe,
 	.playback = {
-		/* The SSI does not support monaural audio. */
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
-		.channels_min = 2,
+		.channels_min = 1,
 		.channels_max = 2,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
-- 
cgit v1.2.2


From 84d2dc3e57bea30fb31397861cb97547069b25e4 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Mon, 2 Dec 2013 22:26:03 -0500
Subject: ALSA: hda - fixup ALC262 to skip depop delay before D3 on Intel
 BayleyBay

This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'.
And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce
codec suspend time.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 537991ccba28..518e6ceeed20 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -374,8 +374,10 @@ static void alc_fixup_no_depop_delay(struct hda_codec *codec,
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (action == HDA_FIXUP_ACT_PROBE)
+	if (action == HDA_FIXUP_ACT_PROBE) {
 		spec->no_depop_delay = 1;
+		codec->depop_delay = 0;
+	}
 }
 
 static int alc_auto_parse_customize_define(struct hda_codec *codec)
-- 
cgit v1.2.2


From 07a9483aaca5d3b5de8ee824ee576321d3f8b4c6 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Tue, 3 Dec 2013 18:38:07 +0800
Subject: ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits

Since we introduced symmetric_channels and symmetric_samplebits, we implement
these two features to fsl_ssi so as to drop some no-more-needed code and make
the driver neat and clean.

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 79 ++++++-------------------------------------------
 1 file changed, 9 insertions(+), 70 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 90ff1071e29c..f9f4569417ed 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
  * @ssi: pointer to the SSI's registers
  * @ssi_phys: physical address of the SSI registers
  * @irq: IRQ of this SSI
- * @first_stream: pointer to the stream that was opened first
- * @second_stream: pointer to second stream
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
  * @cpu_dai: the CPU DAI for this device
@@ -132,8 +130,6 @@ struct fsl_ssi_private {
 	struct ccsr_ssi __iomem *ssi;
 	dma_addr_t ssi_phys;
 	unsigned int irq;
-	struct snd_pcm_substream *first_stream;
-	struct snd_pcm_substream *second_stream;
 	unsigned int fifo_depth;
 	struct snd_soc_dai_driver cpu_dai_drv;
 	struct device_attribute dev_attr;
@@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private =
 		snd_soc_dai_get_drvdata(rtd->cpu_dai);
-	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
 
-	/*
-	 * If this is the first stream opened, then request the IRQ
-	 * and initialize the SSI registers.
+	/* First, we only do fsl_ssi_setup() when SSI is going to be active.
+	 * Second, fsl_ssi_setup was already called by ac97_init earlier if
+	 * the driver is in ac97 mode.
 	 */
-	if (!ssi_private->first_stream) {
-		ssi_private->first_stream = substream;
-
-		/*
-		 * fsl_ssi_setup was already called by ac97_init earlier if
-		 * the driver is in ac97 mode.
-		 */
-		if (!ssi_private->imx_ac97)
-			fsl_ssi_setup(ssi_private);
-	} else {
-		if (synchronous) {
-			struct snd_pcm_runtime *first_runtime =
-				ssi_private->first_stream->runtime;
-			/*
-			 * This is the second stream open, and we're in
-			 * synchronous mode, so we need to impose sample
-			 * sample size constraints. This is because STCCR is
-			 * used for playback and capture in synchronous mode,
-			 * so there's no way to specify different word
-			 * lengths.
-			 *
-			 * Note that this can cause a race condition if the
-			 * second stream is opened before the first stream is
-			 * fully initialized.  We provide some protection by
-			 * checking to make sure the first stream is
-			 * initialized, but it's not perfect.  ALSA sometimes
-			 * re-initializes the driver with a different sample
-			 * rate or size.  If the second stream is opened
-			 * before the first stream has received its final
-			 * parameters, then the second stream may be
-			 * constrained to the wrong sample rate or size.
-			 */
-			if (first_runtime->sample_bits) {
-				snd_pcm_hw_constraint_minmax(substream->runtime,
-						SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-				first_runtime->sample_bits,
-				first_runtime->sample_bits);
-			}
-		}
-
-		ssi_private->second_stream = substream;
-	}
+	if (!dai->active && !ssi_private->imx_ac97)
+		fsl_ssi_setup(ssi_private);
 
 	return 0;
 }
@@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-/**
- * fsl_ssi_shutdown: shutdown the SSI
- *
- * Shutdown the SSI if there are no other substreams open.
- */
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-
-	if (ssi_private->first_stream == substream)
-		ssi_private->first_stream = ssi_private->second_stream;
-
-	ssi_private->second_stream = NULL;
-}
-
 static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 {
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai);
@@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 	.startup	= fsl_ssi_startup,
 	.hw_params	= fsl_ssi_hw_params,
-	.shutdown	= fsl_ssi_shutdown,
 	.trigger	= fsl_ssi_trigger,
 };
 
@@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 
 static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = {
 	.startup	= fsl_ssi_startup,
-	.shutdown	= fsl_ssi_shutdown,
 	.trigger	= fsl_ssi_ac97_trigger,
 };
 
@@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	/* Are the RX and the TX clocks locked? */
-	if (!of_find_property(np, "fsl,ssi-asynchronous", NULL))
+	if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) {
 		ssi_private->cpu_dai_drv.symmetric_rates = 1;
+		ssi_private->cpu_dai_drv.symmetric_channels = 1;
+		ssi_private->cpu_dai_drv.symmetric_samplebits = 1;
+	}
 
 	/* Determine the FIFO depth. */
 	iprop = of_get_property(np, "fsl,fifo-depth", NULL);
-- 
cgit v1.2.2


From a010ff628c0953e6c914ecd09678363848617a88 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 2 Dec 2013 21:26:22 +0100
Subject: ASoC: ssm2602: Use core for applying symmetry constraints

Let the core take care of applying sample rate and sample bits constraints
instead of open-coding this in the driver.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ssm2602.c | 50 ++--------------------------------------------
 1 file changed, 2 insertions(+), 48 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 480074d86476..c6dd48561884 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -53,8 +53,6 @@ enum ssm2602_type {
 struct ssm2602_priv {
 	unsigned int sysclk;
 	struct snd_pcm_hw_constraint_list *sysclk_constraints;
-	struct snd_pcm_substream *master_substream;
-	struct snd_pcm_substream *slave_substream;
 
 	struct regmap *regmap;
 
@@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
 	unsigned int iface;
 
-	if (substream == ssm2602->slave_substream) {
-		dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
-		return 0;
-	}
-
 	if (srate < 0)
 		return srate;
 
@@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	struct snd_pcm_runtime *master_runtime;
-
-	/* The DAI has shared clocks so if we already have a playback or
-	 * capture going then constrain this substream to match it.
-	 * TODO: the ssm2602 allows pairs of non-matching PB/REC rates
-	 */
-	if (ssm2602->master_substream) {
-		master_runtime = ssm2602->master_substream->runtime;
-		dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n",
-			master_runtime->sample_bits,
-			master_runtime->rate);
-
-		if (master_runtime->rate != 0)
-			snd_pcm_hw_constraint_minmax(substream->runtime,
-						     SNDRV_PCM_HW_PARAM_RATE,
-						     master_runtime->rate,
-						     master_runtime->rate);
-
-		if (master_runtime->sample_bits != 0)
-			snd_pcm_hw_constraint_minmax(substream->runtime,
-						     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-						     master_runtime->sample_bits,
-						     master_runtime->sample_bits);
-
-		ssm2602->slave_substream = substream;
-	} else
-		ssm2602->master_substream = substream;
 
 	if (ssm2602->sysclk_constraints) {
 		snd_pcm_hw_constraint_list(substream->runtime, 0,
@@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void ssm2602_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-
-	if (ssm2602->master_substream == substream)
-		ssm2602->master_substream = ssm2602->slave_substream;
-
-	ssm2602->slave_substream = NULL;
-}
-
-
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
@@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 static const struct snd_soc_dai_ops ssm2602_dai_ops = {
 	.startup	= ssm2602_startup,
 	.hw_params	= ssm2602_hw_params,
-	.shutdown	= ssm2602_shutdown,
 	.digital_mute	= ssm2602_mute,
 	.set_sysclk	= ssm2602_set_dai_sysclk,
 	.set_fmt	= ssm2602_set_dai_fmt,
@@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = {
 		.rates = SSM2602_RATES,
 		.formats = SSM2602_FORMATS,},
 	.ops = &ssm2602_dai_ops,
+	.symmetric_rates = 1,
+	.symmetric_samplebits = 1,
 };
 
 static int ssm2602_suspend(struct snd_soc_codec *codec)
-- 
cgit v1.2.2


From b13a714923fe4d444977dccb7a0f7ad1346724cf Mon Sep 17 00:00:00 2001
From: Ulf Hansson <ulf.hansson@linaro.org>
Date: Tue, 3 Dec 2013 11:04:26 +0100
Subject: ALSA: AACI: Convert to modern PM ops

Even if the CONFIG_PM explicity is undefined, let's convert to the
modern PM ops.

Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/arm/aaci.c | 23 +++++++++++------------
 1 file changed, 11 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 1ca8dc2ccb89..c421fdb3c7a1 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -753,7 +753,7 @@ static struct snd_pcm_ops aaci_capture_ops = {
  * Power Management.
  */
 #ifdef CONFIG_PM
-static int aaci_do_suspend(struct snd_card *card, unsigned int state)
+static int aaci_do_suspend(struct snd_card *card)
 {
 	struct aaci *aaci = card->private_data;
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3cold);
@@ -761,28 +761,28 @@ static int aaci_do_suspend(struct snd_card *card, unsigned int state)
 	return 0;
 }
 
-static int aaci_do_resume(struct snd_card *card, unsigned int state)
+static int aaci_do_resume(struct snd_card *card)
 {
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
 
-static int aaci_suspend(struct amba_device *dev, pm_message_t state)
+static int aaci_suspend(struct device *dev)
 {
-	struct snd_card *card = amba_get_drvdata(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	return card ? aaci_do_suspend(card) : 0;
 }
 
-static int aaci_resume(struct amba_device *dev)
+static int aaci_resume(struct device *dev)
 {
-	struct snd_card *card = amba_get_drvdata(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	return card ? aaci_do_resume(card) : 0;
 }
+
+static SIMPLE_DEV_PM_OPS(aaci_dev_pm_ops, aaci_suspend, aaci_resume);
+#define AACI_DEV_PM_OPS (&aaci_dev_pm_ops)
 #else
-#define aaci_do_suspend		NULL
-#define aaci_do_resume		NULL
-#define aaci_suspend		NULL
-#define aaci_resume		NULL
+#define AACI_DEV_PM_OPS NULL
 #endif
 
 
@@ -1100,11 +1100,10 @@ MODULE_DEVICE_TABLE(amba, aaci_ids);
 static struct amba_driver aaci_driver = {
 	.drv		= {
 		.name	= DRIVER_NAME,
+		.pm	= AACI_DEV_PM_OPS,
 	},
 	.probe		= aaci_probe,
 	.remove		= aaci_remove,
-	.suspend	= aaci_suspend,
-	.resume		= aaci_resume,
 	.id_table	= aaci_ids,
 };
 
-- 
cgit v1.2.2


From b84c9ce809c91b3c613c967abcee90ebd6582092 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 3 Dec 2013 18:53:02 +0100
Subject: ASoC: jz4740-i2s: Use managed resources

Makes the code a bit shorter.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/jz4740/jz4740-i2s.c | 85 ++++++++-----------------------------------
 1 file changed, 15 insertions(+), 70 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 4c849a49c72a..a0b6a85d89cf 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = {
 static int jz4740_i2s_dev_probe(struct platform_device *pdev)
 {
 	struct jz4740_i2s *i2s;
+	struct resource *mem;
 	int ret;
 
-	i2s = kzalloc(sizeof(*i2s), GFP_KERNEL);
-
+	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
 	if (!i2s)
 		return -ENOMEM;
 
-	i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!i2s->mem) {
-		ret = -ENOENT;
-		goto err_free;
-	}
-
-	i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem),
-				pdev->name);
-	if (!i2s->mem) {
-		ret = -EBUSY;
-		goto err_free;
-	}
-
-	i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem));
-	if (!i2s->base) {
-		ret = -EBUSY;
-		goto err_release_mem_region;
-	}
+	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	i2s->base = devm_ioremap_resource(&pdev->dev, mem);
+	if (IS_ERR(i2s->base))
+		return PTR_ERR(i2s->base);
 
-	i2s->phys_base = i2s->mem->start;
+	i2s->phys_base = mem->start;
 
-	i2s->clk_aic = clk_get(&pdev->dev, "aic");
-	if (IS_ERR(i2s->clk_aic)) {
-		ret = PTR_ERR(i2s->clk_aic);
-		goto err_iounmap;
-	}
+	i2s->clk_aic = devm_clk_get(&pdev->dev, "aic");
+	if (IS_ERR(i2s->clk_aic))
+		return PTR_ERR(i2s->clk_aic);
 
-	i2s->clk_i2s = clk_get(&pdev->dev, "i2s");
-	if (IS_ERR(i2s->clk_i2s)) {
-		ret = PTR_ERR(i2s->clk_i2s);
-		goto err_clk_put_aic;
-	}
+	i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s");
+	if (IS_ERR(i2s->clk_i2s))
+		return PTR_ERR(i2s->clk_i2s);
 
 	platform_set_drvdata(pdev, i2s);
-	ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component,
-					 &jz4740_i2s_dai, 1);
-
-	if (ret) {
-		dev_err(&pdev->dev, "Failed to register DAI\n");
-		goto err_clk_put_i2s;
-	}
-
-	return 0;
-
-err_clk_put_i2s:
-	clk_put(i2s->clk_i2s);
-err_clk_put_aic:
-	clk_put(i2s->clk_aic);
-err_iounmap:
-	iounmap(i2s->base);
-err_release_mem_region:
-	release_mem_region(i2s->mem->start, resource_size(i2s->mem));
-err_free:
-	kfree(i2s);
-
-	return ret;
-}
-
-static int jz4740_i2s_dev_remove(struct platform_device *pdev)
-{
-	struct jz4740_i2s *i2s = platform_get_drvdata(pdev);
-
-	snd_soc_unregister_component(&pdev->dev);
-
-	clk_put(i2s->clk_i2s);
-	clk_put(i2s->clk_aic);
 
-	iounmap(i2s->base);
-	release_mem_region(i2s->mem->start, resource_size(i2s->mem));
-
-	kfree(i2s);
-
-	return 0;
+	return devm_snd_soc_register_component(&pdev->dev,
+		&jz4740_i2s_component, &jz4740_i2s_dai, 1);
 }
 
 static struct platform_driver jz4740_i2s_driver = {
 	.probe = jz4740_i2s_dev_probe,
-	.remove = jz4740_i2s_dev_remove,
 	.driver = {
 		.name = "jz4740-i2s",
 		.owner = THIS_MODULE,
-- 
cgit v1.2.2


From 0406a40a095ca039e5f5ec63783342253c573d06 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 3 Dec 2013 18:53:03 +0100
Subject: ASoC: jz4740: Use the generic dmaengine PCM driver

Now that there is a dmaengine driver for the jz4740 DMA core we can use the
generic dmaengine PCM driver. This allows us to remove the custom jz4740-pcm
code completely.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/jz4740/Kconfig      |   1 +
 sound/soc/jz4740/jz4740-i2s.c |  54 +++----
 sound/soc/jz4740/jz4740-pcm.c | 358 ------------------------------------------
 sound/soc/jz4740/jz4740-pcm.h |  20 ---
 sound/soc/jz4740/qi_lb60.c    |   2 +-
 5 files changed, 24 insertions(+), 411 deletions(-)
 delete mode 100644 sound/soc/jz4740/jz4740-pcm.c
 delete mode 100644 sound/soc/jz4740/jz4740-pcm.h

(limited to 'sound')

diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig
index 5351cba66c9e..29f76af5d963 100644
--- a/sound/soc/jz4740/Kconfig
+++ b/sound/soc/jz4740/Kconfig
@@ -1,6 +1,7 @@
 config SND_JZ4740_SOC
 	tristate "SoC Audio for Ingenic JZ4740 SoC"
 	depends on MACH_JZ4740 && SND_SOC
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the JZ4740 I2S interface. You will also need to select the audio
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index a0b6a85d89cf..8f220009e0f6 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -29,9 +29,11 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+#include <asm/mach-jz4740/dma.h>
 
 #include "jz4740-i2s.h"
-#include "jz4740-pcm.h"
 
 #define JZ_REG_AIC_CONF		0x00
 #define JZ_REG_AIC_CTRL		0x04
@@ -89,8 +91,8 @@ struct jz4740_i2s {
 	struct clk *clk_aic;
 	struct clk *clk_i2s;
 
-	struct jz4740_pcm_config pcm_config_playback;
-	struct jz4740_pcm_config pcm_config_capture;
+	struct snd_dmaengine_dai_dma_data playback_dma_data;
+	struct snd_dmaengine_dai_dma_data capture_dma_data;
 };
 
 static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s,
@@ -233,8 +235,6 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
 	struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	enum jz4740_dma_width dma_width;
-	struct jz4740_pcm_config *pcm_config;
 	unsigned int sample_size;
 	uint32_t ctrl;
 
@@ -243,11 +243,9 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		sample_size = 0;
-		dma_width = JZ4740_DMA_WIDTH_8BIT;
 		break;
 	case SNDRV_PCM_FORMAT_S16:
 		sample_size = 1;
-		dma_width = JZ4740_DMA_WIDTH_16BIT;
 		break;
 	default:
 		return -EINVAL;
@@ -260,22 +258,13 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
 			ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO;
 		else
 			ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
-
-		pcm_config = &i2s->pcm_config_playback;
-		pcm_config->dma_config.dst_width = dma_width;
-
 	} else {
 		ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK;
 		ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
-
-		pcm_config = &i2s->pcm_config_capture;
-		pcm_config->dma_config.src_width = dma_width;
 	}
 
 	jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
 
-	snd_soc_dai_set_dma_data(dai, substream, pcm_config);
-
 	return 0;
 }
 
@@ -342,25 +331,19 @@ static int jz4740_i2s_resume(struct snd_soc_dai *dai)
 
 static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
 {
-	struct jz4740_dma_config *dma_config;
+	struct snd_dmaengine_dai_dma_data *dma_data;
 
 	/* Playback */
-	dma_config = &i2s->pcm_config_playback.dma_config;
-	dma_config->src_width = JZ4740_DMA_WIDTH_32BIT;
-	dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
-	dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT;
-	dma_config->flags = JZ4740_DMA_SRC_AUTOINC;
-	dma_config->mode = JZ4740_DMA_MODE_SINGLE;
-	i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
+	dma_data = &i2s->playback_dma_data;
+	dma_data->maxburst = 16;
+	dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT;
+	dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO;
 
 	/* Capture */
-	dma_config = &i2s->pcm_config_capture.dma_config;
-	dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT;
-	dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
-	dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE;
-	dma_config->flags = JZ4740_DMA_DST_AUTOINC;
-	dma_config->mode = JZ4740_DMA_MODE_SINGLE;
-	i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
+	dma_data = &i2s->capture_dma_data;
+	dma_data->maxburst = 16;
+	dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE;
+	dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO;
 }
 
 static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai)
@@ -371,6 +354,8 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai)
 	clk_prepare_enable(i2s->clk_aic);
 
 	jz4740_i2c_init_pcm_config(i2s);
+	snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data,
+		&i2s->capture_dma_data);
 
 	conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
 		(8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
@@ -456,8 +441,13 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(pdev, i2s);
 
-	return devm_snd_soc_register_component(&pdev->dev,
+	ret = devm_snd_soc_register_component(&pdev->dev,
 		&jz4740_i2s_component, &jz4740_i2s_dai, 1);
+	if (ret)
+		return ret;
+
+	return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 
 static struct platform_driver jz4740_i2s_driver = {
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
deleted file mode 100644
index 1d7ef28585e1..000000000000
--- a/sound/soc/jz4740/jz4740-pcm.c
+++ /dev/null
@@ -1,358 +0,0 @@
-/*
- *  Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de>
- *
- *  This program is free software; you can redistribute	 it and/or modify it
- *  under  the terms of	 the GNU General  Public License as published by the
- *  Free Software Foundation;  either version 2 of the	License, or (at your
- *  option) any later version.
- *
- *  You should have received a copy of the  GNU General Public License along
- *  with this program; if not, write  to the Free Software Foundation, Inc.,
- *  675 Mass Ave, Cambridge, MA 02139, USA.
- *
- */
-
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/kernel.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-
-#include <linux/dma-mapping.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-jz4740/dma.h>
-#include "jz4740-pcm.h"
-
-struct jz4740_runtime_data {
-	unsigned long dma_period;
-	dma_addr_t dma_start;
-	dma_addr_t dma_pos;
-	dma_addr_t dma_end;
-
-	struct jz4740_dma_chan *dma;
-
-	dma_addr_t fifo_addr;
-};
-
-/* identify hardware playback capabilities */
-static const struct snd_pcm_hardware jz4740_pcm_hardware = {
-	.info = SNDRV_PCM_INFO_MMAP |
-		SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED |
-		SNDRV_PCM_INFO_BLOCK_TRANSFER,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
-
-	.rates			= SNDRV_PCM_RATE_8000_48000,
-	.channels_min		= 1,
-	.channels_max		= 2,
-	.period_bytes_min	= 16,
-	.period_bytes_max	= 2 * PAGE_SIZE,
-	.periods_min		= 2,
-	.periods_max		= 128,
-	.buffer_bytes_max	= 128 * 2 * PAGE_SIZE,
-	.fifo_size		= 32,
-};
-
-static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd,
-	struct snd_pcm_substream *substream)
-{
-	unsigned long count;
-
-	if (prtd->dma_pos == prtd->dma_end)
-		prtd->dma_pos = prtd->dma_start;
-
-	if (prtd->dma_pos + prtd->dma_period > prtd->dma_end)
-		count = prtd->dma_end - prtd->dma_pos;
-	else
-		count = prtd->dma_period;
-
-	jz4740_dma_disable(prtd->dma);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos);
-		jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr);
-	} else {
-		jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr);
-		jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos);
-	}
-
-	jz4740_dma_set_transfer_count(prtd->dma, count);
-
-	prtd->dma_pos += count;
-
-	jz4740_dma_enable(prtd->dma);
-}
-
-static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err,
-	void *dev_id)
-{
-	struct snd_pcm_substream *substream = dev_id;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd = runtime->private_data;
-
-	snd_pcm_period_elapsed(substream);
-
-	jz4740_pcm_start_transfer(prtd, substream);
-}
-
-static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd = runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct jz4740_pcm_config *config;
-
-	config = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	if (!config)
-		return 0;
-
-	if (!prtd->dma) {
-		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-			prtd->dma = jz4740_dma_request(substream, "PCM Capture");
-		else
-			prtd->dma = jz4740_dma_request(substream, "PCM Playback");
-	}
-
-	if (!prtd->dma)
-		return -EBUSY;
-
-	jz4740_dma_configure(prtd->dma, &config->dma_config);
-	prtd->fifo_addr = config->fifo_addr;
-
-	jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done);
-
-	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-	runtime->dma_bytes = params_buffer_bytes(params);
-
-	prtd->dma_period = params_period_bytes(params);
-	prtd->dma_start = runtime->dma_addr;
-	prtd->dma_pos = prtd->dma_start;
-	prtd->dma_end = prtd->dma_start + runtime->dma_bytes;
-
-	return 0;
-}
-
-static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	struct jz4740_runtime_data *prtd = substream->runtime->private_data;
-
-	snd_pcm_set_runtime_buffer(substream, NULL);
-	if (prtd->dma) {
-		jz4740_dma_free(prtd->dma);
-		prtd->dma = NULL;
-	}
-
-	return 0;
-}
-
-static int jz4740_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct jz4740_runtime_data *prtd = substream->runtime->private_data;
-
-	if (!prtd->dma)
-		return -EBUSY;
-
-	prtd->dma_pos = prtd->dma_start;
-
-	return 0;
-}
-
-static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd = runtime->private_data;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		jz4740_pcm_start_transfer(prtd, substream);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		jz4740_dma_disable(prtd->dma);
-		break;
-	default:
-		break;
-	}
-
-	return 0;
-}
-
-static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd = runtime->private_data;
-	unsigned long byte_offset;
-	snd_pcm_uframes_t offset;
-	struct jz4740_dma_chan *dma = prtd->dma;
-
-	/* prtd->dma_pos points to the end of the current transfer. So by
-	 * subtracting prdt->dma_start we get the offset to the end of the
-	 * current period in bytes. By subtracting the residue of the transfer
-	 * we get the current offset in bytes. */
-	byte_offset = prtd->dma_pos - prtd->dma_start;
-	byte_offset -= jz4740_dma_get_residue(dma);
-
-	offset = bytes_to_frames(runtime, byte_offset);
-	if (offset >= runtime->buffer_size)
-		offset = 0;
-
-	return offset;
-}
-
-static int jz4740_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd;
-
-	prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
-	if (prtd == NULL)
-		return -ENOMEM;
-
-	snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware);
-
-	runtime->private_data = prtd;
-
-	return 0;
-}
-
-static int jz4740_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct jz4740_runtime_data *prtd = runtime->private_data;
-
-	kfree(prtd);
-
-	return 0;
-}
-
-static int jz4740_pcm_mmap(struct snd_pcm_substream *substream,
-	struct vm_area_struct *vma)
-{
-	return remap_pfn_range(vma, vma->vm_start,
-			substream->dma_buffer.addr >> PAGE_SHIFT,
-			vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
-static struct snd_pcm_ops jz4740_pcm_ops = {
-	.open		= jz4740_pcm_open,
-	.close		= jz4740_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= jz4740_pcm_hw_params,
-	.hw_free	= jz4740_pcm_hw_free,
-	.prepare	= jz4740_pcm_prepare,
-	.trigger	= jz4740_pcm_trigger,
-	.pointer	= jz4740_pcm_pointer,
-	.mmap		= jz4740_pcm_mmap,
-};
-
-static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size = jz4740_pcm_hardware.buffer_bytes_max;
-
-	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
-	buf->private_data = NULL;
-
-	buf->area = dma_alloc_noncoherent(pcm->card->dev, size,
-					  &buf->addr, GFP_KERNEL);
-	if (!buf->area)
-		return -ENOMEM;
-
-	buf->bytes = size;
-
-	return 0;
-}
-
-static void jz4740_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-	int stream;
-
-	for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) {
-		substream = pcm->streams[stream].substream;
-		if (!substream)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-
-		dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area,
-				buf->addr);
-		buf->area = NULL;
-	}
-}
-
-static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = jz4740_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto err;
-	}
-
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = jz4740_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto err;
-	}
-
-err:
-	return ret;
-}
-
-static struct snd_soc_platform_driver jz4740_soc_platform = {
-		.ops		= &jz4740_pcm_ops,
-		.pcm_new	= jz4740_pcm_new,
-		.pcm_free	= jz4740_pcm_free,
-};
-
-static int jz4740_pcm_probe(struct platform_device *pdev)
-{
-	return snd_soc_register_platform(&pdev->dev, &jz4740_soc_platform);
-}
-
-static int jz4740_pcm_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_platform(&pdev->dev);
-	return 0;
-}
-
-static struct platform_driver jz4740_pcm_driver = {
-	.probe = jz4740_pcm_probe,
-	.remove = jz4740_pcm_remove,
-	.driver = {
-		.name = "jz4740-pcm-audio",
-		.owner = THIS_MODULE,
-	},
-};
-
-module_platform_driver(jz4740_pcm_driver);
-
-MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
-MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h
deleted file mode 100644
index 1220cbb4382c..000000000000
--- a/sound/soc/jz4740/jz4740-pcm.h
+++ /dev/null
@@ -1,20 +0,0 @@
-/*
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _JZ4740_PCM_H
-#define _JZ4740_PCM_H
-
-#include <linux/dma-mapping.h>
-#include <asm/mach-jz4740/dma.h>
-
-
-struct jz4740_pcm_config {
-	struct jz4740_dma_config dma_config;
-	phys_addr_t fifo_addr;
-};
-
-#endif
diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c
index 55fd6b5df55f..82b5f37cd2c7 100644
--- a/sound/soc/jz4740/qi_lb60.c
+++ b/sound/soc/jz4740/qi_lb60.c
@@ -73,7 +73,7 @@ static struct snd_soc_dai_link qi_lb60_dai = {
 	.name = "jz4740",
 	.stream_name = "jz4740",
 	.cpu_dai_name = "jz4740-i2s",
-	.platform_name = "jz4740-pcm-audio",
+	.platform_name = "jz4740-i2s",
 	.codec_dai_name = "jz4740-hifi",
 	.codec_name = "jz4740-codec",
 	.init = qi_lb60_codec_init,
-- 
cgit v1.2.2


From 0b4bbae85e046042af76a65920db4bb5509c97bd Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Wed, 4 Dec 2013 11:18:37 +0800
Subject: ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in
 soc_pcm_close()

This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()

Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..89d594138773 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -390,11 +390,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	if (!codec_dai->active)
 		codec_dai->rate = 0;
 
-	/* Muting the DAC suppresses artifacts caused during digital
-	 * shutdown, for example from stopping clocks.
-	 */
-	snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
-
 	if (cpu_dai->driver->ops->shutdown)
 		cpu_dai->driver->ops->shutdown(substream, cpu_dai);
 
-- 
cgit v1.2.2


From 60dbb4f17417777fc56cc909c3e89f4d338e8ca8 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 3 Dec 2013 22:09:33 -0800
Subject: ASoC: rcar: use devm_clk_get() instead of clk_get()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/ssi.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 477465f9507b..82b04c6f352d 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -652,7 +652,7 @@ int rsnd_ssi_probe(struct platform_device *pdev,
 
 		snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i);
 
-		clk = clk_get(dev, name);
+		clk = devm_clk_get(dev, name);
 		if (IS_ERR(clk))
 			return PTR_ERR(clk);
 
@@ -713,7 +713,6 @@ void rsnd_ssi_remove(struct platform_device *pdev,
 	int i;
 
 	for_each_rsnd_ssi(ssi, priv, i) {
-		clk_put(ssi->clk);
 		if (rsnd_ssi_dma_available(ssi))
 			rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod));
 	}
-- 
cgit v1.2.2


From 1552c32547ca807f0df7d9abca54468033df8764 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 28 Nov 2013 18:11:38 +0000
Subject: ASoC: adsp: Use async writes where possible

This will allow a marginal speed improvement when used with a bus that
supports async I/O by reducing the amount of context thrashing between
writes, allowing the bus to be more fully utilised.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm_adsp.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..6b1c01c66e64 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1468,8 +1468,8 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 	unsigned int val;
 	int ret, count;
 
-	ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
-				 ADSP2_SYS_ENA, ADSP2_SYS_ENA);
+	ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL,
+				       ADSP2_SYS_ENA, ADSP2_SYS_ENA);
 	if (ret != 0)
 		return ret;
 
@@ -1521,9 +1521,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		val = (val & ARIZONA_SYSCLK_FREQ_MASK)
 			>> ARIZONA_SYSCLK_FREQ_SHIFT;
 
-		ret = regmap_update_bits(dsp->regmap,
-					 dsp->base + ADSP2_CLOCKING,
-					 ADSP2_CLK_SEL_MASK, val);
+		ret = regmap_update_bits_async(dsp->regmap,
+					       dsp->base + ADSP2_CLOCKING,
+					       ADSP2_CLK_SEL_MASK, val);
 		if (ret != 0) {
 			adsp_err(dsp, "Failed to set clock rate: %d\n",
 				 ret);
@@ -1586,10 +1586,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		if (ret != 0)
 			goto err;
 
-		ret = regmap_update_bits(dsp->regmap,
-					 dsp->base + ADSP2_CONTROL,
-					 ADSP2_CORE_ENA | ADSP2_START,
-					 ADSP2_CORE_ENA | ADSP2_START);
+		ret = regmap_update_bits_async(dsp->regmap,
+					       dsp->base + ADSP2_CONTROL,
+					       ADSP2_CORE_ENA | ADSP2_START,
+					       ADSP2_CORE_ENA | ADSP2_START);
 		if (ret != 0)
 			goto err;
 
-- 
cgit v1.2.2


From 24f4bd57a7b24d10e52a3807f88adec79824e5d8 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Wed, 4 Dec 2013 20:27:44 -0200
Subject: ASoC: imx-ssi: Check the return value from clk_prepare_enable()

clk_prepare_enable() may fail, so let's check its return value and propagate it
in the case of error.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/imx-ssi.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 6336757e967a..df552fa1aa65 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -535,7 +535,9 @@ static int imx_ssi_probe(struct platform_device *pdev)
 			ret);
 		goto failed_clk;
 	}
-	clk_prepare_enable(ssi->clk);
+	ret = clk_prepare_enable(ssi->clk);
+	if (ret)
+		goto failed_clk;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	ssi->base = devm_ioremap_resource(&pdev->dev, res);
-- 
cgit v1.2.2


From 7637af2e17f18bfe6264d834c6edee7706a0f15c Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 4 Dec 2013 15:19:27 -0700
Subject: ASoC: tegra: add tegra+MAX98090 machine driver

Initially, this binding and driver only describe/support playback to
headphones and speakers, and capture from the external microphone, with
GPIO-based jack detection for the headphone jack only.

This driver is useful for the Venice2 board.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/tegra/Kconfig          |  10 ++
 sound/soc/tegra/Makefile         |   2 +
 sound/soc/tegra/tegra_max98090.c | 275 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 287 insertions(+)
 create mode 100644 sound/soc/tegra/tegra_max98090.c

(limited to 'sound')

diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 8fc653ca3ab4..65a85f542521 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -116,3 +116,13 @@ config SND_SOC_TEGRA_ALC5632
 	help
 	  Say Y or M here if you want to add support for SoC audio on the
 	  Toshiba AC100 netbook.
+
+config SND_SOC_TEGRA_MAX98090
+	tristate "SoC Audio support for Tegra boards using a MAX98090 codec"
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
+	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
+	select SND_SOC_MAX98090
+	help
+	  Say Y or M here if you want to add support for SoC audio on Tegra
+	  boards using the MAX98090 codec, such as Venice2.
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index 21d2550a08a4..5ae588cd96c4 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -24,6 +24,7 @@ snd-soc-tegra-wm8903-objs := tegra_wm8903.o
 snd-soc-tegra-wm9712-objs := tegra_wm9712.o
 snd-soc-tegra-trimslice-objs := trimslice.o
 snd-soc-tegra-alc5632-objs := tegra_alc5632.o
+snd-soc-tegra-max98090-objs := tegra_max98090.o
 
 obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o
 obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o
@@ -31,3 +32,4 @@ obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o
 obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o
 obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o
 obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o
+obj-$(CONFIG_SND_SOC_TEGRA_MAX98090) += snd-soc-tegra-max98090.o
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
new file mode 100644
index 000000000000..0283cfb7c031
--- /dev/null
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -0,0 +1,275 @@
+/*
+ * Tegra machine ASoC driver for boards using a MAX90809 CODEC.
+ *
+ * Copyright (c) 2013, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (C) 2010-2012 - NVIDIA, Inc.
+ * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
+ * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd.
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "tegra_asoc_utils.h"
+
+#define DRV_NAME "tegra-snd-max98090"
+
+struct tegra_max98090 {
+	struct tegra_asoc_utils_data util_data;
+	int gpio_hp_det;
+};
+
+static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct snd_soc_card *card = codec->card;
+	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
+	int srate, mclk;
+	int err;
+
+	srate = params_rate(params);
+	switch (srate) {
+	case 8000:
+	case 16000:
+	case 24000:
+	case 32000:
+	case 48000:
+	case 64000:
+	case 96000:
+		mclk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		mclk = 11289600;
+		break;
+	default:
+		mclk = 12000000;
+		break;
+	}
+
+	err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk);
+	if (err < 0) {
+		dev_err(card->dev, "Can't configure clocks\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
+					SND_SOC_CLOCK_IN);
+	if (err < 0) {
+		dev_err(card->dev, "codec_dai clock not set\n");
+		return err;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops tegra_max98090_ops = {
+	.hw_params = tegra_max98090_asoc_hw_params,
+};
+
+static struct snd_soc_jack tegra_max98090_hp_jack;
+
+static struct snd_soc_jack_pin tegra_max98090_hp_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = {
+	.name = "Headphone detection",
+	.report = SND_JACK_HEADPHONE,
+	.debounce_time = 150,
+	.invert = 1,
+};
+
+static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_SPK("Speakers", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_kcontrol_new tegra_max98090_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Speakers"),
+};
+
+static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(codec->card);
+
+	if (gpio_is_valid(machine->gpio_hp_det)) {
+		snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE,
+				&tegra_max98090_hp_jack);
+		snd_soc_jack_add_pins(&tegra_max98090_hp_jack,
+				ARRAY_SIZE(tegra_max98090_hp_jack_pins),
+				tegra_max98090_hp_jack_pins);
+
+		tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det;
+		snd_soc_jack_add_gpios(&tegra_max98090_hp_jack,
+					1,
+					&tegra_max98090_hp_jack_gpio);
+	}
+
+	return 0;
+}
+
+static struct snd_soc_dai_link tegra_max98090_dai = {
+	.name = "max98090",
+	.stream_name = "max98090 PCM",
+	.codec_dai_name = "HiFi",
+	.init = tegra_max98090_asoc_init,
+	.ops = &tegra_max98090_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+};
+
+static struct snd_soc_card snd_soc_tegra_max98090 = {
+	.name = "tegra-max98090",
+	.owner = THIS_MODULE,
+	.dai_link = &tegra_max98090_dai,
+	.num_links = 1,
+	.controls = tegra_max98090_controls,
+	.num_controls = ARRAY_SIZE(tegra_max98090_controls),
+	.dapm_widgets = tegra_max98090_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tegra_max98090_dapm_widgets),
+	.fully_routed = true,
+};
+
+static int tegra_max98090_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct snd_soc_card *card = &snd_soc_tegra_max98090;
+	struct tegra_max98090 *machine;
+	int ret;
+
+	machine = devm_kzalloc(&pdev->dev,
+			sizeof(struct tegra_max98090), GFP_KERNEL);
+	if (!machine) {
+		dev_err(&pdev->dev, "Can't allocate tegra_max98090\n");
+		return -ENOMEM;
+	}
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, machine);
+
+	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
+	if (machine->gpio_hp_det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
+	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
+	if (ret)
+		goto err;
+
+	ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing");
+	if (ret)
+		goto err;
+
+	tegra_max98090_dai.codec_of_node = of_parse_phandle(np,
+			"nvidia,audio-codec", 0);
+	if (!tegra_max98090_dai.codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_max98090_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
+	if (!tegra_max98090_dai.cpu_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,i2s-controller' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_max98090_dai.platform_of_node = tegra_max98090_dai.cpu_of_node;
+
+	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
+	if (ret)
+		goto err;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		goto err_fini_utils;
+	}
+
+	return 0;
+
+err_fini_utils:
+	tegra_asoc_utils_fini(&machine->util_data);
+err:
+	return ret;
+}
+
+static int tegra_max98090_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card);
+
+	snd_soc_jack_free_gpios(&tegra_max98090_hp_jack, 1,
+				&tegra_max98090_hp_jack_gpio);
+
+	snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&machine->util_data);
+
+	return 0;
+}
+
+static const struct of_device_id tegra_max98090_of_match[] = {
+	{ .compatible = "nvidia,tegra-audio-max98090", },
+	{},
+};
+
+static struct platform_driver tegra_max98090_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = tegra_max98090_of_match,
+	},
+	.probe = tegra_max98090_probe,
+	.remove = tegra_max98090_remove,
+};
+module_platform_driver(tegra_max98090_driver);
+
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra max98090 machine ASoC driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_max98090_of_match);
-- 
cgit v1.2.2


From f4d6a55d7be3f5b944c17e6aafb84d96c4dd09d7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Dec 2013 11:55:05 +0100
Subject: ALSA: hda - Clean up async codec PM using standard async
 infrastructure

This simplifies lots of codes indeed.

Tested-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 64 ++++++++++++++---------------------------------
 sound/pci/hda/hda_codec.h |  6 -----
 2 files changed, 19 insertions(+), 51 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5b7622034eee..2e090c8d7144 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -26,6 +26,7 @@
 #include <linux/pci.h>
 #include <linux/mutex.h>
 #include <linux/module.h>
+#include <linux/async.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include <sound/asoundef.h>
@@ -96,8 +97,6 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
 
 #ifdef CONFIG_PM
 #define codec_in_pm(codec)	((codec)->in_pm)
-static void hda_suspend_work(struct work_struct *work);
-static void hda_resume_work(struct work_struct *work);
 static void hda_power_work(struct work_struct *work);
 static void hda_keep_power_on(struct hda_codec *codec);
 #define hda_codec_is_power_on(codec)	((codec)->power_on)
@@ -842,11 +841,6 @@ static int snd_hda_bus_free(struct hda_bus *bus)
 	if (bus->workq)
 		destroy_workqueue(bus->workq);
 
-#ifdef CONFIG_PM
-	if (bus->pm_wq)
-		destroy_workqueue(bus->pm_wq);
-#endif
-
 	kfree(bus);
 	return 0;
 }
@@ -891,9 +885,6 @@ int snd_hda_bus_new(struct snd_card *card,
 		.dev_register = snd_hda_bus_dev_register,
 		.dev_free = snd_hda_bus_dev_free,
 	};
-#ifdef CONFIG_PM
-	char wqname[16];
-#endif
 
 	if (snd_BUG_ON(!temp))
 		return -EINVAL;
@@ -930,16 +921,6 @@ int snd_hda_bus_new(struct snd_card *card,
 		return -ENOMEM;
 	}
 
-#ifdef CONFIG_PM
-	sprintf(wqname, "hda-pm-wq-%d", card->number);
-	bus->pm_wq = create_workqueue(wqname);
-	if (!bus->pm_wq) {
-		snd_printk(KERN_ERR "cannot create PM workqueue\n");
-		snd_hda_bus_free(bus);
-		return -ENOMEM;
-	}
-#endif
-
 	err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops);
 	if (err < 0) {
 		snd_hda_bus_free(bus);
@@ -1476,8 +1457,6 @@ int snd_hda_codec_new(struct hda_bus *bus,
 #ifdef CONFIG_PM
 	spin_lock_init(&codec->power_lock);
 	INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
-	INIT_WORK(&codec->suspend_work, hda_suspend_work);
-	INIT_WORK(&codec->resume_work, hda_resume_work);
 	/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
 	 * the caller has to power down appropriatley after initialization
 	 * phase.
@@ -1495,9 +1474,6 @@ int snd_hda_codec_new(struct hda_bus *bus,
 
 	list_add_tail(&codec->list, &bus->codec_list);
 	bus->num_codecs++;
-#ifdef CONFIG_PM
-	workqueue_set_max_active(bus->pm_wq, bus->num_codecs);
-#endif
 
 	bus->caddr_tbl[codec_addr] = codec;
 
@@ -5120,22 +5096,6 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
-
-static void hda_suspend_work(struct work_struct *work)
-{
-	struct hda_codec *codec =
-		container_of(work, struct hda_codec, suspend_work);
-
-	hda_call_codec_suspend(codec, false);
-}
-
-static void hda_resume_work(struct work_struct *work)
-{
-	struct hda_codec *codec =
-		container_of(work, struct hda_codec, resume_work);
-
-	hda_call_codec_resume(codec);
-}
 #endif
 
 /*
@@ -5699,6 +5659,17 @@ EXPORT_SYMBOL_HDA(snd_hda_add_imux_item);
  * power management
  */
 
+
+static void hda_async_suspend(void *data, async_cookie_t cookie)
+{
+	hda_call_codec_suspend(data, false);
+}
+
+static void hda_async_resume(void *data, async_cookie_t cookie)
+{
+	hda_call_codec_resume(data);
+}
+
 /**
  * snd_hda_suspend - suspend the codecs
  * @bus: the HDA bus
@@ -5708,19 +5679,21 @@ EXPORT_SYMBOL_HDA(snd_hda_add_imux_item);
 int snd_hda_suspend(struct hda_bus *bus)
 {
 	struct hda_codec *codec;
+	ASYNC_DOMAIN_EXCLUSIVE(domain);
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		if (hda_codec_is_power_on(codec)) {
 			if (bus->num_codecs > 1)
-				queue_work(bus->pm_wq, &codec->suspend_work);
+				async_schedule_domain(hda_async_suspend, codec,
+						      &domain);
 			else
 				hda_call_codec_suspend(codec, false);
 		}
 	}
 
 	if (bus->num_codecs > 1)
-		flush_workqueue(bus->pm_wq);
+		async_synchronize_full_domain(&domain);
 
 	return 0;
 }
@@ -5735,16 +5708,17 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend);
 int snd_hda_resume(struct hda_bus *bus)
 {
 	struct hda_codec *codec;
+	ASYNC_DOMAIN_EXCLUSIVE(domain);
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		if (bus->num_codecs > 1)
-			queue_work(bus->pm_wq, &codec->resume_work);
+			async_schedule_domain(hda_async_resume, codec, &domain);
 		else
 			hda_call_codec_resume(codec);
 	}
 
 	if (bus->num_codecs > 1)
-		flush_workqueue(bus->pm_wq);
+		async_synchronize_full_domain(&domain);
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 3ab4834761a8..260b190adf74 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -684,9 +684,6 @@ struct hda_bus {
 	struct hda_bus_unsolicited *unsol;
 	char workq_name[16];
 	struct workqueue_struct *workq;	/* common workqueue for codecs */
-#ifdef CONFIG_PM
-	struct workqueue_struct *pm_wq;	/* workqueue to parallel codec PM */
-#endif
 
 	/* assigned PCMs */
 	DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES);
@@ -921,9 +918,6 @@ struct hda_codec {
 	unsigned long power_off_acct;
 	unsigned long power_jiffies;
 	spinlock_t power_lock;
-	/* tasks to parallel multi-codec suspend/resume */
-	struct work_struct suspend_work;
-	struct work_struct resume_work;
 #endif
 
 	/* filter the requested power state per nid */
-- 
cgit v1.2.2


From 5b8620bb84b2f66b7a776fbfc0d273f6c78d969c Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Thu, 5 Dec 2013 18:35:48 -0500
Subject: ALSA: hda - skip depop delay before D3 for Haswell and Valleyview2
 display codec

This patch skips the default depop delay before D3 for Haswell (10 ms) and
Valleyview2 (100 ms) display codec, to reduce codec suspend time.

The analog part of display audio is implemented in the external display. Some
displays have weak pop noise while others not when suspending, no matter there
is the default delay or not.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 4098196b2300..1bd637b1dd0f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2247,6 +2247,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		intel_haswell_fixup_enable_dp12(codec);
 	}
 
+	if (is_haswell(codec) || is_valleyview(codec)) {
+		codec->depop_delay = 0;
+	}
+
 	if (hdmi_parse_codec(codec) < 0) {
 		codec->spec = NULL;
 		kfree(spec);
-- 
cgit v1.2.2


From c9e4bdb7557ad152f8a41301e646403d9f0ad92c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Dec 2013 09:30:14 +0100
Subject: ALSA: hda - Allow capture-only configuration

We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations.  The parser shouldn't escape in such a case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 2729e881541d..5ba3106b9712 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -4334,7 +4334,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 			spec->no_analog = 1;
 			goto dig_only;
 		}
-		return 0; /* can't find valid BIOS pin config */
+		if (!cfg->num_inputs && !cfg->dig_in_pin)
+			return 0; /* can't find valid BIOS pin config */
 	}
 
 	if (!spec->no_primary_hp &&
-- 
cgit v1.2.2


From 4f7f67fb774168a25802919493cdff41c8b9e384 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Dec 2013 10:51:37 +0100
Subject: ALSA: hda - Add missing initialization of aamix paths

The loopback mixing paths aren't initialized correctly at init
callback.  Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 18 ++++++++++++++++++
 1 file changed, 18 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 5ba3106b9712..058318de1af4 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5140,6 +5140,23 @@ static void init_multi_io(struct hda_codec *codec)
 	}
 }
 
+static void init_aamix_paths(struct hda_codec *codec)
+{
+	struct hda_gen_spec *spec = codec->spec;
+
+	if (!spec->have_aamix_ctl)
+		return;
+	update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0],
+			   spec->aamix_out_paths[0],
+			   spec->autocfg.line_out_type);
+	update_aamix_paths(codec, spec->aamix_mode, spec->hp_paths[0],
+			   spec->aamix_out_paths[1],
+			   AUTO_PIN_HP_OUT);
+	update_aamix_paths(codec, spec->aamix_mode, spec->speaker_paths[0],
+			   spec->aamix_out_paths[2],
+			   AUTO_PIN_SPEAKER_OUT);
+}
+
 /* set up input pins and loopback paths */
 static void init_analog_input(struct hda_codec *codec)
 {
@@ -5242,6 +5259,7 @@ int snd_hda_gen_init(struct hda_codec *codec)
 	init_multi_out(codec);
 	init_extra_out(codec);
 	init_multi_io(codec);
+	init_aamix_paths(codec);
 	init_analog_input(codec);
 	init_input_src(codec);
 	init_digital(codec);
-- 
cgit v1.2.2


From e8648e5e33e45fa2bb8706107eafc8ef42ed0774 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Dec 2013 17:15:01 +0100
Subject: ALSA: hda - Ignore small negative LPIB delay correction

Sometimes the hardware reports LPIB being advanced than POSBUF.
When this happens, the driver adjusts to a positive value by adding
the buffer size.  Then the driver detects it as an error (greater than
the period size), and stops the LPIB delay account from this point
on.

When I took a close look at these conditions, the values shown are all
very small numbers, and it'd be better to just ignore these values
instead of discontinuing the LPIB delay correction.

In this patch, the driver checks a negative delay value and ignores if
it's a significantly small error.  Currently the threshold is set to
64 frames, but could be smaller.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 19 +++++++++++++++++--
 1 file changed, 17 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 80c5f14e8ecd..af86c71f27bf 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -431,6 +431,8 @@ struct azx_dev {
 	struct timecounter  azx_tc;
 	struct cyclecounter azx_cc;
 
+	int delay_negative_threshold;
+
 #ifdef CONFIG_SND_HDA_DSP_LOADER
 	struct mutex dsp_mutex;
 #endif
@@ -2195,6 +2197,15 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 			goto unlock;
 	}
 
+	/* when LPIB delay correction gives a small negative value,
+	 * we ignore it; currently set the threshold statically to
+	 * 64 frames
+	 */
+	if (runtime->period_size > 64)
+		azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64);
+	else
+		azx_dev->delay_negative_threshold = 0;
+
 	/* wallclk has 24Mhz clock source */
 	azx_dev->period_wallclk = (((runtime->period_size * 24000) /
 						runtime->rate) * 1000);
@@ -2447,8 +2458,12 @@ static unsigned int azx_get_position(struct azx *chip,
 			delay = pos - lpib_pos;
 		else
 			delay = lpib_pos - pos;
-		if (delay < 0)
-			delay += azx_dev->bufsize;
+		if (delay < 0) {
+			if (delay >= azx_dev->delay_negative_threshold)
+				delay = 0;
+			else
+				delay += azx_dev->bufsize;
+		}
 		if (delay >= azx_dev->period_bytes) {
 			snd_printk(KERN_WARNING SFX
 				   "%s: Unstable LPIB (%d >= %d); "
-- 
cgit v1.2.2


From f1e762ddc8035f803d15f8c524151bf788482ec2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 9 Dec 2013 16:02:24 +0100
Subject: ALSA: hda - Enable stereo mix as default for AD and VIA codecs

AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression.  In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint.  This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c  | 19 ++++++++++++++++++-
 sound/pci/hda/patch_analog.c |  1 +
 sound/pci/hda/patch_via.c    |  1 +
 3 files changed, 20 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 058318de1af4..44be167d1cad 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3033,6 +3033,8 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin,
 			spec->imux_pins[imux->num_items] = pin;
 			snd_hda_add_imux_item(imux, label, cfg_idx, NULL);
 			imux_added = true;
+			if (spec->dyn_adc_switch)
+				spec->dyn_adc_idx[imux_idx] = c;
 		}
 	}
 
@@ -3130,7 +3132,9 @@ static int create_input_ctls(struct hda_codec *codec)
 		}
 	}
 
-	if (mixer && spec->add_stereo_mix_input) {
+	/* add stereo mix when explicitly enabled via hint */
+	if (mixer && spec->add_stereo_mix_input &&
+	    snd_hda_get_bool_hint(codec, "add_stereo_mix_input") > 0) {
 		err = parse_capture_source(codec, mixer, CFG_IDX_MIX, num_adcs,
 					   "Stereo Mix", 0);
 		if (err < 0)
@@ -4403,6 +4407,19 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 	if (err < 0)
 		return err;
 
+	/* add stereo mix if available and not enabled yet */
+	if (!spec->auto_mic && spec->mixer_nid &&
+	    spec->add_stereo_mix_input &&
+	    spec->input_mux.num_items > 1 &&
+	    snd_hda_get_bool_hint(codec, "add_stereo_mix_input") < 0) {
+		err = parse_capture_source(codec, spec->mixer_nid,
+					   CFG_IDX_MIX, spec->num_all_adcs,
+					   "Stereo Mix", 0);
+		if (err < 0)
+			return err;
+	}
+
+
 	err = create_capture_mixers(codec);
 	if (err < 0)
 		return err;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 34d86ec5d3dd..05280033c302 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -194,6 +194,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
 	codec->no_sticky_stream = 1;
 
 	spec->gen.indep_hp = 1;
+	spec->gen.add_stereo_mix_input = 1;
 
 	err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0);
 	if (err < 0)
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 0bc20ef5687a..f84195f3ea31 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -138,6 +138,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec)
 	spec->gen.indep_hp = 1;
 	spec->gen.keep_eapd_on = 1;
 	spec->gen.pcm_playback_hook = via_playback_pcm_hook;
+	spec->gen.add_stereo_mix_input = 1;
 	return spec;
 }
 
-- 
cgit v1.2.2


From ed326363d7a7844330325148efe06a994e4bc78f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 9 Dec 2013 14:28:36 +0100
Subject: ALSA: hda - Split verb definitions into sound/hda_verbs.h

Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h | 547 +---------------------------------------------
 1 file changed, 1 insertion(+), 546 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 260b190adf74..3c484c27048b 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -25,552 +25,7 @@
 #include <sound/control.h>
 #include <sound/pcm.h>
 #include <sound/hwdep.h>
-
-/*
- * nodes
- */
-#define	AC_NODE_ROOT		0x00
-
-/*
- * function group types
- */
-enum {
-	AC_GRP_AUDIO_FUNCTION = 0x01,
-	AC_GRP_MODEM_FUNCTION = 0x02,
-};
-	
-/*
- * widget types
- */
-enum {
-	AC_WID_AUD_OUT,		/* Audio Out */
-	AC_WID_AUD_IN,		/* Audio In */
-	AC_WID_AUD_MIX,		/* Audio Mixer */
-	AC_WID_AUD_SEL,		/* Audio Selector */
-	AC_WID_PIN,		/* Pin Complex */
-	AC_WID_POWER,		/* Power */
-	AC_WID_VOL_KNB,		/* Volume Knob */
-	AC_WID_BEEP,		/* Beep Generator */
-	AC_WID_VENDOR = 0x0f	/* Vendor specific */
-};
-
-/*
- * GET verbs
- */
-#define AC_VERB_GET_STREAM_FORMAT		0x0a00
-#define AC_VERB_GET_AMP_GAIN_MUTE		0x0b00
-#define AC_VERB_GET_PROC_COEF			0x0c00
-#define AC_VERB_GET_COEF_INDEX			0x0d00
-#define AC_VERB_PARAMETERS			0x0f00
-#define AC_VERB_GET_CONNECT_SEL			0x0f01
-#define AC_VERB_GET_CONNECT_LIST		0x0f02
-#define AC_VERB_GET_PROC_STATE			0x0f03
-#define AC_VERB_GET_SDI_SELECT			0x0f04
-#define AC_VERB_GET_POWER_STATE			0x0f05
-#define AC_VERB_GET_CONV			0x0f06
-#define AC_VERB_GET_PIN_WIDGET_CONTROL		0x0f07
-#define AC_VERB_GET_UNSOLICITED_RESPONSE	0x0f08
-#define AC_VERB_GET_PIN_SENSE			0x0f09
-#define AC_VERB_GET_BEEP_CONTROL		0x0f0a
-#define AC_VERB_GET_EAPD_BTLENABLE		0x0f0c
-#define AC_VERB_GET_DIGI_CONVERT_1		0x0f0d
-#define AC_VERB_GET_DIGI_CONVERT_2		0x0f0e /* unused */
-#define AC_VERB_GET_VOLUME_KNOB_CONTROL		0x0f0f
-/* f10-f1a: GPIO */
-#define AC_VERB_GET_GPIO_DATA			0x0f15
-#define AC_VERB_GET_GPIO_MASK			0x0f16
-#define AC_VERB_GET_GPIO_DIRECTION		0x0f17
-#define AC_VERB_GET_GPIO_WAKE_MASK		0x0f18
-#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK	0x0f19
-#define AC_VERB_GET_GPIO_STICKY_MASK		0x0f1a
-#define AC_VERB_GET_CONFIG_DEFAULT		0x0f1c
-/* f20: AFG/MFG */
-#define AC_VERB_GET_SUBSYSTEM_ID		0x0f20
-#define AC_VERB_GET_CVT_CHAN_COUNT		0x0f2d
-#define AC_VERB_GET_HDMI_DIP_SIZE		0x0f2e
-#define AC_VERB_GET_HDMI_ELDD			0x0f2f
-#define AC_VERB_GET_HDMI_DIP_INDEX		0x0f30
-#define AC_VERB_GET_HDMI_DIP_DATA		0x0f31
-#define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
-#define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
-#define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
-#define AC_VERB_GET_DEVICE_SEL			0xf35
-#define AC_VERB_GET_DEVICE_LIST			0xf36
-
-/*
- * SET verbs
- */
-#define AC_VERB_SET_STREAM_FORMAT		0x200
-#define AC_VERB_SET_AMP_GAIN_MUTE		0x300
-#define AC_VERB_SET_PROC_COEF			0x400
-#define AC_VERB_SET_COEF_INDEX			0x500
-#define AC_VERB_SET_CONNECT_SEL			0x701
-#define AC_VERB_SET_PROC_STATE			0x703
-#define AC_VERB_SET_SDI_SELECT			0x704
-#define AC_VERB_SET_POWER_STATE			0x705
-#define AC_VERB_SET_CHANNEL_STREAMID		0x706
-#define AC_VERB_SET_PIN_WIDGET_CONTROL		0x707
-#define AC_VERB_SET_UNSOLICITED_ENABLE		0x708
-#define AC_VERB_SET_PIN_SENSE			0x709
-#define AC_VERB_SET_BEEP_CONTROL		0x70a
-#define AC_VERB_SET_EAPD_BTLENABLE		0x70c
-#define AC_VERB_SET_DIGI_CONVERT_1		0x70d
-#define AC_VERB_SET_DIGI_CONVERT_2		0x70e
-#define AC_VERB_SET_VOLUME_KNOB_CONTROL		0x70f
-#define AC_VERB_SET_GPIO_DATA			0x715
-#define AC_VERB_SET_GPIO_MASK			0x716
-#define AC_VERB_SET_GPIO_DIRECTION		0x717
-#define AC_VERB_SET_GPIO_WAKE_MASK		0x718
-#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK	0x719
-#define AC_VERB_SET_GPIO_STICKY_MASK		0x71a
-#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0	0x71c
-#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1	0x71d
-#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2	0x71e
-#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3	0x71f
-#define AC_VERB_SET_EAPD				0x788
-#define AC_VERB_SET_CODEC_RESET			0x7ff
-#define AC_VERB_SET_CVT_CHAN_COUNT		0x72d
-#define AC_VERB_SET_HDMI_DIP_INDEX		0x730
-#define AC_VERB_SET_HDMI_DIP_DATA		0x731
-#define AC_VERB_SET_HDMI_DIP_XMIT		0x732
-#define AC_VERB_SET_HDMI_CP_CTRL		0x733
-#define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
-#define AC_VERB_SET_DEVICE_SEL			0x735
-
-/*
- * Parameter IDs
- */
-#define AC_PAR_VENDOR_ID		0x00
-#define AC_PAR_SUBSYSTEM_ID		0x01
-#define AC_PAR_REV_ID			0x02
-#define AC_PAR_NODE_COUNT		0x04
-#define AC_PAR_FUNCTION_TYPE		0x05
-#define AC_PAR_AUDIO_FG_CAP		0x08
-#define AC_PAR_AUDIO_WIDGET_CAP		0x09
-#define AC_PAR_PCM			0x0a
-#define AC_PAR_STREAM			0x0b
-#define AC_PAR_PIN_CAP			0x0c
-#define AC_PAR_AMP_IN_CAP		0x0d
-#define AC_PAR_CONNLIST_LEN		0x0e
-#define AC_PAR_POWER_STATE		0x0f
-#define AC_PAR_PROC_CAP			0x10
-#define AC_PAR_GPIO_CAP			0x11
-#define AC_PAR_AMP_OUT_CAP		0x12
-#define AC_PAR_VOL_KNB_CAP		0x13
-#define AC_PAR_DEVLIST_LEN		0x15
-#define AC_PAR_HDMI_LPCM_CAP		0x20
-
-/*
- * AC_VERB_PARAMETERS results (32bit)
- */
-
-/* Function Group Type */
-#define AC_FGT_TYPE			(0xff<<0)
-#define AC_FGT_TYPE_SHIFT		0
-#define AC_FGT_UNSOL_CAP		(1<<8)
-
-/* Audio Function Group Capabilities */
-#define AC_AFG_OUT_DELAY		(0xf<<0)
-#define AC_AFG_IN_DELAY			(0xf<<8)
-#define AC_AFG_BEEP_GEN			(1<<16)
-
-/* Audio Widget Capabilities */
-#define AC_WCAP_STEREO			(1<<0)	/* stereo I/O */
-#define AC_WCAP_IN_AMP			(1<<1)	/* AMP-in present */
-#define AC_WCAP_OUT_AMP			(1<<2)	/* AMP-out present */
-#define AC_WCAP_AMP_OVRD		(1<<3)	/* AMP-parameter override */
-#define AC_WCAP_FORMAT_OVRD		(1<<4)	/* format override */
-#define AC_WCAP_STRIPE			(1<<5)	/* stripe */
-#define AC_WCAP_PROC_WID		(1<<6)	/* Proc Widget */
-#define AC_WCAP_UNSOL_CAP		(1<<7)	/* Unsol capable */
-#define AC_WCAP_CONN_LIST		(1<<8)	/* connection list */
-#define AC_WCAP_DIGITAL			(1<<9)	/* digital I/O */
-#define AC_WCAP_POWER			(1<<10)	/* power control */
-#define AC_WCAP_LR_SWAP			(1<<11)	/* L/R swap */
-#define AC_WCAP_CP_CAPS			(1<<12) /* content protection */
-#define AC_WCAP_CHAN_CNT_EXT		(7<<13)	/* channel count ext */
-#define AC_WCAP_DELAY			(0xf<<16)
-#define AC_WCAP_DELAY_SHIFT		16
-#define AC_WCAP_TYPE			(0xf<<20)
-#define AC_WCAP_TYPE_SHIFT		20
-
-/* supported PCM rates and bits */
-#define AC_SUPPCM_RATES			(0xfff << 0)
-#define AC_SUPPCM_BITS_8		(1<<16)
-#define AC_SUPPCM_BITS_16		(1<<17)
-#define AC_SUPPCM_BITS_20		(1<<18)
-#define AC_SUPPCM_BITS_24		(1<<19)
-#define AC_SUPPCM_BITS_32		(1<<20)
-
-/* supported PCM stream format */
-#define AC_SUPFMT_PCM			(1<<0)
-#define AC_SUPFMT_FLOAT32		(1<<1)
-#define AC_SUPFMT_AC3			(1<<2)
-
-/* GP I/O count */
-#define AC_GPIO_IO_COUNT		(0xff<<0)
-#define AC_GPIO_O_COUNT			(0xff<<8)
-#define AC_GPIO_O_COUNT_SHIFT		8
-#define AC_GPIO_I_COUNT			(0xff<<16)
-#define AC_GPIO_I_COUNT_SHIFT		16
-#define AC_GPIO_UNSOLICITED		(1<<30)
-#define AC_GPIO_WAKE			(1<<31)
-
-/* Converter stream, channel */
-#define AC_CONV_CHANNEL			(0xf<<0)
-#define AC_CONV_STREAM			(0xf<<4)
-#define AC_CONV_STREAM_SHIFT		4
-
-/* Input converter SDI select */
-#define AC_SDI_SELECT			(0xf<<0)
-
-/* stream format id */
-#define AC_FMT_CHAN_SHIFT		0
-#define AC_FMT_CHAN_MASK		(0x0f << 0)
-#define AC_FMT_BITS_SHIFT		4
-#define AC_FMT_BITS_MASK		(7 << 4)
-#define AC_FMT_BITS_8			(0 << 4)
-#define AC_FMT_BITS_16			(1 << 4)
-#define AC_FMT_BITS_20			(2 << 4)
-#define AC_FMT_BITS_24			(3 << 4)
-#define AC_FMT_BITS_32			(4 << 4)
-#define AC_FMT_DIV_SHIFT		8
-#define AC_FMT_DIV_MASK			(7 << 8)
-#define AC_FMT_MULT_SHIFT		11
-#define AC_FMT_MULT_MASK		(7 << 11)
-#define AC_FMT_BASE_SHIFT		14
-#define AC_FMT_BASE_48K			(0 << 14)
-#define AC_FMT_BASE_44K			(1 << 14)
-#define AC_FMT_TYPE_SHIFT		15
-#define AC_FMT_TYPE_PCM			(0 << 15)
-#define AC_FMT_TYPE_NON_PCM		(1 << 15)
-
-/* Unsolicited response control */
-#define AC_UNSOL_TAG			(0x3f<<0)
-#define AC_UNSOL_ENABLED		(1<<7)
-#define AC_USRSP_EN			AC_UNSOL_ENABLED
-
-/* Unsolicited responses */
-#define AC_UNSOL_RES_TAG		(0x3f<<26)
-#define AC_UNSOL_RES_TAG_SHIFT		26
-#define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
-#define AC_UNSOL_RES_SUBTAG_SHIFT	21
-#define AC_UNSOL_RES_DE			(0x3f<<15)  /* Device Entry
-						     * (for DP1.2 MST)
-						     */
-#define AC_UNSOL_RES_DE_SHIFT		15
-#define AC_UNSOL_RES_IA			(1<<2)	/* Inactive (for DP1.2 MST) */
-#define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
-#define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
-#define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
-#define AC_UNSOL_RES_CP_READY		(1<<0)	/* content protection */
-
-/* Pin widget capabilies */
-#define AC_PINCAP_IMP_SENSE		(1<<0)	/* impedance sense capable */
-#define AC_PINCAP_TRIG_REQ		(1<<1)	/* trigger required */
-#define AC_PINCAP_PRES_DETECT		(1<<2)	/* presence detect capable */
-#define AC_PINCAP_HP_DRV		(1<<3)	/* headphone drive capable */
-#define AC_PINCAP_OUT			(1<<4)	/* output capable */
-#define AC_PINCAP_IN			(1<<5)	/* input capable */
-#define AC_PINCAP_BALANCE		(1<<6)	/* balanced I/O capable */
-/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification,
- *       but is marked reserved in the Intel HDA specification.
- */
-#define AC_PINCAP_LR_SWAP		(1<<7)	/* L/R swap */
-/* Note: The same bit as LR_SWAP is newly defined as HDMI capability
- *       in HD-audio specification
- */
-#define AC_PINCAP_HDMI			(1<<7)	/* HDMI pin */
-#define AC_PINCAP_DP			(1<<24)	/* DisplayPort pin, can
-						 * coexist with AC_PINCAP_HDMI
-						 */
-#define AC_PINCAP_VREF			(0x37<<8)
-#define AC_PINCAP_VREF_SHIFT		8
-#define AC_PINCAP_EAPD			(1<<16)	/* EAPD capable */
-#define AC_PINCAP_HBR			(1<<27)	/* High Bit Rate */
-/* Vref status (used in pin cap) */
-#define AC_PINCAP_VREF_HIZ		(1<<0)	/* Hi-Z */
-#define AC_PINCAP_VREF_50		(1<<1)	/* 50% */
-#define AC_PINCAP_VREF_GRD		(1<<2)	/* ground */
-#define AC_PINCAP_VREF_80		(1<<4)	/* 80% */
-#define AC_PINCAP_VREF_100		(1<<5)	/* 100% */
-
-/* Amplifier capabilities */
-#define AC_AMPCAP_OFFSET		(0x7f<<0)  /* 0dB offset */
-#define AC_AMPCAP_OFFSET_SHIFT		0
-#define AC_AMPCAP_NUM_STEPS		(0x7f<<8)  /* number of steps */
-#define AC_AMPCAP_NUM_STEPS_SHIFT	8
-#define AC_AMPCAP_STEP_SIZE		(0x7f<<16) /* step size 0-32dB
-						    * in 0.25dB
-						    */
-#define AC_AMPCAP_STEP_SIZE_SHIFT	16
-#define AC_AMPCAP_MUTE			(1<<31)    /* mute capable */
-#define AC_AMPCAP_MUTE_SHIFT		31
-
-/* driver-specific amp-caps: using bits 24-30 */
-#define AC_AMPCAP_MIN_MUTE		(1 << 30) /* min-volume = mute */
-
-/* Connection list */
-#define AC_CLIST_LENGTH			(0x7f<<0)
-#define AC_CLIST_LONG			(1<<7)
-
-/* Supported power status */
-#define AC_PWRST_D0SUP			(1<<0)
-#define AC_PWRST_D1SUP			(1<<1)
-#define AC_PWRST_D2SUP			(1<<2)
-#define AC_PWRST_D3SUP			(1<<3)
-#define AC_PWRST_D3COLDSUP		(1<<4)
-#define AC_PWRST_S3D3COLDSUP		(1<<29)
-#define AC_PWRST_CLKSTOP		(1<<30)
-#define AC_PWRST_EPSS			(1U<<31)
-
-/* Power state values */
-#define AC_PWRST_SETTING		(0xf<<0)
-#define AC_PWRST_ACTUAL			(0xf<<4)
-#define AC_PWRST_ACTUAL_SHIFT		4
-#define AC_PWRST_D0			0x00
-#define AC_PWRST_D1			0x01
-#define AC_PWRST_D2			0x02
-#define AC_PWRST_D3			0x03
-#define AC_PWRST_ERROR                  (1<<8)
-#define AC_PWRST_CLK_STOP_OK            (1<<9)
-#define AC_PWRST_SETTING_RESET          (1<<10)
-
-/* Processing capabilies */
-#define AC_PCAP_BENIGN			(1<<0)
-#define AC_PCAP_NUM_COEF		(0xff<<8)
-#define AC_PCAP_NUM_COEF_SHIFT		8
-
-/* Volume knobs capabilities */
-#define AC_KNBCAP_NUM_STEPS		(0x7f<<0)
-#define AC_KNBCAP_DELTA			(1<<7)
-
-/* HDMI LPCM capabilities */
-#define AC_LPCMCAP_48K_CP_CHNS		(0x0f<<0) /* max channels w/ CP-on */	
-#define AC_LPCMCAP_48K_NO_CHNS		(0x0f<<4) /* max channels w/o CP-on */
-#define AC_LPCMCAP_48K_20BIT		(1<<8)	/* 20b bitrate supported */
-#define AC_LPCMCAP_48K_24BIT		(1<<9)	/* 24b bitrate supported */
-#define AC_LPCMCAP_96K_CP_CHNS		(0x0f<<10) /* max channels w/ CP-on */	
-#define AC_LPCMCAP_96K_NO_CHNS		(0x0f<<14) /* max channels w/o CP-on */
-#define AC_LPCMCAP_96K_20BIT		(1<<18)	/* 20b bitrate supported */
-#define AC_LPCMCAP_96K_24BIT		(1<<19)	/* 24b bitrate supported */
-#define AC_LPCMCAP_192K_CP_CHNS		(0x0f<<20) /* max channels w/ CP-on */	
-#define AC_LPCMCAP_192K_NO_CHNS		(0x0f<<24) /* max channels w/o CP-on */
-#define AC_LPCMCAP_192K_20BIT		(1<<28)	/* 20b bitrate supported */
-#define AC_LPCMCAP_192K_24BIT		(1<<29)	/* 24b bitrate supported */
-#define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
-#define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */
-
-/* Display pin's device list length */
-#define AC_DEV_LIST_LEN_MASK		0x3f
-#define AC_MAX_DEV_LIST_LEN		64
-
-/*
- * Control Parameters
- */
-
-/* Amp gain/mute */
-#define AC_AMP_MUTE			(1<<7)
-#define AC_AMP_GAIN			(0x7f)
-#define AC_AMP_GET_INDEX		(0xf<<0)
-
-#define AC_AMP_GET_LEFT			(1<<13)
-#define AC_AMP_GET_RIGHT		(0<<13)
-#define AC_AMP_GET_OUTPUT		(1<<15)
-#define AC_AMP_GET_INPUT		(0<<15)
-
-#define AC_AMP_SET_INDEX		(0xf<<8)
-#define AC_AMP_SET_INDEX_SHIFT		8
-#define AC_AMP_SET_RIGHT		(1<<12)
-#define AC_AMP_SET_LEFT			(1<<13)
-#define AC_AMP_SET_INPUT		(1<<14)
-#define AC_AMP_SET_OUTPUT		(1<<15)
-
-/* DIGITAL1 bits */
-#define AC_DIG1_ENABLE			(1<<0)
-#define AC_DIG1_V			(1<<1)
-#define AC_DIG1_VCFG			(1<<2)
-#define AC_DIG1_EMPHASIS		(1<<3)
-#define AC_DIG1_COPYRIGHT		(1<<4)
-#define AC_DIG1_NONAUDIO		(1<<5)
-#define AC_DIG1_PROFESSIONAL		(1<<6)
-#define AC_DIG1_LEVEL			(1<<7)
-
-/* DIGITAL2 bits */
-#define AC_DIG2_CC			(0x7f<<0)
-
-/* DIGITAL3 bits */
-#define AC_DIG3_ICT			(0xf<<0)
-#define AC_DIG3_KAE			(1<<7)
-
-/* Pin widget control - 8bit */
-#define AC_PINCTL_EPT			(0x3<<0)
-#define AC_PINCTL_EPT_NATIVE		0
-#define AC_PINCTL_EPT_HBR		3
-#define AC_PINCTL_VREFEN		(0x7<<0)
-#define AC_PINCTL_VREF_HIZ		0	/* Hi-Z */
-#define AC_PINCTL_VREF_50		1	/* 50% */
-#define AC_PINCTL_VREF_GRD		2	/* ground */
-#define AC_PINCTL_VREF_80		4	/* 80% */
-#define AC_PINCTL_VREF_100		5	/* 100% */
-#define AC_PINCTL_IN_EN			(1<<5)
-#define AC_PINCTL_OUT_EN		(1<<6)
-#define AC_PINCTL_HP_EN			(1<<7)
-
-/* Pin sense - 32bit */
-#define AC_PINSENSE_IMPEDANCE_MASK	(0x7fffffff)
-#define AC_PINSENSE_PRESENCE		(1<<31)
-#define AC_PINSENSE_ELDV		(1<<30)	/* ELD valid (HDMI) */
-
-/* EAPD/BTL enable - 32bit */
-#define AC_EAPDBTL_BALANCED		(1<<0)
-#define AC_EAPDBTL_EAPD			(1<<1)
-#define AC_EAPDBTL_LR_SWAP		(1<<2)
-
-/* HDMI ELD data */
-#define AC_ELDD_ELD_VALID		(1<<31)
-#define AC_ELDD_ELD_DATA		0xff
-
-/* HDMI DIP size */
-#define AC_DIPSIZE_ELD_BUF		(1<<3) /* ELD buf size of packet size */
-#define AC_DIPSIZE_PACK_IDX		(0x07<<0) /* packet index */
-
-/* HDMI DIP index */
-#define AC_DIPIDX_PACK_IDX		(0x07<<5) /* packet idnex */
-#define AC_DIPIDX_BYTE_IDX		(0x1f<<0) /* byte index */
-
-/* HDMI DIP xmit (transmit) control */
-#define AC_DIPXMIT_MASK			(0x3<<6)
-#define AC_DIPXMIT_DISABLE		(0x0<<6) /* disable xmit */
-#define AC_DIPXMIT_ONCE			(0x2<<6) /* xmit once then disable */
-#define AC_DIPXMIT_BEST			(0x3<<6) /* best effort */
-
-/* HDMI content protection (CP) control */
-#define AC_CPCTRL_CES			(1<<9) /* current encryption state */
-#define AC_CPCTRL_READY			(1<<8) /* ready bit */
-#define AC_CPCTRL_SUBTAG		(0x1f<<3) /* subtag for unsol-resp */
-#define AC_CPCTRL_STATE			(3<<0) /* current CP request state */
-
-/* Converter channel <-> HDMI slot mapping */
-#define AC_CVTMAP_HDMI_SLOT		(0xf<<0) /* HDMI slot number */
-#define AC_CVTMAP_CHAN			(0xf<<4) /* converter channel number */
-
-/* configuration default - 32bit */
-#define AC_DEFCFG_SEQUENCE		(0xf<<0)
-#define AC_DEFCFG_DEF_ASSOC		(0xf<<4)
-#define AC_DEFCFG_ASSOC_SHIFT		4
-#define AC_DEFCFG_MISC			(0xf<<8)
-#define AC_DEFCFG_MISC_SHIFT		8
-#define AC_DEFCFG_MISC_NO_PRESENCE	(1<<0)
-#define AC_DEFCFG_COLOR			(0xf<<12)
-#define AC_DEFCFG_COLOR_SHIFT		12
-#define AC_DEFCFG_CONN_TYPE		(0xf<<16)
-#define AC_DEFCFG_CONN_TYPE_SHIFT	16
-#define AC_DEFCFG_DEVICE		(0xf<<20)
-#define AC_DEFCFG_DEVICE_SHIFT		20
-#define AC_DEFCFG_LOCATION		(0x3f<<24)
-#define AC_DEFCFG_LOCATION_SHIFT	24
-#define AC_DEFCFG_PORT_CONN		(0x3<<30)
-#define AC_DEFCFG_PORT_CONN_SHIFT	30
-
-/* Display pin's device list entry */
-#define AC_DE_PD			(1<<0)
-#define AC_DE_ELDV			(1<<1)
-#define AC_DE_IA			(1<<2)
-
-/* device device types (0x0-0xf) */
-enum {
-	AC_JACK_LINE_OUT,
-	AC_JACK_SPEAKER,
-	AC_JACK_HP_OUT,
-	AC_JACK_CD,
-	AC_JACK_SPDIF_OUT,
-	AC_JACK_DIG_OTHER_OUT,
-	AC_JACK_MODEM_LINE_SIDE,
-	AC_JACK_MODEM_HAND_SIDE,
-	AC_JACK_LINE_IN,
-	AC_JACK_AUX,
-	AC_JACK_MIC_IN,
-	AC_JACK_TELEPHONY,
-	AC_JACK_SPDIF_IN,
-	AC_JACK_DIG_OTHER_IN,
-	AC_JACK_OTHER = 0xf,
-};
-
-/* jack connection types (0x0-0xf) */
-enum {
-	AC_JACK_CONN_UNKNOWN,
-	AC_JACK_CONN_1_8,
-	AC_JACK_CONN_1_4,
-	AC_JACK_CONN_ATAPI,
-	AC_JACK_CONN_RCA,
-	AC_JACK_CONN_OPTICAL,
-	AC_JACK_CONN_OTHER_DIGITAL,
-	AC_JACK_CONN_OTHER_ANALOG,
-	AC_JACK_CONN_DIN,
-	AC_JACK_CONN_XLR,
-	AC_JACK_CONN_RJ11,
-	AC_JACK_CONN_COMB,
-	AC_JACK_CONN_OTHER = 0xf,
-};
-
-/* jack colors (0x0-0xf) */
-enum {
-	AC_JACK_COLOR_UNKNOWN,
-	AC_JACK_COLOR_BLACK,
-	AC_JACK_COLOR_GREY,
-	AC_JACK_COLOR_BLUE,
-	AC_JACK_COLOR_GREEN,
-	AC_JACK_COLOR_RED,
-	AC_JACK_COLOR_ORANGE,
-	AC_JACK_COLOR_YELLOW,
-	AC_JACK_COLOR_PURPLE,
-	AC_JACK_COLOR_PINK,
-	AC_JACK_COLOR_WHITE = 0xe,
-	AC_JACK_COLOR_OTHER,
-};
-
-/* Jack location (0x0-0x3f) */
-/* common case */
-enum {
-	AC_JACK_LOC_NONE,
-	AC_JACK_LOC_REAR,
-	AC_JACK_LOC_FRONT,
-	AC_JACK_LOC_LEFT,
-	AC_JACK_LOC_RIGHT,
-	AC_JACK_LOC_TOP,
-	AC_JACK_LOC_BOTTOM,
-};
-/* bits 4-5 */
-enum {
-	AC_JACK_LOC_EXTERNAL = 0x00,
-	AC_JACK_LOC_INTERNAL = 0x10,
-	AC_JACK_LOC_SEPARATE = 0x20,
-	AC_JACK_LOC_OTHER    = 0x30,
-};
-enum {
-	/* external on primary chasis */
-	AC_JACK_LOC_REAR_PANEL = 0x07,
-	AC_JACK_LOC_DRIVE_BAY,
-	/* internal */
-	AC_JACK_LOC_RISER = 0x17,
-	AC_JACK_LOC_HDMI,
-	AC_JACK_LOC_ATAPI,
-	/* others */
-	AC_JACK_LOC_MOBILE_IN = 0x37,
-	AC_JACK_LOC_MOBILE_OUT,
-};
-
-/* Port connectivity (0-3) */
-enum {
-	AC_JACK_PORT_COMPLEX,
-	AC_JACK_PORT_NONE,
-	AC_JACK_PORT_FIXED,
-	AC_JACK_PORT_BOTH,
-};
-
-/* max. codec address */
-#define HDA_MAX_CODEC_ADDRESS	0x0f
+#include <sound/hda_verbs.h>
 
 /*
  * generic arrays
-- 
cgit v1.2.2


From 8f2fe346822419ee729c081b71c8835b733e0884 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 6 Dec 2013 11:02:50 +0100
Subject: ASoC: Add support for the Analog Devices AXI-I2S core

This patch adds support for the AXI-I2S softcore. The core implements a simple
bidirectional I2S transceiver and is used by Analog Devices in some of their
reference designs for various FPGA platforms.

The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/Kconfig       |   1 +
 sound/soc/Makefile      |   1 +
 sound/soc/adi/Kconfig   |  13 +++
 sound/soc/adi/Makefile  |   3 +
 sound/soc/adi/axi-i2s.c | 277 ++++++++++++++++++++++++++++++++++++++++++++++++
 5 files changed, 295 insertions(+)
 create mode 100644 sound/soc/adi/Kconfig
 create mode 100644 sound/soc/adi/Makefile
 create mode 100644 sound/soc/adi/axi-i2s.c

(limited to 'sound')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 5138b8493051..866dfec4b6b5 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -31,6 +31,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM
 	select SND_DMAENGINE_PCM
 
 # All the supported SoCs
+source "sound/soc/adi/Kconfig"
 source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
 source "sound/soc/blackfin/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 8b9e70105dd2..c70c7f76d2df 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -8,6 +8,7 @@ endif
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
 obj-$(CONFIG_SND_SOC)	+= generic/
+obj-$(CONFIG_SND_SOC)	+= adi/
 obj-$(CONFIG_SND_SOC)	+= atmel/
 obj-$(CONFIG_SND_SOC)	+= au1x/
 obj-$(CONFIG_SND_SOC)	+= blackfin/
diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig
new file mode 100644
index 000000000000..46f4b79f7a87
--- /dev/null
+++ b/sound/soc/adi/Kconfig
@@ -0,0 +1,13 @@
+config SND_SOC_ADI
+	tristate "Audio support for Analog Devices reference designs"
+	depends on MICROBLAZE || ARCH_ZYNQ || COMPILE_TEST
+	help
+	  Audio support for various reference designs by Analog Devices.
+
+config SND_SOC_ADI_AXI_I2S
+	tristate "AXI-I2S support"
+	depends on SND_SOC_ADI
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  ASoC driver for the Analog Devices AXI-I2S softcore peripheral.
diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile
new file mode 100644
index 000000000000..d32c21ae14b6
--- /dev/null
+++ b/sound/soc/adi/Makefile
@@ -0,0 +1,3 @@
+snd-soc-adi-axi-i2s-objs := axi-i2s.o
+
+obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
new file mode 100644
index 000000000000..0822c7713669
--- /dev/null
+++ b/sound/soc/adi/axi-i2s.c
@@ -0,0 +1,277 @@
+/*
+ * Copyright (C) 2012-2013, Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/clk.h>
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#define AXI_I2S_REG_RESET	0x00
+#define AXI_I2S_REG_CTRL	0x04
+#define AXI_I2S_REG_CLK_CTRL	0x08
+#define AXI_I2S_REG_STATUS	0x10
+
+#define AXI_I2S_REG_RX_FIFO	0x28
+#define AXI_I2S_REG_TX_FIFO	0x2C
+
+#define AXI_I2S_RESET_GLOBAL	BIT(0)
+#define AXI_I2S_RESET_TX_FIFO	BIT(1)
+#define AXI_I2S_RESET_RX_FIFO	BIT(2)
+
+#define AXI_I2S_CTRL_TX_EN	BIT(0)
+#define AXI_I2S_CTRL_RX_EN	BIT(1)
+
+/* The frame size is configurable, but for now we always set it 64 bit */
+#define AXI_I2S_BITS_PER_FRAME 64
+
+struct axi_i2s {
+	struct regmap *regmap;
+	struct clk *clk;
+	struct clk *clk_ref;
+
+	struct snd_soc_dai_driver dai_driver;
+
+	struct snd_dmaengine_dai_dma_data capture_dma_data;
+	struct snd_dmaengine_dai_dma_data playback_dma_data;
+
+	struct snd_ratnum ratnum;
+	struct snd_pcm_hw_constraint_ratnums rate_constraints;
+};
+
+static int axi_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+	struct snd_soc_dai *dai)
+{
+	struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		mask = AXI_I2S_CTRL_RX_EN;
+	else
+		mask = AXI_I2S_CTRL_TX_EN;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = mask;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(i2s->regmap, AXI_I2S_REG_CTRL, mask, val);
+
+	return 0;
+}
+
+static int axi_i2s_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int bclk_div, word_size;
+	unsigned int bclk_rate;
+
+	bclk_rate = params_rate(params) * AXI_I2S_BITS_PER_FRAME;
+
+	word_size = AXI_I2S_BITS_PER_FRAME / 2 - 1;
+	bclk_div = DIV_ROUND_UP(clk_get_rate(i2s->clk_ref), bclk_rate) / 2 - 1;
+
+	regmap_write(i2s->regmap, AXI_I2S_REG_CLK_CTRL, (word_size << 16) |
+		bclk_div);
+
+	return 0;
+}
+
+static int axi_i2s_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	uint32_t mask;
+	int ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		mask = AXI_I2S_RESET_RX_FIFO;
+	else
+		mask = AXI_I2S_RESET_TX_FIFO;
+
+	regmap_write(i2s->regmap, AXI_I2S_REG_RESET, mask);
+
+	ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_RATE,
+			   &i2s->rate_constraints);
+	if (ret)
+		return ret;
+
+	return clk_prepare_enable(i2s->clk_ref);
+}
+
+static void axi_i2s_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable_unprepare(i2s->clk_ref);
+}
+
+static int axi_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data,
+		&i2s->capture_dma_data);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops axi_i2s_dai_ops = {
+	.startup = axi_i2s_startup,
+	.shutdown = axi_i2s_shutdown,
+	.trigger = axi_i2s_trigger,
+	.hw_params = axi_i2s_hw_params,
+};
+
+static struct snd_soc_dai_driver axi_i2s_dai = {
+	.probe = axi_i2s_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_KNOT,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_KNOT,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE,
+	},
+	.ops = &axi_i2s_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static const struct snd_soc_component_driver axi_i2s_component = {
+	.name = "axi-i2s",
+};
+
+static const struct regmap_config axi_i2s_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = AXI_I2S_REG_STATUS,
+};
+
+static int axi_i2s_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct axi_i2s *i2s;
+	void __iomem *base;
+	int ret;
+
+	i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL);
+	if (!i2s)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, i2s);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_request_and_ioremap(&pdev->dev, res);
+	if (!base)
+		return -EBUSY;
+
+	i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+		&axi_i2s_regmap_config);
+	if (IS_ERR(i2s->regmap))
+		return PTR_ERR(i2s->regmap);
+
+	i2s->clk = devm_clk_get(&pdev->dev, "axi");
+	if (IS_ERR(i2s->clk))
+		return PTR_ERR(i2s->clk);
+
+	i2s->clk_ref = devm_clk_get(&pdev->dev, "ref");
+	if (IS_ERR(i2s->clk_ref))
+		return PTR_ERR(i2s->clk_ref);
+
+	ret = clk_prepare_enable(i2s->clk);
+	if (ret)
+		return ret;
+
+	i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO;
+	i2s->playback_dma_data.addr_width = 4;
+	i2s->playback_dma_data.maxburst = 1;
+
+	i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO;
+	i2s->capture_dma_data.addr_width = 4;
+	i2s->capture_dma_data.maxburst = 1;
+
+	i2s->ratnum.num = clk_get_rate(i2s->clk_ref) / 2 / AXI_I2S_BITS_PER_FRAME;
+	i2s->ratnum.den_step = 1;
+	i2s->ratnum.den_min = 1;
+	i2s->ratnum.den_max = 64;
+
+	i2s->rate_constraints.rats = &i2s->ratnum;
+	i2s->rate_constraints.nrats = 1;
+
+	regmap_write(i2s->regmap, AXI_I2S_REG_RESET, AXI_I2S_RESET_GLOBAL);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &axi_i2s_component,
+					 &axi_i2s_dai, 1);
+	if (ret)
+		goto err_clk_disable;
+
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	if (ret)
+		goto err_clk_disable;
+
+err_clk_disable:
+	clk_disable_unprepare(i2s->clk);
+	return ret;
+}
+
+static int axi_i2s_dev_remove(struct platform_device *pdev)
+{
+	struct axi_i2s *i2s = platform_get_drvdata(pdev);
+
+	clk_disable_unprepare(i2s->clk);
+
+	return 0;
+}
+
+static const struct of_device_id axi_i2s_of_match[] = {
+	{ .compatible = "adi,axi-i2s-1.00.a", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, axi_i2s_of_match);
+
+static struct platform_driver axi_i2s_driver = {
+	.driver = {
+		.name = "axi-i2s",
+		.owner = THIS_MODULE,
+		.of_match_table = axi_i2s_of_match,
+	},
+	.probe = axi_i2s_probe,
+	.remove = axi_i2s_dev_remove,
+};
+module_platform_driver(axi_i2s_driver);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("AXI I2S driver");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.2


From 429e4374cc007f260c7d0e2d0df5247deeaf8fbe Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 6 Dec 2013 11:02:52 +0100
Subject: ASoC: Add support for the Analog Devices AXI-SPDIF driver

This patch adds a ASoC driver for the AXI-SPDIF softcore. The core implements a
simple SPDIF transmitter and is used on some Analog Devices' reference designs
for various FPGA platforms. For now the driver only support the PL330 as the the
DMA controller.

The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/adi/Kconfig     |   8 ++
 sound/soc/adi/Makefile    |   2 +
 sound/soc/adi/axi-spdif.c | 272 ++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 282 insertions(+)
 create mode 100644 sound/soc/adi/axi-spdif.c

(limited to 'sound')

diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig
index 46f4b79f7a87..dd763f55edac 100644
--- a/sound/soc/adi/Kconfig
+++ b/sound/soc/adi/Kconfig
@@ -11,3 +11,11 @@ config SND_SOC_ADI_AXI_I2S
 	select REGMAP_MMIO
 	help
 	  ASoC driver for the Analog Devices AXI-I2S softcore peripheral.
+
+config SND_SOC_ADI_AXI_SPDIF
+	tristate "AXI-SPDIF support"
+	depends on SND_SOC_ADI
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	select REGMAP_MMIO
+	help
+	  ASoC driver for the Analog Devices AXI-SPDIF softcore peripheral.
diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile
index d32c21ae14b6..64456c1e5347 100644
--- a/sound/soc/adi/Makefile
+++ b/sound/soc/adi/Makefile
@@ -1,3 +1,5 @@
 snd-soc-adi-axi-i2s-objs := axi-i2s.o
+snd-soc-adi-axi-spdif-objs := axi-spdif.o
 
 obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o
+obj-$(CONFIG_SND_SOC_ADI_AXI_SPDIF) += snd-soc-adi-axi-spdif.o
diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c
new file mode 100644
index 000000000000..d5408d23b438
--- /dev/null
+++ b/sound/soc/adi/axi-spdif.c
@@ -0,0 +1,272 @@
+/*
+ * Copyright (C) 2012-2013, Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/clk.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/dmaengine_pcm.h>
+
+#define AXI_SPDIF_REG_CTRL	0x0
+#define AXI_SPDIF_REG_STAT	0x4
+#define AXI_SPDIF_REG_TX_FIFO	0xc
+
+#define AXI_SPDIF_CTRL_TXDATA BIT(1)
+#define AXI_SPDIF_CTRL_TXEN BIT(0)
+#define AXI_SPDIF_CTRL_CLKDIV_OFFSET 8
+#define AXI_SPDIF_CTRL_CLKDIV_MASK (0xff << 8)
+
+#define AXI_SPDIF_FREQ_44100	(0x0 << 6)
+#define AXI_SPDIF_FREQ_48000	(0x1 << 6)
+#define AXI_SPDIF_FREQ_32000	(0x2 << 6)
+#define AXI_SPDIF_FREQ_NA	(0x3 << 6)
+
+struct axi_spdif {
+	struct regmap *regmap;
+	struct clk *clk;
+	struct clk *clk_ref;
+
+	struct snd_dmaengine_dai_dma_data dma_data;
+
+	struct snd_ratnum ratnum;
+	struct snd_pcm_hw_constraint_ratnums rate_constraints;
+};
+
+static int axi_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+	struct snd_soc_dai *dai)
+{
+	struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	unsigned int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = AXI_SPDIF_CTRL_TXDATA;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL,
+		AXI_SPDIF_CTRL_TXDATA, val);
+
+	return 0;
+}
+
+static int axi_spdif_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	unsigned int rate = params_rate(params);
+	unsigned int clkdiv, stat;
+
+	switch (params_rate(params)) {
+	case 32000:
+		stat = AXI_SPDIF_FREQ_32000;
+		break;
+	case 44100:
+		stat = AXI_SPDIF_FREQ_44100;
+		break;
+	case 48000:
+		stat = AXI_SPDIF_FREQ_48000;
+		break;
+	default:
+		stat = AXI_SPDIF_FREQ_NA;
+		break;
+	}
+
+	clkdiv = DIV_ROUND_CLOSEST(clk_get_rate(spdif->clk_ref),
+			rate * 64 * 2) - 1;
+	clkdiv <<= AXI_SPDIF_CTRL_CLKDIV_OFFSET;
+
+	regmap_write(spdif->regmap, AXI_SPDIF_REG_STAT, stat);
+	regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL,
+		AXI_SPDIF_CTRL_CLKDIV_MASK, clkdiv);
+
+	return 0;
+}
+
+static int axi_spdif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai, &spdif->dma_data, NULL);
+
+	return 0;
+}
+
+static int axi_spdif_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0,
+			   SNDRV_PCM_HW_PARAM_RATE,
+			   &spdif->rate_constraints);
+	if (ret)
+		return ret;
+
+	ret = clk_prepare_enable(spdif->clk_ref);
+	if (ret)
+		return ret;
+
+	regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL,
+		AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN);
+
+	return 0;
+}
+
+static void axi_spdif_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+
+	regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL,
+		AXI_SPDIF_CTRL_TXEN, 0);
+
+	clk_disable_unprepare(spdif->clk_ref);
+}
+
+static const struct snd_soc_dai_ops axi_spdif_dai_ops = {
+	.startup = axi_spdif_startup,
+	.shutdown = axi_spdif_shutdown,
+	.trigger = axi_spdif_trigger,
+	.hw_params = axi_spdif_hw_params,
+};
+
+static struct snd_soc_dai_driver axi_spdif_dai = {
+	.probe = axi_spdif_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_KNOT,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &axi_spdif_dai_ops,
+};
+
+static const struct snd_soc_component_driver axi_spdif_component = {
+	.name = "axi-spdif",
+};
+
+static const struct regmap_config axi_spdif_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = AXI_SPDIF_REG_STAT,
+};
+
+static int axi_spdif_probe(struct platform_device *pdev)
+{
+	struct axi_spdif *spdif;
+	struct resource *res;
+	void __iomem *base;
+	int ret;
+
+	spdif = devm_kzalloc(&pdev->dev, sizeof(*spdif), GFP_KERNEL);
+	if (!spdif)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, spdif);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_request_and_ioremap(&pdev->dev, res);
+	if (!base)
+		return -EBUSY;
+
+	spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+					    &axi_spdif_regmap_config);
+	if (IS_ERR(spdif->regmap))
+		return PTR_ERR(spdif->regmap);
+
+	spdif->clk = devm_clk_get(&pdev->dev, "axi");
+	if (IS_ERR(spdif->clk))
+		return PTR_ERR(spdif->clk);
+
+	spdif->clk_ref = devm_clk_get(&pdev->dev, "ref");
+	if (IS_ERR(spdif->clk_ref))
+		return PTR_ERR(spdif->clk_ref);
+
+	ret = clk_prepare_enable(spdif->clk);
+	if (ret)
+		return ret;
+
+	spdif->dma_data.addr = res->start + AXI_SPDIF_REG_TX_FIFO;
+	spdif->dma_data.addr_width = 4;
+	spdif->dma_data.maxburst = 1;
+
+	spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128;
+	spdif->ratnum.den_step = 1;
+	spdif->ratnum.den_min = 1;
+	spdif->ratnum.den_max = 64;
+
+	spdif->rate_constraints.rats = &spdif->ratnum;
+	spdif->rate_constraints.nrats = 1;
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &axi_spdif_component,
+					 &axi_spdif_dai, 1);
+	if (ret)
+		goto err_clk_disable;
+
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	if (ret)
+		goto err_clk_disable;
+
+	return 0;
+
+err_clk_disable:
+	clk_disable_unprepare(spdif->clk);
+	return ret;
+}
+
+static int axi_spdif_dev_remove(struct platform_device *pdev)
+{
+	struct axi_spdif *spdif = platform_get_drvdata(pdev);
+
+	clk_disable_unprepare(spdif->clk);
+
+	return 0;
+}
+
+static const struct of_device_id axi_spdif_of_match[] = {
+	{ .compatible = "adi,axi-spdif-tx-1.00.a", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, axi_spdif_of_match);
+
+static struct platform_driver axi_spdif_driver = {
+	.driver = {
+		.name = "axi-spdif",
+		.owner = THIS_MODULE,
+		.of_match_table = axi_spdif_of_match,
+	},
+	.probe = axi_spdif_probe,
+	.remove = axi_spdif_dev_remove,
+};
+module_platform_driver(axi_spdif_driver);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("AXI SPDIF driver");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.2


From 58381da687742a3d8bbb98362152de8326a0c077 Mon Sep 17 00:00:00 2001
From: Jan Weitzel <j.weitzel@phytec.de>
Date: Thu, 5 Dec 2013 09:54:02 +0100
Subject: ASoC: tlv320aic3x: no mono controls 3007 model

if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite
registers for class-d amplifier.
classd amplifier controls are only used for AIC3X_MODEL_3007.

Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route
and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007
codecs.

Testet only with AIC3X_MODEL_3007

Signed-off-by: Jan Weitzel <j.weitzel@phytec.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/tlv320aic3x.c | 126 +++++++++++++++++++++++++++--------------
 1 file changed, 83 insertions(+), 43 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 546d16b7d38f..470fbfb4b386 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -350,16 +350,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 			 DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL,
 			 0, 118, 1, output_stage_tlv),
 
-	SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume",
-			 LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
-			 0, 118, 1, output_stage_tlv),
-	SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume",
-			 PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
-			 0, 118, 1, output_stage_tlv),
-	SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
-			 DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
-			 0, 118, 1, output_stage_tlv),
-
 	SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume",
 			 LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL,
 			 0, 118, 1, output_stage_tlv),
@@ -383,7 +373,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	/* Output pin mute controls */
 	SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
 		     0x01, 0),
-	SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
 	SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
 	SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
@@ -412,6 +401,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
+static const struct snd_kcontrol_new aic3x_mono_controls[] = {
+	SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume",
+			 LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume",
+			 PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
+			 DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
+};
+
 /*
  * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps
  */
@@ -565,9 +568,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0),
 
-	/* Mono Output */
-	SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
-
 	/* Inputs to Left ADC */
 	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
 	SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
@@ -626,9 +626,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_right_line_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_right_line_mixer_controls)),
-	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0,
-			   &aic3x_mono_mixer_controls[0],
-			   ARRAY_SIZE(aic3x_mono_mixer_controls)),
 	SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_left_hp_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_left_hp_mixer_controls)),
@@ -644,7 +641,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 
 	SND_SOC_DAPM_OUTPUT("LLOUT"),
 	SND_SOC_DAPM_OUTPUT("RLOUT"),
-	SND_SOC_DAPM_OUTPUT("MONO_LOUT"),
 	SND_SOC_DAPM_OUTPUT("HPLOUT"),
 	SND_SOC_DAPM_OUTPUT("HPROUT"),
 	SND_SOC_DAPM_OUTPUT("HPLCOM"),
@@ -666,6 +662,17 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("Detection"),
 };
 
+static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = {
+	/* Mono Output */
+	SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0,
+			   &aic3x_mono_mixer_controls[0],
+			   ARRAY_SIZE(aic3x_mono_mixer_controls)),
+
+	SND_SOC_DAPM_OUTPUT("MONO_LOUT"),
+};
+
 static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = {
 	/* Class-D outputs */
 	SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0),
@@ -754,17 +761,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right Line Out", NULL, "Right DAC Mux"},
 	{"RLOUT", NULL, "Right Line Out"},
 
-	/* Mono Output */
-	{"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
-	{"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
-	{"Mono Mixer", "DACL1 Switch", "Left DAC Mux"},
-	{"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
-	{"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
-	{"Mono Mixer", "DACR1 Switch", "Right DAC Mux"},
-
-	{"Mono Out", NULL, "Mono Mixer"},
-	{"MONO_LOUT", NULL, "Mono Out"},
-
 	/* Left HP Output */
 	{"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
 	{"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
@@ -820,6 +816,18 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HPRCOM", NULL, "Right HP Com"},
 };
 
+static const struct snd_soc_dapm_route intercon_mono[] = {
+	/* Mono Output */
+	{"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
+	{"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
+	{"Mono Mixer", "DACL1 Switch", "Left DAC Mux"},
+	{"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
+	{"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
+	{"Mono Mixer", "DACR1 Switch", "Right DAC Mux"},
+	{"Mono Out", NULL, "Mono Mixer"},
+	{"MONO_LOUT", NULL, "Mono Out"},
+};
+
 static const struct snd_soc_dapm_route intercon_3007[] = {
 	/* Class-D outputs */
 	{"Left Class-D Out", NULL, "Left Line Out"},
@@ -833,11 +841,20 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
-	if (aic3x->model == AIC3X_MODEL_3007) {
+	switch (aic3x->model) {
+	case AIC3X_MODEL_3X:
+	case AIC3X_MODEL_33:
+		snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets,
+			ARRAY_SIZE(aic3x_dapm_mono_widgets));
+		snd_soc_dapm_add_routes(dapm, intercon_mono,
+					ARRAY_SIZE(intercon_mono));
+		break;
+	case AIC3X_MODEL_3007:
 		snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets,
 			ARRAY_SIZE(aic3007_dapm_widgets));
 		snd_soc_dapm_add_routes(dapm, intercon_3007,
 					ARRAY_SIZE(intercon_3007));
+		break;
 	}
 
 	return 0;
@@ -1218,6 +1235,24 @@ static int aic3x_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static void aic3x_mono_init(struct snd_soc_codec *codec)
+{
+	/* DAC to Mono Line Out default volume and route to Output mixer */
+	snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+	snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+
+	/* unmute all outputs */
+	snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE);
+
+	/* PGA to Mono Line Out default volume, disconnect from Output Mixer */
+	snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL);
+	snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL);
+
+	/* Line2 to Mono Out default volume, disconnect from Output Mixer */
+	snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
+	snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
+}
+
 /*
  * initialise the AIC3X driver
  * register the mixer and dsp interfaces with the kernel
@@ -1241,14 +1276,10 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	/* DAC to Line Out default volume and route to Output mixer */
 	snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
 	snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
-	/* DAC to Mono Line Out default volume and route to Output mixer */
-	snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
-	snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
 
 	/* unmute all outputs */
 	snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE);
 	snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE);
-	snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE);
 	snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE);
 	snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE);
 	snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE);
@@ -1269,9 +1300,6 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	/* PGA to Line Out default volume, disconnect from Output Mixer */
 	snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
 	snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
-	/* PGA to Mono Line Out default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL);
 
 	/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
 	snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
@@ -1281,12 +1309,15 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	/* Line2 Line Out default volume, disconnect from Output Mixer */
 	snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
 	snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
-	/* Line2 to Mono Out default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
-	if (aic3x->model == AIC3X_MODEL_3007) {
+	switch (aic3x->model) {
+	case AIC3X_MODEL_3X:
+	case AIC3X_MODEL_33:
+		aic3x_mono_init(codec);
+		break;
+	case AIC3X_MODEL_3007:
 		snd_soc_write(codec, CLASSD_CTRL, 0);
+		break;
 	}
 
 	return 0;
@@ -1343,8 +1374,17 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 			      (aic3x->setup->gpio_func[1] & 0xf) << 4);
 	}
 
-	if (aic3x->model == AIC3X_MODEL_3007)
-		snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
+	switch (aic3x->model) {
+	case AIC3X_MODEL_3X:
+	case AIC3X_MODEL_33:
+		snd_soc_add_codec_controls(codec, aic3x_mono_controls,
+				ARRAY_SIZE(aic3x_mono_controls));
+		break;
+	case AIC3X_MODEL_3007:
+		snd_soc_add_codec_controls(codec,
+				&aic3x_classd_amp_gain_ctrl, 1);
+		break;
+	}
 
 	/* set mic bias voltage */
 	switch (aic3x->micbias_vg) {
-- 
cgit v1.2.2


From 11b3a7add2841aa698aa0a32396b6db413c22eda Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 3 Dec 2013 14:26:32 -0700
Subject: ASoC: restructure dmaengine_pcm_request_chan_of()

Restructure the internals of dmaengine_pcm_request_chan_of() as a loop
over all channels to be allocated. This makes it easier to add logic
that applies to all allocated channels, without having to duplicate that
logic in each of the half-duplex/full-duplex paths.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 21 +++++++++++++--------
 1 file changed, 13 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 87e86357124e..fbc28a799e06 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -291,21 +291,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	struct device *dev)
 {
 	unsigned int i;
+	const char *name;
 
 	if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT |
 			   SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) ||
 	    !dev->of_node)
 		return;
 
-	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) {
-		pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx");
-		pcm->chan[1] = pcm->chan[0];
-	} else {
-		for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
-			pcm->chan[i] = dma_request_slave_channel(dev,
-					dmaengine_pcm_dma_channel_names[i]);
-		}
+	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+	     i++) {
+		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+			name = "rx-tx";
+		else
+			name = dmaengine_pcm_dma_channel_names[i];
+		pcm->chan[i] = dma_request_slave_channel(dev, name);
+		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+			break;
 	}
+
+	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+		pcm->chan[1] = pcm->chan[0];
 }
 
 /**
-- 
cgit v1.2.2


From 194c7dea00c68c1b1f8ff26304fa937a006f66dd Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 3 Dec 2013 14:26:34 -0700
Subject: ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config

Add fields to struct snd_dmaengine_pcm_config to allow custom:

- DMA channel names.

  This is useful when the default "tx" and "rx" channel names don't
  apply, for example if a HW module supports multiple channels, each
  having different DMA channel names. This is the case with the FIFOs
  in Tegra's AHUB. This new facility can replace
  SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.

- DMA device

  This allows requesting DMA channels for a device other than the device
  which is registering the "PCM" driver. This is quite unusual, but is
  currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
  module contains its own FIFOs which DMA writes to. However, in Tegra30,
  the DMA FIFOs were split out AHUB HW module, which then routes the data
  through a cross-bar, and into the DAI HW modules. However, the current
  ASoC driver structure does not expose this detail, and acts as if the
  FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
  driver is registered with the DAI HW module, yet the DMA channels must
  be looked up in the AHUB HW module's device tree node. This new config
  field allows that to happen. Eventually, the Tegra drivers will be
  reworked to fully expose the AHUB, and this config field can be
  removed.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 18 ++++++++++++++++--
 1 file changed, 16 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 1cb3494cf278..5b70c556fba3 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -288,7 +288,7 @@ static const char * const dmaengine_pcm_dma_channel_names[] = {
 };
 
 static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
-	struct device *dev)
+	struct device *dev, const struct snd_dmaengine_pcm_config *config)
 {
 	unsigned int i;
 	const char *name;
@@ -298,12 +298,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	    !dev->of_node)
 		return;
 
+	if (config->dma_dev) {
+		/*
+		 * If this warning is seen, it probably means that your Linux
+		 * device structure does not match your HW device structure.
+		 * It would be best to refactor the Linux device structure to
+		 * correctly match the HW structure.
+		 */
+		dev_warn(dev, "DMA channels sourced from device %s",
+			 dev_name(config->dma_dev));
+		dev = config->dma_dev;
+	}
+
 	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
 	     i++) {
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			name = "rx-tx";
 		else
 			name = dmaengine_pcm_dma_channel_names[i];
+		if (config->chan_names[i])
+			name = config->chan_names[i];
 		pcm->chan[i] = dma_request_slave_channel(dev, name);
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			break;
@@ -346,7 +360,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	pcm->config = config;
 	pcm->flags = flags;
 
-	dmaengine_pcm_request_chan_of(pcm, dev);
+	dmaengine_pcm_request_chan_of(pcm, dev, config);
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
 		ret = snd_soc_add_platform(dev, &pcm->platform,
-- 
cgit v1.2.2


From 1a1c75a7982ca1181bb84bb9e83f9f7c752fb104 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 9 Dec 2013 17:26:01 -0800
Subject: ASoC: rsnd: gen: fixup Gen2 channel size

Gen2 has 0 - 9, total 10 channels, not 9 channels.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index bf066f73ef05..d0ab203836de 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -217,11 +217,11 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv,
 
 /* single address mapping */
 #define RSND_GEN2_S_REG(gen, reg, id, offset)				\
-	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 9)
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, 0, 10)
 
 /* multi address mapping */
 #define RSND_GEN2_M_REG(gen, reg, id, offset, _id_offset)		\
-	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 9)
+	RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN2_##reg, offset, _id_offset, 10)
 
 static int rsnd_gen2_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 {
-- 
cgit v1.2.2


From 860d0c0dd2e7dbf98b47f38d80793137bc6c6ebc Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:17 +0200
Subject: ASoC: davinci: Kconfig: Remove help section for SND_DAVINCI_SOC

The help text is misleading and the prompt itself explains the purpose of
this config section.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/Kconfig | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 95970f5db3ec..fb9182659de1 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,11 +1,6 @@
 config SND_DAVINCI_SOC
 	tristate "SoC Audio for the TI DAVINCI or AM33XX chip"
 	depends on ARCH_DAVINCI || SOC_AM33XX
-	help
-	  Platform driver for daVinci or AM33xx
-	  Say Y or M if you want to add support for codecs attached to
-	  the DAVINCI AC97, I2S, or McASP interface. You will also need
-	  to select the audio interfaces to support below.
 
 config SND_DAVINCI_SOC_I2S
 	tristate
-- 
cgit v1.2.2


From c3238a4c058edd1528f0bec9a37fe79479e9e1a8 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:18 +0200
Subject: ASoC: davinci: Kconfig/Makefile: Generic EVM machine driver related
 cleanup

We have several boards using the same machine driver for audio support.
All of these machines can select a generic machine driver config option to
build the needed driver while keeping the config options used within the
driver for compile time code path selection.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/Kconfig  | 20 ++++++++++----------
 sound/soc/davinci/Makefile |  8 ++------
 2 files changed, 12 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index fb9182659de1..be667719d44e 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -11,11 +11,15 @@ config SND_DAVINCI_SOC_MCASP
 config SND_DAVINCI_SOC_VCIF
 	tristate
 
+config SND_DAVINCI_SOC_GENERIC_EVM
+	tristate
+	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_MCASP
+
 config SND_AM33XX_SOC_EVM
 	tristate "SoC Audio for the AM33XX chip based boards"
 	depends on SND_DAVINCI_SOC && SOC_AM33XX
-	select SND_SOC_TLV320AIC3X
-	select SND_DAVINCI_SOC_MCASP
+	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
 	  Say Y or M if you want to add support for SoC audio on AM33XX
 	  boards using McASP and TLV320AIC3X codec. For example AM335X-EVM,
@@ -26,8 +30,7 @@ config SND_DAVINCI_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
 	depends on SND_DAVINCI_SOC
 	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM
-	select SND_DAVINCI_SOC_I2S
-	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
 	  DaVinci DM6446, DM355 or DM365 EVM platforms.
@@ -54,8 +57,7 @@ endchoice
 config  SND_DM6467_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6467 EVM"
 	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM
-	select SND_DAVINCI_SOC_MCASP
-	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_GENERIC_EVM
 	select SND_SOC_SPDIF
 
 	help
@@ -64,8 +66,7 @@ config  SND_DM6467_SOC_EVM
 config  SND_DA830_SOC_EVM
 	tristate "SoC Audio support for DA830/OMAP-L137 EVM"
 	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM
-	select SND_DAVINCI_SOC_MCASP
-	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_GENERIC_EVM
 
 	help
 	  Say Y if you want to add support for SoC audio on TI
@@ -74,8 +75,7 @@ config  SND_DA830_SOC_EVM
 config  SND_DA850_SOC_EVM
 	tristate "SoC Audio support for DA850/OMAP-L138 EVM"
 	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM
-	select SND_DAVINCI_SOC_MCASP
-	select SND_SOC_TLV320AIC3X
+	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
 	  DA850/OMAP-L138 EVM
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index bc81e79fc301..744d4d9a0184 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -9,11 +9,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
 obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o
 
-# DAVINCI Machine Support
+# Generic DAVINCI/AM33xx Machine Support
 snd-soc-evm-objs := davinci-evm.o
 
-obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
-obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o
-obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o
-obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o
-obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DAVINCI_SOC_GENERIC_EVM) += snd-soc-evm.o
-- 
cgit v1.2.2


From a42efd97f7471c78617c6329ed39919e2f31a7cc Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:19 +0200
Subject: ASoC: davinci: kconfig: Prepare for AM43xx support

AM43xx have the same McASP IP as AM33xx and both platform uses eDMA. Modify
the Kconfig so it will be possible to add audio support for AM43xx based
boards later.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/Kconfig | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index be667719d44e..a8ec1fc3e4d0 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,6 +1,6 @@
 config SND_DAVINCI_SOC
-	tristate "SoC Audio for the TI DAVINCI or AM33XX chip"
-	depends on ARCH_DAVINCI || SOC_AM33XX
+	tristate "SoC Audio for TI DAVINCI or AM33XX/AM43XX chips"
+	depends on ARCH_DAVINCI || SOC_AM33XX || SOC_AM43XX
 
 config SND_DAVINCI_SOC_I2S
 	tristate
-- 
cgit v1.2.2


From 7b6a772932eca1ed8d22414a1ba985c5232369d0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:20 +0200
Subject: ASoC: davinci-evm: Do not include davinci-mcasp.h

There's no need to include this header file here.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-evm.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 623eb5e7c089..2a00e2dbc9ec 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -28,7 +28,6 @@
 
 #include "davinci-pcm.h"
 #include "davinci-i2s.h"
-#include "davinci-mcasp.h"
 
 struct snd_soc_card_drvdata_davinci {
 	unsigned sysclk;
-- 
cgit v1.2.2


From d38970e1363ddf63fa2f681f0c9b47fc91ca3961 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:21 +0200
Subject: ASoC: davinci-evm: Switch to use .dai_fmt of snd_soc_dai_link(s)

Specify the dai formats to use within the snd_soc_dai_link structures. In
this way we can remove the code dealing with the dai format configuration
from the machin driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-evm.c | 47 ++++++++++++++++-------------------------
 1 file changed, 18 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2a00e2dbc9ec..70ff3772079f 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -33,8 +33,6 @@ struct snd_soc_card_drvdata_davinci {
 	unsigned sysclk;
 };
 
-#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
-		SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
 static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
@@ -47,16 +45,6 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
 			   snd_soc_card_get_drvdata(soc_card))->sysclk;
 
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
-	if (ret < 0)
-		return ret;
-
 	/* set the codec system clock */
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
 	if (ret < 0)
@@ -70,24 +58,10 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int evm_spdif_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-
-	/* set cpu DAI configuration */
-	return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
-}
-
 static struct snd_soc_ops evm_ops = {
 	.hw_params = evm_hw_params,
 };
 
-static struct snd_soc_ops evm_spdif_ops = {
-	.hw_params = evm_spdif_hw_params,
-};
-
 /* davinci-evm machine dapm widgets */
 static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
@@ -164,6 +138,8 @@ static struct snd_soc_dai_link dm6446_evm_dai = {
 	.platform_name = "davinci-mcbsp",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 };
 
 static struct snd_soc_dai_link dm355_evm_dai = {
@@ -175,6 +151,8 @@ static struct snd_soc_dai_link dm355_evm_dai = {
 	.platform_name = "davinci-mcbsp.1",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 };
 
 static struct snd_soc_dai_link dm365_evm_dai = {
@@ -183,10 +161,12 @@ static struct snd_soc_dai_link dm365_evm_dai = {
 	.stream_name = "AIC3X",
 	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.init = evm_aic3x_init,
 	.codec_name = "tlv320aic3x-codec.1-0018",
-	.ops = &evm_ops,
 	.platform_name = "davinci-mcbsp",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 #elif defined(CONFIG_SND_DM365_VOICE_CODEC)
 	.name = "Voice Codec - CQ93VC",
 	.stream_name = "CQ93",
@@ -207,6 +187,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.codec_name = "tlv320aic3x-codec.0-001a",
 		.init = evm_aic3x_init,
 		.ops = &evm_ops,
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+			   SND_SOC_DAIFMT_IB_NF,
 	},
 	{
 		.name = "McASP",
@@ -215,7 +197,8 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.codec_dai_name = "dit-hifi",
 		.codec_name = "spdif_dit",
 		.platform_name = "davinci-mcasp.1",
-		.ops = &evm_spdif_ops,
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+			   SND_SOC_DAIFMT_IB_NF,
 	},
 };
 
@@ -228,6 +211,8 @@ static struct snd_soc_dai_link da830_evm_dai = {
 	.platform_name = "davinci-mcasp.1",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 };
 
 static struct snd_soc_dai_link da850_evm_dai = {
@@ -239,6 +224,8 @@ static struct snd_soc_dai_link da850_evm_dai = {
 	.platform_name = "davinci-mcasp.0",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 };
 
 /* davinci dm6446 evm audio machine driver */
@@ -335,6 +322,8 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = {
 	.codec_dai_name	= "tlv320aic3x-hifi",
 	.ops            = &evm_ops,
 	.init           = evm_aic3x_init,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
 };
 
 static const struct of_device_id davinci_evm_dt_ids[] = {
-- 
cgit v1.2.2


From ed29cd5e8d9941353f82784a2478ba6babc828da Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:22 +0200
Subject: ASoC: davinci-mcasp: Move DAVINCI_MCASP_RATE from header to source
 file

It is not used outside of the .c file.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 2 ++
 sound/soc/davinci/davinci-mcasp.h | 1 -
 2 files changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 71e14bb3a8cd..9763a5d75558 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -955,6 +955,8 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 	.set_sysclk	= davinci_mcasp_set_sysclk,
 };
 
+#define DAVINCI_MCASP_RATES	SNDRV_PCM_RATE_8000_192000
+
 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
 				SNDRV_PCM_FMTBIT_U8 | \
 				SNDRV_PCM_FMTBIT_S16_LE | \
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a2e27e1c32f3..a84e79604dfa 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -23,7 +23,6 @@
 
 #include "davinci-pcm.h"
 
-#define DAVINCI_MCASP_RATES	SNDRV_PCM_RATE_8000_192000
 #define DAVINCI_MCASP_I2S_DAI	0
 #define DAVINCI_MCASP_DIT_DAI	1
 
-- 
cgit v1.2.2


From 18f93506623aacbb269f47cbda9fe90ffc5acda6 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:23 +0200
Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_I2S/DIT_DAI defines

These are not used, probably leftovers from the past.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.h | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index a84e79604dfa..70b089b4e0f5 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -23,9 +23,6 @@
 
 #include "davinci-pcm.h"
 
-#define DAVINCI_MCASP_I2S_DAI	0
-#define DAVINCI_MCASP_DIT_DAI	1
-
 struct davinci_audio_dev {
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
-- 
cgit v1.2.2


From 58e48d9774d4d8fc5e0785dbd2ccf075b248ad96 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:24 +0200
Subject: ASoC: davinci-mcasp: Correct dai driver struct initialization for 2nd
 dai

Add .name when assigning the dai name.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9763a5d75558..6cde1babc119 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -987,7 +987,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 
 	},
 	{
-		"davinci-mcasp.1",
+		.name		= "davinci-mcasp.1",
 		.playback 	= {
 			.channels_min	= 1,
 			.channels_max	= 384,
-- 
cgit v1.2.2


From 02e08d9b6bd67784d4c58e659c21674b31972c34 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:25 +0200
Subject: ASoC: davinci-mcasp: Move register definitions to header file

It is better for readability to have the register definitions out from the
source file.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 267 --------------------------------------
 sound/soc/davinci/davinci-mcasp.h | 267 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 267 insertions(+), 267 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 6cde1babc119..1c1585e18eae 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -35,273 +35,6 @@
 #include "davinci-pcm.h"
 #include "davinci-mcasp.h"
 
-/*
- * McASP register definitions
- */
-#define DAVINCI_MCASP_PID_REG		0x00
-#define DAVINCI_MCASP_PWREMUMGT_REG	0x04
-
-#define DAVINCI_MCASP_PFUNC_REG		0x10
-#define DAVINCI_MCASP_PDIR_REG		0x14
-#define DAVINCI_MCASP_PDOUT_REG		0x18
-#define DAVINCI_MCASP_PDSET_REG		0x1c
-
-#define DAVINCI_MCASP_PDCLR_REG		0x20
-
-#define DAVINCI_MCASP_TLGC_REG		0x30
-#define DAVINCI_MCASP_TLMR_REG		0x34
-
-#define DAVINCI_MCASP_GBLCTL_REG	0x44
-#define DAVINCI_MCASP_AMUTE_REG		0x48
-#define DAVINCI_MCASP_LBCTL_REG		0x4c
-
-#define DAVINCI_MCASP_TXDITCTL_REG	0x50
-
-#define DAVINCI_MCASP_GBLCTLR_REG	0x60
-#define DAVINCI_MCASP_RXMASK_REG	0x64
-#define DAVINCI_MCASP_RXFMT_REG		0x68
-#define DAVINCI_MCASP_RXFMCTL_REG	0x6c
-
-#define DAVINCI_MCASP_ACLKRCTL_REG	0x70
-#define DAVINCI_MCASP_AHCLKRCTL_REG	0x74
-#define DAVINCI_MCASP_RXTDM_REG		0x78
-#define DAVINCI_MCASP_EVTCTLR_REG	0x7c
-
-#define DAVINCI_MCASP_RXSTAT_REG	0x80
-#define DAVINCI_MCASP_RXTDMSLOT_REG	0x84
-#define DAVINCI_MCASP_RXCLKCHK_REG	0x88
-#define DAVINCI_MCASP_REVTCTL_REG	0x8c
-
-#define DAVINCI_MCASP_GBLCTLX_REG	0xa0
-#define DAVINCI_MCASP_TXMASK_REG	0xa4
-#define DAVINCI_MCASP_TXFMT_REG		0xa8
-#define DAVINCI_MCASP_TXFMCTL_REG	0xac
-
-#define DAVINCI_MCASP_ACLKXCTL_REG	0xb0
-#define DAVINCI_MCASP_AHCLKXCTL_REG	0xb4
-#define DAVINCI_MCASP_TXTDM_REG		0xb8
-#define DAVINCI_MCASP_EVTCTLX_REG	0xbc
-
-#define DAVINCI_MCASP_TXSTAT_REG	0xc0
-#define DAVINCI_MCASP_TXTDMSLOT_REG	0xc4
-#define DAVINCI_MCASP_TXCLKCHK_REG	0xc8
-#define DAVINCI_MCASP_XEVTCTL_REG	0xcc
-
-/* Left(even TDM Slot) Channel Status Register File */
-#define DAVINCI_MCASP_DITCSRA_REG	0x100
-/* Right(odd TDM slot) Channel Status Register File */
-#define DAVINCI_MCASP_DITCSRB_REG	0x118
-/* Left(even TDM slot) User Data Register File */
-#define DAVINCI_MCASP_DITUDRA_REG	0x130
-/* Right(odd TDM Slot) User Data Register File */
-#define DAVINCI_MCASP_DITUDRB_REG	0x148
-
-/* Serializer n Control Register */
-#define DAVINCI_MCASP_XRSRCTL_BASE_REG	0x180
-#define DAVINCI_MCASP_XRSRCTL_REG(n)	(DAVINCI_MCASP_XRSRCTL_BASE_REG + \
-						(n << 2))
-
-/* Transmit Buffer for Serializer n */
-#define DAVINCI_MCASP_TXBUF_REG		0x200
-/* Receive Buffer for Serializer n */
-#define DAVINCI_MCASP_RXBUF_REG		0x280
-
-/* McASP FIFO Registers */
-#define DAVINCI_MCASP_WFIFOCTL		(0x1010)
-#define DAVINCI_MCASP_WFIFOSTS		(0x1014)
-#define DAVINCI_MCASP_RFIFOCTL		(0x1018)
-#define DAVINCI_MCASP_RFIFOSTS		(0x101C)
-#define MCASP_VER3_WFIFOCTL		(0x1000)
-#define MCASP_VER3_WFIFOSTS		(0x1004)
-#define MCASP_VER3_RFIFOCTL		(0x1008)
-#define MCASP_VER3_RFIFOSTS		(0x100C)
-
-/*
- * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
- *     Register Bits
- */
-#define MCASP_FREE	BIT(0)
-#define MCASP_SOFT	BIT(1)
-
-/*
- * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits
- */
-#define AXR(n)		(1<<n)
-#define PFUNC_AMUTE	BIT(25)
-#define ACLKX		BIT(26)
-#define AHCLKX		BIT(27)
-#define AFSX		BIT(28)
-#define ACLKR		BIT(29)
-#define AHCLKR		BIT(30)
-#define AFSR		BIT(31)
-
-/*
- * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits
- */
-#define AXR(n)		(1<<n)
-#define PDIR_AMUTE	BIT(25)
-#define ACLKX		BIT(26)
-#define AHCLKX		BIT(27)
-#define AFSX		BIT(28)
-#define ACLKR		BIT(29)
-#define AHCLKR		BIT(30)
-#define AFSR		BIT(31)
-
-/*
- * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits
- */
-#define DITEN	BIT(0)	/* Transmit DIT mode enable/disable */
-#define VA	BIT(2)
-#define VB	BIT(3)
-
-/*
- * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits
- */
-#define TXROT(val)	(val)
-#define TXSEL		BIT(3)
-#define TXSSZ(val)	(val<<4)
-#define TXPBIT(val)	(val<<8)
-#define TXPAD(val)	(val<<13)
-#define TXORD		BIT(15)
-#define FSXDLY(val)	(val<<16)
-
-/*
- * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits
- */
-#define RXROT(val)	(val)
-#define RXSEL		BIT(3)
-#define RXSSZ(val)	(val<<4)
-#define RXPBIT(val)	(val<<8)
-#define RXPAD(val)	(val<<13)
-#define RXORD		BIT(15)
-#define FSRDLY(val)	(val<<16)
-
-/*
- * DAVINCI_MCASP_TXFMCTL_REG -  Transmit Frame Control Register Bits
- */
-#define FSXPOL		BIT(0)
-#define AFSXE		BIT(1)
-#define FSXDUR		BIT(4)
-#define FSXMOD(val)	(val<<7)
-
-/*
- * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits
- */
-#define FSRPOL		BIT(0)
-#define AFSRE		BIT(1)
-#define FSRDUR		BIT(4)
-#define FSRMOD(val)	(val<<7)
-
-/*
- * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits
- */
-#define ACLKXDIV(val)	(val)
-#define ACLKXE		BIT(5)
-#define TX_ASYNC	BIT(6)
-#define ACLKXPOL	BIT(7)
-#define ACLKXDIV_MASK	0x1f
-
-/*
- * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
- */
-#define ACLKRDIV(val)	(val)
-#define ACLKRE		BIT(5)
-#define RX_ASYNC	BIT(6)
-#define ACLKRPOL	BIT(7)
-#define ACLKRDIV_MASK	0x1f
-
-/*
- * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
- *     Register Bits
- */
-#define AHCLKXDIV(val)	(val)
-#define AHCLKXPOL	BIT(14)
-#define AHCLKXE		BIT(15)
-#define AHCLKXDIV_MASK	0xfff
-
-/*
- * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
- *     Register Bits
- */
-#define AHCLKRDIV(val)	(val)
-#define AHCLKRPOL	BIT(14)
-#define AHCLKRE		BIT(15)
-#define AHCLKRDIV_MASK	0xfff
-
-/*
- * DAVINCI_MCASP_XRSRCTL_BASE_REG -  Serializer Control Register Bits
- */
-#define MODE(val)	(val)
-#define DISMOD		(val)(val<<2)
-#define TXSTATE		BIT(4)
-#define RXSTATE		BIT(5)
-#define SRMOD_MASK	3
-#define SRMOD_INACTIVE	0
-
-/*
- * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
- */
-#define LBEN		BIT(0)
-#define LBORD		BIT(1)
-#define LBGENMODE(val)	(val<<2)
-
-/*
- * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration
- */
-#define TXTDMS(n)	(1<<n)
-
-/*
- * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration
- */
-#define RXTDMS(n)	(1<<n)
-
-/*
- * DAVINCI_MCASP_GBLCTL_REG -  Global Control Register Bits
- */
-#define RXCLKRST	BIT(0)	/* Receiver Clock Divider Reset */
-#define RXHCLKRST	BIT(1)	/* Receiver High Frequency Clock Divider */
-#define RXSERCLR	BIT(2)	/* Receiver Serializer Clear */
-#define RXSMRST		BIT(3)	/* Receiver State Machine Reset */
-#define RXFSRST		BIT(4)	/* Frame Sync Generator Reset */
-#define TXCLKRST	BIT(8)	/* Transmitter Clock Divider Reset */
-#define TXHCLKRST	BIT(9)	/* Transmitter High Frequency Clock Divider*/
-#define TXSERCLR	BIT(10)	/* Transmit Serializer Clear */
-#define TXSMRST		BIT(11)	/* Transmitter State Machine Reset */
-#define TXFSRST		BIT(12)	/* Frame Sync Generator Reset */
-
-/*
- * DAVINCI_MCASP_AMUTE_REG -  Mute Control Register Bits
- */
-#define MUTENA(val)	(val)
-#define MUTEINPOL	BIT(2)
-#define MUTEINENA	BIT(3)
-#define MUTEIN		BIT(4)
-#define MUTER		BIT(5)
-#define MUTEX		BIT(6)
-#define MUTEFSR		BIT(7)
-#define MUTEFSX		BIT(8)
-#define MUTEBADCLKR	BIT(9)
-#define MUTEBADCLKX	BIT(10)
-#define MUTERXDMAERR	BIT(11)
-#define MUTETXDMAERR	BIT(12)
-
-/*
- * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits
- */
-#define RXDATADMADIS	BIT(0)
-
-/*
- * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits
- */
-#define TXDATADMADIS	BIT(0)
-
-/*
- * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
- */
-#define FIFO_ENABLE	BIT(16)
-#define NUMEVT_MASK	(0xFF << 8)
-#define NUMDMA_MASK	(0xFF)
-
 #define DAVINCI_MCASP_NUM_SERIALIZER	16
 
 static inline void mcasp_set_bits(void __iomem *reg, u32 val)
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 70b089b4e0f5..619b98befc84 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -23,6 +23,273 @@
 
 #include "davinci-pcm.h"
 
+/*
+ * McASP register definitions
+ */
+#define DAVINCI_MCASP_PID_REG		0x00
+#define DAVINCI_MCASP_PWREMUMGT_REG	0x04
+
+#define DAVINCI_MCASP_PFUNC_REG		0x10
+#define DAVINCI_MCASP_PDIR_REG		0x14
+#define DAVINCI_MCASP_PDOUT_REG		0x18
+#define DAVINCI_MCASP_PDSET_REG		0x1c
+
+#define DAVINCI_MCASP_PDCLR_REG		0x20
+
+#define DAVINCI_MCASP_TLGC_REG		0x30
+#define DAVINCI_MCASP_TLMR_REG		0x34
+
+#define DAVINCI_MCASP_GBLCTL_REG	0x44
+#define DAVINCI_MCASP_AMUTE_REG		0x48
+#define DAVINCI_MCASP_LBCTL_REG		0x4c
+
+#define DAVINCI_MCASP_TXDITCTL_REG	0x50
+
+#define DAVINCI_MCASP_GBLCTLR_REG	0x60
+#define DAVINCI_MCASP_RXMASK_REG	0x64
+#define DAVINCI_MCASP_RXFMT_REG		0x68
+#define DAVINCI_MCASP_RXFMCTL_REG	0x6c
+
+#define DAVINCI_MCASP_ACLKRCTL_REG	0x70
+#define DAVINCI_MCASP_AHCLKRCTL_REG	0x74
+#define DAVINCI_MCASP_RXTDM_REG		0x78
+#define DAVINCI_MCASP_EVTCTLR_REG	0x7c
+
+#define DAVINCI_MCASP_RXSTAT_REG	0x80
+#define DAVINCI_MCASP_RXTDMSLOT_REG	0x84
+#define DAVINCI_MCASP_RXCLKCHK_REG	0x88
+#define DAVINCI_MCASP_REVTCTL_REG	0x8c
+
+#define DAVINCI_MCASP_GBLCTLX_REG	0xa0
+#define DAVINCI_MCASP_TXMASK_REG	0xa4
+#define DAVINCI_MCASP_TXFMT_REG		0xa8
+#define DAVINCI_MCASP_TXFMCTL_REG	0xac
+
+#define DAVINCI_MCASP_ACLKXCTL_REG	0xb0
+#define DAVINCI_MCASP_AHCLKXCTL_REG	0xb4
+#define DAVINCI_MCASP_TXTDM_REG		0xb8
+#define DAVINCI_MCASP_EVTCTLX_REG	0xbc
+
+#define DAVINCI_MCASP_TXSTAT_REG	0xc0
+#define DAVINCI_MCASP_TXTDMSLOT_REG	0xc4
+#define DAVINCI_MCASP_TXCLKCHK_REG	0xc8
+#define DAVINCI_MCASP_XEVTCTL_REG	0xcc
+
+/* Left(even TDM Slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRA_REG	0x100
+/* Right(odd TDM slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRB_REG	0x118
+/* Left(even TDM slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRA_REG	0x130
+/* Right(odd TDM Slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRB_REG	0x148
+
+/* Serializer n Control Register */
+#define DAVINCI_MCASP_XRSRCTL_BASE_REG	0x180
+#define DAVINCI_MCASP_XRSRCTL_REG(n)	(DAVINCI_MCASP_XRSRCTL_BASE_REG + \
+						(n << 2))
+
+/* Transmit Buffer for Serializer n */
+#define DAVINCI_MCASP_TXBUF_REG		0x200
+/* Receive Buffer for Serializer n */
+#define DAVINCI_MCASP_RXBUF_REG		0x280
+
+/* McASP FIFO Registers */
+#define DAVINCI_MCASP_WFIFOCTL		(0x1010)
+#define DAVINCI_MCASP_WFIFOSTS		(0x1014)
+#define DAVINCI_MCASP_RFIFOCTL		(0x1018)
+#define DAVINCI_MCASP_RFIFOSTS		(0x101C)
+#define MCASP_VER3_WFIFOCTL		(0x1000)
+#define MCASP_VER3_WFIFOSTS		(0x1004)
+#define MCASP_VER3_RFIFOCTL		(0x1008)
+#define MCASP_VER3_RFIFOSTS		(0x100C)
+
+/*
+ * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
+ *     Register Bits
+ */
+#define MCASP_FREE	BIT(0)
+#define MCASP_SOFT	BIT(1)
+
+/*
+ * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits
+ */
+#define AXR(n)		(1<<n)
+#define PFUNC_AMUTE	BIT(25)
+#define ACLKX		BIT(26)
+#define AHCLKX		BIT(27)
+#define AFSX		BIT(28)
+#define ACLKR		BIT(29)
+#define AHCLKR		BIT(30)
+#define AFSR		BIT(31)
+
+/*
+ * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits
+ */
+#define AXR(n)		(1<<n)
+#define PDIR_AMUTE	BIT(25)
+#define ACLKX		BIT(26)
+#define AHCLKX		BIT(27)
+#define AFSX		BIT(28)
+#define ACLKR		BIT(29)
+#define AHCLKR		BIT(30)
+#define AFSR		BIT(31)
+
+/*
+ * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits
+ */
+#define DITEN	BIT(0)	/* Transmit DIT mode enable/disable */
+#define VA	BIT(2)
+#define VB	BIT(3)
+
+/*
+ * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits
+ */
+#define TXROT(val)	(val)
+#define TXSEL		BIT(3)
+#define TXSSZ(val)	(val<<4)
+#define TXPBIT(val)	(val<<8)
+#define TXPAD(val)	(val<<13)
+#define TXORD		BIT(15)
+#define FSXDLY(val)	(val<<16)
+
+/*
+ * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits
+ */
+#define RXROT(val)	(val)
+#define RXSEL		BIT(3)
+#define RXSSZ(val)	(val<<4)
+#define RXPBIT(val)	(val<<8)
+#define RXPAD(val)	(val<<13)
+#define RXORD		BIT(15)
+#define FSRDLY(val)	(val<<16)
+
+/*
+ * DAVINCI_MCASP_TXFMCTL_REG -  Transmit Frame Control Register Bits
+ */
+#define FSXPOL		BIT(0)
+#define AFSXE		BIT(1)
+#define FSXDUR		BIT(4)
+#define FSXMOD(val)	(val<<7)
+
+/*
+ * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits
+ */
+#define FSRPOL		BIT(0)
+#define AFSRE		BIT(1)
+#define FSRDUR		BIT(4)
+#define FSRMOD(val)	(val<<7)
+
+/*
+ * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits
+ */
+#define ACLKXDIV(val)	(val)
+#define ACLKXE		BIT(5)
+#define TX_ASYNC	BIT(6)
+#define ACLKXPOL	BIT(7)
+#define ACLKXDIV_MASK	0x1f
+
+/*
+ * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
+ */
+#define ACLKRDIV(val)	(val)
+#define ACLKRE		BIT(5)
+#define RX_ASYNC	BIT(6)
+#define ACLKRPOL	BIT(7)
+#define ACLKRDIV_MASK	0x1f
+
+/*
+ * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
+ *     Register Bits
+ */
+#define AHCLKXDIV(val)	(val)
+#define AHCLKXPOL	BIT(14)
+#define AHCLKXE		BIT(15)
+#define AHCLKXDIV_MASK	0xfff
+
+/*
+ * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
+ *     Register Bits
+ */
+#define AHCLKRDIV(val)	(val)
+#define AHCLKRPOL	BIT(14)
+#define AHCLKRE		BIT(15)
+#define AHCLKRDIV_MASK	0xfff
+
+/*
+ * DAVINCI_MCASP_XRSRCTL_BASE_REG -  Serializer Control Register Bits
+ */
+#define MODE(val)	(val)
+#define DISMOD		(val)(val<<2)
+#define TXSTATE		BIT(4)
+#define RXSTATE		BIT(5)
+#define SRMOD_MASK	3
+#define SRMOD_INACTIVE	0
+
+/*
+ * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
+ */
+#define LBEN		BIT(0)
+#define LBORD		BIT(1)
+#define LBGENMODE(val)	(val<<2)
+
+/*
+ * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration
+ */
+#define TXTDMS(n)	(1<<n)
+
+/*
+ * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration
+ */
+#define RXTDMS(n)	(1<<n)
+
+/*
+ * DAVINCI_MCASP_GBLCTL_REG -  Global Control Register Bits
+ */
+#define RXCLKRST	BIT(0)	/* Receiver Clock Divider Reset */
+#define RXHCLKRST	BIT(1)	/* Receiver High Frequency Clock Divider */
+#define RXSERCLR	BIT(2)	/* Receiver Serializer Clear */
+#define RXSMRST		BIT(3)	/* Receiver State Machine Reset */
+#define RXFSRST		BIT(4)	/* Frame Sync Generator Reset */
+#define TXCLKRST	BIT(8)	/* Transmitter Clock Divider Reset */
+#define TXHCLKRST	BIT(9)	/* Transmitter High Frequency Clock Divider*/
+#define TXSERCLR	BIT(10)	/* Transmit Serializer Clear */
+#define TXSMRST		BIT(11)	/* Transmitter State Machine Reset */
+#define TXFSRST		BIT(12)	/* Frame Sync Generator Reset */
+
+/*
+ * DAVINCI_MCASP_AMUTE_REG -  Mute Control Register Bits
+ */
+#define MUTENA(val)	(val)
+#define MUTEINPOL	BIT(2)
+#define MUTEINENA	BIT(3)
+#define MUTEIN		BIT(4)
+#define MUTER		BIT(5)
+#define MUTEX		BIT(6)
+#define MUTEFSR		BIT(7)
+#define MUTEFSX		BIT(8)
+#define MUTEBADCLKR	BIT(9)
+#define MUTEBADCLKX	BIT(10)
+#define MUTERXDMAERR	BIT(11)
+#define MUTETXDMAERR	BIT(12)
+
+/*
+ * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits
+ */
+#define RXDATADMADIS	BIT(0)
+
+/*
+ * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits
+ */
+#define TXDATADMADIS	BIT(0)
+
+/*
+ * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
+ */
+#define FIFO_ENABLE	BIT(16)
+#define NUMEVT_MASK	(0xFF << 8)
+#define NUMDMA_MASK	(0xFF)
+
 struct davinci_audio_dev {
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
-- 
cgit v1.2.2


From 21400a72d6bba3d233161ad131e36b6b7f78381e Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:26 +0200
Subject: ASoC: davinci-mcasp: Move private struct definition to source file

Since it is a private struct strictly used by the davinci-mcasp driver it
can be moved from header file to the source file.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++
 sound/soc/davinci/davinci-mcasp.h | 35 -----------------------------------
 2 files changed, 30 insertions(+), 35 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 1c1585e18eae..70107956dd0c 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -37,6 +37,36 @@
 
 #define DAVINCI_MCASP_NUM_SERIALIZER	16
 
+struct davinci_audio_dev {
+	struct davinci_pcm_dma_params dma_params[2];
+	void __iomem *base;
+	struct device *dev;
+
+	/* McASP specific data */
+	int	tdm_slots;
+	u8	op_mode;
+	u8	num_serializer;
+	u8	*serial_dir;
+	u8	version;
+	u16	bclk_lrclk_ratio;
+
+	/* McASP FIFO related */
+	u8	txnumevt;
+	u8	rxnumevt;
+
+#ifdef CONFIG_PM_SLEEP
+	struct {
+		u32	txfmtctl;
+		u32	rxfmtctl;
+		u32	txfmt;
+		u32	rxfmt;
+		u32	aclkxctl;
+		u32	aclkrctl;
+		u32	pdir;
+	} context;
+#endif
+};
+
 static inline void mcasp_set_bits(void __iomem *reg, u32 val)
 {
 	__raw_writel(__raw_readl(reg) | val, reg);
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 619b98befc84..80e5a1846687 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -18,11 +18,6 @@
 #ifndef DAVINCI_MCASP_H
 #define DAVINCI_MCASP_H
 
-#include <linux/io.h>
-#include <linux/platform_data/davinci_asp.h>
-
-#include "davinci-pcm.h"
-
 /*
  * McASP register definitions
  */
@@ -290,34 +285,4 @@
 #define NUMEVT_MASK	(0xFF << 8)
 #define NUMDMA_MASK	(0xFF)
 
-struct davinci_audio_dev {
-	struct davinci_pcm_dma_params dma_params[2];
-	void __iomem *base;
-	struct device *dev;
-
-	/* McASP specific data */
-	int	tdm_slots;
-	u8	op_mode;
-	u8	num_serializer;
-	u8	*serial_dir;
-	u8	version;
-	u16	bclk_lrclk_ratio;
-
-	/* McASP FIFO related */
-	u8	txnumevt;
-	u8	rxnumevt;
-
-#ifdef CONFIG_PM_SLEEP
-	struct {
-		u32	txfmtctl;
-		u32	rxfmtctl;
-		u32	txfmt;
-		u32	rxfmt;
-		u32	aclkxctl;
-		u32	aclkrctl;
-		u32	pdir;
-	} context;
-#endif
-};
-
 #endif	/* DAVINCI_MCASP_H */
-- 
cgit v1.2.2


From 57f439b8676b2dad63a148de4e61d80b4e196c2a Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:27 +0200
Subject: ASoC: davinci-mcasp: Remove unused DAVINCI_MCASP_NUM_SERIALIZER
 define

It is not used in the code.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 70107956dd0c..e4c0fb4b9784 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -35,8 +35,6 @@
 #include "davinci-pcm.h"
 #include "davinci-mcasp.h"
 
-#define DAVINCI_MCASP_NUM_SERIALIZER	16
-
 struct davinci_audio_dev {
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
-- 
cgit v1.2.2


From eba0ecf067913d60768bb3d11f861b949f072a93 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:28 +0200
Subject: ASoC: davinci-mcasp: Do not inline the mcasp_set_ctl_reg function

It brings no benefit to inline this function due to it's size and function.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index e4c0fb4b9784..ce1607b3c6df 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -90,7 +90,7 @@ static inline u32 mcasp_get_reg(void __iomem *reg)
 	return (unsigned int)__raw_readl(reg);
 }
 
-static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
+static void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
 {
 	int i = 0;
 
-- 
cgit v1.2.2


From 70091a3e6aa2e7a05eaefcaec1a43c27a5023eb7 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:29 +0200
Subject: ASoC: davinci-mcasp: Rename private struct and it's users (dev ->
 mcasp)

Rename the private struct from davinci_audio_dev to davinci_mcasp.
Change the local use of the pointer to this struct from *dev to *mcasp.

The aim is to have better readable code for the first look since having
dev->xxxx in the code when using the local private struct is a bit
surprising.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 418 +++++++++++++++++++-------------------
 1 file changed, 209 insertions(+), 209 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index ce1607b3c6df..bd85c98bf5a5 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -35,7 +35,7 @@
 #include "davinci-pcm.h"
 #include "davinci-mcasp.h"
 
-struct davinci_audio_dev {
+struct davinci_mcasp {
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
 	struct device *dev;
@@ -107,36 +107,36 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
 		printk(KERN_ERR "GBLCTL write error\n");
 }
 
-static void mcasp_start_rx(struct davinci_audio_dev *dev)
+static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 {
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
 }
 
-static void mcasp_start_tx(struct davinci_audio_dev *dev)
+static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 {
 	u8 offset = 0, i;
 	u32 cnt;
 
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
-	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
-	for (i = 0; i < dev->num_serializer; i++) {
-		if (dev->serial_dir[i] == TX_MODE) {
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
+	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	for (i = 0; i < mcasp->num_serializer; i++) {
+		if (mcasp->serial_dir[i] == TX_MODE) {
 			offset = i;
 			break;
 		}
@@ -144,116 +144,116 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
 
 	/* wait for TX ready */
 	cnt = 0;
-	while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) &
+	while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) &
 		 TXSTATE) && (cnt < 100000))
 		cnt++;
 
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
 }
 
-static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
+static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 {
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (dev->txnumevt) {	/* enable FIFO */
-			switch (dev->version) {
+		if (mcasp->txnumevt) {	/* enable FIFO */
+			switch (mcasp->version) {
 			case MCASP_VERSION_3:
-				mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL,
-								FIFO_ENABLE);
-				mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL,
-								FIFO_ENABLE);
+				mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
+					       FIFO_ENABLE);
+				mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
+					       FIFO_ENABLE);
 				break;
 			default:
-				mcasp_clr_bits(dev->base +
+				mcasp_clr_bits(mcasp->base +
 					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
-				mcasp_set_bits(dev->base +
+				mcasp_set_bits(mcasp->base +
 					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
 			}
 		}
-		mcasp_start_tx(dev);
+		mcasp_start_tx(mcasp);
 	} else {
-		if (dev->rxnumevt) {	/* enable FIFO */
-			switch (dev->version) {
+		if (mcasp->rxnumevt) {	/* enable FIFO */
+			switch (mcasp->version) {
 			case MCASP_VERSION_3:
-				mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL,
-								FIFO_ENABLE);
-				mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL,
-								FIFO_ENABLE);
+				mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
+					       FIFO_ENABLE);
+				mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
+					       FIFO_ENABLE);
 				break;
 			default:
-				mcasp_clr_bits(dev->base +
+				mcasp_clr_bits(mcasp->base +
 					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
-				mcasp_set_bits(dev->base +
+				mcasp_set_bits(mcasp->base +
 					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
 			}
 		}
-		mcasp_start_rx(dev);
+		mcasp_start_rx(mcasp);
 	}
 }
 
-static void mcasp_stop_rx(struct davinci_audio_dev *dev)
+static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
 {
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
 }
 
-static void mcasp_stop_tx(struct davinci_audio_dev *dev)
+static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
 {
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
 }
 
-static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
+static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
 {
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (dev->txnumevt) {	/* disable FIFO */
-			switch (dev->version) {
+		if (mcasp->txnumevt) {	/* disable FIFO */
+			switch (mcasp->version) {
 			case MCASP_VERSION_3:
-				mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL,
-								FIFO_ENABLE);
+				mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
+					       FIFO_ENABLE);
 				break;
 			default:
-				mcasp_clr_bits(dev->base +
+				mcasp_clr_bits(mcasp->base +
 					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
 			}
 		}
-		mcasp_stop_tx(dev);
+		mcasp_stop_tx(mcasp);
 	} else {
-		if (dev->rxnumevt) {	/* disable FIFO */
-			switch (dev->version) {
+		if (mcasp->rxnumevt) {	/* disable FIFO */
+			switch (mcasp->version) {
 			case MCASP_VERSION_3:
-				mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL,
-								FIFO_ENABLE);
+				mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
+					       FIFO_ENABLE);
 			break;
 
 			default:
-				mcasp_clr_bits(dev->base +
+				mcasp_clr_bits(mcasp->base +
 					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
 			}
 		}
-		mcasp_stop_rx(dev);
+		mcasp_stop_rx(mcasp);
 	}
 }
 
 static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 					 unsigned int fmt)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
-	void __iomem *base = dev->base;
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	void __iomem *base = mcasp->base;
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_AC97:
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 		break;
 	default:
 		/* configure a full-word SYNC pulse (LRCLK) */
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 
 		/* make 1st data bit occur one ACLK cycle after the frame sync */
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
 		break;
 	}
 
@@ -342,25 +342,25 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
 static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
 	switch (div_id) {
 	case 0:		/* MCLK divider */
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG,
 			       AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG,
 			       AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
 		break;
 
 	case 1:		/* BCLK divider */
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG,
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG,
 			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
 		break;
 
 	case 2:		/* BCLK/LRCLK ratio */
-		dev->bclk_lrclk_ratio = div;
+		mcasp->bclk_lrclk_ratio = div;
 		break;
 
 	default:
@@ -373,22 +373,22 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 				    unsigned int freq, int dir)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
 	if (dir == SND_SOC_CLOCK_OUT) {
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
 	} else {
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
 	}
 
 	return 0;
 }
 
-static int davinci_config_channel_size(struct davinci_audio_dev *dev,
+static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 				       int word_length)
 {
 	u32 fmt;
@@ -405,70 +405,70 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 	 * both left and right channels), so it has to be divided by number of
 	 * tdm-slots (for I2S - divided by 2).
 	 */
-	if (dev->bclk_lrclk_ratio)
-		word_length = dev->bclk_lrclk_ratio / dev->tdm_slots;
+	if (mcasp->bclk_lrclk_ratio)
+		word_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots;
 
 	/* mapping of the XSSZ bit-field as described in the datasheet */
 	fmt = (word_length >> 1) - 1;
 
-	if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) {
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
+	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
 				RXSSZ(fmt), RXSSZ(0x0F));
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
 				TXSSZ(fmt), TXSSZ(0x0F));
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
 				TXROT(tx_rotate), TXROT(7));
-		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
+		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
 				RXROT(rx_rotate), RXROT(7));
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG,
 				mask);
 	}
 
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask);
 
 	return 0;
 }
 
-static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream,
+static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 				    int channels)
 {
 	int i;
 	u8 tx_ser = 0;
 	u8 rx_ser = 0;
 	u8 ser;
-	u8 slots = dev->tdm_slots;
+	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
 	/* Default configuration */
-	mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
+	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
 
 	/* All PINS as McASP */
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG,
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG,
 				TXDATADMADIS);
 	} else {
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG,
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG,
 				RXDATADMADIS);
 	}
 
-	for (i = 0; i < dev->num_serializer; i++) {
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
-					dev->serial_dir[i]);
-		if (dev->serial_dir[i] == TX_MODE &&
+	for (i = 0; i < mcasp->num_serializer; i++) {
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i),
+					mcasp->serial_dir[i]);
+		if (mcasp->serial_dir[i] == TX_MODE &&
 					tx_ser < max_active_serializers) {
-			mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
+			mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG,
 					AXR(i));
 			tx_ser++;
-		} else if (dev->serial_dir[i] == RX_MODE &&
+		} else if (mcasp->serial_dir[i] == RX_MODE &&
 					rx_ser < max_active_serializers) {
-			mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
+			mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG,
 					AXR(i));
 			rx_ser++;
 		} else {
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i),
 					SRMOD_INACTIVE, SRMOD_MASK);
 		}
 	}
@@ -479,127 +479,127 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream,
 		ser = rx_ser;
 
 	if (ser < max_active_serializers) {
-		dev_warn(dev->dev, "stream has more channels (%d) than are "
+		dev_warn(mcasp->dev, "stream has more channels (%d) than are "
 			"enabled in mcasp (%d)\n", channels, ser * slots);
 		return -EINVAL;
 	}
 
-	if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (dev->txnumevt * tx_ser > 64)
-			dev->txnumevt = 1;
+	if (mcasp->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (mcasp->txnumevt * tx_ser > 64)
+			mcasp->txnumevt = 1;
 
-		switch (dev->version) {
+		switch (mcasp->version) {
 		case MCASP_VERSION_3:
-			mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser,
+			mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser,
 								NUMDMA_MASK);
-			mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL,
-				((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
+			mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
+				((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
 			break;
 		default:
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL,
 							tx_ser,	NUMDMA_MASK);
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
-				((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL,
+				((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
 		}
 	}
 
-	if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
-		if (dev->rxnumevt * rx_ser > 64)
-			dev->rxnumevt = 1;
-		switch (dev->version) {
+	if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (mcasp->rxnumevt * rx_ser > 64)
+			mcasp->rxnumevt = 1;
+		switch (mcasp->version) {
 		case MCASP_VERSION_3:
-			mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser,
+			mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser,
 								NUMDMA_MASK);
-			mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL,
-				((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
+			mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
+				((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
 			break;
 		default:
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL,
 							rx_ser,	NUMDMA_MASK);
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
-				((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL,
+				((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
 		}
 	}
 
 	return 0;
 }
 
-static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
+static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 {
 	int i, active_slots;
 	u32 mask = 0;
 
-	active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots;
+	active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
 	for (i = 0; i < active_slots; i++)
 		mask |= (1 << i);
 
-	mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
+	mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* bit stream is MSB first  with no delay */
 		/* DSP_B mode */
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
 
-		if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
-					FSXMOD(dev->tdm_slots), FSXMOD(0x1FF));
+		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG,
+					FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
 		else
 			printk(KERN_ERR "playback tdm slot %d not supported\n",
-				dev->tdm_slots);
+				mcasp->tdm_slots);
 	} else {
 		/* bit stream is MSB first with no delay */
 		/* DSP_B mode */
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask);
 
-		if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
-			mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG,
-					FSRMOD(dev->tdm_slots), FSRMOD(0x1FF));
+		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
+			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG,
+					FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
 		else
 			printk(KERN_ERR "capture tdm slot %d not supported\n",
-				dev->tdm_slots);
+				mcasp->tdm_slots);
 	}
 }
 
 /* S/PDIF */
-static void davinci_hw_dit_param(struct davinci_audio_dev *dev)
+static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
 {
 	/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
 	   and LSB first */
-	mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
+	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
 						TXROT(6) | TXSSZ(15));
 
 	/* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG,
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG,
 						AFSXE | FSXMOD(0x180));
 
 	/* Set the TX tdm : for all the slots */
-	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
 
 	/* Set the TX clock controls : div = 1 and internal */
-	mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
+	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG,
 						ACLKXE | TX_ASYNC);
 
-	mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+	mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
 
 	/* Only 44100 and 48000 are valid, both have the same setting */
-	mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
+	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
 
 	/* Enable the DIT */
-	mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN);
 }
 
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params,
 					struct snd_soc_dai *cpu_dai)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 	struct davinci_pcm_dma_params *dma_params =
-					&dev->dma_params[substream->stream];
+					&mcasp->dma_params[substream->stream];
 	int word_length;
 	u8 fifo_level;
-	u8 slots = dev->tdm_slots;
+	u8 slots = mcasp->tdm_slots;
 	u8 active_serializers;
 	int channels;
 	struct snd_interval *pcm_channels = hw_param_interval(params,
@@ -608,17 +608,17 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 
 	active_serializers = (channels + slots - 1) / slots;
 
-	if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL)
+	if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL)
 		return -EINVAL;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		fifo_level = dev->txnumevt * active_serializers;
+		fifo_level = mcasp->txnumevt * active_serializers;
 	else
-		fifo_level = dev->rxnumevt * active_serializers;
+		fifo_level = mcasp->rxnumevt * active_serializers;
 
-	if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
-		davinci_hw_dit_param(dev);
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		davinci_hw_dit_param(mcasp);
 	else
-		davinci_hw_param(dev, substream->stream);
+		davinci_hw_param(mcasp, substream->stream);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U8:
@@ -652,13 +652,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (dev->version == MCASP_VERSION_2 && !fifo_level)
+	if (mcasp->version == MCASP_VERSION_2 && !fifo_level)
 		dma_params->acnt = 4;
 	else
 		dma_params->acnt = dma_params->data_type;
 
 	dma_params->fifo_level = fifo_level;
-	davinci_config_channel_size(dev, word_length);
+	davinci_config_channel_size(mcasp, word_length);
 
 	return 0;
 }
@@ -666,29 +666,29 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 				     int cmd, struct snd_soc_dai *cpu_dai)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 	int ret = 0;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ret = pm_runtime_get_sync(dev->dev);
+		ret = pm_runtime_get_sync(mcasp->dev);
 		if (IS_ERR_VALUE(ret))
-			dev_err(dev->dev, "pm_runtime_get_sync() failed\n");
-		davinci_mcasp_start(dev, substream->stream);
+			dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n");
+		davinci_mcasp_start(mcasp, substream->stream);
 		break;
 
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		davinci_mcasp_stop(dev, substream->stream);
-		ret = pm_runtime_put_sync(dev->dev);
+		davinci_mcasp_stop(mcasp, substream->stream);
+		ret = pm_runtime_put_sync(mcasp->dev);
 		if (IS_ERR_VALUE(ret))
-			dev_err(dev->dev, "pm_runtime_put_sync() failed\n");
+			dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n");
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		davinci_mcasp_stop(dev, substream->stream);
+		davinci_mcasp_stop(mcasp, substream->stream);
 		break;
 
 	default:
@@ -701,9 +701,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
-	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
-	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params);
 	return 0;
 }
 
@@ -915,7 +915,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	struct davinci_pcm_dma_params *dma_data;
 	struct resource *mem, *ioarea, *res, *dat;
 	struct snd_platform_data *pdata;
-	struct davinci_audio_dev *dev;
+	struct davinci_mcasp *mcasp;
 	int ret;
 
 	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -923,9 +923,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev),
+	mcasp = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcasp),
 			   GFP_KERNEL);
-	if (!dev)
+	if (!mcasp)
 		return	-ENOMEM;
 
 	pdata = davinci_mcasp_set_pdata_from_of(pdev);
@@ -936,7 +936,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
 	if (!mem) {
-		dev_warn(dev->dev,
+		dev_warn(mcasp->dev,
 			 "\"mpu\" mem resource not found, using index 0\n");
 		mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 		if (!mem) {
@@ -960,27 +960,27 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
-	if (!dev->base) {
+	mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem));
+	if (!mcasp->base) {
 		dev_err(&pdev->dev, "ioremap failed\n");
 		ret = -ENOMEM;
 		goto err_release_clk;
 	}
 
-	dev->op_mode = pdata->op_mode;
-	dev->tdm_slots = pdata->tdm_slots;
-	dev->num_serializer = pdata->num_serializer;
-	dev->serial_dir = pdata->serial_dir;
-	dev->version = pdata->version;
-	dev->txnumevt = pdata->txnumevt;
-	dev->rxnumevt = pdata->rxnumevt;
-	dev->dev = &pdev->dev;
+	mcasp->op_mode = pdata->op_mode;
+	mcasp->tdm_slots = pdata->tdm_slots;
+	mcasp->num_serializer = pdata->num_serializer;
+	mcasp->serial_dir = pdata->serial_dir;
+	mcasp->version = pdata->version;
+	mcasp->txnumevt = pdata->txnumevt;
+	mcasp->rxnumevt = pdata->rxnumevt;
+	mcasp->dev = &pdev->dev;
 
 	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
 	if (!dat)
 		dat = mem;
 
-	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
@@ -993,7 +993,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	else
 		dma_data->channel = pdata->tx_dma_channel;
 
-	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+	dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
@@ -1006,7 +1006,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	else
 		dma_data->channel = pdata->rx_dma_channel;
 
-	dev_set_drvdata(&pdev->dev, dev);
+	dev_set_drvdata(&pdev->dev, mcasp);
 	ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
 					 &davinci_mcasp_dai[pdata->op_mode], 1);
 
@@ -1044,32 +1044,32 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 #ifdef CONFIG_PM_SLEEP
 static int davinci_mcasp_suspend(struct device *dev)
 {
-	struct davinci_audio_dev *a = dev_get_drvdata(dev);
-	void __iomem *base = a->base;
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	void __iomem *base = mcasp->base;
 
-	a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG);
-	a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG);
-	a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG);
-	a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG);
-	a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG);
-	a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG);
-	a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG);
+	mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG);
+	mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG);
+	mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG);
+	mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG);
+	mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG);
+	mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG);
+	mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG);
 
 	return 0;
 }
 
 static int davinci_mcasp_resume(struct device *dev)
 {
-	struct davinci_audio_dev *a = dev_get_drvdata(dev);
-	void __iomem *base = a->base;
-
-	mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt);
-	mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt);
-	mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir);
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	void __iomem *base = mcasp->base;
+
+	mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt);
+	mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt);
+	mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl);
+	mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir);
 
 	return 0;
 }
-- 
cgit v1.2.2


From 8f113b77b511c9e26706d4eb077af0ba30893ee4 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:30 +0200
Subject: ASoC: davinci-mcasp: Be consistent with the use of base in
 davinci_mcasp_set_dai_fmt

Replace mcasp->base use with plain base in the davinci_mcasp_set_dai_fmt()
function since it has been already used by the remaining part of the function.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index bd85c98bf5a5..1341f327df83 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -243,17 +243,17 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_AC97:
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 		break;
 	default:
 		/* configure a full-word SYNC pulse (LRCLK) */
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 
 		/* make 1st data bit occur one ACLK cycle after the frame sync */
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
+		mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
+		mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
 		break;
 	}
 
-- 
cgit v1.2.2


From 487dce8823cdcb70e645e5312a0d4f7081e1ad13 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:31 +0200
Subject: ASoC: davinci-mcasp: Simplify FIFO configuration code

The FIFO registers base address is different in dm646x compared to newer
SoCs with McASP IP. Instead of using two paths (switch/case) to handle the
difference we can simply pick the correct base address beforehand and use
offsets to address the register we need to configure.
With this change the indentation depth can be reduced as well.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 101 +++++++++++---------------------------
 sound/soc/davinci/davinci-mcasp.h |  16 +++---
 2 files changed, 38 insertions(+), 79 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 1341f327df83..72ea45893abf 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -38,6 +38,7 @@
 struct davinci_mcasp {
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
+	u32 fifo_base;
 	struct device *dev;
 
 	/* McASP specific data */
@@ -153,38 +154,20 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 
 static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 {
+	u32 reg;
+
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* enable FIFO */
-			switch (mcasp->version) {
-			case MCASP_VERSION_3:
-				mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
-					       FIFO_ENABLE);
-				mcasp_set_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
-					       FIFO_ENABLE);
-				break;
-			default:
-				mcasp_clr_bits(mcasp->base +
-					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
-				mcasp_set_bits(mcasp->base +
-					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
-			}
+			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE);
 		}
 		mcasp_start_tx(mcasp);
 	} else {
 		if (mcasp->rxnumevt) {	/* enable FIFO */
-			switch (mcasp->version) {
-			case MCASP_VERSION_3:
-				mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
-					       FIFO_ENABLE);
-				mcasp_set_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
-					       FIFO_ENABLE);
-				break;
-			default:
-				mcasp_clr_bits(mcasp->base +
-					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
-				mcasp_set_bits(mcasp->base +
-					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
-			}
+			reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE);
 		}
 		mcasp_start_rx(mcasp);
 	}
@@ -204,31 +187,18 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
 
 static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
 {
+	u32 reg;
+
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* disable FIFO */
-			switch (mcasp->version) {
-			case MCASP_VERSION_3:
-				mcasp_clr_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
-					       FIFO_ENABLE);
-				break;
-			default:
-				mcasp_clr_bits(mcasp->base +
-					DAVINCI_MCASP_WFIFOCTL,	FIFO_ENABLE);
-			}
+			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
 		}
 		mcasp_stop_tx(mcasp);
 	} else {
 		if (mcasp->rxnumevt) {	/* disable FIFO */
-			switch (mcasp->version) {
-			case MCASP_VERSION_3:
-				mcasp_clr_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
-					       FIFO_ENABLE);
-			break;
-
-			default:
-				mcasp_clr_bits(mcasp->base +
-					DAVINCI_MCASP_RFIFOCTL,	FIFO_ENABLE);
-			}
+			reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
 		}
 		mcasp_stop_rx(mcasp);
 	}
@@ -438,6 +408,7 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 	u8 ser;
 	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
+	u32 reg;
 	/* Default configuration */
 	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
 
@@ -488,37 +459,20 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 		if (mcasp->txnumevt * tx_ser > 64)
 			mcasp->txnumevt = 1;
 
-		switch (mcasp->version) {
-		case MCASP_VERSION_3:
-			mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL, tx_ser,
-								NUMDMA_MASK);
-			mcasp_mod_bits(mcasp->base + MCASP_VER3_WFIFOCTL,
-				((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
-			break;
-		default:
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL,
-							tx_ser,	NUMDMA_MASK);
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_WFIFOCTL,
-				((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
-		}
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK);
+		mcasp_mod_bits(mcasp->base + reg,
+			       ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
 	}
 
 	if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
 		if (mcasp->rxnumevt * rx_ser > 64)
 			mcasp->rxnumevt = 1;
-		switch (mcasp->version) {
-		case MCASP_VERSION_3:
-			mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL, rx_ser,
-								NUMDMA_MASK);
-			mcasp_mod_bits(mcasp->base + MCASP_VER3_RFIFOCTL,
-				((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
-			break;
-		default:
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL,
-							rx_ser,	NUMDMA_MASK);
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RFIFOCTL,
-				((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
-		}
+
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK);
+		mcasp_mod_bits(mcasp->base + reg,
+			       ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
 	}
 
 	return 0;
@@ -974,6 +928,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->version = pdata->version;
 	mcasp->txnumevt = pdata->txnumevt;
 	mcasp->rxnumevt = pdata->rxnumevt;
+	if (mcasp->version < MCASP_VERSION_3)
+		mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
+	else
+		mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
+
 	mcasp->dev = &pdev->dev;
 
 	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 80e5a1846687..8fed757d6087 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -90,14 +90,14 @@
 #define DAVINCI_MCASP_RXBUF_REG		0x280
 
 /* McASP FIFO Registers */
-#define DAVINCI_MCASP_WFIFOCTL		(0x1010)
-#define DAVINCI_MCASP_WFIFOSTS		(0x1014)
-#define DAVINCI_MCASP_RFIFOCTL		(0x1018)
-#define DAVINCI_MCASP_RFIFOSTS		(0x101C)
-#define MCASP_VER3_WFIFOCTL		(0x1000)
-#define MCASP_VER3_WFIFOSTS		(0x1004)
-#define MCASP_VER3_RFIFOCTL		(0x1008)
-#define MCASP_VER3_RFIFOSTS		(0x100C)
+#define DAVINCI_MCASP_V2_AFIFO_BASE	(0x1010)
+#define DAVINCI_MCASP_V3_AFIFO_BASE	(0x1000)
+
+/* FIFO register offsets from AFIFO base */
+#define MCASP_WFIFOCTL_OFFSET		(0x0)
+#define MCASP_WFIFOSTS_OFFSET		(0x4)
+#define MCASP_RFIFOCTL_OFFSET		(0x8)
+#define MCASP_RFIFOSTS_OFFSET		(0xc)
 
 /*
  * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
-- 
cgit v1.2.2


From cbc7956c81eea644c0d99aee43f1632897703300 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:32 +0200
Subject: ASoC: davinci-mcasp: Data source (bus) selection support

The audio data to/from McASP can be sent/received via two method:
Via the data port (preferred) or via the configuration bus.
Currently the driver assumes that all data communication will be done via
the data port.
This patch adds support for selecting the configuration port as data
interface.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 38 ++++++++++++++++++++++++++++----------
 1 file changed, 28 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 72ea45893abf..35a6292889a5 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -53,6 +53,8 @@ struct davinci_mcasp {
 	u8	txnumevt;
 	u8	rxnumevt;
 
+	bool	dat_port;
+
 #ifdef CONFIG_PM_SLEEP
 	struct {
 		u32	txfmtctl;
@@ -482,6 +484,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 {
 	int i, active_slots;
 	u32 mask = 0;
+	u32 busel = 0;
 
 	active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
 	for (i = 0; i < active_slots; i++)
@@ -489,11 +492,15 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 
 	mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
 
+	if (!mcasp->dat_port)
+		busel = TXSEL;
+
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* bit stream is MSB first  with no delay */
 		/* DSP_B mode */
 		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask);
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
+			       busel | TXORD);
 
 		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
 			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG,
@@ -504,7 +511,8 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 	} else {
 		/* bit stream is MSB first with no delay */
 		/* DSP_B mode */
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
+			       busel | RXORD);
 		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask);
 
 		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
@@ -928,23 +936,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->version = pdata->version;
 	mcasp->txnumevt = pdata->txnumevt;
 	mcasp->rxnumevt = pdata->rxnumevt;
-	if (mcasp->version < MCASP_VERSION_3)
-		mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
-	else
-		mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
 
 	mcasp->dev = &pdev->dev;
 
 	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
-	if (!dat)
-		dat = mem;
+	if (dat)
+		mcasp->dat_port = true;
 
 	dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
 	dma_data->sram_size = pdata->sram_size_playback;
-	dma_data->dma_addr = dat->start + pdata->tx_dma_offset;
+	if (dat)
+		dma_data->dma_addr = dat->start;
+	else
+		dma_data->dma_addr = mem->start + pdata->tx_dma_offset;
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (res)
@@ -957,7 +964,18 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->sram_pool = pdata->sram_pool;
 	dma_data->sram_size = pdata->sram_size_capture;
-	dma_data->dma_addr = dat->start + pdata->rx_dma_offset;
+	if (dat)
+		dma_data->dma_addr = dat->start;
+	else
+		dma_data->dma_addr = mem->start + pdata->rx_dma_offset;
+
+	if (mcasp->version < MCASP_VERSION_3) {
+		mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
+		/* dma_data->dma_addr is pointing to the data port address */
+		mcasp->dat_port = true;
+	} else {
+		mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
+	}
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (res)
-- 
cgit v1.2.2


From 4dcb5a0bffaa7dc51e738c4f651a31993b1eb08b Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:33 +0200
Subject: ASoC: davinci-mcasp: Fix synchronous master receive mode

In synchronous mode both transmit and receive sections are using the TX
clocks. In setup like this the TX clocks need to be enabled when capture
is running.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 48 ++++++++++++++++++++++++++++++++++++++-
 1 file changed, 47 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 35a6292889a5..93f2e294d649 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -48,6 +48,7 @@ struct davinci_mcasp {
 	u8	*serial_dir;
 	u8	version;
 	u16	bclk_lrclk_ratio;
+	int	streams;
 
 	/* McASP FIFO related */
 	u8	txnumevt;
@@ -110,10 +111,31 @@ static void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
 		printk(KERN_ERR "GBLCTL write error\n");
 }
 
+static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp)
+{
+	u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG);
+	u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG);
+
+	return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE;
+}
+
 static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 {
 	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
 	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
+
+	/*
+	 * When ASYNC == 0 the transmit and receive sections operate
+	 * synchronously from the transmit clock and frame sync. We need to make
+	 * sure that the TX signlas are enabled when starting reception.
+	 */
+	if (mcasp_is_synchronous(mcasp)) {
+		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
+				  TXHCLKRST);
+		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
+				  TXCLKRST);
+	}
+
 	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
 	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0);
 
@@ -123,6 +145,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 
 	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
 	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+
+	if (mcasp_is_synchronous(mcasp))
+		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
+				  TXFSRST);
 }
 
 static void mcasp_start_tx(struct davinci_mcasp *mcasp)
@@ -158,6 +184,8 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 {
 	u32 reg;
 
+	mcasp->streams++;
+
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* enable FIFO */
 			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
@@ -177,13 +205,29 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 
 static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
 {
+	/*
+	 * In synchronous mode stop the TX clocks if no other stream is
+	 * running
+	 */
+	if (mcasp_is_synchronous(mcasp) && !mcasp->streams)
+		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
+
 	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
 	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
 }
 
 static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
 {
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
+	u32 val = 0;
+
+	/*
+	 * In synchronous mode keep TX clocks running if the capture stream is
+	 * still running.
+	 */
+	if (mcasp_is_synchronous(mcasp) && mcasp->streams)
+		val =  TXHCLKRST | TXCLKRST | TXFSRST;
+
+	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val);
 	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
 }
 
@@ -191,6 +235,8 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
 {
 	u32 reg;
 
+	mcasp->streams--;
+
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* disable FIFO */
 			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-- 
cgit v1.2.2


From 453c499028bf2ecf3b31ccb7c3657fe1b0b28943 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:34 +0200
Subject: ASoC: davinci-mcasp: Support for McASP version found in DRA7xx

The IP in DRA7xx is similar to the IP found in TI81xxAM3xxx/AM4xxx type of
SoCs but it is is integrated with sDMA instead of eDMA. The suitable pcm
driver for DRA7xx is the omap-pcm driver which is using dmaengine.
In the driver we can configure both dma related structures used for eDMA and
sDMA. The only thing we need to make sure that we set the correct dma_data
at startup with snd_soc_dai_set_dma_data()

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 52 +++++++++++++++++++++++++++++++++------
 1 file changed, 45 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 93f2e294d649..fc8c13d2f31e 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -31,12 +31,14 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "davinci-pcm.h"
 #include "davinci-mcasp.h"
 
 struct davinci_mcasp {
 	struct davinci_pcm_dma_params dma_params[2];
+	struct snd_dmaengine_dai_dma_data dma_data[2];
 	void __iomem *base;
 	u32 fifo_base;
 	struct device *dev;
@@ -458,7 +460,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 	u8 max_active_serializers = (channels + slots - 1) / slots;
 	u32 reg;
 	/* Default configuration */
-	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
+	if (mcasp->version != MCASP_VERSION_4)
+		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG,
+			       MCASP_SOFT);
 
 	/* All PINS as McASP */
 	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
@@ -605,6 +609,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 	struct davinci_pcm_dma_params *dma_params =
 					&mcasp->dma_params[substream->stream];
+	struct snd_dmaengine_dai_dma_data *dma_data =
+					&mcasp->dma_data[substream->stream];
 	int word_length;
 	u8 fifo_level;
 	u8 slots = mcasp->tdm_slots;
@@ -666,6 +672,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		dma_params->acnt = dma_params->data_type;
 
 	dma_params->fifo_level = fifo_level;
+	dma_data->maxburst = fifo_level;
+
 	davinci_config_channel_size(mcasp, word_length);
 
 	return 0;
@@ -711,7 +719,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
-	snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params);
+	if (mcasp->version == MCASP_VERSION_4)
+		snd_soc_dai_set_dma_data(dai, substream,
+					&mcasp->dma_data[substream->stream]);
+	else
+		snd_soc_dai_set_dma_data(dai, substream, mcasp->dma_params);
+
 	return 0;
 }
 
@@ -794,6 +807,13 @@ static struct snd_platform_data omap2_mcasp_pdata = {
 	.version = MCASP_VERSION_3,
 };
 
+static struct snd_platform_data dra7_mcasp_pdata = {
+	.tx_dma_offset = 0x200,
+	.rx_dma_offset = 0x284,
+	.asp_chan_q = EVENTQ_0,
+	.version = MCASP_VERSION_4,
+};
+
 static const struct of_device_id mcasp_dt_ids[] = {
 	{
 		.compatible = "ti,dm646x-mcasp-audio",
@@ -807,6 +827,10 @@ static const struct of_device_id mcasp_dt_ids[] = {
 		.compatible = "ti,am33xx-mcasp-audio",
 		.data = &omap2_mcasp_pdata,
 	},
+	{
+		.compatible = "ti,dra7-mcasp-audio",
+		.data = &dra7_mcasp_pdata,
+	},
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, mcasp_dt_ids);
@@ -999,6 +1023,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	else
 		dma_data->dma_addr = mem->start + pdata->tx_dma_offset;
 
+	/* Unconditional dmaengine stuff */
+	mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr;
+
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (res)
 		dma_data->channel = res->start;
@@ -1015,6 +1042,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	else
 		dma_data->dma_addr = mem->start + pdata->rx_dma_offset;
 
+	/* Unconditional dmaengine stuff */
+	mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr;
+
 	if (mcasp->version < MCASP_VERSION_3) {
 		mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
 		/* dma_data->dma_addr is pointing to the data port address */
@@ -1029,6 +1059,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	else
 		dma_data->channel = pdata->rx_dma_channel;
 
+	/* Unconditional dmaengine stuff */
+	mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx";
+	mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx";
+
 	dev_set_drvdata(&pdev->dev, mcasp);
 	ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
 					 &davinci_mcasp_dai[pdata->op_mode], 1);
@@ -1036,10 +1070,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	if (ret != 0)
 		goto err_release_clk;
 
-	ret = davinci_soc_platform_register(&pdev->dev);
-	if (ret) {
-		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
-		goto err_unregister_component;
+	if (mcasp->version != MCASP_VERSION_4) {
+		ret = davinci_soc_platform_register(&pdev->dev);
+		if (ret) {
+			dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+			goto err_unregister_component;
+		}
 	}
 
 	return 0;
@@ -1054,9 +1090,11 @@ err_release_clk:
 
 static int davinci_mcasp_remove(struct platform_device *pdev)
 {
+	struct davinci_mcasp *mcasp = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	davinci_soc_platform_unregister(&pdev->dev);
+	if (mcasp->version != MCASP_VERSION_4)
+		davinci_soc_platform_unregister(&pdev->dev);
 
 	pm_runtime_put_sync(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
-- 
cgit v1.2.2


From f68205a7f8c0b1fd02cec6116bbb66bb4fd7bc51 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:36 +0200
Subject: ASoC: davinci-mcasp: Change IO functions parameter list

Instead of passing __iomem address (mcasp->base + register_offset) pass
the main mcasp structure and only access the mcasp->base in the low level
IO functions.
In most cases this helps with code readability and it will make it easier
to switch over to regmap in the future.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 336 ++++++++++++++++++--------------------
 1 file changed, 161 insertions(+), 175 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index fc8c13d2f31e..19c66625af4e 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -71,60 +71,67 @@ struct davinci_mcasp {
 #endif
 };
 
-static inline void mcasp_set_bits(void __iomem *reg, u32 val)
+static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val)
 {
+	void __iomem *reg = mcasp->base + offset;
 	__raw_writel(__raw_readl(reg) | val, reg);
 }
 
-static inline void mcasp_clr_bits(void __iomem *reg, u32 val)
+static inline void mcasp_clr_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val)
 {
+	void __iomem *reg = mcasp->base + offset;
 	__raw_writel((__raw_readl(reg) & ~(val)), reg);
 }
 
-static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask)
+static inline void mcasp_mod_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val, u32 mask)
 {
+	void __iomem *reg = mcasp->base + offset;
 	__raw_writel((__raw_readl(reg) & ~mask) | val, reg);
 }
 
-static inline void mcasp_set_reg(void __iomem *reg, u32 val)
+static inline void mcasp_set_reg(struct davinci_mcasp *mcasp, u32 offset,
+				 u32 val)
 {
-	__raw_writel(val, reg);
+	__raw_writel(val, mcasp->base + offset);
 }
 
-static inline u32 mcasp_get_reg(void __iomem *reg)
+static inline u32 mcasp_get_reg(struct davinci_mcasp *mcasp, u32 offset)
 {
-	return (unsigned int)__raw_readl(reg);
+	return (u32)__raw_readl(mcasp->base + offset);
 }
 
-static void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
+static void mcasp_set_ctl_reg(struct davinci_mcasp *mcasp, u32 ctl_reg, u32 val)
 {
 	int i = 0;
 
-	mcasp_set_bits(regs, val);
+	mcasp_set_bits(mcasp, ctl_reg, val);
 
 	/* programming GBLCTL needs to read back from GBLCTL and verfiy */
 	/* loop count is to avoid the lock-up */
 	for (i = 0; i < 1000; i++) {
-		if ((mcasp_get_reg(regs) & val) == val)
+		if ((mcasp_get_reg(mcasp, ctl_reg) & val) == val)
 			break;
 	}
 
-	if (i == 1000 && ((mcasp_get_reg(regs) & val) != val))
+	if (i == 1000 && ((mcasp_get_reg(mcasp, ctl_reg) & val) != val))
 		printk(KERN_ERR "GBLCTL write error\n");
 }
 
 static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp)
 {
-	u32 rxfmctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG);
-	u32 aclkxctl = mcasp_get_reg(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG);
+	u32 rxfmctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
+	u32 aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
 
 	return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE;
 }
 
 static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 {
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
 
 	/*
 	 * When ASYNC == 0 the transmit and receive sections operate
@@ -132,25 +139,22 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 	 * sure that the TX signlas are enabled when starting reception.
 	 */
 	if (mcasp_is_synchronous(mcasp)) {
-		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
-				  TXHCLKRST);
-		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
-				  TXCLKRST);
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
 	}
 
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
 
 	if (mcasp_is_synchronous(mcasp))
-		mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG,
-				  TXFSRST);
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
 }
 
 static void mcasp_start_tx(struct davinci_mcasp *mcasp)
@@ -158,14 +162,14 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 	u8 offset = 0, i;
 	u32 cnt;
 
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
 
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
-	mcasp_set_ctl_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
 	for (i = 0; i < mcasp->num_serializer; i++) {
 		if (mcasp->serial_dir[i] == TX_MODE) {
 			offset = i;
@@ -175,11 +179,11 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 
 	/* wait for TX ready */
 	cnt = 0;
-	while (!(mcasp_get_reg(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) &
+	while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(offset)) &
 		 TXSTATE) && (cnt < 100000))
 		cnt++;
 
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXBUF_REG, 0);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0);
 }
 
 static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
@@ -191,15 +195,15 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* enable FIFO */
 			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
-			mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+			mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
 		}
 		mcasp_start_tx(mcasp);
 	} else {
 		if (mcasp->rxnumevt) {	/* enable FIFO */
 			reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
-			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
-			mcasp_set_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+			mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
 		}
 		mcasp_start_rx(mcasp);
 	}
@@ -212,10 +216,10 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
 	 * running
 	 */
 	if (mcasp_is_synchronous(mcasp) && !mcasp->streams)
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, 0);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, 0);
 
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLR_REG, 0);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
 }
 
 static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
@@ -229,8 +233,8 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
 	if (mcasp_is_synchronous(mcasp) && mcasp->streams)
 		val =  TXHCLKRST | TXCLKRST | TXFSRST;
 
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_GBLCTLX_REG, val);
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
 }
 
 static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
@@ -242,13 +246,13 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (mcasp->txnumevt) {	/* disable FIFO */
 			reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
 		}
 		mcasp_stop_tx(mcasp);
 	} else {
 		if (mcasp->rxnumevt) {	/* disable FIFO */
 			reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
-			mcasp_clr_bits(mcasp->base + reg, FIFO_ENABLE);
+			mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
 		}
 		mcasp_stop_rx(mcasp);
 	}
@@ -258,62 +262,57 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 					 unsigned int fmt)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
-	void __iomem *base = mcasp->base;
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_AC97:
-		mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 		break;
 	default:
 		/* configure a full-word SYNC pulse (LRCLK) */
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 
 		/* make 1st data bit occur one ACLK cycle after the frame sync */
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
 		break;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
 		/* codec is clock and frame slave */
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
-				ACLKX | ACLKR);
-		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
-				AFSX | AFSR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		/* codec is clock master and frame slave */
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
-				ACLKX | ACLKR);
-		mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
-				AFSX | AFSR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* codec is clock and frame master */
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
-		mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
-		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG,
-				ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
+			       ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
 		break;
 
 	default:
@@ -322,35 +321,35 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_IB_NF:
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
-		mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
-		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
 		break;
 
 	case SND_SOC_DAIFMT_NB_IF:
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
 
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
 		break;
 
 	case SND_SOC_DAIFMT_IB_IF:
-		mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
-		mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
-		mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
 		break;
 
 	case SND_SOC_DAIFMT_NB_NF:
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
-		mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
 
-		mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
-		mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
 		break;
 
 	default:
@@ -366,16 +365,16 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 
 	switch (div_id) {
 	case 0:		/* MCLK divider */
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG,
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
 			       AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG,
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
 			       AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
 		break;
 
 	case 1:		/* BCLK divider */
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG,
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG,
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_ACLKRCTL_REG,
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
 			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
 		break;
 
@@ -396,13 +395,13 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
 	if (dir == SND_SOC_CLOCK_OUT) {
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX);
 	} else {
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX);
 	}
 
 	return 0;
@@ -432,19 +431,18 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 	fmt = (word_length >> 1) - 1;
 
 	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
-				RXSSZ(fmt), RXSSZ(0x0F));
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
-				TXSSZ(fmt), TXSSZ(0x0F));
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
-				TXROT(tx_rotate), TXROT(7));
-		mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
-				RXROT(rx_rotate), RXROT(7));
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXMASK_REG,
-				mask);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
+			       RXSSZ(0x0F));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt),
+			       TXSSZ(0x0F));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate),
+			       TXROT(7));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate),
+			       RXROT(7));
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask);
 	}
 
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXMASK_REG, mask);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask);
 
 	return 0;
 }
@@ -461,38 +459,33 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 	u32 reg;
 	/* Default configuration */
 	if (mcasp->version != MCASP_VERSION_4)
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PWREMUMGT_REG,
-			       MCASP_SOFT);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
 
 	/* All PINS as McASP */
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000);
 
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG,
-				TXDATADMADIS);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
 	} else {
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
-		mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_REVTCTL_REG,
-				RXDATADMADIS);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS);
 	}
 
 	for (i = 0; i < mcasp->num_serializer; i++) {
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i),
-					mcasp->serial_dir[i]);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			       mcasp->serial_dir[i]);
 		if (mcasp->serial_dir[i] == TX_MODE &&
 					tx_ser < max_active_serializers) {
-			mcasp_set_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG,
-					AXR(i));
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i));
 			tx_ser++;
 		} else if (mcasp->serial_dir[i] == RX_MODE &&
 					rx_ser < max_active_serializers) {
-			mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_PDIR_REG,
-					AXR(i));
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i));
 			rx_ser++;
 		} else {
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_XRSRCTL_REG(i),
-					SRMOD_INACTIVE, SRMOD_MASK);
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+				       SRMOD_INACTIVE, SRMOD_MASK);
 		}
 	}
 
@@ -512,9 +505,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 			mcasp->txnumevt = 1;
 
 		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
-		mcasp_mod_bits(mcasp->base + reg, tx_ser, NUMDMA_MASK);
-		mcasp_mod_bits(mcasp->base + reg,
-			       ((mcasp->txnumevt * tx_ser) << 8), NUMEVT_MASK);
+		mcasp_mod_bits(mcasp, reg, tx_ser, NUMDMA_MASK);
+		mcasp_mod_bits(mcasp, reg, ((mcasp->txnumevt * tx_ser) << 8),
+			       NUMEVT_MASK);
 	}
 
 	if (mcasp->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -522,9 +515,9 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 			mcasp->rxnumevt = 1;
 
 		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
-		mcasp_mod_bits(mcasp->base + reg, rx_ser, NUMDMA_MASK);
-		mcasp_mod_bits(mcasp->base + reg,
-			       ((mcasp->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
+		mcasp_mod_bits(mcasp, reg, rx_ser, NUMDMA_MASK);
+		mcasp_mod_bits(mcasp, reg, ((mcasp->rxnumevt * rx_ser) << 8),
+			       NUMEVT_MASK);
 	}
 
 	return 0;
@@ -540,7 +533,7 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 	for (i = 0; i < active_slots; i++)
 		mask |= (1 << i);
 
-	mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
 
 	if (!mcasp->dat_port)
 		busel = TXSEL;
@@ -548,26 +541,24 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* bit stream is MSB first  with no delay */
 		/* DSP_B mode */
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, mask);
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
-			       busel | TXORD);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
 
 		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG,
-					FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+				       FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
 		else
 			printk(KERN_ERR "playback tdm slot %d not supported\n",
 				mcasp->tdm_slots);
 	} else {
 		/* bit stream is MSB first with no delay */
 		/* DSP_B mode */
-		mcasp_set_bits(mcasp->base + DAVINCI_MCASP_RXFMT_REG,
-			       busel | RXORD);
-		mcasp_set_reg(mcasp->base + DAVINCI_MCASP_RXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
 
 		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
-			mcasp_mod_bits(mcasp->base + DAVINCI_MCASP_RXFMCTL_REG,
-					FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+				       FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
 		else
 			printk(KERN_ERR "capture tdm slot %d not supported\n",
 				mcasp->tdm_slots);
@@ -579,27 +570,24 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
 {
 	/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
 	   and LSB first */
-	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXFMT_REG,
-						TXROT(6) | TXSSZ(15));
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15));
 
 	/* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXFMCTL_REG,
-						AFSXE | FSXMOD(0x180));
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180));
 
 	/* Set the TX tdm : for all the slots */
-	mcasp_set_reg(mcasp->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
 
 	/* Set the TX clock controls : div = 1 and internal */
-	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_ACLKXCTL_REG,
-						ACLKXE | TX_ASYNC);
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC);
 
-	mcasp_clr_bits(mcasp->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
 
 	/* Only 44100 and 48000 are valid, both have the same setting */
-	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3));
 
 	/* Enable the DIT */
-	mcasp_set_bits(mcasp->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
 }
 
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -1106,15 +1094,14 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 static int davinci_mcasp_suspend(struct device *dev)
 {
 	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-	void __iomem *base = mcasp->base;
 
-	mcasp->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG);
-	mcasp->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG);
-	mcasp->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG);
-	mcasp->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG);
-	mcasp->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG);
-	mcasp->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG);
-	mcasp->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG);
+	mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
+	mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
+	mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
+	mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
+	mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
+	mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
+	mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
 
 	return 0;
 }
@@ -1122,15 +1109,14 @@ static int davinci_mcasp_suspend(struct device *dev)
 static int davinci_mcasp_resume(struct device *dev)
 {
 	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-	void __iomem *base = mcasp->base;
-
-	mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt);
-	mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt);
-	mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl);
-	mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir);
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir);
 
 	return 0;
 }
-- 
cgit v1.2.2


From b14899da9ddeb8501db13fd08d0d1a8af61529c5 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:37 +0200
Subject: ASoC: davinci-mcasp: Correct am33xx snd_platform_data name

An earlier patch overlooked this when the compatible has been changed from
omap2 -> am33x.
Rename omap2_mcasp_pdata to am33xx_mcasp_pdata.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 19c66625af4e..8ec879548488 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -788,7 +788,7 @@ static struct snd_platform_data da830_mcasp_pdata = {
 	.version = MCASP_VERSION_2,
 };
 
-static struct snd_platform_data omap2_mcasp_pdata = {
+static struct snd_platform_data am33xx_mcasp_pdata = {
 	.tx_dma_offset = 0,
 	.rx_dma_offset = 0,
 	.asp_chan_q = EVENTQ_0,
@@ -813,7 +813,7 @@ static const struct of_device_id mcasp_dt_ids[] = {
 	},
 	{
 		.compatible = "ti,am33xx-mcasp-audio",
-		.data = &omap2_mcasp_pdata,
+		.data = &am33xx_mcasp_pdata,
 	},
 	{
 		.compatible = "ti,dra7-mcasp-audio",
-- 
cgit v1.2.2


From ae726e93946403b14f8cad20d5cbd22d015c9106 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 14 Nov 2013 11:35:35 +0200
Subject: ASoC: davinci-mcasp: Support for fck reparenting

Optional DT property to specify the desired parent clock for the McASP fck
clock.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-mcasp.c | 44 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 44 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 8ec879548488..b7858bfa0295 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -21,6 +21,7 @@
 #include <linux/slab.h>
 #include <linux/delay.h>
 #include <linux/io.h>
+#include <linux/clk.h>
 #include <linux/pm_runtime.h>
 #include <linux/of.h>
 #include <linux/of_platform.h>
@@ -823,6 +824,46 @@ static const struct of_device_id mcasp_dt_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, mcasp_dt_ids);
 
+static int mcasp_reparent_fck(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct clk *gfclk, *parent_clk;
+	const char *parent_name;
+	int ret;
+
+	if (!node)
+		return 0;
+
+	parent_name = of_get_property(node, "fck_parent", NULL);
+	if (!parent_name)
+		return 0;
+
+	gfclk = clk_get(&pdev->dev, "fck");
+	if (IS_ERR(gfclk)) {
+		dev_err(&pdev->dev, "failed to get fck\n");
+		return PTR_ERR(gfclk);
+	}
+
+	parent_clk = clk_get(NULL, parent_name);
+	if (IS_ERR(parent_clk)) {
+		dev_err(&pdev->dev, "failed to get parent clock\n");
+		ret = PTR_ERR(parent_clk);
+		goto err1;
+	}
+
+	ret = clk_set_parent(gfclk, parent_clk);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to reparent fck\n");
+		goto err2;
+	}
+
+err2:
+	clk_put(parent_clk);
+err1:
+	clk_put(gfclk);
+	return ret;
+}
+
 static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
 						struct platform_device *pdev)
 {
@@ -1052,6 +1093,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx";
 
 	dev_set_drvdata(&pdev->dev, mcasp);
+
+	mcasp_reparent_fck(pdev);
+
 	ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
 					 &davinci_mcasp_dai[pdata->op_mode], 1);
 
-- 
cgit v1.2.2


From f62438ac53ae852eca8e98a5a96742aac41403c7 Mon Sep 17 00:00:00 2001
From: Paul Walmsley <pwalmsley@nvidia.com>
Date: Mon, 9 Dec 2013 18:40:48 -0800
Subject: ALSA: at73c213: clk_round_rate() can return a zero upon error

Treat both negative and zero return values from clk_round_rate()
as errors.  This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.

Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error.  All other values will be
considered valid rates.  The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.

Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/spi/at73c213.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 8e3d9a6c7a3b..25c38afaee49 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -174,7 +174,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip)
 		dac_rate_new = 8 * (ssc_rate / ssc_div);
 
 		status = clk_round_rate(chip->board->dac_clk, dac_rate_new);
-		if (status < 0)
+		if (status <= 0)
 			return status;
 
 		/* Ignore difference smaller than 256 Hz. */
-- 
cgit v1.2.2


From 337bb336b95bd7884fa3a194eafbdf52a0216b2e Mon Sep 17 00:00:00 2001
From: Paul Walmsley <pwalmsley@nvidia.com>
Date: Mon, 9 Dec 2013 18:49:13 -0800
Subject: ALSA: atmel_abdac: clk_round_rate() can return a zero upon error

Treat both negative and zero return values from clk_round_rate()
as errors.  This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.

Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error; all other values will be
considered valid rates.  The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.

Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/atmel/abdac.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 872d59e35ee2..b78a3bc6f35e 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -354,7 +354,7 @@ static int set_sample_rates(struct atmel_abdac *dac)
 	/* we start at 192 kHz and work our way down to 5112 Hz */
 	while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) {
 		new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate);
-		if (new_rate < 0)
+		if (new_rate <= 0)
 			break;
 		/* make sure we are below the ABDAC clock */
 		if (new_rate <= clk_get_rate(dac->pclk)) {
-- 
cgit v1.2.2


From a073278228836d7d18fdd6c40b619919c0befb64 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Tue, 10 Dec 2013 20:46:59 -0800
Subject: ASoC: fsi: remove original filter from fsi_dma_probe()

Remove original filter from fsi_dma_probe(),
and use SH-DMA suitable filter.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/fsi.c | 36 +++++++++++++++++++++++-------------
 1 file changed, 23 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index b33ca7cd085b..6101055aae1d 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -232,9 +232,9 @@ struct fsi_stream {
 	 * these are for DMAEngine
 	 */
 	struct dma_chan		*chan;
-	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
 	struct work_struct	work;
 	dma_addr_t		dma;
+	int			dma_id;
 	int			loop_cnt;
 	int			additional_pos;
 };
@@ -1410,15 +1410,6 @@ static void fsi_dma_do_work(struct work_struct *work)
 	}
 }
 
-static bool fsi_dma_filter(struct dma_chan *chan, void *param)
-{
-	struct sh_dmae_slave *slave = param;
-
-	chan->private = slave;
-
-	return true;
-}
-
 static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 {
 	schedule_work(&io->work);
@@ -1446,15 +1437,34 @@ static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
 static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev)
 {
 	dma_cap_mask_t mask;
+	int is_play = fsi_stream_is_play(fsi, io);
 
 	dma_cap_zero(mask);
 	dma_cap_set(DMA_SLAVE, mask);
 
-	io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave);
+	io->chan = dma_request_slave_channel_compat(mask,
+				shdma_chan_filter, (void *)io->dma_id,
+				dev, is_play ? "tx" : "rx");
+	if (io->chan) {
+		struct dma_slave_config cfg;
+		int ret;
+
+		cfg.slave_id	= io->dma_id;
+		cfg.dst_addr	= 0; /* use default addr */
+		cfg.src_addr	= 0; /* use default addr */
+		cfg.direction	= is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+
+		ret = dmaengine_slave_config(io->chan, &cfg);
+		if (ret < 0) {
+			dma_release_channel(io->chan);
+			io->chan = NULL;
+		}
+	}
+
 	if (!io->chan) {
 
 		/* switch to PIO handler */
-		if (fsi_stream_is_play(fsi, io))
+		if (is_play)
 			fsi->playback.handler	= &fsi_pio_push_handler;
 		else
 			fsi->capture.handler	= &fsi_pio_pop_handler;
@@ -1960,7 +1970,7 @@ static void fsi_handler_init(struct fsi_priv *fsi,
 	fsi->capture.priv	= fsi;
 
 	if (info->tx_id) {
-		fsi->playback.slave.shdma_slave.slave_id = info->tx_id;
+		fsi->playback.dma_id  = info->tx_id;
 		fsi->playback.handler = &fsi_dma_push_handler;
 	}
 }
-- 
cgit v1.2.2


From 5eda87b890f867b098e5566b5543642851e8b9c3 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 11:11:02 -0700
Subject: ASoC: dmaengine: support deferred probe for DMA channels

Enhance dmaengine_pcm_request_chan_of() to support deferred probe for
DMA channels, by using the new dma_request_slave_channel_or_err() API.
This prevents snd_dmaengine_pcm_register() from succeeding without
acquiring DMA channels due to the relevant DMA controller not yet being
registered.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 20 ++++++++++++++++----
 1 file changed, 16 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 5b70c556fba3..585eaa69e8c3 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -287,16 +287,17 @@ static const char * const dmaengine_pcm_dma_channel_names[] = {
 	[SNDRV_PCM_STREAM_CAPTURE] = "rx",
 };
 
-static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
+static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	struct device *dev, const struct snd_dmaengine_pcm_config *config)
 {
 	unsigned int i;
 	const char *name;
+	struct dma_chan *chan;
 
 	if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT |
 			   SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) ||
 	    !dev->of_node)
-		return;
+		return 0;
 
 	if (config->dma_dev) {
 		/*
@@ -318,13 +319,22 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 			name = dmaengine_pcm_dma_channel_names[i];
 		if (config->chan_names[i])
 			name = config->chan_names[i];
-		pcm->chan[i] = dma_request_slave_channel(dev, name);
+		chan = dma_request_slave_channel_reason(dev, name);
+		if (IS_ERR(chan)) {
+			if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER)
+				return -EPROBE_DEFER;
+			pcm->chan[i] = NULL;
+		} else {
+			pcm->chan[i] = chan;
+		}
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			break;
 	}
 
 	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 		pcm->chan[1] = pcm->chan[0];
+
+	return 0;
 }
 
 static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
@@ -360,7 +370,9 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	pcm->config = config;
 	pcm->flags = flags;
 
-	dmaengine_pcm_request_chan_of(pcm, dev, config);
+	ret = dmaengine_pcm_request_chan_of(pcm, dev, config);
+	if (ret)
+		goto err_free_dma;
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
 		ret = snd_soc_add_platform(dev, &pcm->platform,
-- 
cgit v1.2.2


From e9036c2a60f368b8ddc349fbbcde918ed32ab597 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 11 Dec 2013 11:20:50 -0700
Subject: ASoC: dmaengine: fix deferred probe detection

Check the return value of dma_request_slave_channel_reason() to see if
deferred probe happens, not the variable the return value will be
assigned to later.

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 5eda87b890f8 ("ASoC: dmaengine: support deferred probe for DMA channels")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 585eaa69e8c3..7483922f6ee3 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -321,7 +321,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 			name = config->chan_names[i];
 		chan = dma_request_slave_channel_reason(dev, name);
 		if (IS_ERR(chan)) {
-			if (PTR_ERR(pcm->chan[i]) == -EPROBE_DEFER)
+			if (PTR_ERR(chan) == -EPROBE_DEFER)
 				return -EPROBE_DEFER;
 			pcm->chan[i] = NULL;
 		} else {
-- 
cgit v1.2.2


From 3688569e8173e84cd95d98f158245e17bca4f593 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Sat, 19 Oct 2013 15:23:15 +0100
Subject: ASoC: samsung: Provide helper for DMA init

In preparation for using the dmaengine helpers in ASoC rather than the
dmaengine wrappers for the Samsung API wrap the configuration of dma_data.
The dmaengine code expects different data to that used by the legacy API.

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/ac97.c | 51 +++++++++++++++---------------------------------
 sound/soc/samsung/dma.c  |  8 ++++++++
 sound/soc/samsung/dma.h  |  3 +++
 sound/soc/samsung/i2s.c  |  7 +++++--
 sound/soc/samsung/pcm.c  | 18 +++++++++--------
 5 files changed, 42 insertions(+), 45 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index 350ba23a9893..4a88e36c82ec 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = {
 	.reset      = s3c_ac97_cold_reset,
 };
 
-static int s3c_ac97_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *params,
-				  struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct s3c_dma_params *dma_data;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dma_data = &s3c_ac97_pcm_out;
-	else
-		dma_data = &s3c_ac97_pcm_in;
-
-	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
-
-	return 0;
-}
-
 static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 				struct snd_soc_dai *dai)
 {
@@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 	return 0;
 }
 
-static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-				      struct snd_pcm_hw_params *params,
-				      struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return -ENODEV;
-	else
-		snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in);
-
-	return 0;
-}
-
 static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 				    int cmd, struct snd_soc_dai *dai)
 {
@@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 }
 
 static const struct snd_soc_dai_ops s3c_ac97_dai_ops = {
-	.hw_params	= s3c_ac97_hw_params,
 	.trigger	= s3c_ac97_trigger,
 };
 
 static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = {
-	.hw_params	= s3c_ac97_hw_mic_params,
 	.trigger	= s3c_ac97_mic_trigger,
 };
 
+static int s3c_ac97_dai_probe(struct snd_soc_dai *dai)
+{
+	samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in);
+
+	return 0;
+}
+
+static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai)
+{
+	samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in);
+
+	return 0;
+}
+
 static struct snd_soc_dai_driver s3c_ac97_dai[] = {
 	[S3C_AC97_DAI_PCM] = {
 		.name =	"samsung-ac97",
@@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = {
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_8000_48000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.probe = s3c_ac97_dai_probe,
 		.ops = &s3c_ac97_dai_ops,
 	},
 	[S3C_AC97_DAI_MIC] = {
@@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = {
 			.channels_max = 1,
 			.rates = SNDRV_PCM_RATE_8000_48000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.probe = s3c_ac97_mic_dai_probe,
 		.ops = &s3c_ac97_mic_dai_ops,
 	},
 };
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index fe2748b494d4..ee23194f7ab8 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = {
 	.pcm_free	= dma_free_dma_buffers,
 };
 
+void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
+				struct s3c_dma_params *playback,
+				struct s3c_dma_params *capture)
+{
+	snd_soc_dai_init_dma_data(dai, playback, capture);
+}
+EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data);
+
 int samsung_asoc_dma_platform_register(struct device *dev)
 {
 	return snd_soc_register_platform(dev, &samsung_asoc_platform);
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index 0e86315a3eaf..fb09a1c5f75b 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -22,6 +22,9 @@ struct s3c_dma_params {
 	char *ch_name;
 };
 
+void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
+				struct s3c_dma_params *playback,
+				struct s3c_dma_params *capture);
 int samsung_asoc_dma_platform_register(struct device *dev);
 void samsung_asoc_dma_platform_unregister(struct device *dev);
 
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index a5cbdb4f1655..eab0050d4579 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai)
 	struct i2s_dai *i2s = to_info(dai);
 	struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai;
 
-	if (other && other->clk) /* If this is probe on secondary */
+	if (other && other->clk) { /* If this is probe on secondary */
+		samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback,
+					   NULL);
 		goto probe_exit;
+	}
 
 	i2s->addr = ioremap(i2s->base, 0x100);
 	if (i2s->addr == NULL) {
@@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai)
 	}
 	clk_prepare_enable(i2s->clk);
 
-	snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture);
+	samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture);
 
 	if (other) {
 		other->addr = i2s->addr;
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index e54256fc4b2c..6a5e4bf6ac96 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-	struct s3c_dma_params *dma_data;
 	void __iomem *regs = pcm->regs;
 	struct clk *clk;
 	int sclk_div, sync_div;
@@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	dev_dbg(pcm->dev, "Entered %s\n", __func__);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dma_data = pcm->dma_playback;
-	else
-		dma_data = pcm->dma_capture;
-
-	snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data);
-
 	/* Strictly check for sample size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = {
 	.set_fmt	= s3c_pcm_set_fmt,
 };
 
+static int s3c_pcm_dai_probe(struct snd_soc_dai *dai)
+{
+	struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture);
+
+	return 0;
+}
+
 #define S3C_PCM_RATES  SNDRV_PCM_RATE_8000_96000
 
 #define S3C_PCM_DAI_DECLARE			\
 	.symmetric_rates = 1,					\
+	.probe = s3c_pcm_dai_probe,				\
 	.ops = &s3c_pcm_dai_ops,				\
 	.playback = {						\
 		.channels_min	= 2,				\
-- 
cgit v1.2.2


From d37bdf736d9b7a198d35aaaf611e96ddc2e00ddf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 5 Dec 2013 14:14:52 +0000
Subject: ASoC: samsung: Use ASoC dmaengine code where possible

Since all Exynos platforms have been converted to dmaengine and many of
the older platforms are in the process of conversion they do not need to
use the legacy s3c-dma APIs for DMA but can instead use the standard ASoC
dmaengine helpers. This both allows them to benefit from improvements
implemented in the generic code and supports multiplatform.

This patch includes some fixes from Padma for Exynos SoCs, her testing
was on a slightly earlier version of the patch due to unrelated breakage
preventing testing.

Signed-off-by: Mark Brown <broonie@linaro.org>
Tested By: Padmavathi Venna <padma.v@samsung.com>
---
 sound/soc/samsung/Kconfig     | 13 +++++--
 sound/soc/samsung/Makefile    |  6 ++--
 sound/soc/samsung/dma.h       |  3 ++
 sound/soc/samsung/dmaengine.c | 84 +++++++++++++++++++++++++++++++++++++++++++
 sound/soc/samsung/i2s.c       |  2 ++
 5 files changed, 104 insertions(+), 4 deletions(-)
 create mode 100644 sound/soc/samsung/dmaengine.c

(limited to 'sound')

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 37459dfd168d..27930fc432dc 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,13 +1,22 @@
 config SND_SOC_SAMSUNG
 	tristate "ASoC support for Samsung"
 	depends on PLAT_SAMSUNG
-	select S3C64XX_DMA if ARCH_S3C64XX
-	select S3C24XX_DMA if ARCH_S3C24XX
+	select S3C2410_DMA if ARCH_S3C24XX
+	select S3C64XX_PL080 if ARCH_S3C64XX
+	select SND_S3C_DMA if !ARCH_S3C24XX
+	select SND_S3C_DMA_LEGACY if ARCH_S3C24XX
+	select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Samsung SoCs' Audio interfaces. You will also need to
 	  select the audio interfaces to support below.
 
+config SND_S3C_DMA
+	tristate
+
+config SND_S3C_DMA_LEGACY
+	tristate
+
 config SND_S3C24XX_I2S
 	tristate
 	select S3C2410_DMA
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 709f6059ad67..86715d8efee6 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -1,5 +1,6 @@
 # S3c24XX Platform Support
-snd-soc-s3c24xx-objs := dma.o
+snd-soc-s3c-dma-objs := dmaengine.o
+snd-soc-s3c-dma-legacy-objs := dma.o
 snd-soc-idma-objs := idma.o
 snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
@@ -9,7 +10,8 @@ snd-soc-samsung-spdif-objs := spdif.o
 snd-soc-pcm-objs := pcm.o
 snd-soc-i2s-objs := i2s.o
 
-obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c24xx.o
+obj-$(CONFIG_SND_S3C_DMA) += snd-soc-s3c-dma.o
+obj-$(CONFIG_SND_S3C_DMA_LEGACY) += snd-soc-s3c-dma-legacy.o
 obj-$(CONFIG_SND_S3C24XX_I2S) += snd-soc-s3c24xx-i2s.o
 obj-$(CONFIG_SND_SAMSUNG_AC97) += snd-soc-ac97.o
 obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index fb09a1c5f75b..225e5378014e 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -12,6 +12,8 @@
 #ifndef _S3C_AUDIO_H
 #define _S3C_AUDIO_H
 
+#include <sound/dmaengine_pcm.h>
+
 struct s3c_dma_params {
 	struct s3c2410_dma_client *client;	/* stream identifier */
 	int channel;				/* Channel ID */
@@ -20,6 +22,7 @@ struct s3c_dma_params {
 	unsigned ch;
 	struct samsung_dma_ops *ops;
 	char *ch_name;
+	struct snd_dmaengine_dai_dma_data dma_data;
 };
 
 void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c
new file mode 100644
index 000000000000..3be479d51b9b
--- /dev/null
+++ b/sound/soc/samsung/dmaengine.c
@@ -0,0 +1,84 @@
+/*
+ * dmaengine.c - Samsung dmaengine wrapper
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2013 Linaro
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/amba/pl08x.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "dma.h"
+
+#ifdef CONFIG_ARCH_S3C64XX
+#define filter_fn pl08x_filter_id
+#else
+#define filter_fn NULL
+#endif
+
+static const struct snd_dmaengine_pcm_config samsung_dmaengine_pcm_config = {
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+	.compat_filter_fn = filter_fn,
+};
+
+void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
+				struct s3c_dma_params *playback,
+				struct s3c_dma_params *capture)
+{
+	struct snd_dmaengine_dai_dma_data *playback_data = NULL;
+	struct snd_dmaengine_dai_dma_data *capture_data = NULL;
+
+	if (playback) {
+		playback_data = &playback->dma_data;
+		playback_data->filter_data = (void *)playback->channel;
+		playback_data->chan_name = playback->ch_name;
+		playback_data->addr = playback->dma_addr;
+		playback_data->addr_width = playback->dma_size;
+	}
+	if (capture) {
+		capture_data = &capture->dma_data;
+		capture_data->filter_data = (void *)capture->channel;
+		capture_data->chan_name = capture->ch_name;
+		capture_data->addr = capture->dma_addr;
+		capture_data->addr_width = capture->dma_size;
+	}
+
+	snd_soc_dai_init_dma_data(dai, playback_data, capture_data);
+}
+EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data);
+
+int samsung_asoc_dma_platform_register(struct device *dev)
+{
+	return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config,
+					  SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME |
+					  SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
+					  SND_DMAENGINE_PCM_FLAG_COMPAT);
+}
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register);
+
+void samsung_asoc_dma_platform_unregister(struct device *dev)
+{
+	return snd_dmaengine_pcm_unregister(dev);
+}
+EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister);
+
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_DESCRIPTION("Samsung dmaengine ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index eab0050d4579..92f64363427d 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -702,6 +702,8 @@ static int i2s_hw_params(struct snd_pcm_substream *substream,
 	}
 	writel(mod, i2s->addr + I2SMOD);
 
+	samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture);
+
 	i2s->frmclk = params_rate(params);
 
 	return 0;
-- 
cgit v1.2.2


From 753834cb593da03b4efc468bb8cb76dbc0743b31 Mon Sep 17 00:00:00 2001
From: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Date: Sat, 14 Dec 2013 13:29:11 +0800
Subject: ASoC: tegra20-ac97: add missing clk_disable_unprepare() on error path

Add the missing clk_disable_unprepare() before return from
tegra20_ac97_platform_probe() in the error handling case.

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/tegra/tegra20_ac97.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index ae27bcd586d2..088518d7694a 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -404,7 +404,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 	ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops);
 	if (ret) {
 		dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret);
-		goto err_asoc_utils_fini;
+		goto err_clk_disable_unprepare;
 	}
 
 	ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component,
@@ -412,7 +412,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev, "Could not register DAI: %d\n", ret);
 		ret = -ENOMEM;
-		goto err_asoc_utils_fini;
+		goto err_clk_disable_unprepare;
 	}
 
 	ret = tegra_pcm_platform_register(&pdev->dev);
@@ -428,6 +428,8 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
 
 err_unregister_component:
 	snd_soc_unregister_component(&pdev->dev);
+err_clk_disable_unprepare:
+	clk_disable_unprepare(ac97->clk_ac97);
 err_asoc_utils_fini:
 	tegra_asoc_utils_fini(&ac97->util_data);
 err_clk_put:
-- 
cgit v1.2.2


From 527659766778bcb0a2b739867ce4d95b4c57c7d7 Mon Sep 17 00:00:00 2001
From: Nenghua Cao <nhcao@marvell.com>
Date: Fri, 13 Dec 2013 20:13:49 +0800
Subject: ASoC: dapm: update DPCM runtime when mixer/mux changes

    DPCM can dynamically alter the FE to BE PCM links at runtime based
on mixer/mux setting updates. Add soc_dpcm_runtime_update() calling in
get/put function for mixer/mux to support this feature.

Signed-off-by: Nenghua Cao <nhcao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-dapm.c | 28 ++++++++++++++++++++++++----
 1 file changed, 24 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dcade130157f..67e63ab1f11e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2868,6 +2868,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	int connect, change;
 	struct snd_soc_dapm_update update;
+	int ret = 0;
 
 	if (snd_soc_volsw_is_stereo(mc))
 		dev_warn(codec->dapm.dev,
@@ -2901,12 +2902,16 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 			card->update = &update;
 		}
 
-		soc_dapm_mixer_update_power(card, kcontrol, connect);
+		ret = soc_dapm_mixer_update_power(card, kcontrol, connect);
 
 		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
+
+	if (ret > 0)
+		soc_dpcm_runtime_update(card);
+
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
@@ -2955,6 +2960,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	unsigned int val, mux, change;
 	unsigned int mask;
 	struct snd_soc_dapm_update update;
+	int ret = 0;
 
 	if (ucontrol->value.enumerated.item[0] > e->max - 1)
 		return -EINVAL;
@@ -2978,12 +2984,16 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 		update.val = val;
 		card->update = &update;
 
-		soc_dapm_mux_update_power(card, kcontrol, mux, e);
+		ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
 
 		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
+
+	if (ret > 0)
+		soc_dpcm_runtime_update(card);
+
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
@@ -3019,6 +3029,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e =
 		(struct soc_enum *)kcontrol->private_value;
 	int change;
+	int ret = 0;
 
 	if (ucontrol->value.enumerated.item[0] >= e->max)
 		return -EINVAL;
@@ -3028,9 +3039,13 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
 	value = ucontrol->value.enumerated.item[0];
 	change = dapm_kcontrol_set_value(kcontrol, value);
 	if (change)
-		soc_dapm_mux_update_power(card, kcontrol, value, e);
+		ret = soc_dapm_mux_update_power(card, kcontrol, value, e);
 
 	mutex_unlock(&card->dapm_mutex);
+
+	if (ret > 0)
+		soc_dpcm_runtime_update(card);
+
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
@@ -3097,6 +3112,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 	unsigned int val, mux, change;
 	unsigned int mask;
 	struct snd_soc_dapm_update update;
+	int ret = 0;
 
 	if (ucontrol->value.enumerated.item[0] > e->max - 1)
 		return -EINVAL;
@@ -3120,12 +3136,16 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 		update.val = val;
 		card->update = &update;
 
-		soc_dapm_mux_update_power(card, kcontrol, mux, e);
+		ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
 
 		card->update = NULL;
 	}
 
 	mutex_unlock(&card->dapm_mutex);
+
+	if (ret > 0)
+		soc_dpcm_runtime_update(card);
+
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
-- 
cgit v1.2.2


From 252e91ff1094eefacd25b401c3b77e549803cae6 Mon Sep 17 00:00:00 2001
From: Shawn Guo <shawn.guo@linaro.org>
Date: Fri, 13 Dec 2013 14:43:02 +0800
Subject: ASoC: sgtl5000: read chip revision for once

Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe()
reads it out from register, so that we can use it in
sgtl5000_enable_regulators() with no need to read register again.

Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/sgtl5000.c | 16 +++-------------
 1 file changed, 3 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 1f4093f3f3a1..bd291d2b327d 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -115,6 +115,7 @@ struct sgtl5000_priv {
 	struct ldo_regulator *ldo;
 	struct regmap *regmap;
 	struct clk *mclk;
+	int revision;
 };
 
 /*
@@ -1300,9 +1301,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec)
 
 static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 {
-	int reg;
 	int ret;
-	int rev;
 	int i;
 	int external_vddd = 0;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
@@ -1332,14 +1331,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 	 * workaround for revision 0x11 and later,
 	 * roll back to use internal LDO
 	 */
-
-	ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg);
-	if (ret)
-		goto err_regulator_disable;
-
-	rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
-
-	if (external_vddd && rev >= 0x11) {
+	if (external_vddd && sgtl5000->revision >= 0x11) {
 		/* disable all regulator first */
 		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
@@ -1362,9 +1354,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 
 	return 0;
 
-err_regulator_disable:
-	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
 err_regulator_free:
 	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
 				sgtl5000->supplies);
@@ -1566,6 +1555,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 
 	rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
 	dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev);
+	sgtl5000->revision = rev;
 
 	i2c_set_clientdata(client, sgtl5000);
 
-- 
cgit v1.2.2


From 7fd7a48bd584a66cc4b0e3b92bb75b061578e19e Mon Sep 17 00:00:00 2001
From: Fengguang Wu <fengguang.wu@intel.com>
Date: Mon, 16 Dec 2013 13:07:23 +0100
Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of
 devm_request_and_ioremap()

devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:

	sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195

Generated by: coccinelle/api/devm_ioremap_resource.cocci

Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/adi/axi-i2s.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 0822c7713669..7f91a86dd734 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -192,9 +192,9 @@ static int axi_i2s_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, i2s);
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_request_and_ioremap(&pdev->dev, res);
-	if (!base)
-		return -EBUSY;
+	base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(base))
+		return PTR_ERR(base);
 
 	i2s->regmap = devm_regmap_init_mmio(&pdev->dev, base,
 		&axi_i2s_regmap_config);
-- 
cgit v1.2.2


From bbe580302d33cff282129e26c44f9c3450d6a086 Mon Sep 17 00:00:00 2001
From: Fengguang Wu <fengguang.wu@intel.com>
Date: Mon, 16 Dec 2013 13:07:24 +0100
Subject: ASoC: axi-spdif: Use devm_ioremap_resource() instead of
 devm_request_and_ioremap()

devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:

	sound/soc/adi/axi-spdif.c:194:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 194

Generated by: coccinelle/api/devm_ioremap_resource.cocci

Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/adi/axi-spdif.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c
index d5408d23b438..8db7a9920695 100644
--- a/sound/soc/adi/axi-spdif.c
+++ b/sound/soc/adi/axi-spdif.c
@@ -191,9 +191,9 @@ static int axi_spdif_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, spdif);
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	base = devm_request_and_ioremap(&pdev->dev, res);
-	if (!base)
-		return -EBUSY;
+	base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(base))
+		return PTR_ERR(base);
 
 	spdif->regmap = devm_regmap_init_mmio(&pdev->dev, base,
 					    &axi_spdif_regmap_config);
-- 
cgit v1.2.2


From 2b67f8ba41ac7acf01c8d5c742c713ead8b589cd Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 17 Dec 2013 15:16:40 +0800
Subject: ASoC: dmaengine: Add config parameter NULL check.

Because the "ASoC: dmaengine-pcm: Provide default config" has provided
us one defualt config of DMA. When using this, the config parameter of
devm_snd_dmaengine_pcm_register() will be NULL, so here we need to have
a check before using it.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 7483922f6ee3..2a6c569d991f 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -299,7 +299,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	    !dev->of_node)
 		return 0;
 
-	if (config->dma_dev) {
+	if (config && config->dma_dev) {
 		/*
 		 * If this warning is seen, it probably means that your Linux
 		 * device structure does not match your HW device structure.
@@ -317,7 +317,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 			name = "rx-tx";
 		else
 			name = dmaengine_pcm_dma_channel_names[i];
-		if (config->chan_names[i])
+		if (config && config->chan_names[i])
 			name = config->chan_names[i];
 		chan = dma_request_slave_channel_reason(dev, name);
 		if (IS_ERR(chan)) {
-- 
cgit v1.2.2


From 3c43c69537daa044c61965fad24e24ad392c4166 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 12 Dec 2013 00:49:22 +0000
Subject: ASoC: arizona: Use async writes

Where possible write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.

Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c | 163 +++++++++++++++++++++++++--------------------
 1 file changed, 89 insertions(+), 74 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index eb9f5d4d8928..6bfd8031c0c9 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -93,7 +93,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		if (!priv->spk_ena && manual_ena) {
-			snd_soc_write(codec, 0x4f5, 0x25a);
+			regmap_write_async(arizona->regmap, 0x4f5, 0x25a);
 			priv->spk_ena_pending = true;
 		}
 		break;
@@ -105,12 +105,13 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 			return -EBUSY;
 		}
 
-		snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
-				    1 << w->shift, 1 << w->shift);
+		regmap_update_bits_async(arizona->regmap,
+					 ARIZONA_OUTPUT_ENABLES_1,
+					 1 << w->shift, 1 << w->shift);
 
 		if (priv->spk_ena_pending) {
 			msleep(75);
-			snd_soc_write(codec, 0x4f5, 0xda);
+			regmap_write_async(arizona->regmap, 0x4f5, 0xda);
 			priv->spk_ena_pending = false;
 			priv->spk_ena++;
 		}
@@ -119,16 +120,19 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 		if (manual_ena) {
 			priv->spk_ena--;
 			if (!priv->spk_ena)
-				snd_soc_write(codec, 0x4f5, 0x25a);
+				regmap_write_async(arizona->regmap,
+						   0x4f5, 0x25a);
 		}
 
-		snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
-				    1 << w->shift, 0);
+		regmap_update_bits_async(arizona->regmap,
+					 ARIZONA_OUTPUT_ENABLES_1,
+					 1 << w->shift, 0);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
 		if (manual_ena) {
 			if (!priv->spk_ena)
-				snd_soc_write(codec, 0x4f5, 0x0da);
+				regmap_write_async(arizona->regmap,
+						   0x4f5, 0x0da);
 		}
 		break;
 	}
@@ -687,6 +691,7 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w,
 		   int event)
 {
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
+	struct arizona *arizona = priv->arizona;
 	unsigned int mask = 1 << w->shift;
 	unsigned int val;
 
@@ -709,7 +714,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w,
 	if (priv->arizona->hpdet_magic)
 		val = 0;
 
-	snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val);
+	regmap_update_bits_async(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1,
+				 mask, val);
 
 	return arizona_out_ev(w, kcontrol, event);
 }
@@ -864,6 +870,8 @@ EXPORT_SYMBOL_GPL(arizona_set_sysclk);
 static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
 	int lrclk, bclk, mode, base;
 
 	base = dai->driver->base;
@@ -920,17 +928,19 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
-			    ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR,
-			    bclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL,
-			    ARIZONA_AIF1TX_LRCLK_INV |
-			    ARIZONA_AIF1TX_LRCLK_MSTR, lrclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL,
-			    ARIZONA_AIF1RX_LRCLK_INV |
-			    ARIZONA_AIF1RX_LRCLK_MSTR, lrclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT,
-			    ARIZONA_AIF1_FMT_MASK, mode);
+	regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_BCLK_CTRL,
+				 ARIZONA_AIF1_BCLK_INV |
+				 ARIZONA_AIF1_BCLK_MSTR,
+				 bclk);
+	regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_TX_PIN_CTRL,
+				 ARIZONA_AIF1TX_LRCLK_INV |
+				 ARIZONA_AIF1TX_LRCLK_MSTR, lrclk);
+	regmap_update_bits_async(arizona->regmap,
+				 base + ARIZONA_AIF_RX_PIN_CTRL,
+				 ARIZONA_AIF1RX_LRCLK_INV |
+				 ARIZONA_AIF1RX_LRCLK_MSTR, lrclk);
+	regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FORMAT,
+			   ARIZONA_AIF1_FMT_MASK, mode);
 
 	return 0;
 }
@@ -1182,18 +1192,22 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	if (ret != 0)
 		return ret;
 
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
-			    ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
-			    ARIZONA_AIF1TX_BCPF_MASK, lrclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
-			    ARIZONA_AIF1RX_BCPF_MASK, lrclk);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
-			    ARIZONA_AIF1TX_WL_MASK |
-			    ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
-	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
-			    ARIZONA_AIF1RX_WL_MASK |
-			    ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+	regmap_update_bits_async(arizona->regmap,
+				 base + ARIZONA_AIF_BCLK_CTRL,
+				 ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
+	regmap_update_bits_async(arizona->regmap,
+				 base + ARIZONA_AIF_TX_BCLK_RATE,
+				 ARIZONA_AIF1TX_BCPF_MASK, lrclk);
+	regmap_update_bits_async(arizona->regmap,
+				 base + ARIZONA_AIF_RX_BCLK_RATE,
+				 ARIZONA_AIF1RX_BCPF_MASK, lrclk);
+	regmap_update_bits_async(arizona->regmap,
+				 base + ARIZONA_AIF_FRAME_CTRL_1,
+				 ARIZONA_AIF1TX_WL_MASK |
+				 ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
+	regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2,
+			   ARIZONA_AIF1RX_WL_MASK |
+			   ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
 
 	return 0;
 }
@@ -1446,31 +1460,31 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
 			      struct arizona_fll_cfg *cfg, int source,
 			      bool sync)
 {
-	regmap_update_bits(arizona->regmap, base + 3,
-			   ARIZONA_FLL1_THETA_MASK, cfg->theta);
-	regmap_update_bits(arizona->regmap, base + 4,
-			   ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda);
-	regmap_update_bits(arizona->regmap, base + 5,
-			   ARIZONA_FLL1_FRATIO_MASK,
-			   cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT);
-	regmap_update_bits(arizona->regmap, base + 6,
-			   ARIZONA_FLL1_CLK_REF_DIV_MASK |
-			   ARIZONA_FLL1_CLK_REF_SRC_MASK,
-			   cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
-			   source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
+	regmap_update_bits_async(arizona->regmap, base + 3,
+				 ARIZONA_FLL1_THETA_MASK, cfg->theta);
+	regmap_update_bits_async(arizona->regmap, base + 4,
+				 ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda);
+	regmap_update_bits_async(arizona->regmap, base + 5,
+				 ARIZONA_FLL1_FRATIO_MASK,
+				 cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT);
+	regmap_update_bits_async(arizona->regmap, base + 6,
+				 ARIZONA_FLL1_CLK_REF_DIV_MASK |
+				 ARIZONA_FLL1_CLK_REF_SRC_MASK,
+				 cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
+				 source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
 
 	if (sync)
-		regmap_update_bits(arizona->regmap, base + 0x7,
-				   ARIZONA_FLL1_GAIN_MASK,
-				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+		regmap_update_bits_async(arizona->regmap, base + 0x7,
+					 ARIZONA_FLL1_GAIN_MASK,
+					 cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
 	else
-		regmap_update_bits(arizona->regmap, base + 0x9,
-				   ARIZONA_FLL1_GAIN_MASK,
-				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+		regmap_update_bits_async(arizona->regmap, base + 0x9,
+					 ARIZONA_FLL1_GAIN_MASK,
+					 cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
 
-	regmap_update_bits(arizona->regmap, base + 2,
-			   ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
-			   ARIZONA_FLL1_CTRL_UPD | cfg->n);
+	regmap_update_bits_async(arizona->regmap, base + 2,
+				 ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
+				 ARIZONA_FLL1_CTRL_UPD | cfg->n);
 }
 
 static bool arizona_is_enabled_fll(struct arizona_fll *fll)
@@ -1503,9 +1517,9 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 	 */
 	if (fll->ref_src >= 0 && fll->ref_freq &&
 	    fll->ref_src != fll->sync_src) {
-		regmap_update_bits(arizona->regmap, fll->base + 5,
-				   ARIZONA_FLL1_OUTDIV_MASK,
-				   ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+		regmap_update_bits_async(arizona->regmap, fll->base + 5,
+					 ARIZONA_FLL1_OUTDIV_MASK,
+					 ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
 
 		arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
 				  false);
@@ -1515,15 +1529,15 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 			use_sync = true;
 		}
 	} else if (fll->sync_src >= 0) {
-		regmap_update_bits(arizona->regmap, fll->base + 5,
-				   ARIZONA_FLL1_OUTDIV_MASK,
-				   sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+		regmap_update_bits_async(arizona->regmap, fll->base + 5,
+					 ARIZONA_FLL1_OUTDIV_MASK,
+					 sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
 
 		arizona_apply_fll(arizona, fll->base, sync,
 				  fll->sync_src, false);
 
-		regmap_update_bits(arizona->regmap, fll->base + 0x11,
-				   ARIZONA_FLL1_SYNC_ENA, 0);
+		regmap_update_bits_async(arizona->regmap, fll->base + 0x11,
+					 ARIZONA_FLL1_SYNC_ENA, 0);
 	} else {
 		arizona_fll_err(fll, "No clocks provided\n");
 		return;
@@ -1534,11 +1548,12 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 	 * sync source.
 	 */
 	if (use_sync && fll->sync_freq > 100000)
-		regmap_update_bits(arizona->regmap, fll->base + 0x17,
-				   ARIZONA_FLL1_SYNC_BW, 0);
+		regmap_update_bits_async(arizona->regmap, fll->base + 0x17,
+					 ARIZONA_FLL1_SYNC_BW, 0);
 	else
-		regmap_update_bits(arizona->regmap, fll->base + 0x17,
-				   ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW);
+		regmap_update_bits_async(arizona->regmap, fll->base + 0x17,
+					 ARIZONA_FLL1_SYNC_BW,
+					 ARIZONA_FLL1_SYNC_BW);
 
 	if (!arizona_is_enabled_fll(fll))
 		pm_runtime_get(arizona->dev);
@@ -1546,14 +1561,14 @@ static void arizona_enable_fll(struct arizona_fll *fll,
 	/* Clear any pending completions */
 	try_wait_for_completion(&fll->ok);
 
-	regmap_update_bits(arizona->regmap, fll->base + 1,
-			   ARIZONA_FLL1_FREERUN, 0);
-	regmap_update_bits(arizona->regmap, fll->base + 1,
-			   ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
+	regmap_update_bits_async(arizona->regmap, fll->base + 1,
+				 ARIZONA_FLL1_FREERUN, 0);
+	regmap_update_bits_async(arizona->regmap, fll->base + 1,
+				 ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
 	if (use_sync)
-		regmap_update_bits(arizona->regmap, fll->base + 0x11,
-				   ARIZONA_FLL1_SYNC_ENA,
-				   ARIZONA_FLL1_SYNC_ENA);
+		regmap_update_bits_async(arizona->regmap, fll->base + 0x11,
+					 ARIZONA_FLL1_SYNC_ENA,
+					 ARIZONA_FLL1_SYNC_ENA);
 
 	ret = wait_for_completion_timeout(&fll->ok,
 					  msecs_to_jiffies(250));
@@ -1566,8 +1581,8 @@ static void arizona_disable_fll(struct arizona_fll *fll)
 	struct arizona *arizona = fll->arizona;
 	bool change;
 
-	regmap_update_bits(arizona->regmap, fll->base + 1,
-			   ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN);
+	regmap_update_bits_async(arizona->regmap, fll->base + 1,
+				 ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN);
 	regmap_update_bits_check(arizona->regmap, fll->base + 1,
 				 ARIZONA_FLL1_ENA, 0, &change);
 	regmap_update_bits(arizona->regmap, fll->base + 0x11,
-- 
cgit v1.2.2


From bd4893492bfc6081da55e445d200c1a832770a06 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 12 Dec 2013 00:49:50 +0000
Subject: ASoC: wm5102: Use async writes

When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.

Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5102.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index a08e8bf6d07c..ce9c8e14d4bd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -601,8 +601,8 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w,
 	case SND_SOC_DAPM_POST_PMU:
 		if (patch)
 			for (i = 0; i < patch_size; i++)
-				regmap_write(regmap, patch[i].reg,
-					     patch[i].def);
+				regmap_write_async(regmap, patch[i].reg,
+						   patch[i].def);
 		break;
 
 	default:
-- 
cgit v1.2.2


From 959e4083cd9c43fb3e818984926f9c590ee0aa2b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 12 Dec 2013 00:49:56 +0000
Subject: ASoC: wm5110: Use async writes

When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.

Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5110.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 0e63d8ce533f..ebcbe7831e57 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -105,8 +105,8 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
 	case SND_SOC_DAPM_POST_PMU:
 		if (patch)
 			for (i = 0; i < patch_size; i++)
-				regmap_write(regmap, patch[i].reg,
-					     patch[i].def);
+				regmap_write_async(regmap, patch[i].reg,
+						   patch[i].def);
 		break;
 
 	default:
-- 
cgit v1.2.2


From 1f4fe272f068813377845a959cab5ce786a155bf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Thu, 12 Dec 2013 00:50:04 +0000
Subject: ASoC: wm8997: Use async writes

When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.

Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8997.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 1392bb3c9254..555115ee2159 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -103,8 +103,8 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w,
 	case SND_SOC_DAPM_POST_PMU:
 		if (patch)
 			for (i = 0; i < patch_size; i++)
-				regmap_write(regmap, patch[i].reg,
-					     patch[i].def);
+				regmap_write_async(regmap, patch[i].reg,
+						   patch[i].def);
 		break;
 	default:
 		break;
-- 
cgit v1.2.2


From 48b752ac2f80f483a3059ae109f9de02dcc054dd Mon Sep 17 00:00:00 2001
From: Qiao Zhou <zhouqiao@marvell.com>
Date: Tue, 17 Dec 2013 16:22:24 +0800
Subject: ASoC: mmp-pcm: config pcm slave via generic dmaengine

use snd_dmaengine_pcm_prepare_slave_config to set slave config,
and remove the max_burst_size = 4 hard code.

select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm.

Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/pxa/Kconfig   |  2 +-
 sound/soc/pxa/mmp-pcm.c | 18 +++---------------
 2 files changed, 4 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 4db74a083db1..6473052b6899 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -11,7 +11,7 @@ config SND_PXA2XX_SOC
 config SND_MMP_SOC
 	bool "Soc Audio for Marvell MMP chips"
 	depends on ARCH_MMP
-	select SND_DMAENGINE_PCM
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select SND_ARM
 	help
 	  Say Y if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 7929e19b0ef5..682ee52942bf 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -67,27 +67,15 @@ static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
 {
 	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_dmaengine_dai_dma_data *dma_params;
 	struct dma_slave_config slave_config;
 	int ret;
 
-	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-	if (!dma_params)
-		return 0;
-
-	ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
+	ret =
+	    snd_dmaengine_pcm_prepare_slave_config(substream, params,
+						   &slave_config);
 	if (ret)
 		return ret;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		slave_config.dst_addr     = dma_params->addr;
-		slave_config.dst_maxburst = 4;
-	} else {
-		slave_config.src_addr	  = dma_params->addr;
-		slave_config.src_maxburst = 4;
-	}
-
 	ret = dmaengine_slave_config(chan, &slave_config);
 	if (ret)
 		return ret;
-- 
cgit v1.2.2


From 58e354be337dff43dbb66c4564bb9479354cc5dd Mon Sep 17 00:00:00 2001
From: Lucas Stach <dev@lynxeye.de>
Date: Tue, 17 Dec 2013 11:52:43 +0100
Subject: ASoC: tegra: Tweak matching of AC97 components

Matching works completely based on the cpu of_node.

Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/tegra/tegra_wm9712.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c
index 5e119630b0e0..45b57892b6a5 100644
--- a/sound/soc/tegra/tegra_wm9712.c
+++ b/sound/soc/tegra/tegra_wm9712.c
@@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link tegra_wm9712_dai = {
 	.name = "AC97 HiFi",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai_name = "tegra20-ac97",
 	.codec_dai_name = "wm9712-hifi",
 	.codec_name = "wm9712-codec",
 	.init = tegra_wm9712_init,
-- 
cgit v1.2.2


From 5095f55d7cc327026daaa3fa583aa4c1388ca556 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Wed, 18 Dec 2013 09:25:48 +0000
Subject: ASoC: wm_adsp: Remove duplicate info message for DSP RAM ready

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm_adsp.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 6b1c01c66e64..8f720ded27c4 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1488,7 +1488,6 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 	}
 
 	adsp_dbg(dsp, "RAM ready after %d polls\n", count);
-	adsp_info(dsp, "RAM ready after %d polls\n", count);
 
 	return 0;
 }
-- 
cgit v1.2.2


From da4a7a3926d09c13ae052ede67feb7285e01e3f5 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 18 Dec 2013 10:46:04 +0100
Subject: ALSA: hda - Explicitly keep codec powered up in hdmi_present_sense

This should help us avoid the following mutex deadlock:

[] mutex_lock+0x2a/0x50
[] hdmi_present_sense+0x53/0x3a0 [snd_hda_codec_hdmi]
[] generic_hdmi_resume+0x5a/0x70 [snd_hda_codec_hdmi]
[] hda_call_codec_resume+0xec/0x1d0 [snd_hda_codec]
[] snd_hda_power_save+0x1e4/0x280 [snd_hda_codec]
[] codec_exec_verb+0x5f/0x290 [snd_hda_codec]
[] snd_hda_codec_read+0x5b/0x90 [snd_hda_codec]
[] snd_hdmi_get_eld_size+0x1e/0x20 [snd_hda_codec_hdmi]
[] snd_hdmi_get_eld+0x2c/0xd0 [snd_hda_codec_hdmi]
[] hdmi_present_sense+0x9a/0x3a0 [snd_hda_codec_hdmi]
[] hdmi_repoll_eld+0x34/0x50 [snd_hda_codec_hdmi]

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index f5060fc7c303..977db17db26c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1496,11 +1496,14 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	 * specification worked this way. Hence, we just ignore the data in
 	 * the unsolicited response to avoid custom WARs.
 	 */
-	int present = snd_hda_pin_sense(codec, pin_nid);
+	int present;
 	bool update_eld = false;
 	bool eld_changed = false;
 	bool ret;
 
+	snd_hda_power_up(codec);
+	present = snd_hda_pin_sense(codec, pin_nid);
+
 	mutex_lock(&per_pin->lock);
 	pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE);
 	if (pin_eld->monitor_present)
@@ -1573,6 +1576,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 		jack->block_report = !ret;
 
 	mutex_unlock(&per_pin->lock);
+	snd_hda_power_down(codec);
 	return ret;
 }
 
-- 
cgit v1.2.2


From dd407a3243234c6a17ba624d698e6824067003c9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 18 Dec 2013 13:50:10 +0000
Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language

The kernel as a number of cases of gendered language. The majority of these
refer to objects that don't have gender in English, and so I've replaced
them with "it" and "its". Some refer to people (developers or users), and
I've replaced these with the singular "they" variant. Some are simply
typos that I've fixed up.

I've left cases where gendered language was used to refer to specific
individuals, was a quote or is part of license text.

Signed-off-by: Matthew Garrett <matthew.garrett@nebula.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_dma.c  |  2 +-
 sound/soc/mxs/mxs-saif.c | 10 +++++-----
 2 files changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index fb9bb9eb5ca3..d570f8c81dc6 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm)
 }
 
 /**
- * find_ssi_node -- returns the SSI node that points to his DMA channel node
+ * find_ssi_node -- returns the SSI node that points to its DMA channel node
  *
  * Although this DMA driver attempts to operate independently of the other
  * devices, it still needs to determine some information about the SSI device
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 54e622acac33..d6cb9a51dd57 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2];
  * This also means that both SAIFs must operate at the same sample rate.
  *
  * We abstract this as each saif has a master, the master could be
- * himself or other saifs. In the generic saif driver, saif does not need
- * to know the different clkmux. Saif only needs to know who is his master
- * and operating his master to generate the proper clock rate for him.
+ * itself or other saifs. In the generic saif driver, saif does not need
+ * to know the different clkmux. Saif only needs to know who is its master
+ * and operating its master to generate the proper clock rate for it.
  * The master id is provided in mach-specific layer according to different
  * clkmux setting.
  */
@@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
  * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK
  * is provided by other SAIF, we provide a interface here to get its master
  * from its master_id.
- * Note that the master could be himself.
+ * Note that the master could be itself.
  */
 static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif)
 {
@@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
 		}
 
 		/*
-		 * If the saif's master is not himself, we also need to enable
+		 * If the saif's master is not itself, we also need to enable
 		 * itself clk for its internal basic logic to work.
 		 */
 		if (saif != master_saif) {
-- 
cgit v1.2.2


From 4355082149429d1f87b6fbfc3ebc6305a5372ce2 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 17 Dec 2013 11:24:38 +0800
Subject: ASoC: Add SAI SoC Digital Audio Interface driver.

This adds Freescale SAI ASoC Audio support.
This implementation is only compatible with device tree definition.
Features:
o Supports playback/capture
o Supports 16/20/24 bit PCM
o Supports 8k - 96k sample rates
o Supports master and slave mode.

Signed-off-by: Alison Wang <b18965@freescale.com>
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig   |   4 +
 sound/soc/fsl/Makefile  |   4 +-
 sound/soc/fsl/fsl_sai.c | 492 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_sai.h | 114 +++++++++++
 4 files changed, 613 insertions(+), 1 deletion(-)
 create mode 100644 sound/soc/fsl/fsl_sai.c
 create mode 100644 sound/soc/fsl/fsl_sai.h

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index b7ab71f2ccc1..ac4fe4ea15a9 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,3 +1,7 @@
+config SND_SOC_FSL_SAI
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
 config SND_SOC_FSL_SSI
 	tristate
 
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8db705b0fdf9..aaccbee17006 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,11 +10,13 @@ obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
 snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
-# Freescale PowerPC SSI/DMA Platform Support
+# Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
new file mode 100644
index 000000000000..50a797e65781
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.c
@@ -0,0 +1,492 @@
+/*
+ * Freescale ALSA SoC Digital Audio Interface (SAI) driver.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software, you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation, either version 2 of the License, or(at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_address.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_sai.h"
+
+static inline u32 sai_readl(struct fsl_sai *sai,
+		const void __iomem *addr)
+{
+	u32 val;
+
+	val = __raw_readl(addr);
+
+	if (likely(sai->big_endian_regs))
+		val = be32_to_cpu(val);
+	else
+		val = le32_to_cpu(val);
+	rmb();
+
+	return val;
+}
+
+static inline void sai_writel(struct fsl_sai *sai,
+		u32 val, void __iomem *addr)
+{
+	wmb();
+	if (likely(sai->big_endian_regs))
+		val = cpu_to_be32(val);
+	else
+		val = cpu_to_le32(val);
+
+	__raw_writel(val, addr);
+}
+
+static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
+		int clk_id, unsigned int freq, int fsl_dir)
+{
+	u32 val_cr2, reg_cr2;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (fsl_dir == FSL_FMT_TRANSMITTER)
+		reg_cr2 = FSL_SAI_TCR2;
+	else
+		reg_cr2 = FSL_SAI_RCR2;
+
+	val_cr2 = sai_readl(sai, sai->base + reg_cr2);
+	switch (clk_id) {
+	case FSL_SAI_CLK_BUS:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_BUS;
+		break;
+	case FSL_SAI_CLK_MAST1:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1;
+		break;
+	case FSL_SAI_CLK_MAST2:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2;
+		break;
+	case FSL_SAI_CLK_MAST3:
+		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3;
+		break;
+	default:
+		return -EINVAL;
+	}
+	sai_writel(sai, val_cr2, sai->base + reg_cr2);
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return 0;
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
+	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
+	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
+
+	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+					FSL_FMT_TRANSMITTER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set SAI's transmitter sysclk: %d\n",
+				ret);
+		return ret;
+	}
+
+	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
+					FSL_FMT_RECEIVER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set SAI's receiver sysclk: %d\n",
+				ret);
+		return ret;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
+				unsigned int fmt, int fsl_dir)
+{
+	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (fsl_dir == FSL_FMT_TRANSMITTER) {
+		reg_cr2 = FSL_SAI_TCR2;
+		reg_cr3 = FSL_SAI_TCR3;
+		reg_cr4 = FSL_SAI_TCR4;
+	} else {
+		reg_cr2 = FSL_SAI_RCR2;
+		reg_cr3 = FSL_SAI_RCR3;
+		reg_cr4 = FSL_SAI_RCR4;
+	}
+
+	val_cr2 = sai_readl(sai, sai->base + reg_cr2);
+	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+
+	if (sai->big_endian_data)
+		val_cr4 |= FSL_SAI_CR4_MF;
+	else
+		val_cr4 &= ~FSL_SAI_CR4_MF;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		val_cr4 |= FSL_SAI_CR4_FSE;
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		val_cr2 &= ~FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		val_cr4 &= ~FSL_SAI_CR4_FSP;
+		val_cr2 &= ~FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		val_cr4 |= FSL_SAI_CR4_FSP;
+		val_cr2 |= FSL_SAI_CR2_BCP;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		val_cr4 &= ~FSL_SAI_CR4_FSP;
+		val_cr2 |= FSL_SAI_CR2_BCP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
+		val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val_cr3 |= FSL_SAI_CR3_TRCE;
+
+	if (fsl_dir == FSL_FMT_RECEIVER)
+		val_cr2 |= FSL_SAI_CR2_SYNC;
+
+	sai_writel(sai, val_cr2, sai->base + reg_cr2);
+	sai_writel(sai, val_cr3, sai->base + reg_cr3);
+	sai_writel(sai, val_cr4, sai->base + reg_cr4);
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set SAI's transmitter format: %d\n",
+				ret);
+		return ret;
+	}
+
+	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
+	if (ret) {
+		dev_err(cpu_dai->dev,
+				"Cannot set SAI's receiver format: %d\n",
+				ret);
+		return ret;
+	}
+
+	clk_disable_unprepare(sai->clk);
+
+	return 0;
+}
+
+static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *cpu_dai)
+{
+	u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width;
+	unsigned int channels = params_channels(params);
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		reg_cr4 = FSL_SAI_TCR4;
+		reg_cr5 = FSL_SAI_TCR5;
+		reg_mr = FSL_SAI_TMR;
+	} else {
+		reg_cr4 = FSL_SAI_RCR4;
+		reg_cr5 = FSL_SAI_RCR5;
+		reg_mr = FSL_SAI_RMR;
+	}
+
+	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+	val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
+	val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
+
+	val_cr5 = sai_readl(sai, sai->base + reg_cr5);
+	val_cr5 &= ~FSL_SAI_CR5_WNW_MASK;
+	val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
+	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_width = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		word_width = 20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		word_width = 24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
+	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
+
+	if (sai->big_endian_data)
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
+	else
+		val_cr5 |= FSL_SAI_CR5_FBT(0);
+
+	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+	if (channels == 2 || channels == 1)
+		val_mr = ~0UL - ((1 << channels) - 1);
+	else
+		return -EINVAL;
+
+	sai_writel(sai, val_cr4, sai->base + reg_cr4);
+	sai_writel(sai, val_cr5, sai->base + reg_cr5);
+	sai_writel(sai, val_mr, sai->base + reg_mr);
+
+	return 0;
+}
+
+static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int tcsr, rcsr;
+
+	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
+	rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		tcsr |= FSL_SAI_CSR_FRDE;
+		rcsr &= ~FSL_SAI_CSR_FRDE;
+	} else {
+		rcsr |= FSL_SAI_CSR_FRDE;
+		tcsr &= ~FSL_SAI_CSR_FRDE;
+	}
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		tcsr |= FSL_SAI_CSR_TERE;
+		rcsr |= FSL_SAI_CSR_TERE;
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!(cpu_dai->playback_active || cpu_dai->capture_active)) {
+			tcsr &= ~FSL_SAI_CSR_TERE;
+			rcsr &= ~FSL_SAI_CSR_TERE;
+		}
+		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int fsl_sai_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	int ret;
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ret = clk_prepare_enable(sai->clk);
+
+	return ret;
+}
+
+static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	clk_disable_unprepare(sai->clk);
+}
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
+	.set_sysclk	= fsl_sai_set_dai_sysclk,
+	.set_fmt	= fsl_sai_set_dai_fmt,
+	.hw_params	= fsl_sai_hw_params,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+	.shutdown	= fsl_sai_shutdown,
+};
+
+static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+
+	cpu_dai->playback_dma_data = &sai->dma_params_tx;
+	cpu_dai->capture_dma_data = &sai->dma_params_rx;
+
+	snd_soc_dai_set_drvdata(cpu_dai, sai);
+
+	return 0;
+}
+
+static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
+{
+	cpu_dai->playback_dma_data = NULL;
+	cpu_dai->capture_dma_data = NULL;
+
+	snd_soc_dai_set_drvdata(cpu_dai, NULL);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver fsl_sai_dai = {
+	.probe = fsl_sai_dai_probe,
+	.remove = fsl_sai_dai_remove,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = FSL_SAI_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = FSL_SAI_FORMATS,
+	},
+	.ops = &fsl_sai_pcm_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_component = {
+	.name           = "fsl-sai",
+};
+
+static int fsl_sai_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct fsl_sai *sai;
+	struct resource *res;
+	struct device_node *np = pdev->dev.of_node;
+
+	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
+	if (!sai)
+		return -ENOMEM;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	sai->base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(sai->base))
+		return PTR_ERR(sai->base);
+
+	sai->clk = devm_clk_get(&pdev->dev, "sai");
+	if (IS_ERR(sai->clk)) {
+		dev_err(&pdev->dev, "Cannot get SAI's clock\n");
+		return PTR_ERR(sai->clk);
+	}
+
+	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
+	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
+	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
+	sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
+
+	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
+	platform_set_drvdata(pdev, sai);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+			&fsl_sai_dai, 1);
+	if (ret)
+		return ret;
+
+	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL,
+			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static int fsl_sai_remove(struct platform_device *pdev)
+{
+	snd_dmaengine_pcm_unregister(&pdev->dev);
+
+	return 0;
+}
+
+static const struct of_device_id fsl_sai_ids[] = {
+	{ .compatible = "fsl,vf610-sai", },
+	{ /* sentinel */ }
+};
+
+static struct platform_driver fsl_sai_driver = {
+	.probe = fsl_sai_probe,
+	.remove = fsl_sai_remove,
+
+	.driver = {
+		.name = "fsl-sai",
+		.owner = THIS_MODULE,
+		.of_match_table = fsl_sai_ids,
+	},
+};
+module_platform_driver(fsl_sai_driver);
+
+MODULE_DESCRIPTION("Freescale Soc SAI Interface");
+MODULE_AUTHOR("Xiubo Li, <Li.Xiubo@freescale.com>");
+MODULE_ALIAS("platform:fsl-sai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
new file mode 100644
index 000000000000..41bb62e69361
--- /dev/null
+++ b/sound/soc/fsl/fsl_sai.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __FSL_SAI_H
+#define __FSL_SAI_H
+
+#include <sound/dmaengine_pcm.h>
+
+#define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE |\
+			 SNDRV_PCM_FMTBIT_S24_LE)
+
+/* SAI Transmit/Recieve Control Register */
+#define FSL_SAI_TCSR		0x00
+#define FSL_SAI_RCSR		0x80
+#define FSL_SAI_CSR_TERE	BIT(31)
+#define FSL_SAI_CSR_FWF		BIT(17)
+#define FSL_SAI_CSR_FRIE	BIT(8)
+#define FSL_SAI_CSR_FRDE	BIT(0)
+
+/* SAI Transmit Data/FIFO/MASK Register */
+#define FSL_SAI_TDR		0x20
+#define FSL_SAI_TFR		0x40
+#define FSL_SAI_TMR		0x60
+
+/* SAI Recieve Data/FIFO/MASK Register */
+#define FSL_SAI_RDR		0xa0
+#define FSL_SAI_RFR		0xc0
+#define FSL_SAI_RMR		0xe0
+
+/* SAI Transmit and Recieve Configuration 1 Register */
+#define FSL_SAI_TCR1		0x04
+#define FSL_SAI_RCR1		0x84
+
+/* SAI Transmit and Recieve Configuration 2 Register */
+#define FSL_SAI_TCR2		0x08
+#define FSL_SAI_RCR2		0x88
+#define FSL_SAI_CR2_SYNC	BIT(30)
+#define FSL_SAI_CR2_MSEL_MASK	(0xff << 26)
+#define FSL_SAI_CR2_MSEL_BUS	0
+#define FSL_SAI_CR2_MSEL_MCLK1	BIT(26)
+#define FSL_SAI_CR2_MSEL_MCLK2	BIT(27)
+#define FSL_SAI_CR2_MSEL_MCLK3	(BIT(26) | BIT(27))
+#define FSL_SAI_CR2_BCP		BIT(25)
+#define FSL_SAI_CR2_BCD_MSTR	BIT(24)
+
+/* SAI Transmit and Recieve Configuration 3 Register */
+#define FSL_SAI_TCR3		0x0c
+#define FSL_SAI_RCR3		0x8c
+#define FSL_SAI_CR3_TRCE	BIT(16)
+#define FSL_SAI_CR3_WDFL(x)	(x)
+#define FSL_SAI_CR3_WDFL_MASK	0x1f
+
+/* SAI Transmit and Recieve Configuration 4 Register */
+#define FSL_SAI_TCR4		0x10
+#define FSL_SAI_RCR4		0x90
+#define FSL_SAI_CR4_FRSZ(x)	(((x) - 1) << 16)
+#define FSL_SAI_CR4_FRSZ_MASK	(0x1f << 16)
+#define FSL_SAI_CR4_SYWD(x)	(((x) - 1) << 8)
+#define FSL_SAI_CR4_SYWD_MASK	(0x1f << 8)
+#define FSL_SAI_CR4_MF		BIT(4)
+#define FSL_SAI_CR4_FSE		BIT(3)
+#define FSL_SAI_CR4_FSP		BIT(1)
+#define FSL_SAI_CR4_FSD_MSTR	BIT(0)
+
+/* SAI Transmit and Recieve Configuration 5 Register */
+#define FSL_SAI_TCR5		0x14
+#define FSL_SAI_RCR5		0x94
+#define FSL_SAI_CR5_WNW(x)	(((x) - 1) << 24)
+#define FSL_SAI_CR5_WNW_MASK	(0x1f << 24)
+#define FSL_SAI_CR5_W0W(x)	(((x) - 1) << 16)
+#define FSL_SAI_CR5_W0W_MASK	(0x1f << 16)
+#define FSL_SAI_CR5_FBT(x)	((x) << 8)
+#define FSL_SAI_CR5_FBT_MASK	(0x1f << 8)
+
+/* SAI type */
+#define FSL_SAI_DMA		BIT(0)
+#define FSL_SAI_USE_AC97	BIT(1)
+#define FSL_SAI_NET		BIT(2)
+#define FSL_SAI_TRA_SYN		BIT(3)
+#define FSL_SAI_REC_SYN		BIT(4)
+#define FSL_SAI_USE_I2S_SLAVE	BIT(5)
+
+#define FSL_FMT_TRANSMITTER	0
+#define FSL_FMT_RECEIVER	1
+
+/* SAI clock sources */
+#define FSL_SAI_CLK_BUS		0
+#define FSL_SAI_CLK_MAST1	1
+#define FSL_SAI_CLK_MAST2	2
+#define FSL_SAI_CLK_MAST3	3
+
+/* SAI data transfer numbers per DMA request */
+#define FSL_SAI_MAXBURST_TX 6
+#define FSL_SAI_MAXBURST_RX 6
+
+struct fsl_sai {
+	struct clk *clk;
+
+	void __iomem *base;
+
+	bool big_endian_regs;
+	bool big_endian_data;
+
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+};
+
+#endif /* __FSL_SAI_H */
-- 
cgit v1.2.2


From 11db0da831b1e6ae3c1f8743599434281db294db Mon Sep 17 00:00:00 2001
From: Shawn Guo <shawn.guo@linaro.org>
Date: Fri, 13 Dec 2013 14:43:03 +0800
Subject: ASoC: sgtl5000: clean up sgtl5000_enable_regulators()

Function sgtl5000_enable_regulators() is somehow odd in handling the
optional external VDDD supply.  The driver can only enable this supply
on SGTL5000 chip before revision 0x11, and of course when this external
VDDD is present.  It currently does something like below.

1. Check if regulator_bulk_get() on VDDA, VDDIO and VDDD will fail.  If
   it fails, VDDD must be absent and it falls on internal LDO by calling
   sgtl5000_replace_vddd_with_ldo().  Otherwise, VDDD is used.  And in
   either case, regulator_bulk_enable() will be called to enable
   3 supplies.

2. In case that SGTL5000 revision is later than 0x11, even if external
   VDDD is present, it has to roll back the 'enable' and 'get' calls
   with regulator_bulk_disable() and regulator_bulk_free(), and starts
   over again by calling sgtl5000_replace_vddd_with_ldo() and
   regulator_bulk_enable().

Such back and forth calls sequence is complicated and unnecessary.
Also, since commit 4ddfebd (regulator: core: Provide a dummy regulator
with full constraints), regulator_bulk_get() will always succeeds
because of the dummy regulator.  Thus the VDDD detection is broken.

The patch changes the flow to something like the following, which should
be more reasonable and clear, and also fix the VDDD detection breakage.

1. Check if we're running a chip before revision 0x11, on which an
   external VDDD can possibly be an option.

2. If it is an early revision, call regulator_get_optional() to detect
   whether an external VDDD supply is available.

3. If external VDDD is present, call sgtl5000_replace_vddd_with_ldo() to
   update sgtl5000->supplies info.

4. Drop regulator_bulk_get() call in sgtl5000_replace_vddd_with_ldo(),
   and call it in sgtl5000_enable_regulators() no matter it's an
   external VDDD or internal LDO.

5. Call regulator_bulk_enable() to enable these 3 regulators.

Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/sgtl5000.c | 62 ++++++++++++++++-----------------------------
 1 file changed, 22 insertions(+), 40 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bd291d2b327d..0fcbe90f3ef2 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1286,15 +1286,6 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec)
 
 	sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
 
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
-			sgtl5000->supplies);
-
-	if (ret) {
-		ldo_regulator_remove(codec);
-		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
-		return ret;
-	}
-
 	dev_info(codec->dev, "Using internal LDO instead of VDDD\n");
 	return 0;
 }
@@ -1305,20 +1296,35 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 	int i;
 	int external_vddd = 0;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	struct regulator *vddd;
 
 	for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++)
 		sgtl5000->supplies[i].supply = supply_names[i];
 
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
-	if (!ret)
-		external_vddd = 1;
-	else {
+	/* External VDDD only works before revision 0x11 */
+	if (sgtl5000->revision < 0x11) {
+		vddd = regulator_get_optional(codec->dev, "VDDD");
+		if (IS_ERR(vddd)) {
+			/* See if it's just not registered yet */
+			if (PTR_ERR(vddd) == -EPROBE_DEFER)
+				return -EPROBE_DEFER;
+		} else {
+			external_vddd = 1;
+			regulator_put(vddd);
+		}
+	}
+
+	if (!external_vddd) {
 		ret = sgtl5000_replace_vddd_with_ldo(codec);
 		if (ret)
 			return ret;
 	}
 
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+				 sgtl5000->supplies);
+	if (ret)
+		goto err_ldo_remove;
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
 	if (ret)
@@ -1327,37 +1333,13 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 	/* wait for all power rails bring up */
 	udelay(10);
 
-	/*
-	 * workaround for revision 0x11 and later,
-	 * roll back to use internal LDO
-	 */
-	if (external_vddd && sgtl5000->revision >= 0x11) {
-		/* disable all regulator first */
-		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
-					sgtl5000->supplies);
-		/* free VDDD regulator */
-		regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
-					sgtl5000->supplies);
-
-		ret = sgtl5000_replace_vddd_with_ldo(codec);
-		if (ret)
-			return ret;
-
-		ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
-						sgtl5000->supplies);
-		if (ret)
-			goto err_regulator_free;
-
-		/* wait for all power rails bring up */
-		udelay(10);
-	}
-
 	return 0;
 
 err_regulator_free:
 	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
 				sgtl5000->supplies);
-	if (external_vddd)
+err_ldo_remove:
+	if (!external_vddd)
 		ldo_regulator_remove(codec);
 	return ret;
 
-- 
cgit v1.2.2


From 6f2032a18969d22740a865e0b4f2e48cf5338f36 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 12:34:45 -0700
Subject: ASoC: ep93xx: get rid of ep93xx-pcm-audio struct device

Modify the ep93xx PCM driver so that it's a utility library that can be
registered on each DAI, rather than a separate struct device. This is
more in line with how many recent DT-converted platforms operate, and
avoids the need for yet another struct device.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/cirrus/edb93xx.c     |  2 +-
 sound/soc/cirrus/ep93xx-ac97.c |  8 ++++++++
 sound/soc/cirrus/ep93xx-i2s.c  |  8 ++++++++
 sound/soc/cirrus/ep93xx-pcm.c  | 19 +++++--------------
 sound/soc/cirrus/ep93xx-pcm.h  | 22 ++++++++++++++++++++++
 sound/soc/cirrus/simone.c      |  2 +-
 sound/soc/cirrus/snappercl15.c |  2 +-
 7 files changed, 46 insertions(+), 17 deletions(-)
 create mode 100644 sound/soc/cirrus/ep93xx-pcm.h

(limited to 'sound')

diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c
index c43fb214558a..4f900efc437c 100644
--- a/sound/soc/cirrus/edb93xx.c
+++ b/sound/soc/cirrus/edb93xx.c
@@ -63,7 +63,7 @@ static struct snd_soc_ops edb93xx_ops = {
 static struct snd_soc_dai_link edb93xx_dai = {
 	.name		= "CS4271",
 	.stream_name	= "CS4271 HiFi",
-	.platform_name	= "ep93xx-pcm-audio",
+	.platform_name	= "ep93xx-i2s",
 	.cpu_dai_name	= "ep93xx-i2s",
 	.codec_name	= "spi0.0",
 	.codec_dai_name	= "cs4271-hifi",
diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index efa75b5086a4..cc5583da5380 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -24,6 +24,8 @@
 
 #include <linux/platform_data/dma-ep93xx.h>
 
+#include "ep93xx-pcm.h"
+
 /*
  * Per channel (1-4) registers.
  */
@@ -394,8 +396,14 @@ static int ep93xx_ac97_probe(struct platform_device *pdev)
 	if (ret)
 		goto fail;
 
+	ret = devm_ep93xx_pcm_platform_register(&pdev->dev);
+	if (ret)
+		goto fail_unregister;
+
 	return 0;
 
+fail_unregister:
+	snd_soc_unregister_component(&pdev->dev);
 fail:
 	ep93xx_ac97_info = NULL;
 	snd_soc_set_ac97_ops(NULL);
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index a57643d6402f..167728a73da2 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -30,6 +30,8 @@
 #include <mach/ep93xx-regs.h>
 #include <linux/platform_data/dma-ep93xx.h>
 
+#include "ep93xx-pcm.h"
+
 #define EP93XX_I2S_TXCLKCFG		0x00
 #define EP93XX_I2S_RXCLKCFG		0x04
 #define EP93XX_I2S_GLCTRL		0x0C
@@ -405,8 +407,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	if (err)
 		goto fail_put_lrclk;
 
+	err = devm_ep93xx_pcm_platform_register(&pdev->dev);
+	if (err)
+		goto fail_unregister;
+
 	return 0;
 
+fail_unregister:
+	snd_soc_unregister_component(&pdev->dev);
 fail_put_lrclk:
 	clk_put(info->lrclk);
 fail_put_sclk:
diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index fdb8b8feb4ed..198c5405f2ce 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -23,6 +23,8 @@
 
 #include <linux/platform_data/dma-ep93xx.h>
 
+#include "ep93xx-pcm.h"
+
 static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
 	.info			= (SNDRV_PCM_INFO_MMAP		|
 				   SNDRV_PCM_INFO_MMAP_VALID	|
@@ -76,27 +78,16 @@ static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = {
 	.prealloc_buffer_size = 131072,
 };
 
-static int ep93xx_soc_platform_probe(struct platform_device *pdev)
+int devm_ep93xx_pcm_platform_register(struct device *dev)
 {
-	return devm_snd_dmaengine_pcm_register(&pdev->dev,
+	return devm_snd_dmaengine_pcm_register(dev,
 		&ep93xx_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
-
-static struct platform_driver ep93xx_pcm_driver = {
-	.driver = {
-			.name = "ep93xx-pcm-audio",
-			.owner = THIS_MODULE,
-	},
-
-	.probe = ep93xx_soc_platform_probe,
-};
-
-module_platform_driver(ep93xx_pcm_driver);
+EXPORT_SYMBOL_GPL(devm_ep93xx_pcm_platform_register);
 
 MODULE_AUTHOR("Ryan Mallon");
 MODULE_DESCRIPTION("EP93xx ALSA PCM interface");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:ep93xx-pcm-audio");
diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h
new file mode 100644
index 000000000000..b7a12a2fae9c
--- /dev/null
+++ b/sound/soc/cirrus/ep93xx-pcm.h
@@ -0,0 +1,22 @@
+/*
+ * Copyright (c) 2013, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef __EP93XX_PCM_H__
+#define __EP93XX_PCM_H__
+
+int devm_ep93xx_pcm_platform_register(struct device *dev);
+
+#endif
diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c
index 4d094d00c34a..822a19a89e74 100644
--- a/sound/soc/cirrus/simone.c
+++ b/sound/soc/cirrus/simone.c
@@ -27,7 +27,7 @@ static struct snd_soc_dai_link simone_dai = {
 	.cpu_dai_name	= "ep93xx-ac97",
 	.codec_dai_name	= "ac97-hifi",
 	.codec_name	= "ac97-codec",
-	.platform_name	= "ep93xx-pcm-audio",
+	.platform_name	= "ep93xx-ac97",
 };
 
 static struct snd_soc_card snd_soc_simone = {
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 69041074f2c1..29238a7476dd 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -83,7 +83,7 @@ static struct snd_soc_dai_link snappercl15_dai = {
 	.cpu_dai_name	= "ep93xx-i2s",
 	.codec_dai_name	= "tlv320aic23-hifi",
 	.codec_name	= "tlv320aic23-codec.0-001a",
-	.platform_name	=  "ep93xx-pcm-audio",
+	.platform_name	= "ep93xx-i2s",
 	.init		= snappercl15_tlv320aic23_init,
 	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
 			  SND_SOC_DAIFMT_CBS_CFS,
-- 
cgit v1.2.2


From a8983d4b0a8c439bddae0c9fd1e8a4cf7c402262 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 12:34:46 -0700
Subject: ASoC: ep93xx: remove custom DMA alloc compat function

ep93xx_compat_request_channel() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:

a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
   pointer rather than some custom type.

b) dma_data->filter_data rather than dma_data should be passed to
   snd_dmaengine_pcm_request_channel() as the filter data.

Make minor changes to the ep93xx DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/cirrus/ep93xx-ac97.c | 12 ++++++++++--
 sound/soc/cirrus/ep93xx-i2s.c  | 14 ++++++++++++--
 sound/soc/cirrus/ep93xx-pcm.c  | 13 -------------
 3 files changed, 22 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c
index cc5583da5380..f30dadf85b99 100644
--- a/sound/soc/cirrus/ep93xx-ac97.c
+++ b/sound/soc/cirrus/ep93xx-ac97.c
@@ -19,6 +19,7 @@
 #include <linux/slab.h>
 
 #include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/ac97_codec.h>
 #include <sound/soc.h>
 
@@ -97,6 +98,8 @@ struct ep93xx_ac97_info {
 	struct device		*dev;
 	void __iomem		*regs;
 	struct completion	done;
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
 
 /* currently ALSA only supports a single AC97 device */
@@ -317,8 +320,13 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
 
 static int ep93xx_ac97_dai_probe(struct snd_soc_dai *dai)
 {
-	dai->playback_dma_data = &ep93xx_ac97_pcm_out;
-	dai->capture_dma_data = &ep93xx_ac97_pcm_in;
+	struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
+
+	info->dma_params_tx.filter_data = &ep93xx_ac97_pcm_out;
+	info->dma_params_rx.filter_data = &ep93xx_ac97_pcm_in;
+
+	dai->playback_dma_data = &info->dma_params_tx;
+	dai->capture_dma_data = &info->dma_params_rx;
 
 	return 0;
 }
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index 167728a73da2..943145f9d1b6 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -21,6 +21,7 @@
 #include <linux/io.h>
 
 #include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
@@ -63,6 +64,8 @@ struct ep93xx_i2s_info {
 	struct clk			*sclk;
 	struct clk			*lrclk;
 	void __iomem			*regs;
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
 
 static struct ep93xx_dma_data ep93xx_i2s_dma_data[] = {
@@ -142,8 +145,15 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
 
 static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai)
 {
-	dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK];
-	dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE];
+	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
+
+	info->dma_params_tx.filter_data =
+		&ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	info->dma_params_rx.filter_data =
+		&ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	dai->playback_dma_data = &info->dma_params_tx;
+	dai->capture_dma_data = &info->dma_params_rx;
 
 	return 0;
 }
diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index 198c5405f2ce..ca6698d75736 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -59,22 +59,9 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param)
 	return false;
 }
 
-static struct dma_chan *ep93xx_compat_request_channel(
-	struct snd_soc_pcm_runtime *rtd,
-	struct snd_pcm_substream *substream)
-{
-	struct snd_dmaengine_dai_dma_data *dma_data;
-
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter,
-						 dma_data);
-}
-
 static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = {
 	.pcm_hardware = &ep93xx_pcm_hardware,
 	.compat_filter_fn = ep93xx_pcm_dma_filter,
-	.compat_request_channel = ep93xx_compat_request_channel,
 	.prealloc_buffer_size = 131072,
 };
 
-- 
cgit v1.2.2


From ede38884ac25ed78e43f3480056670963a9980f0 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 12:35:24 -0700
Subject: ASoC: SPEAr: get rid of spear-pcm-audio struct device

Modify the SPEAr PCM driver so that it's a utility library that can be
registered on each DAI, rather than a separate struct device. This is
more in line with how many recent DT-converted platforms operate, and
avoids the need for yet another struct device.

This is also required as a pre-cursor to removing
spear_pcm_request_chan().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/spear/spdif_in.c  |  9 +++++++--
 sound/soc/spear/spdif_out.c | 10 ++++++++--
 sound/soc/spear/spear_pcm.c | 18 ++++--------------
 sound/soc/spear/spear_pcm.h | 22 ++++++++++++++++++++++
 4 files changed, 41 insertions(+), 18 deletions(-)
 create mode 100644 sound/soc/spear/spear_pcm.h

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 21a8c954af1c..4627110f3441 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -24,6 +24,7 @@
 #include <sound/spear_dma.h>
 #include <sound/spear_spdif.h>
 #include "spdif_in_regs.h"
+#include "spear_pcm.h"
 
 struct spdif_in_params {
 	u32 format;
@@ -257,8 +258,12 @@ static int spdif_in_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component,
-					       &spdif_in_dai, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &spdif_in_component,
+					      &spdif_in_dai, 1);
+	if (ret)
+		return ret;
+
+	return devm_spear_pcm_platform_register(&pdev->dev);
 }
 
 static struct platform_driver spdif_in_driver = {
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index b6ef6f78dc78..731a1e0cfeaa 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -22,6 +22,7 @@
 #include <sound/spear_dma.h>
 #include <sound/spear_spdif.h>
 #include "spdif_out_regs.h"
+#include "spear_pcm.h"
 
 struct spdif_out_params {
 	u32 rate;
@@ -280,6 +281,7 @@ static int spdif_out_probe(struct platform_device *pdev)
 	struct spdif_out_dev *host;
 	struct spear_spdif_platform_data *pdata;
 	struct resource *res;
+	int ret;
 
 	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
 	if (!host) {
@@ -306,8 +308,12 @@ static int spdif_out_probe(struct platform_device *pdev)
 
 	dev_set_drvdata(&pdev->dev, host);
 
-	return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component,
-					       &spdif_out_dai, 1);
+	ret = devm_snd_soc_register_component(&pdev->dev, &spdif_out_component,
+					      &spdif_out_dai, 1);
+	if (ret)
+		return ret;
+
+	return devm_spear_pcm_platform_register(&pdev->dev);
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 9a02141666ea..f288724961da 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -18,6 +18,7 @@
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/spear_dma.h>
+#include "spear_pcm.h"
 
 static const struct snd_pcm_hardware spear_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -47,26 +48,15 @@ static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = {
 	.prealloc_buffer_size = 16 * 1024,
 };
 
-static int spear_soc_platform_probe(struct platform_device *pdev)
+int devm_spear_pcm_platform_register(struct device *dev)
 {
-	return devm_snd_dmaengine_pcm_register(&pdev->dev,
+	return devm_snd_dmaengine_pcm_register(dev,
 		&spear_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
-
-static struct platform_driver spear_pcm_driver = {
-	.driver = {
-		.name = "spear-pcm-audio",
-		.owner = THIS_MODULE,
-	},
-
-	.probe = spear_soc_platform_probe,
-};
-
-module_platform_driver(spear_pcm_driver);
+EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register);
 
 MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
 MODULE_DESCRIPTION("SPEAr PCM DMA module");
 MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:spear-pcm-audio");
diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h
new file mode 100644
index 000000000000..631e2aa1fb33
--- /dev/null
+++ b/sound/soc/spear/spear_pcm.h
@@ -0,0 +1,22 @@
+/*
+ * Copyright (c) 2013, NVIDIA CORPORATION.  All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef __SPEAR_PCM_H__
+#define __SPEAR_PCM_H__
+
+int devm_spear_pcm_platform_register(struct device *dev);
+
+#endif
-- 
cgit v1.2.2


From e1771bcf99b0dc91f4ba645c1740fd5031702f49 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 10 Dec 2013 12:35:25 -0700
Subject: ASoC: SPEAr: remove custom DMA alloc compat function

spear_pcm_request_chan() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:

a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
   pointer rather than some custom type.

b) dma_data->filter_data rather than dma_data should be passed to
   snd_dmaengine_pcm_request_channel() as the filter data.

Make minor changes to the SPEAr DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/spear/spdif_in.c  | 10 +++++++---
 sound/soc/spear/spdif_out.c | 10 +++++++---
 sound/soc/spear/spear_pcm.c | 21 +++++++--------------
 sound/soc/spear/spear_pcm.h |  4 +++-
 4 files changed, 24 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
index 4627110f3441..4ab442a63d7e 100644
--- a/sound/soc/spear/spdif_in.c
+++ b/sound/soc/spear/spdif_in.c
@@ -18,6 +18,7 @@
 #include <linux/ioport.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -38,6 +39,8 @@ struct spdif_in_dev {
 	struct device *dev;
 	void (*reset_perip)(void);
 	int irq;
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_pcm_config config;
 };
 
 static void spdif_in_configure(struct spdif_in_dev *host)
@@ -54,7 +57,8 @@ static int spdif_in_dai_probe(struct snd_soc_dai *dai)
 {
 	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
 
-	dai->capture_dma_data = &host->dma_params;
+	host->dma_params_rx.filter_data = &host->dma_params;
+	dai->capture_dma_data = &host->dma_params_rx;
 
 	return 0;
 }
@@ -245,7 +249,6 @@ static int spdif_in_probe(struct platform_device *pdev)
 	host->dma_params.addr = res_fifo->start;
 	host->dma_params.max_burst = 16;
 	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
-	host->dma_params.filter = pdata->filter;
 	host->reset_perip = pdata->reset_perip;
 
 	host->dev = &pdev->dev;
@@ -263,7 +266,8 @@ static int spdif_in_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return devm_spear_pcm_platform_register(&pdev->dev);
+	return devm_spear_pcm_platform_register(&pdev->dev, &host->config,
+						pdata->filter);
 }
 
 static struct platform_driver spdif_in_driver = {
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 731a1e0cfeaa..fe99f461aff0 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -18,6 +18,7 @@
 #include <linux/ioport.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/dmaengine_pcm.h>
 #include <sound/soc.h>
 #include <sound/spear_dma.h>
 #include <sound/spear_spdif.h>
@@ -36,6 +37,8 @@ struct spdif_out_dev {
 	struct spdif_out_params saved_params;
 	u32 running;
 	void __iomem *io_base;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+	struct snd_dmaengine_pcm_config config;
 };
 
 static void spdif_out_configure(struct spdif_out_dev *host)
@@ -245,7 +248,8 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai)
 {
 	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
 
-	dai->playback_dma_data = &host->dma_params;
+	host->dma_params_tx.filter_data = &host->dma_params;
+	dai->playback_dma_data = &host->dma_params_tx;
 
 	return snd_soc_add_dai_controls(dai, spdif_out_controls,
 				ARRAY_SIZE(spdif_out_controls));
@@ -304,7 +308,6 @@ static int spdif_out_probe(struct platform_device *pdev)
 	host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA;
 	host->dma_params.max_burst = 16;
 	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
-	host->dma_params.filter = pdata->filter;
 
 	dev_set_drvdata(&pdev->dev, host);
 
@@ -313,7 +316,8 @@ static int spdif_out_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	return devm_spear_pcm_platform_register(&pdev->dev);
+	return devm_spear_pcm_platform_register(&pdev->dev, &host->config,
+						pdata->filter);
 }
 
 #ifdef CONFIG_PM
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index f288724961da..0e5a8f35d0ad 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -32,26 +32,19 @@ static const struct snd_pcm_hardware spear_pcm_hardware = {
 	.fifo_size = 0, /* fifo size in bytes */
 };
 
-static struct dma_chan *spear_pcm_request_chan(struct snd_soc_pcm_runtime *rtd,
-	struct snd_pcm_substream *substream)
-{
-	struct spear_dma_data *dma_data;
-
-	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
-	return snd_dmaengine_pcm_request_channel(dma_data->filter, dma_data);
-}
-
 static const struct snd_dmaengine_pcm_config spear_dmaengine_pcm_config = {
 	.pcm_hardware = &spear_pcm_hardware,
-	.compat_request_channel = spear_pcm_request_chan,
 	.prealloc_buffer_size = 16 * 1024,
 };
 
-int devm_spear_pcm_platform_register(struct device *dev)
+int devm_spear_pcm_platform_register(struct device *dev,
+			struct snd_dmaengine_pcm_config *config,
+			bool (*filter)(struct dma_chan *chan, void *slave))
 {
-	return devm_snd_dmaengine_pcm_register(dev,
-		&spear_dmaengine_pcm_config,
+	*config = spear_dmaengine_pcm_config;
+	config->compat_filter_fn = filter;
+
+	return snd_dmaengine_pcm_register(dev, config,
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
diff --git a/sound/soc/spear/spear_pcm.h b/sound/soc/spear/spear_pcm.h
index 631e2aa1fb33..9b0ca62d6f02 100644
--- a/sound/soc/spear/spear_pcm.h
+++ b/sound/soc/spear/spear_pcm.h
@@ -17,6 +17,8 @@
 #ifndef __SPEAR_PCM_H__
 #define __SPEAR_PCM_H__
 
-int devm_spear_pcm_platform_register(struct device *dev);
+int devm_spear_pcm_platform_register(struct device *dev,
+			struct snd_dmaengine_pcm_config *config,
+			bool (*filter)(struct dma_chan *chan, void *slave));
 
 #endif
-- 
cgit v1.2.2


From a4c0d2735c66c2363100575324a2f7659fb1f684 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 19 Dec 2013 09:11:02 +0000
Subject: ASoC: wm5110: Expose switch controls for DRE

Certain use-cases require the DRE to be disabled so expose controls for
the enables.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm5110.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index ebcbe7831e57..3487ffa2e950 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -371,6 +371,13 @@ SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
 SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
 	   ARIZONA_SPK2R_MUTE_SHIFT, 1, 1),
 
+SOC_DOUBLE("HPOUT1 DRE Switch", ARIZONA_DRE_ENABLE,
+	   ARIZONA_DRE1L_ENA_SHIFT, ARIZONA_DRE1R_ENA_SHIFT, 1, 0),
+SOC_DOUBLE("HPOUT2 DRE Switch", ARIZONA_DRE_ENABLE,
+	   ARIZONA_DRE2L_ENA_SHIFT, ARIZONA_DRE2R_ENA_SHIFT, 1, 0),
+SOC_DOUBLE("HPOUT3 DRE Switch", ARIZONA_DRE_ENABLE,
+	   ARIZONA_DRE3L_ENA_SHIFT, ARIZONA_DRE3R_ENA_SHIFT, 1, 0),
+
 SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp),
 SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp),
 
-- 
cgit v1.2.2


From fbedc8cbc3c40281cf52ed0e2e5998dea98e2992 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 19 Dec 2013 09:30:12 +0000
Subject: ASoC: wm5110: Add FSH for ISRCs

Currently, the driver only supports configuration of the lower sample
rate (FSL) on the ISRCs. With the higher rate being fixed a SYSCLK, this
patch adds support for configuring the higher sample rate (FSH).

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/arizona.c | 16 ++++++++++++++++
 sound/soc/codecs/arizona.h |  1 +
 sound/soc/codecs/wm5110.c  |  3 +++
 3 files changed, 20 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 6bfd8031c0c9..56d3ff570ad4 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -498,6 +498,22 @@ const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = {
 EXPORT_SYMBOL_GPL(arizona_rate_val);
 
 
+const struct soc_enum arizona_isrc_fsh[] = {
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_1,
+			      ARIZONA_ISRC1_FSH_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_1,
+			      ARIZONA_ISRC2_FSH_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_1,
+			      ARIZONA_ISRC3_FSH_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+};
+EXPORT_SYMBOL_GPL(arizona_isrc_fsh);
+
 const struct soc_enum arizona_isrc_fsl[] = {
 	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2,
 			      ARIZONA_ISRC1_FSL_SHIFT, 0xf,
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 6641f3d64194..99a97c06d46a 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -186,6 +186,7 @@ extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
 extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
 
 extern const struct soc_enum arizona_isrc_fsl[];
+extern const struct soc_enum arizona_isrc_fsh[];
 
 extern const struct soc_enum arizona_in_vi_ramp;
 extern const struct soc_enum arizona_in_vd_ramp;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 3487ffa2e950..57be82261c2f 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -280,6 +280,9 @@ SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
 SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]),
 SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]),
 SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]),
+SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]),
+SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]),
+SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]),
 
 ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE),
-- 
cgit v1.2.2


From 56d37d85438df38e150282baafe52dcd588854c7 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 19 Dec 2013 09:30:13 +0000
Subject: ASoC: wm5110: Add support for ASRC RATE 1

Add support for configuring the sample rate on the SYSCLK side of the
ASRC.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/arizona.c | 7 +++++++
 sound/soc/codecs/arizona.h | 1 +
 sound/soc/codecs/wm5110.c  | 1 +
 3 files changed, 9 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 56d3ff570ad4..e4295fee8f13 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -530,6 +530,13 @@ const struct soc_enum arizona_isrc_fsl[] = {
 };
 EXPORT_SYMBOL_GPL(arizona_isrc_fsl);
 
+const struct soc_enum arizona_asrc_rate1 =
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ASRC_RATE1,
+			      ARIZONA_ASRC_RATE1_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE - 1,
+			      arizona_rate_text, arizona_rate_val);
+EXPORT_SYMBOL_GPL(arizona_asrc_rate1);
+
 static const char *arizona_vol_ramp_text[] = {
 	"0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB",
 	"15ms/6dB", "30ms/6dB",
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 99a97c06d46a..10b398477203 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -187,6 +187,7 @@ extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
 
 extern const struct soc_enum arizona_isrc_fsl[];
 extern const struct soc_enum arizona_isrc_fsh[];
+extern const struct soc_enum arizona_asrc_rate1;
 
 extern const struct soc_enum arizona_in_vi_ramp;
 extern const struct soc_enum arizona_in_vd_ramp;
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 57be82261c2f..f3d96eae031a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -283,6 +283,7 @@ SOC_VALUE_ENUM("ISRC3 FSL", arizona_isrc_fsl[2]),
 SOC_VALUE_ENUM("ISRC1 FSH", arizona_isrc_fsh[0]),
 SOC_VALUE_ENUM("ISRC2 FSH", arizona_isrc_fsh[1]),
 SOC_VALUE_ENUM("ISRC3 FSH", arizona_isrc_fsh[2]),
+SOC_VALUE_ENUM("ASRC RATE 1", arizona_asrc_rate1),
 
 ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE),
-- 
cgit v1.2.2


From e1acb40a3addc9aceb4600f04c9c86b50770b9b8 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Thu, 19 Dec 2013 11:59:54 +0800
Subject: ASoC: simple-card: Use devm_snd_soc_register_card()

Makes the code slightly shorter.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 10 +---------
 1 file changed, 1 insertion(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 7a9b6b4898c0..3d190d05ad4a 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -234,14 +234,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	cinfo->snd_card.num_links	= 1;
 	cinfo->snd_card.dev		= &pdev->dev;
 
-	return snd_soc_register_card(&cinfo->snd_card);
-}
-
-static int asoc_simple_card_remove(struct platform_device *pdev)
-{
-	struct asoc_simple_card_info *cinfo = pdev->dev.platform_data;
-
-	return snd_soc_unregister_card(&cinfo->snd_card);
+	return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card);
 }
 
 static const struct of_device_id asoc_simple_of_match[] = {
@@ -257,7 +250,6 @@ static struct platform_driver asoc_simple_card = {
 		.of_match_table = asoc_simple_of_match,
 	},
 	.probe		= asoc_simple_card_probe,
-	.remove		= asoc_simple_card_remove,
 };
 
 module_platform_driver(asoc_simple_card);
-- 
cgit v1.2.2


From 595fe1b702c3fa739ab395ab0471054875f20b90 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Dec 2013 07:35:11 +0100
Subject: ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate

So far, CONFIG_SND_HDA_CODEC_* kconfigs have been booleans due to
historical reasons.  The major reason was that the automatic codec
driver probing wouldn't work if user sets a codec driver as a module
while the controller driver as a built-in.  And, another reason was to
avoid exporting symbols of the helper codes when all drivers are built
in.

But, this sort of "kindness" rather confuses people in the end,
especially makes the config refinement via localmodconfig unhappy.
Also, a codec module would still work if you re-bind the controller
driver via sysfs (although it's no automatic loading), so there might
be a slight use case.

That said, better to let people fallen into a pitfall than being too
smart and restrict something.  Let's make things straightforward: now
all CONFIG_SND_HDA_CODEC_* become tristate, and all symbols exported
unconditionally.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig     | 115 ++++++++++------------------------------------
 sound/pci/hda/Makefile    |  50 ++++++--------------
 sound/pci/hda/hda_codec.h |  13 ------
 3 files changed, 37 insertions(+), 141 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 4cdd9ded4563..b0c0481c3339 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -87,126 +87,72 @@ config SND_HDA_PATCH_LOADER
 	  This option turns on hwdep and reconfig features automatically.
 
 config SND_HDA_CODEC_REALTEK
-	bool "Build Realtek HD-audio codec support"
-	default y
+	tristate "Build Realtek HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include Realtek HD-audio codec support in
+	  Say Y or M here to include Realtek HD-audio codec support in
 	  snd-hda-intel driver, such as ALC880.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-realtek.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_ANALOG
-	bool "Build Analog Device HD-audio codec support"
-	default y
+	tristate "Build Analog Device HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include Analog Device HD-audio codec support in
+	  Say Y or M here to include Analog Device HD-audio codec support in
 	  snd-hda-intel driver, such as AD1986A.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-analog.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_SIGMATEL
-	bool "Build IDT/Sigmatel HD-audio codec support"
-	default y
+	tristate "Build IDT/Sigmatel HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include IDT (Sigmatel) HD-audio codec support in
+	  Say Y or M here to include IDT (Sigmatel) HD-audio codec support in
 	  snd-hda-intel driver, such as STAC9200.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-idt.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_VIA
-	bool "Build VIA HD-audio codec support"
-	default y
+	tristate "Build VIA HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include VIA HD-audio codec support in
+	  Say Y or M here to include VIA HD-audio codec support in
 	  snd-hda-intel driver, such as VT1708.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-via.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_HDMI
-	bool "Build HDMI/DisplayPort HD-audio codec support"
-	default y
+	tristate "Build HDMI/DisplayPort HD-audio codec support"
 	help
-	  Say Y here to include HDMI and DisplayPort HD-audio codec
+	  Say Y or M here to include HDMI and DisplayPort HD-audio codec
 	  support in snd-hda-intel driver.  This includes all AMD/ATI,
 	  Intel and Nvidia HDMI/DisplayPort codecs.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-hdmi.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_I915
 	bool
 	default y
 	depends on DRM_I915
 
 config SND_HDA_CODEC_CIRRUS
-	bool "Build Cirrus Logic codec support"
-	default y
+	tristate "Build Cirrus Logic codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include Cirrus Logic codec support in
+	  Say Y or M here to include Cirrus Logic codec support in
 	  snd-hda-intel driver, such as CS4206.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-cirrus.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_CONEXANT
-	bool "Build Conexant HD-audio codec support"
-	default y
+	tristate "Build Conexant HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include Conexant HD-audio codec support in
+	  Say Y or M here to include Conexant HD-audio codec support in
 	  snd-hda-intel driver, such as CX20549.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-conexant.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_CA0110
-	bool "Build Creative CA0110-IBG codec support"
-	default y
+	tristate "Build Creative CA0110-IBG codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include Creative CA0110-IBG codec support in
+	  Say Y or M here to include Creative CA0110-IBG codec support in
 	  snd-hda-intel driver, found on some Creative X-Fi cards.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-ca0110.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_CA0132
-	bool "Build Creative CA0132 codec support"
-	default y
+	tristate "Build Creative CA0132 codec support"
 	help
-	  Say Y here to include Creative CA0132 codec support in
+	  Say Y or M here to include Creative CA0132 codec support in
 	  snd-hda-intel driver.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-ca0132.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_CA0132_DSP
 	bool "Support new DSP code for CA0132 codec"
 	depends on SND_HDA_CODEC_CA0132
@@ -220,35 +166,22 @@ config SND_HDA_CODEC_CA0132_DSP
 	  (ctefx.bin).
 
 config SND_HDA_CODEC_CMEDIA
-	bool "Build C-Media HD-audio codec support"
-	default y
+	tristate "Build C-Media HD-audio codec support"
 	select SND_HDA_GENERIC
 	help
-	  Say Y here to include C-Media HD-audio codec support in
+	  Say Y or M here to include C-Media HD-audio codec support in
 	  snd-hda-intel driver, such as CMI9880.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-cmedia.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_CODEC_SI3054
-	bool "Build Silicon Labs 3054 HD-modem codec support"
-	default y
+	tristate "Build Silicon Labs 3054 HD-modem codec support"
 	help
-	  Say Y here to include Silicon Labs 3054 HD-modem codec
+	  Say Y or M here to include Silicon Labs 3054 HD-modem codec
 	  (and compatibles) support in snd-hda-intel driver.
 
-	  When the HD-audio driver is built as a module, the codec
-	  support code is also built as another module,
-	  snd-hda-codec-si3054.
-	  This module is automatically loaded at probing.
-
 config SND_HDA_GENERIC
-	bool "Enable generic HD-audio codec parser"
-	default y
+	tristate "Enable generic HD-audio codec parser"
 	help
-	  Say Y here to enable the generic HD-audio codec parser
+	  Say Y or M here to enable the generic HD-audio codec parser
 	  in snd-hda-intel driver.
 
 config SND_HDA_POWER_SAVE_DEFAULT
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 5a40c652df41..1fcb118e480a 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -27,43 +27,19 @@ snd-hda-codec-hdmi-objs :=	patch_hdmi.o hda_eld.o
 # common driver
 obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o
 
-# codec drivers (note: CONFIG_SND_HDA_CODEC_XXX are booleans)
-ifdef CONFIG_SND_HDA_GENERIC
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-generic.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_REALTEK
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-realtek.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_CMEDIA
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cmedia.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_ANALOG
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-analog.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-idt.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_SI3054
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_CIRRUS
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_CA0110
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_CA0132
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0132.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_CONEXANT
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_VIA
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o
-endif
-ifdef CONFIG_SND_HDA_CODEC_HDMI
-obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-hdmi.o
-endif
+# codec drivers
+obj-$(CONFIG_SND_HDA_GENERIC) += snd-hda-codec-generic.o
+obj-$(CONFIG_SND_HDA_CODEC_REALTEK) += snd-hda-codec-realtek.o
+obj-$(CONFIG_SND_HDA_CODEC_CMEDIA) += snd-hda-codec-cmedia.o
+obj-$(CONFIG_SND_HDA_CODEC_ANALOG) += snd-hda-codec-analog.o
+obj-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += snd-hda-codec-idt.o
+obj-$(CONFIG_SND_HDA_CODEC_SI3054) += snd-hda-codec-si3054.o
+obj-$(CONFIG_SND_HDA_CODEC_CIRRUS) += snd-hda-codec-cirrus.o
+obj-$(CONFIG_SND_HDA_CODEC_CA0110) += snd-hda-codec-ca0110.o
+obj-$(CONFIG_SND_HDA_CODEC_CA0132) += snd-hda-codec-ca0132.o
+obj-$(CONFIG_SND_HDA_CODEC_CONEXANT) += snd-hda-codec-conexant.o
+obj-$(CONFIG_SND_HDA_CODEC_VIA) += snd-hda-codec-via.o
+obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o
 
 # this must be the last entry after codec drivers;
 # otherwise the codec patches won't be hooked before the PCI probe
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 3c484c27048b..2b5d19e48a27 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -681,19 +681,6 @@ snd_hda_codec_load_dsp_cleanup(struct hda_codec *codec,
 				struct snd_dma_buffer *dmab) {}
 #endif
 
-/*
- * Codec modularization
- */
-
-/* Export symbols only for communication with codec drivers;
- * When built in kernel, all HD-audio drivers are supposed to be statically
- * linked to the kernel.  Thus, the symbols don't have to (or shouldn't) be
- * exported unless it's built as a module.
- */
-#ifdef MODULE
 #define EXPORT_SYMBOL_HDA(sym) EXPORT_SYMBOL_GPL(sym)
-#else
-#define EXPORT_SYMBOL_HDA(sym)
-#endif
 
 #endif /* __SOUND_HDA_CODEC_H */
-- 
cgit v1.2.2


From 2698ea98059bb01a9b5d9cfd20e23d55f68c0a17 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Dec 2013 07:45:52 +0100
Subject: ALSA: hda - Kill EXPORT_SYMBOL_HDA()

Replace all with the standard EXPORT_SYMBOL_GPL().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c |  18 ++--
 sound/pci/hda/hda_beep.c        |  10 +-
 sound/pci/hda/hda_codec.c       | 214 ++++++++++++++++++++--------------------
 sound/pci/hda/hda_generic.c     |  44 ++++-----
 sound/pci/hda/hda_hwdep.c       |   8 +-
 sound/pci/hda/hda_jack.c        |  30 +++---
 sound/pci/hda/patch_hdmi.c      |   2 +-
 7 files changed, 163 insertions(+), 163 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 853c6a69e29e..47ad31c6aa70 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -414,7 +414,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
+EXPORT_SYMBOL_GPL(snd_hda_parse_pin_defcfg);
 
 int snd_hda_get_input_pin_attr(unsigned int def_conf)
 {
@@ -435,7 +435,7 @@ int snd_hda_get_input_pin_attr(unsigned int def_conf)
 		return INPUT_PIN_ATTR_FRONT;
 	return INPUT_PIN_ATTR_NORMAL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
+EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr);
 
 /**
  * hda_get_input_pin_label - Give a label for the given input pin
@@ -547,7 +547,7 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec,
 				       cfg->inputs[input].pin,
 				       has_multiple_pins);
 }
-EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
+EXPORT_SYMBOL_GPL(hda_get_autocfg_input_label);
 
 /* return the position of NID in the list, or -1 if not found */
 static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
@@ -721,7 +721,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
 	strlcpy(label, name, maxlen);
 	return 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+EXPORT_SYMBOL_GPL(snd_hda_get_pin_label);
 
 int snd_hda_add_verbs(struct hda_codec *codec,
 		      const struct hda_verb *list)
@@ -733,7 +733,7 @@ int snd_hda_add_verbs(struct hda_codec *codec,
 	*v = list;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_verbs);
+EXPORT_SYMBOL_GPL(snd_hda_add_verbs);
 
 void snd_hda_apply_verbs(struct hda_codec *codec)
 {
@@ -743,7 +743,7 @@ void snd_hda_apply_verbs(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, *v);
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_apply_verbs);
+EXPORT_SYMBOL_GPL(snd_hda_apply_verbs);
 
 void snd_hda_apply_pincfgs(struct hda_codec *codec,
 			   const struct hda_pintbl *cfg)
@@ -751,7 +751,7 @@ void snd_hda_apply_pincfgs(struct hda_codec *codec,
 	for (; cfg->nid; cfg++)
 		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 }
-EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs);
+EXPORT_SYMBOL_GPL(snd_hda_apply_pincfgs);
 
 static void set_pin_targets(struct hda_codec *codec,
 			    const struct hda_pintbl *cfg)
@@ -822,7 +822,7 @@ void snd_hda_apply_fixup(struct hda_codec *codec, int action)
 	if (codec->fixup_list)
 		apply_fixup(codec, codec->fixup_id, action, 0);
 }
-EXPORT_SYMBOL_HDA(snd_hda_apply_fixup);
+EXPORT_SYMBOL_GPL(snd_hda_apply_fixup);
 
 void snd_hda_pick_fixup(struct hda_codec *codec,
 			const struct hda_model_fixup *models,
@@ -880,4 +880,4 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 		codec->fixup_name = name;
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_pick_fixup);
+EXPORT_SYMBOL_GPL(snd_hda_pick_fixup);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 98bce9830be0..0589b39cda6e 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -194,7 +194,7 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device);
+EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device);
 
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 {
@@ -231,7 +231,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device);
+EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device);
 
 void snd_hda_detach_beep_device(struct hda_codec *codec)
 {
@@ -243,7 +243,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 		kfree(beep);
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
+EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device);
 
 static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
 {
@@ -265,7 +265,7 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 	}
 	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get_beep);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get_beep);
 
 int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
@@ -288,4 +288,4 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 		return 0;
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2e090c8d7144..ab2a991d916f 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -84,7 +84,7 @@ int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset)
 	mutex_unlock(&preset_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_codec_preset);
+EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset);
 
 int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset)
 {
@@ -93,7 +93,7 @@ int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset)
 	mutex_unlock(&preset_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset);
+EXPORT_SYMBOL_GPL(snd_hda_delete_codec_preset);
 
 #ifdef CONFIG_PM
 #define codec_in_pm(codec)	((codec)->in_pm)
@@ -152,7 +152,7 @@ const char *snd_hda_get_jack_location(u32 cfg)
 	}
 	return "UNKNOWN";
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_jack_location);
+EXPORT_SYMBOL_GPL(snd_hda_get_jack_location);
 
 /**
  * snd_hda_get_jack_connectivity - Give a connectivity string of the jack
@@ -167,7 +167,7 @@ const char *snd_hda_get_jack_connectivity(u32 cfg)
 
 	return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3];
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity);
+EXPORT_SYMBOL_GPL(snd_hda_get_jack_connectivity);
 
 /**
  * snd_hda_get_jack_type - Give a type string of the jack
@@ -188,7 +188,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
 	return jack_types[(cfg & AC_DEFCFG_DEVICE)
 				>> AC_DEFCFG_DEVICE_SHIFT];
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_jack_type);
+EXPORT_SYMBOL_GPL(snd_hda_get_jack_type);
 
 /*
  * Compose a 32bit command word to be sent to the HD-audio controller
@@ -284,7 +284,7 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 		return -1;
 	return res;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_read);
+EXPORT_SYMBOL_GPL(snd_hda_codec_read);
 
 /**
  * snd_hda_codec_write - send a single command without waiting for response
@@ -306,7 +306,7 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
 	return codec_exec_verb(codec, cmd, flags,
 			       codec->bus->sync_write ? &res : NULL);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_write);
+EXPORT_SYMBOL_GPL(snd_hda_codec_write);
 
 /**
  * snd_hda_sequence_write - sequence writes
@@ -321,7 +321,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq)
 	for (; seq->nid; seq++)
 		snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param);
 }
-EXPORT_SYMBOL_HDA(snd_hda_sequence_write);
+EXPORT_SYMBOL_GPL(snd_hda_sequence_write);
 
 /**
  * snd_hda_get_sub_nodes - get the range of sub nodes
@@ -343,7 +343,7 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
 	*start_id = (parm >> 16) & 0x7fff;
 	return (int)(parm & 0x7fff);
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes);
+EXPORT_SYMBOL_GPL(snd_hda_get_sub_nodes);
 
 /* connection list element */
 struct hda_conn_list {
@@ -453,7 +453,7 @@ int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
 		added = true;
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
+EXPORT_SYMBOL_GPL(snd_hda_get_conn_list);
 
 /**
  * snd_hda_get_connections - copy connection list
@@ -485,7 +485,7 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 
 	return len;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_connections);
+EXPORT_SYMBOL_GPL(snd_hda_get_connections);
 
 /* return CONNLIST_LEN parameter of the given widget */
 static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid)
@@ -634,7 +634,7 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int len,
 
 	return add_conn_list(codec, nid, len, list);
 }
-EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
+EXPORT_SYMBOL_GPL(snd_hda_override_conn_list);
 
 /**
  * snd_hda_get_conn_index - get the connection index of the given NID
@@ -673,7 +673,7 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 	}
 	return -1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
+EXPORT_SYMBOL_GPL(snd_hda_get_conn_index);
 
 
 /* return DEVLIST_LEN parameter of the given widget */
@@ -769,7 +769,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_queue_unsol_event);
+EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event);
 
 /*
  * process queued unsolicited events
@@ -930,7 +930,7 @@ int snd_hda_bus_new(struct snd_card *card,
 		*busp = bus;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_bus_new);
+EXPORT_SYMBOL_GPL(snd_hda_bus_new);
 
 #ifdef CONFIG_SND_HDA_GENERIC
 #define is_generic_config(codec) \
@@ -1170,7 +1170,7 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec,
 {
 	return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg);
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_pincfg);
 
 /**
  * snd_hda_codec_get_pincfg - Obtain a pin-default configuration
@@ -1205,7 +1205,7 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
 		return pin->cfg;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg);
+EXPORT_SYMBOL_GPL(snd_hda_codec_get_pincfg);
 
 /* remember the current pinctl target value */
 int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid,
@@ -1219,7 +1219,7 @@ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid,
 	pin->target = val;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_set_pin_target);
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_pin_target);
 
 /* return the current pinctl target value */
 int snd_hda_codec_get_pin_target(struct hda_codec *codec, hda_nid_t nid)
@@ -1231,7 +1231,7 @@ int snd_hda_codec_get_pin_target(struct hda_codec *codec, hda_nid_t nid)
 		return 0;
 	return pin->target;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_get_pin_target);
+EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target);
 
 /**
  * snd_hda_shutup_pins - Shut up all pins
@@ -1256,7 +1256,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec)
 	}
 	codec->pins_shutup = 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
+EXPORT_SYMBOL_GPL(snd_hda_shutup_pins);
 
 #ifdef CONFIG_PM
 /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
@@ -1544,7 +1544,7 @@ int snd_hda_codec_new(struct hda_bus *bus,
 	snd_hda_codec_free(codec);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_new);
+EXPORT_SYMBOL_GPL(snd_hda_codec_new);
 
 int snd_hda_codec_update_widgets(struct hda_codec *codec)
 {
@@ -1567,7 +1567,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec)
 
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_update_widgets);
+EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets);
 
 
 #ifdef CONFIG_SND_HDA_CODEC_HDMI
@@ -1651,7 +1651,7 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 			 "%s %s", codec->vendor_name, codec->chip_name);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
+EXPORT_SYMBOL_GPL(snd_hda_codec_configure);
 
 /* update the stream-id if changed */
 static void update_pcm_stream_id(struct hda_codec *codec,
@@ -1738,7 +1738,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 		}
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream);
+EXPORT_SYMBOL_GPL(snd_hda_codec_setup_stream);
 
 static void really_cleanup_stream(struct hda_codec *codec,
 				  struct hda_cvt_setup *q);
@@ -1773,7 +1773,7 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
 			p->active = 0;
 	}
 }
-EXPORT_SYMBOL_HDA(__snd_hda_codec_cleanup_stream);
+EXPORT_SYMBOL_GPL(__snd_hda_codec_cleanup_stream);
 
 static void really_cleanup_stream(struct hda_codec *codec,
 				  struct hda_cvt_setup *q)
@@ -1961,7 +1961,7 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
 			       HDA_HASH_KEY(nid, direction, 0),
 			       read_amp_cap);
 }
-EXPORT_SYMBOL_HDA(query_amp_caps);
+EXPORT_SYMBOL_GPL(query_amp_caps);
 
 /**
  * snd_hda_override_amp_caps - Override the AMP capabilities
@@ -1981,7 +1981,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 {
 	return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps);
 }
-EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
+EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps);
 
 static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid,
 				 int dir)
@@ -2005,7 +2005,7 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
 	return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid),
 			       read_pin_cap);
 }
-EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
+EXPORT_SYMBOL_GPL(snd_hda_query_pin_caps);
 
 /**
  * snd_hda_override_pin_caps - Override the pin capabilities
@@ -2022,7 +2022,7 @@ int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
 {
 	return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps);
 }
-EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
+EXPORT_SYMBOL_GPL(snd_hda_override_pin_caps);
 
 /* read or sync the hash value with the current value;
  * call within hash_mutex
@@ -2103,7 +2103,7 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
 	mutex_unlock(&codec->hash_mutex);
 	return val;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
+EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read);
 
 static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 			    int direction, int idx, int mask, int val,
@@ -2155,7 +2155,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 {
 	return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update);
+EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update);
 
 /**
  * snd_hda_codec_amp_stereo - update the AMP stereo values
@@ -2181,7 +2181,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
 						idx, mask, val);
 	return ret;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo);
+EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo);
 
 /* Works like snd_hda_codec_amp_update() but it writes the value only at
  * the first access.  If the amp was already initialized / updated beforehand,
@@ -2192,7 +2192,7 @@ int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 {
 	return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_amp_init);
+EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init);
 
 int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
 				  int dir, int idx, int mask, int val)
@@ -2206,7 +2206,7 @@ int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
 					      idx, mask, val);
 	return ret;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_amp_init_stereo);
+EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init_stereo);
 
 /**
  * snd_hda_codec_resume_amp - Resume all AMP commands from the cache
@@ -2249,7 +2249,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
 	}
 	mutex_unlock(&codec->hash_mutex);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
+EXPORT_SYMBOL_GPL(snd_hda_codec_resume_amp);
 
 static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir,
 			     unsigned int ofs)
@@ -2289,7 +2289,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_info);
 
 
 static inline unsigned int
@@ -2346,7 +2346,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 		*valp = read_amp_value(codec, nid, 1, dir, idx, ofs);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get);
 
 /**
  * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume
@@ -2376,7 +2376,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 	snd_hda_power_down(codec);
 	return change;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put);
 
 /**
  * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume
@@ -2414,7 +2414,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 		return -EFAULT;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_tlv);
 
 /**
  * snd_hda_set_vmaster_tlv - Set TLV for a virtual master control
@@ -2442,7 +2442,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
 	tlv[2] = -nums * step;
 	tlv[3] = step;
 }
-EXPORT_SYMBOL_HDA(snd_hda_set_vmaster_tlv);
+EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv);
 
 /* find a mixer control element with the given name */
 static struct snd_kcontrol *
@@ -2471,7 +2471,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 {
 	return find_mixer_ctl(codec, name, 0, 0);
 }
-EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl);
+EXPORT_SYMBOL_GPL(snd_hda_find_mixer_ctl);
 
 static int find_empty_mixer_ctl_idx(struct hda_codec *codec, const char *name,
 				    int start_idx)
@@ -2531,7 +2531,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid,
 	item->flags = flags;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_ctl_add);
+EXPORT_SYMBOL_GPL(snd_hda_ctl_add);
 
 /**
  * snd_hda_add_nid - Assign a NID to a control element
@@ -2562,7 +2562,7 @@ int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl,
 	       kctl->id.name, kctl->id.index, index);
 	return -EINVAL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_nid);
+EXPORT_SYMBOL_GPL(snd_hda_add_nid);
 
 /**
  * snd_hda_ctls_clear - Clear all controls assigned to the given codec
@@ -2613,7 +2613,7 @@ int snd_hda_lock_devices(struct hda_bus *bus)
 	spin_unlock(&card->files_lock);
 	return -EINVAL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_lock_devices);
+EXPORT_SYMBOL_GPL(snd_hda_lock_devices);
 
 void snd_hda_unlock_devices(struct hda_bus *bus)
 {
@@ -2624,7 +2624,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus)
 	card->shutdown = 0;
 	spin_unlock(&card->files_lock);
 }
-EXPORT_SYMBOL_HDA(snd_hda_unlock_devices);
+EXPORT_SYMBOL_GPL(snd_hda_unlock_devices);
 
 /**
  * snd_hda_codec_reset - Clear all objects assigned to the codec
@@ -2848,7 +2848,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 		*ctl_ret = kctl;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster);
+EXPORT_SYMBOL_GPL(__snd_hda_add_vmaster);
 
 /*
  * mute-LED control using vmaster
@@ -2925,7 +2925,7 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec,
 		return -ENOMEM;
 	return snd_hda_ctl_add(codec, 0, kctl);
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook);
+EXPORT_SYMBOL_GPL(snd_hda_add_vmaster_hook);
 
 /*
  * Call the hook with the current value for synchronization
@@ -2949,7 +2949,7 @@ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
 		break;
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook);
+EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook);
 
 
 /**
@@ -2969,7 +2969,7 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.integer.max = 1;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info);
 
 /**
  * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch
@@ -2995,7 +2995,7 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
 			 HDA_AMP_MUTE) ? 0 : 1;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get);
 
 /**
  * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch
@@ -3029,7 +3029,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 	snd_hda_power_down(codec);
 	return change;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put);
 
 /*
  * bound volume controls
@@ -3061,7 +3061,7 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get);
 
 /**
  * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control
@@ -3091,7 +3091,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->control_mutex);
 	return err < 0 ? err : change;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put);
 
 /**
  * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control
@@ -3114,7 +3114,7 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info);
 
 /**
  * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control
@@ -3137,7 +3137,7 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get);
 
 /**
  * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control
@@ -3166,7 +3166,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
 	mutex_unlock(&codec->control_mutex);
 	return err < 0 ? err : change;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put);
 
 /**
  * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control
@@ -3189,7 +3189,7 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv);
+EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_tlv);
 
 struct hda_ctl_ops snd_hda_bind_vol = {
 	.info = snd_hda_mixer_amp_volume_info,
@@ -3197,7 +3197,7 @@ struct hda_ctl_ops snd_hda_bind_vol = {
 	.put = snd_hda_mixer_amp_volume_put,
 	.tlv = snd_hda_mixer_amp_tlv
 };
-EXPORT_SYMBOL_HDA(snd_hda_bind_vol);
+EXPORT_SYMBOL_GPL(snd_hda_bind_vol);
 
 struct hda_ctl_ops snd_hda_bind_sw = {
 	.info = snd_hda_mixer_amp_switch_info,
@@ -3205,7 +3205,7 @@ struct hda_ctl_ops snd_hda_bind_sw = {
 	.put = snd_hda_mixer_amp_switch_put,
 	.tlv = snd_hda_mixer_amp_tlv
 };
-EXPORT_SYMBOL_HDA(snd_hda_bind_sw);
+EXPORT_SYMBOL_GPL(snd_hda_bind_sw);
 
 /*
  * SPDIF out controls
@@ -3509,7 +3509,7 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec,
 	spdif->status = convert_to_spdif_status(spdif->ctls);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_create_dig_out_ctls);
+EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls);
 
 /* get the hda_spdif_out entry from the given NID
  * call within spdif_mutex lock
@@ -3526,7 +3526,7 @@ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
 	}
 	return NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_spdif_out_of_nid);
+EXPORT_SYMBOL_GPL(snd_hda_spdif_out_of_nid);
 
 void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 {
@@ -3537,7 +3537,7 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
 	spdif->nid = (u16)-1;
 	mutex_unlock(&codec->spdif_mutex);
 }
-EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_unassign);
+EXPORT_SYMBOL_GPL(snd_hda_spdif_ctls_unassign);
 
 void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
 {
@@ -3553,7 +3553,7 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
 	}
 	mutex_unlock(&codec->spdif_mutex);
 }
-EXPORT_SYMBOL_HDA(snd_hda_spdif_ctls_assign);
+EXPORT_SYMBOL_GPL(snd_hda_spdif_ctls_assign);
 
 /*
  * SPDIF sharing with analog output
@@ -3601,7 +3601,7 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
 	/* ATTENTION: here mout is passed as private_data, instead of codec */
 	return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl);
 }
-EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
+EXPORT_SYMBOL_GPL(snd_hda_create_spdif_share_sw);
 
 /*
  * SPDIF input
@@ -3709,7 +3709,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 		AC_DIG1_ENABLE;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls);
+EXPORT_SYMBOL_GPL(snd_hda_create_spdif_in_ctls);
 
 /*
  * command cache
@@ -3760,7 +3760,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
 	mutex_unlock(&codec->bus->cmd_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache);
+EXPORT_SYMBOL_GPL(snd_hda_codec_write_cache);
 
 /**
  * snd_hda_codec_update_cache - check cache and write the cmd only when needed
@@ -3795,7 +3795,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid,
 	mutex_unlock(&codec->bus->cmd_mutex);
 	return snd_hda_codec_write_cache(codec, nid, flags, verb, parm);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache);
+EXPORT_SYMBOL_GPL(snd_hda_codec_update_cache);
 
 /**
  * snd_hda_codec_resume_cache - Resume the all commands from the cache
@@ -3827,7 +3827,7 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec)
 	}
 	mutex_unlock(&codec->hash_mutex);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_resume_cache);
+EXPORT_SYMBOL_GPL(snd_hda_codec_resume_cache);
 
 /**
  * snd_hda_sequence_write_cache - sequence writes with caching
@@ -3845,7 +3845,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec,
 		snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
 					  seq->param);
 }
-EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache);
+EXPORT_SYMBOL_GPL(snd_hda_sequence_write_cache);
 
 /**
  * snd_hda_codec_flush_cache - Execute all pending (cached) amps / verbs
@@ -3856,7 +3856,7 @@ void snd_hda_codec_flush_cache(struct hda_codec *codec)
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_flush_cache);
+EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache);
 
 void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
 				    unsigned int power_state)
@@ -3878,7 +3878,7 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
 				    state);
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all);
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_power_to_all);
 
 /*
  *  supported power states check
@@ -3937,7 +3937,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
 	}
 	return power_state;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_eapd_power_filter);
+EXPORT_SYMBOL_GPL(snd_hda_codec_eapd_power_filter);
 
 /*
  * set power state of the codec, and return the power state
@@ -4124,7 +4124,7 @@ int snd_hda_build_controls(struct hda_bus *bus)
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_build_controls);
+EXPORT_SYMBOL_GPL(snd_hda_build_controls);
 
 /*
  * add standard channel maps if not specified
@@ -4297,7 +4297,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 
 	return val;
 }
-EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
+EXPORT_SYMBOL_GPL(snd_hda_calc_stream_format);
 
 static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid,
 				  int dir)
@@ -4443,7 +4443,7 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_query_supported_pcm);
+EXPORT_SYMBOL_GPL(snd_hda_query_supported_pcm);
 
 /**
  * snd_hda_is_supported_format - Check the validity of the format
@@ -4510,7 +4510,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 
 	return 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_is_supported_format);
+EXPORT_SYMBOL_GPL(snd_hda_is_supported_format);
 
 /*
  * PCM stuff
@@ -4588,7 +4588,7 @@ int snd_hda_codec_prepare(struct hda_codec *codec,
 	mutex_unlock(&codec->bus->prepare_mutex);
 	return ret;
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_prepare);
+EXPORT_SYMBOL_GPL(snd_hda_codec_prepare);
 
 void snd_hda_codec_cleanup(struct hda_codec *codec,
 			   struct hda_pcm_stream *hinfo,
@@ -4598,7 +4598,7 @@ void snd_hda_codec_cleanup(struct hda_codec *codec,
 	hinfo->ops.cleanup(hinfo, codec, substream);
 	mutex_unlock(&codec->bus->prepare_mutex);
 }
-EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup);
+EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup);
 
 /* global */
 const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = {
@@ -4756,7 +4756,7 @@ int snd_hda_build_pcms(struct hda_bus *bus)
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_build_pcms);
+EXPORT_SYMBOL_GPL(snd_hda_build_pcms);
 
 /**
  * snd_hda_check_board_config - compare the current codec with the config table
@@ -4812,7 +4812,7 @@ int snd_hda_check_board_config(struct hda_codec *codec,
 	}
 	return -1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_check_board_config);
+EXPORT_SYMBOL_GPL(snd_hda_check_board_config);
 
 /**
  * snd_hda_check_board_codec_sid_config - compare the current codec
@@ -4873,7 +4873,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
 	}
 	return -1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config);
+EXPORT_SYMBOL_GPL(snd_hda_check_board_codec_sid_config);
 
 /**
  * snd_hda_add_new_ctls - create controls from the array
@@ -4923,7 +4923,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec,
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls);
+EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls);
 
 #ifdef CONFIG_PM
 static void hda_power_work(struct work_struct *work)
@@ -5045,7 +5045,7 @@ void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait)
 		__snd_hda_power_down(codec);
 	spin_unlock(&codec->power_lock);
 }
-EXPORT_SYMBOL_HDA(snd_hda_power_save);
+EXPORT_SYMBOL_GPL(snd_hda_power_save);
 
 /**
  * snd_hda_check_amp_list_power - Check the amp list and update the power
@@ -5095,7 +5095,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
+EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power);
 #endif
 
 /*
@@ -5119,7 +5119,7 @@ int snd_hda_ch_mode_info(struct hda_codec *codec,
 		chmode[uinfo->value.enumerated.item].channels);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info);
+EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info);
 
 /**
  * snd_hda_ch_mode_get - Get callback helper for the channel mode enum
@@ -5140,7 +5140,7 @@ int snd_hda_ch_mode_get(struct hda_codec *codec,
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get);
+EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get);
 
 /**
  * snd_hda_ch_mode_put - Put callback helper for the channel mode enum
@@ -5164,7 +5164,7 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
 		snd_hda_sequence_write_cache(codec, chmode[mode].sequence);
 	return 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put);
+EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put);
 
 /*
  * input MUX helper
@@ -5189,7 +5189,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux,
 	strcpy(uinfo->value.enumerated.name, imux->items[index].label);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_input_mux_info);
+EXPORT_SYMBOL_GPL(snd_hda_input_mux_info);
 
 /**
  * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum
@@ -5214,7 +5214,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 	*cur_val = idx;
 	return 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_input_mux_put);
+EXPORT_SYMBOL_GPL(snd_hda_input_mux_put);
 
 
 /*
@@ -5243,7 +5243,7 @@ int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
 	       texts[uinfo->value.enumerated.item]);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_enum_helper_info);
+EXPORT_SYMBOL_GPL(snd_hda_enum_helper_info);
 
 /*
  * Multi-channel / digital-out PCM helper functions
@@ -5309,7 +5309,7 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus)
 			codec->patch_ops.reboot_notify(codec);
 	}
 }
-EXPORT_SYMBOL_HDA(snd_hda_bus_reboot_notify);
+EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify);
 
 /**
  * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode
@@ -5325,7 +5325,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open);
 
 /**
  * snd_hda_multi_out_dig_prepare - prepare the digital out stream
@@ -5341,7 +5341,7 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare);
 
 /**
  * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream
@@ -5354,7 +5354,7 @@ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_cleanup);
 
 /**
  * snd_hda_multi_out_dig_close - release the digital out stream
@@ -5367,7 +5367,7 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close);
 
 /**
  * snd_hda_multi_out_analog_open - open analog outputs
@@ -5417,7 +5417,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
 	return snd_pcm_hw_constraint_step(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open);
 
 /**
  * snd_hda_multi_out_analog_prepare - Preapre the analog outputs.
@@ -5494,7 +5494,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare);
 
 /**
  * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out
@@ -5525,7 +5525,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
 	mutex_unlock(&codec->spdif_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
+EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup);
 
 /**
  * snd_hda_get_default_vref - Get the default (mic) VREF pin bits
@@ -5552,7 +5552,7 @@ unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin)
 		return AC_PINCTL_VREF_GRD;
 	return AC_PINCTL_VREF_HIZ;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_default_vref);
+EXPORT_SYMBOL_GPL(snd_hda_get_default_vref);
 
 /* correct the pin ctl value for matching with the pin cap */
 unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec,
@@ -5603,7 +5603,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec,
 
 	return val;
 }
-EXPORT_SYMBOL_HDA(snd_hda_correct_pin_ctl);
+EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl);
 
 int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int val, bool cached)
@@ -5617,7 +5617,7 @@ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
 		return snd_hda_codec_write(codec, pin, 0,
 					   AC_VERB_SET_PIN_WIDGET_CONTROL, val);
 }
-EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl);
+EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl);
 
 /**
  * snd_hda_add_imux_item - Add an item to input_mux
@@ -5651,7 +5651,7 @@ int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
 	imux->num_items++;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_imux_item);
+EXPORT_SYMBOL_GPL(snd_hda_add_imux_item);
 
 
 #ifdef CONFIG_PM
@@ -5697,7 +5697,7 @@ int snd_hda_suspend(struct hda_bus *bus)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_suspend);
+EXPORT_SYMBOL_GPL(snd_hda_suspend);
 
 /**
  * snd_hda_resume - resume the codecs
@@ -5722,7 +5722,7 @@ int snd_hda_resume(struct hda_bus *bus)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_resume);
+EXPORT_SYMBOL_GPL(snd_hda_resume);
 #endif /* CONFIG_PM */
 
 /*
@@ -5756,7 +5756,7 @@ void *snd_array_new(struct snd_array *array)
 	}
 	return snd_array_elem(array, array->used++);
 }
-EXPORT_SYMBOL_HDA(snd_array_new);
+EXPORT_SYMBOL_GPL(snd_array_new);
 
 /**
  * snd_array_free - free the given array elements
@@ -5769,7 +5769,7 @@ void snd_array_free(struct snd_array *array)
 	array->alloced = 0;
 	array->list = NULL;
 }
-EXPORT_SYMBOL_HDA(snd_array_free);
+EXPORT_SYMBOL_GPL(snd_array_free);
 
 /**
  * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer
@@ -5790,7 +5790,7 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen)
 
 	buf[j] = '\0'; /* necessary when j == 0 */
 }
-EXPORT_SYMBOL_HDA(snd_print_pcm_bits);
+EXPORT_SYMBOL_GPL(snd_print_pcm_bits);
 
 MODULE_DESCRIPTION("HDA codec core");
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8cebdcfdcfdc..ce004326d54b 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -48,7 +48,7 @@ int snd_hda_gen_spec_init(struct hda_gen_spec *spec)
 	mutex_init(&spec->pcm_mutex);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_spec_init);
+EXPORT_SYMBOL_GPL(snd_hda_gen_spec_init);
 
 struct snd_kcontrol_new *
 snd_hda_gen_add_kctl(struct hda_gen_spec *spec, const char *name,
@@ -66,7 +66,7 @@ snd_hda_gen_add_kctl(struct hda_gen_spec *spec, const char *name,
 		return NULL;
 	return knew;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_add_kctl);
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_kctl);
 
 static void free_kctls(struct hda_gen_spec *spec)
 {
@@ -87,7 +87,7 @@ void snd_hda_gen_spec_free(struct hda_gen_spec *spec)
 	snd_array_free(&spec->paths);
 	snd_array_free(&spec->loopback_list);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_spec_free);
+EXPORT_SYMBOL_GPL(snd_hda_gen_spec_free);
 
 /*
  * store user hints
@@ -267,7 +267,7 @@ struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec,
 {
 	return get_nid_path(codec, from_nid, to_nid, 0);
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_nid_path);
+EXPORT_SYMBOL_GPL(snd_hda_get_nid_path);
 
 /* get the index number corresponding to the path instance;
  * the index starts from 1, for easier checking the invalid value
@@ -285,7 +285,7 @@ int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path)
 		return 0;
 	return idx + 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_path_idx);
+EXPORT_SYMBOL_GPL(snd_hda_get_path_idx);
 
 /* get the path instance corresponding to the given index number */
 struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx)
@@ -296,7 +296,7 @@ struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx)
 		return NULL;
 	return snd_array_elem(&spec->paths, idx - 1);
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_path_from_idx);
+EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx);
 
 /* check whether the given DAC is already found in any existing paths */
 static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid)
@@ -433,7 +433,7 @@ bool snd_hda_parse_nid_path(struct hda_codec *codec, hda_nid_t from_nid,
 	}
 	return false;
 }
-EXPORT_SYMBOL_HDA(snd_hda_parse_nid_path);
+EXPORT_SYMBOL_GPL(snd_hda_parse_nid_path);
 
 /*
  * parse the path between the given NIDs and add to the path list.
@@ -464,7 +464,7 @@ snd_hda_add_new_path(struct hda_codec *codec, hda_nid_t from_nid,
 	spec->paths.used--;
 	return NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_add_new_path);
+EXPORT_SYMBOL_GPL(snd_hda_add_new_path);
 
 /* clear the given path as invalid so that it won't be picked up later */
 static void invalidate_nid_path(struct hda_codec *codec, int idx)
@@ -774,7 +774,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
 	if (enable)
 		path->active = true;
 }
-EXPORT_SYMBOL_HDA(snd_hda_activate_path);
+EXPORT_SYMBOL_GPL(snd_hda_activate_path);
 
 /* if the given path is inactive, put widgets into D3 (only if suitable) */
 static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
@@ -1150,7 +1150,7 @@ const struct badness_table hda_main_out_badness = {
 	.shared_clfe = BAD_SHARED_CLFE,
 	.shared_surr_main = BAD_SHARED_SURROUND,
 };
-EXPORT_SYMBOL_HDA(hda_main_out_badness);
+EXPORT_SYMBOL_GPL(hda_main_out_badness);
 
 const struct badness_table hda_extra_out_badness = {
 	.no_primary_dac = BAD_NO_DAC,
@@ -1160,7 +1160,7 @@ const struct badness_table hda_extra_out_badness = {
 	.shared_clfe = BAD_SHARED_EXTRA_SURROUND,
 	.shared_surr_main = BAD_NO_EXTRA_SURR_DAC,
 };
-EXPORT_SYMBOL_HDA(hda_extra_out_badness);
+EXPORT_SYMBOL_GPL(hda_extra_out_badness);
 
 /* get the DAC of the primary output corresponding to the given array index */
 static hda_nid_t get_primary_out(struct hda_codec *codec, int idx)
@@ -3921,7 +3921,7 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 		    spec->autocfg.line_out_pins, paths, on);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
+EXPORT_SYMBOL_GPL(snd_hda_gen_update_outputs);
 
 static void call_update_outputs(struct hda_codec *codec)
 {
@@ -3954,7 +3954,7 @@ void snd_hda_gen_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
 		return;
 	call_update_outputs(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_hp_automute);
+EXPORT_SYMBOL_GPL(snd_hda_gen_hp_automute);
 
 /* standard line-out-automute helper */
 void snd_hda_gen_line_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
@@ -3974,7 +3974,7 @@ void snd_hda_gen_line_automute(struct hda_codec *codec, struct hda_jack_tbl *jac
 		return;
 	call_update_outputs(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_line_automute);
+EXPORT_SYMBOL_GPL(snd_hda_gen_line_automute);
 
 /* standard mic auto-switch helper */
 void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *jack)
@@ -3997,7 +3997,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 	}
 	mux_select(codec, 0, spec->am_entry[0].idx);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch);
+EXPORT_SYMBOL_GPL(snd_hda_gen_mic_autoswitch);
 
 /* call appropriate hooks */
 static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack)
@@ -4513,7 +4513,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 
 	return 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_parse_auto_config);
+EXPORT_SYMBOL_GPL(snd_hda_gen_parse_auto_config);
 
 
 /*
@@ -4595,7 +4595,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_build_controls);
+EXPORT_SYMBOL_GPL(snd_hda_gen_build_controls);
 
 
 /*
@@ -5128,7 +5128,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec)
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_build_pcms);
+EXPORT_SYMBOL_GPL(snd_hda_gen_build_pcms);
 
 
 /*
@@ -5339,7 +5339,7 @@ int snd_hda_gen_init(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_init);
+EXPORT_SYMBOL_GPL(snd_hda_gen_init);
 
 /*
  * free the generic spec;
@@ -5352,7 +5352,7 @@ void snd_hda_gen_free(struct hda_codec *codec)
 	kfree(codec->spec);
 	codec->spec = NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_free);
+EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
 #ifdef CONFIG_PM
 /*
@@ -5364,7 +5364,7 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 	struct hda_gen_spec *spec = codec->spec;
 	return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
 }
-EXPORT_SYMBOL_HDA(snd_hda_gen_check_power_status);
+EXPORT_SYMBOL_GPL(snd_hda_gen_check_power_status);
 #endif
 
 
@@ -5409,7 +5409,7 @@ error:
 	snd_hda_gen_free(codec);
 	return err;
 }
-EXPORT_SYMBOL_HDA(snd_hda_parse_generic_codec);
+EXPORT_SYMBOL_GPL(snd_hda_parse_generic_codec);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Generic HD-audio codec parser");
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index fe0bda19de15..0ec6fb1e9846 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -616,7 +616,7 @@ const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
 	struct hda_hint *hint = get_hint(codec, key);
 	return hint ? hint->val : NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_hint);
+EXPORT_SYMBOL_GPL(snd_hda_get_hint);
 
 int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 {
@@ -642,7 +642,7 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 	mutex_unlock(&codec->user_mutex);
 	return ret;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
+EXPORT_SYMBOL_GPL(snd_hda_get_bool_hint);
 
 int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 {
@@ -663,7 +663,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 	mutex_unlock(&codec->user_mutex);
 	return ret;
 }
-EXPORT_SYMBOL_HDA(snd_hda_get_int_hint);
+EXPORT_SYMBOL_GPL(snd_hda_get_int_hint);
 #endif /* CONFIG_SND_HDA_RECONFIG */
 
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
@@ -851,5 +851,5 @@ int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf)
 	}
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_load_patch);
+EXPORT_SYMBOL_GPL(snd_hda_load_patch);
 #endif /* CONFIG_SND_HDA_PATCH_LOADER */
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index afe594411a56..9746d73cec52 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -34,7 +34,7 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
 		return false;
 	return true;
 }
-EXPORT_SYMBOL_HDA(is_jack_detectable);
+EXPORT_SYMBOL_GPL(is_jack_detectable);
 
 /* execute pin sense measurement */
 static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
@@ -71,7 +71,7 @@ snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid)
 			return jack;
 	return NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_get);
+EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get);
 
 /**
  * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag
@@ -89,7 +89,7 @@ snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag)
 			return jack;
 	return NULL;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_get_from_tag);
+EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag);
 
 /**
  * snd_hda_jack_tbl_new - create a jack-table entry for the given NID
@@ -108,7 +108,7 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid)
 	jack->tag = codec->jacktbl.used;
 	return jack;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new);
+EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_new);
 
 void snd_hda_jack_tbl_clear(struct hda_codec *codec)
 {
@@ -172,7 +172,7 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec)
 		if (jack->nid)
 			jack->jack_dirty = 1;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_set_dirty_all);
+EXPORT_SYMBOL_GPL(snd_hda_jack_set_dirty_all);
 
 /**
  * snd_hda_pin_sense - execute pin sense measurement
@@ -191,7 +191,7 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 	}
 	return read_pin_sense(codec, nid);
 }
-EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
+EXPORT_SYMBOL_GPL(snd_hda_pin_sense);
 
 /**
  * snd_hda_jack_detect_state - query pin Presence Detect status
@@ -211,7 +211,7 @@ int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
 	else
 		return HDA_JACK_NOT_PRESENT;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
+EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
@@ -236,14 +236,14 @@ int snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid,
 					 AC_VERB_SET_UNSOLICITED_ENABLE,
 					 AC_USRSP_EN | jack->tag);
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable_callback);
+EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback);
 
 int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
 			       unsigned char action)
 {
 	return snd_hda_jack_detect_enable_callback(codec, nid, action, NULL);
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
+EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable);
 
 /**
  * snd_hda_jack_set_gating_jack - Set gating jack.
@@ -264,7 +264,7 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_set_gating_jack);
+EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack);
 
 /**
  * snd_hda_jack_report_sync - sync the states of all jacks and report if changed
@@ -297,7 +297,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec)
 #endif
 		}
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync);
+EXPORT_SYMBOL_GPL(snd_hda_jack_report_sync);
 
 #ifdef CONFIG_SND_HDA_INPUT_JACK
 /* guess the jack type from the pin-config */
@@ -377,7 +377,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 {
 	return __snd_hda_jack_add_kctl(codec, nid, name, idx, false);
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
+EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl);
 
 /* get the unique index number for the given kctl name */
 static int get_unique_index(struct hda_codec *codec, const char *name, int idx)
@@ -493,7 +493,7 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
 		return err;
 	return 0;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctls);
+EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctls);
 
 static void call_jack_callback(struct hda_codec *codec,
 			       struct hda_jack_tbl *jack)
@@ -521,7 +521,7 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res)
 	call_jack_callback(codec, event);
 	snd_hda_jack_report_sync(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_unsol_event);
+EXPORT_SYMBOL_GPL(snd_hda_jack_unsol_event);
 
 void snd_hda_jack_poll_all(struct hda_codec *codec)
 {
@@ -542,5 +542,5 @@ void snd_hda_jack_poll_all(struct hda_codec *codec)
 	if (changes)
 		snd_hda_jack_report_sync(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_poll_all);
+EXPORT_SYMBOL_GPL(snd_hda_jack_poll_all);
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 977db17db26c..59b83f4085a8 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3217,7 +3217,7 @@ int snd_hda_parse_hdmi_codec(struct hda_codec *codec)
 {
 	return patch_generic_hdmi(codec);
 }
-EXPORT_SYMBOL_HDA(snd_hda_parse_hdmi_codec);
+EXPORT_SYMBOL_GPL(snd_hda_parse_hdmi_codec);
 
 /*
  * patch entries
-- 
cgit v1.2.2


From d8f66c71d547538e1c61622139455dd231238aef Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Dec 2013 15:09:14 +0100
Subject: ALSA: hda - Add warning texts when codec driver Kconfig doesn't match

When a Kconfig of a codec driver doesn't match with the controller
(CONFIG_SND_HDA_INTEL), it'll result in the non-working automatic
probing.  Unfortunately kbuild can't give such a restriction, but at
least, it's possible to show a warning if such a condition is found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig | 36 ++++++++++++++++++++++++++++++++++++
 1 file changed, 36 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index b0c0481c3339..0e53634dbbd8 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -93,6 +93,9 @@ config SND_HDA_CODEC_REALTEK
 	  Say Y or M here to include Realtek HD-audio codec support in
 	  snd-hda-intel driver, such as ALC880.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_REALTEK=m
+
 config SND_HDA_CODEC_ANALOG
 	tristate "Build Analog Device HD-audio codec support"
 	select SND_HDA_GENERIC
@@ -100,6 +103,9 @@ config SND_HDA_CODEC_ANALOG
 	  Say Y or M here to include Analog Device HD-audio codec support in
 	  snd-hda-intel driver, such as AD1986A.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_ANALOG=m
+
 config SND_HDA_CODEC_SIGMATEL
 	tristate "Build IDT/Sigmatel HD-audio codec support"
 	select SND_HDA_GENERIC
@@ -107,6 +113,9 @@ config SND_HDA_CODEC_SIGMATEL
 	  Say Y or M here to include IDT (Sigmatel) HD-audio codec support in
 	  snd-hda-intel driver, such as STAC9200.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SIGMATEL=m
+
 config SND_HDA_CODEC_VIA
 	tristate "Build VIA HD-audio codec support"
 	select SND_HDA_GENERIC
@@ -114,6 +123,9 @@ config SND_HDA_CODEC_VIA
 	  Say Y or M here to include VIA HD-audio codec support in
 	  snd-hda-intel driver, such as VT1708.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_VIA=m
+
 config SND_HDA_CODEC_HDMI
 	tristate "Build HDMI/DisplayPort HD-audio codec support"
 	help
@@ -121,6 +133,9 @@ config SND_HDA_CODEC_HDMI
 	  support in snd-hda-intel driver.  This includes all AMD/ATI,
 	  Intel and Nvidia HDMI/DisplayPort codecs.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_HDMI=m
+
 config SND_HDA_I915
 	bool
 	default y
@@ -133,6 +148,9 @@ config SND_HDA_CODEC_CIRRUS
 	  Say Y or M here to include Cirrus Logic codec support in
 	  snd-hda-intel driver, such as CS4206.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CIRRUS=m
+
 config SND_HDA_CODEC_CONEXANT
 	tristate "Build Conexant HD-audio codec support"
 	select SND_HDA_GENERIC
@@ -140,6 +158,9 @@ config SND_HDA_CODEC_CONEXANT
 	  Say Y or M here to include Conexant HD-audio codec support in
 	  snd-hda-intel driver, such as CX20549.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CONEXANT=m
+
 config SND_HDA_CODEC_CA0110
 	tristate "Build Creative CA0110-IBG codec support"
 	select SND_HDA_GENERIC
@@ -147,12 +168,18 @@ config SND_HDA_CODEC_CA0110
 	  Say Y or M here to include Creative CA0110-IBG codec support in
 	  snd-hda-intel driver, found on some Creative X-Fi cards.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0110=m
+
 config SND_HDA_CODEC_CA0132
 	tristate "Build Creative CA0132 codec support"
 	help
 	  Say Y or M here to include Creative CA0132 codec support in
 	  snd-hda-intel driver.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CA0132=m
+
 config SND_HDA_CODEC_CA0132_DSP
 	bool "Support new DSP code for CA0132 codec"
 	depends on SND_HDA_CODEC_CA0132
@@ -172,18 +199,27 @@ config SND_HDA_CODEC_CMEDIA
 	  Say Y or M here to include C-Media HD-audio codec support in
 	  snd-hda-intel driver, such as CMI9880.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_CMEDIA=m
+
 config SND_HDA_CODEC_SI3054
 	tristate "Build Silicon Labs 3054 HD-modem codec support"
 	help
 	  Say Y or M here to include Silicon Labs 3054 HD-modem codec
 	  (and compatibles) support in snd-hda-intel driver.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_CODEC_SI3054=m
+
 config SND_HDA_GENERIC
 	tristate "Enable generic HD-audio codec parser"
 	help
 	  Say Y or M here to enable the generic HD-audio codec parser
 	  in snd-hda-intel driver.
 
+comment "Set to Y if you want auto-loading the codec driver"
+	depends on SND_HDA_INTEL=y && SND_HDA_GENERIC=m
+
 config SND_HDA_POWER_SAVE_DEFAULT
 	int "Default time-out for HD-audio power-save mode"
 	depends on PM
-- 
cgit v1.2.2


From 05004cb4cd06127bb8ff70d5ab5a915103828e9d Mon Sep 17 00:00:00 2001
From: Matthew Garrett <matthew.garrett@nebula.com>
Date: Wed, 18 Dec 2013 13:50:10 +0000
Subject: ASoC: fsl/mxs: Remove unnecessarily gendered language

The kernel as a number of cases of gendered language. The majority of these
refer to objects that don't have gender in English, and so I've replaced
them with "it" and "its". Some refer to people (developers or users), and
I've replaced these with the singular "they" variant. Some are simply
typos that I've fixed up.

I've left cases where gendered language was used to refer to specific
individuals, was a quote or is part of license text.

Signed-off-by: Matthew Garrett <matthew.garrett@nebula.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_dma.c  |  2 +-
 sound/soc/mxs/mxs-saif.c | 10 +++++-----
 2 files changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index fb9bb9eb5ca3..d570f8c81dc6 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -852,7 +852,7 @@ static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm)
 }
 
 /**
- * find_ssi_node -- returns the SSI node that points to his DMA channel node
+ * find_ssi_node -- returns the SSI node that points to its DMA channel node
  *
  * Although this DMA driver attempts to operate independently of the other
  * devices, it still needs to determine some information about the SSI device
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 54e622acac33..d6cb9a51dd57 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -50,9 +50,9 @@ static struct mxs_saif *mxs_saif[2];
  * This also means that both SAIFs must operate at the same sample rate.
  *
  * We abstract this as each saif has a master, the master could be
- * himself or other saifs. In the generic saif driver, saif does not need
- * to know the different clkmux. Saif only needs to know who is his master
- * and operating his master to generate the proper clock rate for him.
+ * itself or other saifs. In the generic saif driver, saif does not need
+ * to know the different clkmux. Saif only needs to know who is its master
+ * and operating its master to generate the proper clock rate for it.
  * The master id is provided in mach-specific layer according to different
  * clkmux setting.
  */
@@ -76,7 +76,7 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
  * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK
  * is provided by other SAIF, we provide a interface here to get its master
  * from its master_id.
- * Note that the master could be himself.
+ * Note that the master could be itself.
  */
 static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif)
 {
@@ -516,7 +516,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
 		}
 
 		/*
-		 * If the saif's master is not himself, we also need to enable
+		 * If the saif's master is not itself, we also need to enable
 		 * itself clk for its internal basic logic to work.
 		 */
 		if (saif != master_saif) {
-- 
cgit v1.2.2


From aafa85e71a75fdea9076c5e0f7fe09e12568c9a4 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <b42378@freescale.com>
Date: Thu, 12 Dec 2013 18:44:45 +0800
Subject: ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series

This patch adds three main functions for DAI master mode: set_dai_fmt(),
set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock
accordingly. After appending this patch, the fsl_ssi driver on i.MX series
has the ability to derive LRCLK and BCLK from baud clock source so as to
support some audio Codecs which can only be used in slave mode.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++-
 sound/soc/fsl/fsl_ssi.h |   2 +
 2 files changed, 278 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index f9f4569417ed..b2ebaf811599 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -38,6 +38,7 @@
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
+#include <linux/spinlock.h>
 #include <linux/of_address.h>
 #include <linux/of_irq.h>
 #include <linux/of_platform.h>
@@ -139,7 +140,10 @@ struct fsl_ssi_private {
 	bool ssi_on_imx;
 	bool imx_ac97;
 	bool use_dma;
+	bool baudclk_locked;
 	u8 i2s_mode;
+	spinlock_t baudclk_lock;
+	struct clk *baudclk;
 	struct clk *clk;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
@@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private =
 		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	unsigned long flags;
 
 	/* First, we only do fsl_ssi_setup() when SSI is going to be active.
 	 * Second, fsl_ssi_setup was already called by ac97_init earlier if
 	 * the driver is in ac97 mode.
 	 */
-	if (!dai->active && !ssi_private->imx_ac97)
+	if (!dai->active && !ssi_private->imx_ac97) {
 		fsl_ssi_setup(ssi_private);
+		spin_lock_irqsave(&ssi_private->baudclk_lock, flags);
+		ssi_private->baudclk_locked = false;
+		spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags);
+	}
 
 	return 0;
 }
@@ -501,6 +510,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+/**
+ * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format.
+ */
+static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	u32 strcr = 0, stcr, srcr, scr, mask;
+
+	scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK);
+	scr |= CCSR_SSI_SCR_NET;
+
+	mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR |
+		CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL |
+		CCSR_SSI_STCR_TEFS;
+	stcr = read_ssi(&ssi->stcr) & ~mask;
+	srcr = read_ssi(&ssi->srcr) & ~mask;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBS_CFS:
+			ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER;
+			break;
+		case SND_SOC_DAIFMT_CBM_CFM:
+			ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE;
+			break;
+		default:
+			return -EINVAL;
+		}
+		scr |= ssi_private->i2s_mode;
+
+		/* Data on rising edge of bclk, frame low, 1clk before data */
+		strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP |
+			CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* Data on rising edge of bclk, frame high */
+		strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* Data on rising edge of bclk, frame high, 1clk before data */
+		strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP |
+			CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* Data on rising edge of bclk, frame high */
+		strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP |
+			CCSR_SSI_STCR_TXBIT0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* Nothing to do for both normal cases */
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Invert bit clock */
+		strcr ^= CCSR_SSI_STCR_TSCKP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/* Invert frame clock */
+		strcr ^= CCSR_SSI_STCR_TFSI;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		/* Invert both clocks */
+		strcr ^= CCSR_SSI_STCR_TSCKP;
+		strcr ^= CCSR_SSI_STCR_TFSI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR;
+		scr |= CCSR_SSI_SCR_SYS_CLK_EN;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		scr &= ~CCSR_SSI_SCR_SYS_CLK_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	stcr |= strcr;
+	srcr |= strcr;
+
+	if (ssi_private->cpu_dai_drv.symmetric_rates) {
+		/* Need to clear RXDIR when using SYNC mode */
+		srcr &= ~CCSR_SSI_SRCR_RXDIR;
+		scr |= CCSR_SSI_SCR_SYN;
+	}
+
+	write_ssi(stcr, &ssi->stcr);
+	write_ssi(srcr, &ssi->srcr);
+	write_ssi(scr, &ssi->scr);
+
+	return 0;
+}
+
+/**
+ * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock
+ *
+ * Note: This function can be only called when using SSI as DAI master
+ *
+ * Quick instruction for parameters:
+ * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels
+ * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK.
+ */
+static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret;
+	u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
+	unsigned long flags, clkrate, baudrate, tmprate;
+	u64 sub, savesub = 100000;
+
+	/* Don't apply it to any non-baudclk circumstance */
+	if (IS_ERR(ssi_private->baudclk))
+		return -EINVAL;
+
+	/* It should be already enough to divide clock by setting pm alone */
+	psr = 0;
+	div2 = 0;
+
+	factor = (div2 + 1) * (7 * psr + 1) * 2;
+
+	for (i = 0; i < 255; i++) {
+		/* The bclk rate must be smaller than 1/5 sysclk rate */
+		if (factor * (i + 1) < 5)
+			continue;
+
+		tmprate = freq * factor * (i + 2);
+		clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
+
+		do_div(clkrate, factor);
+		afreq = (u32)clkrate / (i + 1);
+
+		if (freq == afreq)
+			sub = 0;
+		else if (freq / afreq == 1)
+			sub = freq - afreq;
+		else if (afreq / freq == 1)
+			sub = afreq - freq;
+		else
+			continue;
+
+		/* Calculate the fraction */
+		sub *= 100000;
+		do_div(sub, freq);
+
+		if (sub < savesub) {
+			baudrate = tmprate;
+			savesub = sub;
+			pm = i;
+		}
+
+		/* We are lucky */
+		if (savesub == 0)
+			break;
+	}
+
+	/* No proper pm found if it is still remaining the initial value */
+	if (pm == 999) {
+		dev_err(cpu_dai->dev, "failed to handle the required sysclk\n");
+		return -EINVAL;
+	}
+
+	stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) |
+		(psr ? CCSR_SSI_SxCCR_PSR : 0);
+	mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR;
+
+	if (dir == SND_SOC_CLOCK_OUT || synchronous)
+		write_ssi_mask(&ssi->stccr, mask, stccr);
+	else
+		write_ssi_mask(&ssi->srccr, mask, stccr);
+
+	spin_lock_irqsave(&ssi_private->baudclk_lock, flags);
+	if (!ssi_private->baudclk_locked) {
+		ret = clk_set_rate(ssi_private->baudclk, baudrate);
+		if (ret) {
+			spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags);
+			dev_err(cpu_dai->dev, "failed to set baudclk rate\n");
+			return -EINVAL;
+		}
+		ssi_private->baudclk_locked = true;
+	}
+	spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags);
+
+	return 0;
+}
+
+/**
+ * fsl_ssi_set_dai_tdm_slot - set TDM slot number
+ *
+ * Note: This function can be only called when using SSI as DAI master
+ */
+static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
+				u32 rx_mask, int slots, int slot_width)
+{
+	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	u32 val;
+
+	/* The slot number should be >= 2 if using Network mode or I2S mode */
+	val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET);
+	if (val && slots < 2) {
+		dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n");
+		return -EINVAL;
+	}
+
+	write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK,
+			CCSR_SSI_SxCCR_DC(slots));
+	write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK,
+			CCSR_SSI_SxCCR_DC(slots));
+
+	/* The register SxMSKs needs SSI to provide essential clock due to
+	 * hardware design. So we here temporarily enable SSI to set them.
+	 */
+	val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
+	write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN);
+
+	write_ssi(tx_mask, &ssi->stmsk);
+	write_ssi(rx_mask, &ssi->srmsk);
+
+	write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val);
+
+	return 0;
+}
+
 /**
  * fsl_ssi_trigger: start and stop the DMA transfer.
  *
@@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
 	unsigned int sier_bits;
+	unsigned long flags;
 
 	/*
 	 *  Enable only the interrupts and DMA requests
@@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
 
 		if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) &
-					(CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0)
+					(CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) {
 			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
+			spin_lock_irqsave(&ssi_private->baudclk_lock, flags);
+			ssi_private->baudclk_locked = false;
+			spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags);
+		}
 		break;
 
 	default:
@@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai)
 static const struct snd_soc_dai_ops fsl_ssi_dai_ops = {
 	.startup	= fsl_ssi_startup,
 	.hw_params	= fsl_ssi_hw_params,
+	.set_fmt	= fsl_ssi_set_dai_fmt,
+	.set_sysclk	= fsl_ssi_set_dai_sysclk,
+	.set_tdm_slot	= fsl_ssi_set_dai_tdm_slot,
 	.trigger	= fsl_ssi_trigger,
 };
 
@@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;
 
+	ssi_private->baudclk_locked = false;
+	spin_lock_init(&ssi_private->baudclk_lock);
+
 	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) {
 		u32 dma_events[2];
 		ssi_private->ssi_on_imx = true;
@@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 			goto error_irqmap;
 		}
 
+		/* For those SLAVE implementations, we ingore non-baudclk cases
+		 * and, instead, abandon MASTER mode that needs baud clock.
+		 */
+		ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud");
+		if (IS_ERR(ssi_private->baudclk))
+			dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret);
+		else
+			clk_prepare_enable(ssi_private->baudclk);
+
 		/*
 		 * We have burstsize be "fifo_depth - 2" to match the SSI
 		 * watermark setting in fsl_ssi_startup().
@@ -1059,8 +1325,11 @@ error_dev:
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
-	if (ssi_private->ssi_on_imx)
+	if (ssi_private->ssi_on_imx) {
+		if (!IS_ERR(ssi_private->baudclk))
+			clk_disable_unprepare(ssi_private->baudclk);
 		clk_disable_unprepare(ssi_private->clk);
+	}
 
 error_irqmap:
 	irq_dispose_mapping(ssi_private->irq);
@@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 		platform_device_unregister(ssi_private->pdev);
 	snd_soc_unregister_component(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
-	if (ssi_private->ssi_on_imx)
+	if (ssi_private->ssi_on_imx) {
+		if (!IS_ERR(ssi_private->baudclk))
+			clk_disable_unprepare(ssi_private->baudclk);
 		clk_disable_unprepare(ssi_private->clk);
+	}
 	irq_dispose_mapping(ssi_private->irq);
 
 	return 0;
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index e6b9a69e2a68..e6b63240a3d7 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -125,7 +125,9 @@ struct ccsr_ssi {
 #define CCSR_SSI_SRCR_REFS		0x00000001
 
 /* STCCR and SRCCR */
+#define CCSR_SSI_SxCCR_DIV2_SHIFT	18
 #define CCSR_SSI_SxCCR_DIV2		0x00040000
+#define CCSR_SSI_SxCCR_PSR_SHIFT	17
 #define CCSR_SSI_SxCCR_PSR		0x00020000
 #define CCSR_SSI_SxCCR_WL_SHIFT		13
 #define CCSR_SSI_SxCCR_WL_MASK		0x0001E000
-- 
cgit v1.2.2


From 0d35d080ac93f317b6c47180d75c8e1a8109b4c4 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 29 Nov 2013 16:03:43 +0200
Subject: ASoC: twl6040: Rename twl6040_is_path_unmuted ->
 twl6040_can_write_to_chip

Matches more precisely of the functionality.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl6040.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index f2f4bcb2ff71..ef13a501a7b4 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -219,8 +219,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec,
 	return value;
 }
 
-static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec,
-				    unsigned int reg)
+static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec,
+				  unsigned int reg)
 {
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
@@ -250,7 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec,
 		return -EIO;
 
 	twl6040_write_reg_cache(codec, reg, value);
-	if (twl6040_is_path_unmuted(codec, reg))
+	if (twl6040_can_write_to_chip(codec, reg))
 		return twl6040_reg_write(twl6040, reg, value);
 	else
 		return 0;
-- 
cgit v1.2.2


From 53509108f7372f786576d7d43f8f881cdf51d38a Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 29 Nov 2013 16:03:44 +0200
Subject: ASoC: twl6040: Custom caching for sensitive DL1/2 path registers

Introduce a small register cache for registers which needs special caching
to reduce pop noise:
TWL6040_REG_HSLCTL, TWL6040_REG_HSRCTL, TWL6040_REG_EARCTL, TWL6040_REG_HFLCTL
and TWL6040_REG_HFRCTL.
Switch over and use the new small cache for these registers instead of the
main reg_cache.
This is in preparation to remove the local ASoC reg_cache from the driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl6040.c | 51 ++++++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 49 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index ef13a501a7b4..fb8c65bd6e5d 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -72,6 +72,7 @@ struct twl6040_data {
 	int hs_power_mode_locked;
 	bool dl1_unmuted;
 	bool dl2_unmuted;
+	u8 dl12_cache[TWL6040_REG_HFRCTL - TWL6040_REG_HSLCTL + 1];
 	unsigned int clk_in;
 	unsigned int sysclk;
 	struct twl6040_jack_data hs_jack;
@@ -174,18 +175,62 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
 	{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
 };
 
+static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec,
+					   u8 reg, u8 *value)
+{
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	switch (reg) {
+	case TWL6040_REG_HSLCTL:
+	case TWL6040_REG_HSRCTL:
+	case TWL6040_REG_EARCTL:
+	case TWL6040_REG_HFLCTL:
+	case TWL6040_REG_HFRCTL:
+		*value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL];
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
 /*
  * read twl6040 register cache
  */
 static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec,
-						unsigned int reg)
+						  unsigned int reg)
 {
 	u8 *cache = codec->reg_cache;
+	u8 value;
 
 	if (reg >= TWL6040_CACHEREGNUM)
 		return -EIO;
 
-	return cache[reg];
+	if (twl6040_read_dl12_cache(codec, reg, &value))
+		value = cache[reg];
+
+	return value;
+}
+
+static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec,
+					     u8 reg, u8 value)
+{
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (reg) {
+	case TWL6040_REG_HSLCTL:
+	case TWL6040_REG_HSRCTL:
+	case TWL6040_REG_EARCTL:
+	case TWL6040_REG_HFLCTL:
+	case TWL6040_REG_HFRCTL:
+		priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value;
+		break;
+	default:
+		break;
+	}
 }
 
 /*
@@ -199,6 +244,8 @@ static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec,
 	if (reg >= TWL6040_CACHEREGNUM)
 		return;
 	cache[reg] = value;
+
+	twl6040_update_dl12_cache(codec, reg, value);
 }
 
 /*
-- 
cgit v1.2.2


From 79ae5130381cb7d3e74c162123392ba6067c218e Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 29 Nov 2013 16:03:46 +0200
Subject: ASoC: twl6040: Remove register restore functionality

The MFD core takes care of the restore via standard regmap API, no need to
do this anymore here.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl6040.c | 29 -----------------------------
 1 file changed, 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index fb8c65bd6e5d..b07839b2d912 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -133,22 +133,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
 	0x00, /* REG_STATUS	0x2E (ro) */
 };
 
-/* List of registers to be restored after power up */
-static const int twl6040_restore_list[] = {
-	TWL6040_REG_MICLCTL,
-	TWL6040_REG_MICRCTL,
-	TWL6040_REG_MICGAIN,
-	TWL6040_REG_LINEGAIN,
-	TWL6040_REG_HSLCTL,
-	TWL6040_REG_HSRCTL,
-	TWL6040_REG_HSGAIN,
-	TWL6040_REG_EARCTL,
-	TWL6040_REG_HFLCTL,
-	TWL6040_REG_HFLGAIN,
-	TWL6040_REG_HFRCTL,
-	TWL6040_REG_HFRGAIN,
-};
-
 /* set of rates for each pll: low-power and high-performance */
 static unsigned int lp_rates[] = {
 	8000,
@@ -335,17 +319,6 @@ static void twl6040_init_chip(struct snd_soc_codec *codec)
 	twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0);
 }
 
-static void twl6040_restore_regs(struct snd_soc_codec *codec)
-{
-	u8 *cache = codec->reg_cache;
-	int reg, i;
-
-	for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) {
-		reg = twl6040_restore_list[i];
-		twl6040_write(codec, reg, cache[reg]);
-	}
-}
-
 /* set headset dac and driver power mode */
 static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
 {
@@ -978,8 +951,6 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 
 		priv->codec_powered = 1;
 
-		twl6040_restore_regs(codec);
-
 		/* Set external boost GPO */
 		twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02);
 		break;
-- 
cgit v1.2.2


From 626bcacb89f93b2145f3a705a342067a77347a99 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 29 Nov 2013 16:03:47 +0200
Subject: ASoC: twl6040: Remove self managed local reg_cache support

We can rely on mfd driver to manage the register caching via regmap. The
driver still need to cache some registers associated with DL1/2 routes.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl6040.c | 211 +++++++++++----------------------------------
 1 file changed, 49 insertions(+), 162 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index b07839b2d912..0afe8bef6765 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -80,59 +80,6 @@ struct twl6040_data {
 	struct mutex mutex;
 };
 
-/*
- * twl6040 register cache & default register settings
- */
-static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
-	0x00, /* not used	0x00	*/
-	0x4B, /* REG_ASICID	0x01 (ro) */
-	0x00, /* REG_ASICREV	0x02 (ro) */
-	0x00, /* REG_INTID	0x03	*/
-	0x00, /* REG_INTMR	0x04	*/
-	0x00, /* REG_NCPCTRL	0x05	*/
-	0x00, /* REG_LDOCTL	0x06	*/
-	0x60, /* REG_HPPLLCTL	0x07	*/
-	0x00, /* REG_LPPLLCTL	0x08	*/
-	0x4A, /* REG_LPPLLDIV	0x09	*/
-	0x00, /* REG_AMICBCTL	0x0A	*/
-	0x00, /* REG_DMICBCTL	0x0B	*/
-	0x00, /* REG_MICLCTL	0x0C	*/
-	0x00, /* REG_MICRCTL	0x0D	*/
-	0x00, /* REG_MICGAIN	0x0E	*/
-	0x1B, /* REG_LINEGAIN	0x0F	*/
-	0x00, /* REG_HSLCTL	0x10	*/
-	0x00, /* REG_HSRCTL	0x11	*/
-	0x00, /* REG_HSGAIN	0x12	*/
-	0x00, /* REG_EARCTL	0x13	*/
-	0x00, /* REG_HFLCTL	0x14	*/
-	0x00, /* REG_HFLGAIN	0x15	*/
-	0x00, /* REG_HFRCTL	0x16	*/
-	0x00, /* REG_HFRGAIN	0x17	*/
-	0x00, /* REG_VIBCTLL	0x18	*/
-	0x00, /* REG_VIBDATL	0x19	*/
-	0x00, /* REG_VIBCTLR	0x1A	*/
-	0x00, /* REG_VIBDATR	0x1B	*/
-	0x00, /* REG_HKCTL1	0x1C	*/
-	0x00, /* REG_HKCTL2	0x1D	*/
-	0x00, /* REG_GPOCTL	0x1E	*/
-	0x00, /* REG_ALB	0x1F	*/
-	0x00, /* REG_DLB	0x20	*/
-	0x00, /* not used	0x21	*/
-	0x00, /* not used	0x22	*/
-	0x00, /* not used	0x23	*/
-	0x00, /* not used	0x24	*/
-	0x00, /* not used	0x25	*/
-	0x00, /* not used	0x26	*/
-	0x00, /* not used	0x27	*/
-	0x00, /* REG_TRIM1	0x28	*/
-	0x00, /* REG_TRIM2	0x29	*/
-	0x00, /* REG_TRIM3	0x2A	*/
-	0x00, /* REG_HSOTRIM	0x2B	*/
-	0x00, /* REG_HFOTRIM	0x2C	*/
-	0x09, /* REG_ACCCTL	0x2D	*/
-	0x00, /* REG_STATUS	0x2E (ro) */
-};
-
 /* set of rates for each pll: low-power and high-performance */
 static unsigned int lp_rates[] = {
 	8000,
@@ -159,11 +106,14 @@ static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
 	{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
 };
 
-static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec,
-					   u8 reg, u8 *value)
+static unsigned int twl6040_read(struct snd_soc_codec *codec, unsigned int reg)
 {
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
+	struct twl6040 *twl6040 = codec->control_data;
+	u8 value;
+
+	if (reg >= TWL6040_CACHEREGNUM)
+		return -EIO;
 
 	switch (reg) {
 	case TWL6040_REG_HSLCTL:
@@ -171,36 +121,18 @@ static inline int twl6040_read_dl12_cache(struct snd_soc_codec *codec,
 	case TWL6040_REG_EARCTL:
 	case TWL6040_REG_HFLCTL:
 	case TWL6040_REG_HFRCTL:
-		*value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL];
+		value = priv->dl12_cache[reg - TWL6040_REG_HSLCTL];
 		break;
 	default:
-		ret = -EINVAL;
+		value = twl6040_reg_read(twl6040, reg);
 		break;
 	}
 
-	return ret;
-}
-
-/*
- * read twl6040 register cache
- */
-static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec,
-						  unsigned int reg)
-{
-	u8 *cache = codec->reg_cache;
-	u8 value;
-
-	if (reg >= TWL6040_CACHEREGNUM)
-		return -EIO;
-
-	if (twl6040_read_dl12_cache(codec, reg, &value))
-		value = cache[reg];
-
 	return value;
 }
 
-static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec,
-					     u8 reg, u8 value)
+static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec,
+				  unsigned int reg)
 {
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
@@ -208,50 +140,18 @@ static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec,
 	case TWL6040_REG_HSLCTL:
 	case TWL6040_REG_HSRCTL:
 	case TWL6040_REG_EARCTL:
+		/* DL1 path */
+		return priv->dl1_unmuted;
 	case TWL6040_REG_HFLCTL:
 	case TWL6040_REG_HFRCTL:
-		priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value;
-		break;
+		return priv->dl2_unmuted;
 	default:
-		break;
+		return 1;
 	}
 }
 
-/*
- * write twl6040 register cache
- */
-static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec,
-						u8 reg, u8 value)
-{
-	u8 *cache = codec->reg_cache;
-
-	if (reg >= TWL6040_CACHEREGNUM)
-		return;
-	cache[reg] = value;
-
-	twl6040_update_dl12_cache(codec, reg, value);
-}
-
-/*
- * read from twl6040 hardware register
- */
-static int twl6040_read_reg_volatile(struct snd_soc_codec *codec,
-			unsigned int reg)
-{
-	struct twl6040 *twl6040 = codec->control_data;
-	u8 value;
-
-	if (reg >= TWL6040_CACHEREGNUM)
-		return -EIO;
-
-	value = twl6040_reg_read(twl6040, reg);
-	twl6040_write_reg_cache(codec, reg, value);
-
-	return value;
-}
-
-static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec,
-				  unsigned int reg)
+static inline void twl6040_update_dl12_cache(struct snd_soc_codec *codec,
+					     u8 reg, u8 value)
 {
 	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
@@ -259,19 +159,15 @@ static bool twl6040_can_write_to_chip(struct snd_soc_codec *codec,
 	case TWL6040_REG_HSLCTL:
 	case TWL6040_REG_HSRCTL:
 	case TWL6040_REG_EARCTL:
-		/* DL1 path */
-		return priv->dl1_unmuted;
 	case TWL6040_REG_HFLCTL:
 	case TWL6040_REG_HFRCTL:
-		return priv->dl2_unmuted;
+		priv->dl12_cache[reg - TWL6040_REG_HSLCTL] = value;
+		break;
 	default:
-		return 1;
+		break;
 	}
 }
 
-/*
- * write to the twl6040 register space
- */
 static int twl6040_write(struct snd_soc_codec *codec,
 			unsigned int reg, unsigned int value)
 {
@@ -280,7 +176,7 @@ static int twl6040_write(struct snd_soc_codec *codec,
 	if (reg >= TWL6040_CACHEREGNUM)
 		return -EIO;
 
-	twl6040_write_reg_cache(codec, reg, value);
+	twl6040_update_dl12_cache(codec, reg, value);
 	if (twl6040_can_write_to_chip(codec, reg))
 		return twl6040_reg_write(twl6040, reg, value);
 	else
@@ -289,34 +185,27 @@ static int twl6040_write(struct snd_soc_codec *codec,
 
 static void twl6040_init_chip(struct snd_soc_codec *codec)
 {
-	struct twl6040 *twl6040 = codec->control_data;
-	u8 val;
-
-	/* Update reg_cache: ASICREV, and TRIM values */
-	val = twl6040_get_revid(twl6040);
-	twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val);
-
-	twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM1);
-	twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM2);
-	twl6040_read_reg_volatile(codec, TWL6040_REG_TRIM3);
-	twl6040_read_reg_volatile(codec, TWL6040_REG_HSOTRIM);
-	twl6040_read_reg_volatile(codec, TWL6040_REG_HFOTRIM);
+	twl6040_read(codec, TWL6040_REG_TRIM1);
+	twl6040_read(codec, TWL6040_REG_TRIM2);
+	twl6040_read(codec, TWL6040_REG_TRIM3);
+	twl6040_read(codec, TWL6040_REG_HSOTRIM);
+	twl6040_read(codec, TWL6040_REG_HFOTRIM);
 
 	/* Change chip defaults */
 	/* No imput selected for microphone amplifiers */
-	twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18);
-	twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18);
+	twl6040_write(codec, TWL6040_REG_MICLCTL, 0x18);
+	twl6040_write(codec, TWL6040_REG_MICRCTL, 0x18);
 
 	/*
 	 * We need to lower the default gain values, so the ramp code
 	 * can work correctly for the first playback.
 	 * This reduces the pop noise heard at the first playback.
 	 */
-	twl6040_write_reg_cache(codec, TWL6040_REG_HSGAIN, 0xff);
-	twl6040_write_reg_cache(codec, TWL6040_REG_EARCTL, 0x1e);
-	twl6040_write_reg_cache(codec, TWL6040_REG_HFLGAIN, 0x1d);
-	twl6040_write_reg_cache(codec, TWL6040_REG_HFRGAIN, 0x1d);
-	twl6040_write_reg_cache(codec, TWL6040_REG_LINEGAIN, 0);
+	twl6040_write(codec, TWL6040_REG_HSGAIN, 0xff);
+	twl6040_write(codec, TWL6040_REG_EARCTL, 0x1e);
+	twl6040_write(codec, TWL6040_REG_HFLGAIN, 0x1d);
+	twl6040_write(codec, TWL6040_REG_HFRGAIN, 0x1d);
+	twl6040_write(codec, TWL6040_REG_LINEGAIN, 0);
 }
 
 /* set headset dac and driver power mode */
@@ -325,8 +214,8 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
 	int hslctl, hsrctl;
 	int mask = TWL6040_HSDRVMODE | TWL6040_HSDACMODE;
 
-	hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
-	hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
+	hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL);
+	hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL);
 
 	if (high_perf) {
 		hslctl &= ~mask;
@@ -353,8 +242,8 @@ static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
 	 * Both HS DAC need to be turned on (before the HS driver) and off at
 	 * the same time.
 	 */
-	hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
-	hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
+	hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL);
+	hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL);
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		hslctl |= TWL6040_HSDACENA;
 		hsrctl |= TWL6040_HSDACENA;
@@ -399,7 +288,7 @@ static void twl6040_hs_jack_report(struct snd_soc_codec *codec,
 	mutex_lock(&priv->mutex);
 
 	/* Sync status */
-	status = twl6040_read_reg_volatile(codec, TWL6040_REG_STATUS);
+	status = twl6040_read(codec, TWL6040_REG_STATUS);
 	if (status & TWL6040_PLUGCOMP)
 		snd_soc_jack_report(jack, report, report);
 	else
@@ -451,7 +340,7 @@ static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 
 	/* Do not allow changes while Input/FF efect is running */
-	val = twl6040_read_reg_volatile(codec, e->reg);
+	val = twl6040_read(codec, e->reg);
 	if (val & TWL6040_VIBENA && !(val & TWL6040_VIBSEL))
 		return -EBUSY;
 
@@ -676,7 +565,7 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim)
 	if (unlikely(trim >= TWL6040_TRIM_INVAL))
 		return -EINVAL;
 
-	return twl6040_read_reg_cache(codec, TWL6040_REG_TRIM1 + trim);
+	return twl6040_read(codec, TWL6040_REG_TRIM1 + trim);
 }
 EXPORT_SYMBOL_GPL(twl6040_get_trim_value);
 
@@ -1071,9 +960,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i
 
 	switch (id) {
 	case TWL6040_DAI_DL1:
-		hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
-		hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
-		earctl = twl6040_read_reg_cache(codec, TWL6040_REG_EARCTL);
+		hslctl = twl6040_read(codec, TWL6040_REG_HSLCTL);
+		hsrctl = twl6040_read(codec, TWL6040_REG_HSRCTL);
+		earctl = twl6040_read(codec, TWL6040_REG_EARCTL);
 
 		if (mute) {
 			/* Power down drivers and DACs */
@@ -1089,8 +978,8 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i
 		priv->dl1_unmuted = !mute;
 		break;
 	case TWL6040_DAI_DL2:
-		hflctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFLCTL);
-		hfrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HFRCTL);
+		hflctl = twl6040_read(codec, TWL6040_REG_HFLCTL);
+		hfrctl = twl6040_read(codec, TWL6040_REG_HFRCTL);
 
 		if (mute) {
 			/* Power down drivers and DACs */
@@ -1227,6 +1116,7 @@ static int twl6040_resume(struct snd_soc_codec *codec)
 static int twl6040_probe(struct snd_soc_codec *codec)
 {
 	struct twl6040_data *priv;
+	struct twl6040 *twl6040 = dev_get_drvdata(codec->dev->parent);
 	struct platform_device *pdev = container_of(codec->dev,
 						   struct platform_device, dev);
 	int ret = 0;
@@ -1238,7 +1128,7 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 	snd_soc_codec_set_drvdata(codec, priv);
 
 	priv->codec = codec;
-	codec->control_data = dev_get_drvdata(codec->dev->parent);
+	codec->control_data = twl6040;
 
 	priv->plug_irq = platform_get_irq(pdev, 0);
 	if (priv->plug_irq < 0) {
@@ -1258,10 +1148,10 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
+	twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	twl6040_init_chip(codec);
 
-	/* power on device */
-	return twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
 }
 
 static int twl6040_remove(struct snd_soc_codec *codec)
@@ -1279,12 +1169,9 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
 	.remove = twl6040_remove,
 	.suspend = twl6040_suspend,
 	.resume = twl6040_resume,
-	.read = twl6040_read_reg_cache,
+	.read = twl6040_read,
 	.write = twl6040_write,
 	.set_bias_level = twl6040_set_bias_level,
-	.reg_cache_size = ARRAY_SIZE(twl6040_reg),
-	.reg_word_size = sizeof(u8),
-	.reg_cache_default = twl6040_reg,
 	.ignore_pmdown_time = true,
 
 	.controls = twl6040_snd_controls,
-- 
cgit v1.2.2


From d171ebc5b9d197c1ec76c4b21c91981649db4c65 Mon Sep 17 00:00:00 2001
From: Lv Zheng <lv.zheng@intel.com>
Date: Fri, 20 Dec 2013 16:30:42 +0800
Subject: ALSA: hda - Fix wrong <acpi/acpi.h> inclusion in Thinkpad ACPI users.

CONFIG_ACPI dependent code should include <linux/acpi.h> instead of
directly including <acpi/acpi.h>.  This patch cleans up such wrong
inclusions for Thinkpad ACPI users.

Signed-off-by: Lv Zheng <lv.zheng@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 2 +-
 sound/pci/hda/patch_realtek.c  | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3fbf2883e06e..6df4c35b48c5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3243,7 +3243,7 @@ enum {
 #if IS_ENABLED(CONFIG_THINKPAD_ACPI)
 
 #include <linux/thinkpad_acpi.h>
-#include <acpi/acpi.h>
+#include <linux/acpi.h>
 
 static int (*led_set_func)(int, bool);
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3578f11c8e12..ef43171c0bd0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3779,7 +3779,7 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec,
 #if IS_ENABLED(CONFIG_THINKPAD_ACPI)
 
 #include <linux/thinkpad_acpi.h>
-#include <acpi/acpi.h>
+#include <linux/acpi.h>
 
 static int (*led_set_func)(int, bool);
 
-- 
cgit v1.2.2


From 0f4881dc1fd858649589edae76efd9c318bb562f Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 20 Dec 2013 16:08:13 +0100
Subject: ALSA: hda - Enable subwoofer on Dell Vostro 5460/5470

In this case, there are two DACs, and DAC 0x03 is mono. In order
to make headphones and front speaker use DAC 0x02, and subwoofer use
DAC 0x03, we artificially cut the connection from nodes 0x14 and 0x15
to node 0x03, so they can only use DAC 0x02.

In addition, the 5460 and 5470 differs in the sense that 5470 also
needs a headset mic patch, whereas 5460 has individual detection for
headphone and headset mic.

BugLink: https://bugs.launchpad.net/bugs/1211920
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++++------
 1 file changed, 37 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ef43171c0bd0..a84e0493499d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3770,10 +3770,14 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 static void alc290_fixup_mono_speakers(struct hda_codec *codec,
 				       const struct hda_fixup *fix, int action)
 {
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
-		/* Remove DAC node 0x03, as it seems to be
-		   giving mono output */
-		snd_hda_override_wcaps(codec, 0x03, 0);
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		/* DAC node 0x03 is giving mono output. We therefore want to
+		   make sure 0x14 (front speaker) and 0x15 (headphones) use the
+		   stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */
+		hda_nid_t conn1[2] = { 0x0c };
+		snd_hda_override_conn_list(codec, 0x14, 1, conn1);
+		snd_hda_override_conn_list(codec, 0x15, 1, conn1);
+	}
 }
 
 #if IS_ENABLED(CONFIG_THINKPAD_ACPI)
@@ -3913,6 +3917,9 @@ enum {
 	ALC282_FIXUP_ASUS_TX300,
 	ALC283_FIXUP_INT_MIC,
 	ALC290_FIXUP_MONO_SPEAKERS,
+	ALC290_FIXUP_MONO_SPEAKERS_HSJACK,
+	ALC290_FIXUP_SUBWOOFER,
+	ALC290_FIXUP_SUBWOOFER_HSJACK,
 	ALC269_FIXUP_THINKPAD_ACPI,
 	ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC255_FIXUP_HEADSET_MODE,
@@ -4235,9 +4242,31 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
 	},
+	[ALC290_FIXUP_SUBWOOFER_HSJACK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x17, 0x90170112 }, /* subwoofer */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC290_FIXUP_MONO_SPEAKERS_HSJACK,
+	},
+	[ALC290_FIXUP_SUBWOOFER] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x17, 0x90170112 }, /* subwoofer */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC290_FIXUP_MONO_SPEAKERS,
+	},
 	[ALC290_FIXUP_MONO_SPEAKERS] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc290_fixup_mono_speakers,
+	},
+	[ALC290_FIXUP_MONO_SPEAKERS_HSJACK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc290_fixup_mono_speakers,
 		.chained = true,
 		.chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
 	},
@@ -4282,6 +4311,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -4303,10 +4333,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
+	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
 	SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK),
 	SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
-- 
cgit v1.2.2


From 19570d747795f0ecff79403698b868dc5ad7fa86 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Dec 2013 14:32:53 +0100
Subject: ALSA: usb-audio: Add a quirk for Plantronics Gamecom 780

Plantronics Gamecom 780 headset has a firmware problem, and when the
FU 0x09 volume is changed, it results in either too loud or silence
except for a very narrow range.  This patch provides a workaround,
ignoring the node, initialize the volume in a sane value and keep
untouched.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_maps.c | 10 ++++++++++
 sound/usb/quirks.c     | 14 ++++++++++++++
 2 files changed, 24 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index cc2dd1f0decb..32af6b741ef5 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -322,6 +322,12 @@ static struct usbmix_name_map hercules_usb51_map[] = {
 	{ 0 }				/* terminator */
 };
 
+/* Plantronics Gamecom 780 has a broken volume control, better to disable it */
+static struct usbmix_name_map gamecom780_map[] = {
+	{ 9, NULL }, /* FU, speaker out */
+	{}
+};
+
 /*
  * Control map entries
  */
@@ -358,6 +364,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x046d, 0x09a4),
 		.ignore_ctl_error = 1,
 	},
+	{	/* Plantronics GameCom 780 */
+		.id = USB_ID(0x047f, 0xc010),
+		.map = gamecom780_map,
+	},
 	{
 		/* Hercules DJ Console (Windows Edition) */
 		.id = USB_ID(0x06f8, 0xb000),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 0df9ede99dfd..ede753685491 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -660,6 +660,18 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
 	return err;
 }
 
+/* quirk for Plantronics GameCom 780 with CM6302 chip */
+static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev)
+{
+	/* set the initial volume and don't change; other values are either
+	 * too loud or silent due to firmware bug (bko#65251)
+	 */
+	u8 buf[2] = { 0x74, 0xdc };
+	return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
+			USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+			UAC_FU_VOLUME << 8, 9 << 8, buf, 2);
+}
+
 /*
  * Novation Twitch DJ controller
  */
@@ -986,6 +998,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 		return snd_usb_nativeinstruments_boot_quirk(dev);
 	case USB_ID(0x0763, 0x2012):  /* M-Audio Fast Track Pro USB */
 		return snd_usb_fasttrackpro_boot_quirk(dev);
+	case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */
+		return snd_usb_gamecon780_boot_quirk(dev);
 	}
 
 	return 0;
-- 
cgit v1.2.2


From f467a0f513ad81998f0cad1022684a273d5743f7 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:08 +0100
Subject: ASoC: au1x: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/au1x/dbdma2.c |  9 ---------
 sound/soc/au1x/dma.c    | 14 --------------
 2 files changed, 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 3b4eafaf30d3..17a24d804734 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -65,19 +65,10 @@ struct au1xpsc_audio_dmadata {
 #define AU1XPSC_PERIOD_MIN_BYTES	1024
 #define AU1XPSC_BUFFER_MIN_BYTES	65536
 
-#define AU1XPSC_PCM_FMTS					\
-	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_U8 |	\
-	 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |	\
-	 SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |	\
-	 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |	\
-	 SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |	\
-	 0)
-
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
-	.formats	  = AU1XPSC_PCM_FMTS,
 	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
 	.period_bytes_max = 4096 * 1024 - 1,
 	.periods_min	  = 2,
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
index befd1074f9bd..e920b60bf6c2 100644
--- a/sound/soc/au1x/dma.c
+++ b/sound/soc/au1x/dma.c
@@ -21,14 +21,6 @@
 
 #include "psc.h"
 
-#define ALCHEMY_PCM_FMTS					\
-	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_U8 |	\
-	 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |	\
-	 SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |	\
-	 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |	\
-	 SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |	\
-	 0)
-
 struct pcm_period {
 	u32 start;
 	u32 relative_end;	/* relative to start of buffer */
@@ -171,12 +163,6 @@ static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
 static const struct snd_pcm_hardware alchemy_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
-	.formats	  = ALCHEMY_PCM_FMTS,
-	.rates		  = SNDRV_PCM_RATE_8000_192000,
-	.rate_min	  = SNDRV_PCM_RATE_8000,
-	.rate_max	  = SNDRV_PCM_RATE_192000,
-	.channels_min	  = 2,
-	.channels_max	  = 2,
 	.period_bytes_min = 1024,
 	.period_bytes_max = 16 * 1024 - 1,
 	.periods_min	  = 4,
-- 
cgit v1.2.2


From 5e8154332f48f92f37824577c88e400b5e0cd56d Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:22 +0100
Subject: ASoC: tegra: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/tegra/tegra_pcm.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 7b2d23ba69b3..c09ffd18791b 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -42,9 +42,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_INTERLEAVED,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
-	.channels_min		= 2,
-	.channels_max		= 2,
 	.period_bytes_min	= 1024,
 	.period_bytes_max	= PAGE_SIZE,
 	.periods_min		= 2,
-- 
cgit v1.2.2


From a6af47ae5399baf4f5a2426b2121c1bcb9da4019 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 20 Dec 2013 12:17:38 +0800
Subject: ASoC: fsl-sai: Remove fsl_sai_remove()

There is no need of this function and makes the code slightly shorter

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 11 -----------
 1 file changed, 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 50a797e65781..1868ec34be10 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -385,19 +385,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
-static int fsl_sai_dai_remove(struct snd_soc_dai *cpu_dai)
-{
-	cpu_dai->playback_dma_data = NULL;
-	cpu_dai->capture_dma_data = NULL;
-
-	snd_soc_dai_set_drvdata(cpu_dai, NULL);
-
-	return 0;
-}
-
 static struct snd_soc_dai_driver fsl_sai_dai = {
 	.probe = fsl_sai_dai_probe,
-	.remove = fsl_sai_dai_remove,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-- 
cgit v1.2.2


From e5180df3960b6130f17f3c5ab50d23674cdb2b5a Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 20 Dec 2013 12:30:26 +0800
Subject: ASoC: fsl-sai: Use devm_snd_dmaengine_pcm_register()

Makes the code slightly shorter

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 15 +--------------
 1 file changed, 1 insertion(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1868ec34be10..262d3107892e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -443,19 +443,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (ret)
 		return ret;
 
-	ret = snd_dmaengine_pcm_register(&pdev->dev, NULL,
+	return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
 			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
-	if (ret)
-		return ret;
-
-	return 0;
-}
-
-static int fsl_sai_remove(struct platform_device *pdev)
-{
-	snd_dmaengine_pcm_unregister(&pdev->dev);
-
-	return 0;
 }
 
 static const struct of_device_id fsl_sai_ids[] = {
@@ -465,8 +454,6 @@ static const struct of_device_id fsl_sai_ids[] = {
 
 static struct platform_driver fsl_sai_driver = {
 	.probe = fsl_sai_probe,
-	.remove = fsl_sai_remove,
-
 	.driver = {
 		.name = "fsl-sai",
 		.owner = THIS_MODULE,
-- 
cgit v1.2.2


From dd9f40602e96353c210805a99abd9af6abd28473 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 20 Dec 2013 12:35:33 +0800
Subject: ASoC: fsl-sai: Use snd_soc_dai_init_dma_data()

Makes the code slightly shorter

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 262d3107892e..b8cdbf8660fe 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -377,8 +377,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
 
-	cpu_dai->playback_dma_data = &sai->dma_params_tx;
-	cpu_dai->capture_dma_data = &sai->dma_params_rx;
+	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
+				&sai->dma_params_rx);
 
 	snd_soc_dai_set_drvdata(cpu_dai, sai);
 
-- 
cgit v1.2.2


From dd41e0c4e35abc5a5679762ab108a6fcd9a0ead9 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 20 Dec 2013 14:39:50 +0800
Subject: ASoC: simple-card: Add cpu_dai and codec_dai names NULL check

The name of cpu DAI maybe omitted, and then strlen() will lead
kernel panic.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 7a9b6b4898c0..be7c1db5388f 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -142,6 +142,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 	if (ret < 0)
 		return ret;
 
+	if (!info->cpu_dai.name || !info->codec_dai.name)
+		return -EINVAL;
+
 	/* card name is created from CPU/CODEC dai name */
 	name = devm_kzalloc(dev,
 			    strlen(info->cpu_dai.name)   +
-- 
cgit v1.2.2


From 1fb2d9d7465bcbb519c582fa4a3bd04ff4fce2d2 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:00 +0800
Subject: ASoC: fsl_sai: Keep symmetry for clk_enable() and clk_disable()

There are two functions haven't clk_disable_unprepare() if having error.
Thus fix them.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 14 ++++++++------
 1 file changed, 8 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index b8cdbf8660fe..69a375f48efe 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -111,7 +111,7 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		dev_err(cpu_dai->dev,
 				"Cannot set SAI's transmitter sysclk: %d\n",
 				ret);
-		return ret;
+		goto err_clk;
 	}
 
 	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
@@ -120,12 +120,13 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		dev_err(cpu_dai->dev,
 				"Cannot set SAI's receiver sysclk: %d\n",
 				ret);
-		return ret;
+		goto err_clk;
 	}
 
+err_clk:
 	clk_disable_unprepare(sai->clk);
 
-	return 0;
+	return ret;
 }
 
 static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
@@ -222,7 +223,7 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		dev_err(cpu_dai->dev,
 				"Cannot set SAI's transmitter format: %d\n",
 				ret);
-		return ret;
+		goto err_clk;
 	}
 
 	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
@@ -230,12 +231,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 		dev_err(cpu_dai->dev,
 				"Cannot set SAI's receiver format: %d\n",
 				ret);
-		return ret;
+		goto err_clk;
 	}
 
+err_clk:
 	clk_disable_unprepare(sai->clk);
 
-	return 0;
+	return ret;
 }
 
 static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
-- 
cgit v1.2.2


From 1d7003092771bd2feec30e2f3e5a06aa33479e08 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:01 +0800
Subject: ASoC: fsl_sai: Use snd_pcm_format_width()

Use common helper function snd_pcm_format_width() to make code neater.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 17 ++---------------
 1 file changed, 2 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 69a375f48efe..e68102e63521 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -244,9 +244,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params,
 		struct snd_soc_dai *cpu_dai)
 {
-	u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr, word_width;
+	u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr;
 	unsigned int channels = params_channels(params);
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 word_width = snd_pcm_format_width(params_format(params));
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		reg_cr4 = FSL_SAI_TCR4;
@@ -267,20 +268,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
 	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		word_width = 16;
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		word_width = 20;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		word_width = 24;
-		break;
-	default:
-		return -EINVAL;
-	}
-
 	val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
 	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
 	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
-- 
cgit v1.2.2


From d22e28cce80a93578787d273bf1fa26a2be2636b Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:02 +0800
Subject: ASoC: fsl_sai: Drop useless channels check in hw_params()

SAi only supports two data channels on hardware level and the driver also does
register the min->1 and max->2, so no need to check channels.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index e68102e63521..8450bff6fb13 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -278,10 +278,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		val_cr5 |= FSL_SAI_CR5_FBT(0);
 
 	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
-	if (channels == 2 || channels == 1)
-		val_mr = ~0UL - ((1 << channels) - 1);
-	else
-		return -EINVAL;
+	val_mr = ~0UL - ((1 << channels) - 1);
 
 	sai_writel(sai, val_cr4, sai->base + reg_cr4);
 	sai_writel(sai, val_cr5, sai->base + reg_cr5);
-- 
cgit v1.2.2


From 15b29dae6604d2d2daf586429ff12f26272a868a Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:03 +0800
Subject: ASoC: fsl_sai: Drop useless ret in startup()

We can save this ret to make the code neater.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 8450bff6fb13..fc4cd95ad0a9 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -334,12 +334,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 static int fsl_sai_startup(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *cpu_dai)
 {
-	int ret;
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
 
-	ret = clk_prepare_enable(sai->clk);
-
-	return ret;
+	return clk_prepare_enable(sai->clk);
 }
 
 static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
-- 
cgit v1.2.2


From 190af12dad975f2ea7d69d1c5c9d36fec64da767 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:04 +0800
Subject: ASoC: fsl_sai: Make dev_err information neater

Since using dev_err() there's no need to mention SAI any more, it will
print the full name of the driver -- fsl_sai.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 16 ++++------------
 1 file changed, 4 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index fc4cd95ad0a9..68d666b491de 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -108,18 +108,14 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
 					FSL_FMT_TRANSMITTER);
 	if (ret) {
-		dev_err(cpu_dai->dev,
-				"Cannot set SAI's transmitter sysclk: %d\n",
-				ret);
+		dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret);
 		goto err_clk;
 	}
 
 	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
 					FSL_FMT_RECEIVER);
 	if (ret) {
-		dev_err(cpu_dai->dev,
-				"Cannot set SAI's receiver sysclk: %d\n",
-				ret);
+		dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret);
 		goto err_clk;
 	}
 
@@ -220,17 +216,13 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 
 	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
 	if (ret) {
-		dev_err(cpu_dai->dev,
-				"Cannot set SAI's transmitter format: %d\n",
-				ret);
+		dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret);
 		goto err_clk;
 	}
 
 	ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
 	if (ret) {
-		dev_err(cpu_dai->dev,
-				"Cannot set SAI's receiver format: %d\n",
-				ret);
+		dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret);
 		goto err_clk;
 	}
 
-- 
cgit v1.2.2


From 4e3a99f5b004b30bc604d82e5498700649148e0d Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 20 Dec 2013 16:41:05 +0800
Subject: ASoC: fsl_sai: Sort local variable in general way

Generally we would write code for local variable like:
static new_func()
{
	struct xxx *yyy;
	...
	int ret;
}

But this driver only follows this pattern for some functions, not all.
Thus this patch sorts the local variable in the general way.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 68d666b491de..b72132fa70de 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -53,8 +53,8 @@ static inline void sai_writel(struct fsl_sai *sai,
 static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int fsl_dir)
 {
-	u32 val_cr2, reg_cr2;
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 val_cr2, reg_cr2;
 
 	if (fsl_dir == FSL_FMT_TRANSMITTER)
 		reg_cr2 = FSL_SAI_TCR2;
@@ -90,8 +90,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int dir)
 {
-	int ret;
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret;
 
 	if (dir == SND_SOC_CLOCK_IN)
 		return 0;
@@ -128,8 +128,8 @@ err_clk:
 static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 				unsigned int fmt, int fsl_dir)
 {
-	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
 
 	if (fsl_dir == FSL_FMT_TRANSMITTER) {
 		reg_cr2 = FSL_SAI_TCR2;
@@ -207,8 +207,8 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 
 static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 {
-	int ret;
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret;
 
 	ret = clk_prepare_enable(sai->clk);
 	if (ret)
@@ -236,9 +236,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params,
 		struct snd_soc_dai *cpu_dai)
 {
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr;
 	unsigned int channels = params_channels(params);
-	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
 	u32 word_width = snd_pcm_format_width(params_format(params));
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -383,10 +383,10 @@ static const struct snd_soc_component_driver fsl_component = {
 
 static int fsl_sai_probe(struct platform_device *pdev)
 {
-	int ret;
+	struct device_node *np = pdev->dev.of_node;
 	struct fsl_sai *sai;
 	struct resource *res;
-	struct device_node *np = pdev->dev.of_node;
+	int ret;
 
 	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
 	if (!sai)
-- 
cgit v1.2.2


From 150116bcfbd9004f01e682535baddaca2b90d3b2 Mon Sep 17 00:00:00 2001
From: Michael Trimarchi <michael@amarulasolutions.com>
Date: Mon, 23 Dec 2013 11:16:43 +0100
Subject: ALSA: hiface: Fix typo in 352800 rate definition

The Vaughan device support the 352800 rate and not
the 352000

Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/hiface/pcm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c
index c21a3df9a0df..2c44139b4041 100644
--- a/sound/usb/hiface/pcm.c
+++ b/sound/usb/hiface/pcm.c
@@ -110,7 +110,7 @@ static const struct snd_pcm_hardware pcm_hw = {
 #define HIFACE_RATE_96000  0x4a
 #define HIFACE_RATE_176400 0x40
 #define HIFACE_RATE_192000 0x48
-#define HIFACE_RATE_352000 0x58
+#define HIFACE_RATE_352800 0x58
 #define HIFACE_RATE_384000 0x68
 
 static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate)
@@ -141,8 +141,8 @@ static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate)
 	case 192000:
 		rate_value = HIFACE_RATE_192000;
 		break;
-	case 352000:
-		rate_value = HIFACE_RATE_352000;
+	case 352800:
+		rate_value = HIFACE_RATE_352800;
 		break;
 	case 384000:
 		rate_value = HIFACE_RATE_384000;
-- 
cgit v1.2.2


From d754fa9ad18d16209c276fc6241aa2d10f819ede Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:09 +0100
Subject: ASoC: blackfin: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 -
 sound/soc/blackfin/bf5xx-i2s-pcm.c  | 3 ---
 2 files changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 1d4c676eb6cc..cdb8ee75ded9 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -107,7 +107,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
 #endif
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
 
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 0x10000,
 	.periods_min		= 1,
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 2a5b43417fd5..a3881c4381c9 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -52,9 +52,6 @@ static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				   SNDRV_PCM_INFO_MMAP_VALID |
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				   SNDRV_PCM_FMTBIT_S24_LE |
-				   SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 0x10000,
 	.periods_min		= 1,
-- 
cgit v1.2.2


From f52c91921553be26c7a0de13daa0d18ef46655ff Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:10 +0100
Subject: ASoC: davinci: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/davinci-pcm.c | 28 ----------------------------
 1 file changed, 28 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index fb5d107f5603..14145cdf8a11 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -46,33 +46,11 @@ static void print_buf_info(int slot, char *name)
 }
 #endif
 
-#define DAVINCI_PCM_FMTBITS	(\
-				SNDRV_PCM_FMTBIT_S8	|\
-				SNDRV_PCM_FMTBIT_U8	|\
-				SNDRV_PCM_FMTBIT_S16_LE	|\
-				SNDRV_PCM_FMTBIT_S16_BE	|\
-				SNDRV_PCM_FMTBIT_U16_LE	|\
-				SNDRV_PCM_FMTBIT_U16_BE	|\
-				SNDRV_PCM_FMTBIT_S24_LE	|\
-				SNDRV_PCM_FMTBIT_S24_BE	|\
-				SNDRV_PCM_FMTBIT_U24_LE	|\
-				SNDRV_PCM_FMTBIT_U24_BE	|\
-				SNDRV_PCM_FMTBIT_S32_LE	|\
-				SNDRV_PCM_FMTBIT_S32_BE	|\
-				SNDRV_PCM_FMTBIT_U32_LE	|\
-				SNDRV_PCM_FMTBIT_U32_BE)
-
 static struct snd_pcm_hardware pcm_hardware_playback = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
 		 SNDRV_PCM_INFO_BATCH),
-	.formats = DAVINCI_PCM_FMTBITS,
-	.rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT,
-	.rate_min = 8000,
-	.rate_max = 192000,
-	.channels_min = 2,
-	.channels_max = 384,
 	.buffer_bytes_max = 128 * 1024,
 	.period_bytes_min = 32,
 	.period_bytes_max = 8 * 1024,
@@ -86,12 +64,6 @@ static struct snd_pcm_hardware pcm_hardware_capture = {
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_PAUSE |
 		 SNDRV_PCM_INFO_BATCH),
-	.formats = DAVINCI_PCM_FMTBITS,
-	.rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT,
-	.rate_min = 8000,
-	.rate_max = 192000,
-	.channels_min = 2,
-	.channels_max = 384,
 	.buffer_bytes_max = 128 * 1024,
 	.period_bytes_min = 32,
 	.period_bytes_max = 8 * 1024,
-- 
cgit v1.2.2


From 3cec159cfb3fc69190f3ccdc2d1329c66775529f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:19 +0100
Subject: ASoC: s6000: Don't set unused struct snd_pcm_hardware fields
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Daniel Glöckner <daniel-gl@gmx.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/s6000/s6000-pcm.c | 7 -------
 1 file changed, 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index d219880815c0..fb8461e1b1f6 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -33,13 +33,6 @@ static struct snd_pcm_hardware s6000_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX),
-	.formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE),
-	.rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
-		  SNDRV_PCM_RATE_8000_192000),
-	.rate_min = 0,
-	.rate_max = 1562500,
-	.channels_min = 2,
-	.channels_max = 8,
 	.buffer_bytes_max = 0x7ffffff0,
 	.period_bytes_min = 16,
 	.period_bytes_max = 0xfffff0,
-- 
cgit v1.2.2


From 60e21d2873440fc005c88970960bed678138217e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:17 +0100
Subject: ASoC: omap: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/omap/omap-pcm.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index b8fa9862e54c..07b8b7bc9d20 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -45,8 +45,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 64 * 1024,
 	.periods_min		= 2,
-- 
cgit v1.2.2


From e6dc12d7198eddba2e3e7a13feab5c7edde7ba1d Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Wed, 25 Dec 2013 11:20:14 +0800
Subject: ASoC: fsl_sai: Move the global registers setting to _dai_probe()

Because we cannot make sure which one of _dai_fmt() and _dai_sysclk()
will be firstly called. So move the RCSR/TCSR and TCR1/RCR1's
initialization to _dai_probe(), and this can make sure that before any
of {T,R}CR{1~5} register to be set the RCSR/TCSR's RE/TE bit has been
cleared for the hareware limitation.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 17 ++++++++++++-----
 1 file changed, 12 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index b72132fa70de..596aabbf9037 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -100,11 +100,6 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	if (ret)
 		return ret;
 
-	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
-	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
-	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
-	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
-
 	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
 					FSL_FMT_TRANSMITTER);
 	if (ret) {
@@ -351,6 +346,18 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
 static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
+	int ret;
+
+	ret = clk_prepare_enable(sai->clk);
+	if (ret)
+		return ret;
+
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
+	sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
+	sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
+	sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
+
+	clk_disable_unprepare(sai->clk);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
 				&sai->dma_params_rx);
-- 
cgit v1.2.2


From 61c66c60c75bd7f9650e27fb0a017bc39f2629a2 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Thu, 26 Dec 2013 11:41:29 +0530
Subject: ASoC: samsung: Trivial cleanups in header files

commit 5d229ce569be ("ASoC: samsung: move plat/ headers to local directory")
moved the header files but forgot to clean the pointers to their old locaton.
Remove them now.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Jingoo Han <jg1.han@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/regs-ac97.h | 9 ++++-----
 sound/soc/samsung/regs-iis.h  | 9 ++++-----
 2 files changed, 8 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/regs-ac97.h b/sound/soc/samsung/regs-ac97.h
index c3878f7acb83..a71be45bbffc 100644
--- a/sound/soc/samsung/regs-ac97.h
+++ b/sound/soc/samsung/regs-ac97.h
@@ -1,5 +1,4 @@
-/* arch/arm/mach-s3c2410/include/mach/regs-ac97.h
- *
+/*
  * Copyright (c) 2006 Simtec Electronics <linux@simtec.co.uk>
  *		http://www.simtec.co.uk/products/SWLINUX/
  *
@@ -10,8 +9,8 @@
  * S3C2440 AC97 Controller
 */
 
-#ifndef __ASM_ARCH_REGS_AC97_H
-#define __ASM_ARCH_REGS_AC97_H __FILE__
+#ifndef __SAMSUNG_REGS_AC97_H__
+#define __SAMSUNG_REGS_AC97_H__
 
 #define S3C_AC97_GLBCTRL				(0x00)
 
@@ -64,4 +63,4 @@
 #define S3C_AC97_PCM_DATA				(0x18)
 #define S3C_AC97_MIC_DATA				(0x1C)
 
-#endif /* __ASM_ARCH_REGS_AC97_H */
+#endif /* __SAMSUNG_REGS_AC97_H__ */
diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h
index a18d35e7a735..dc6cbbe9c4f0 100644
--- a/sound/soc/samsung/regs-iis.h
+++ b/sound/soc/samsung/regs-iis.h
@@ -1,5 +1,4 @@
-/* arch/arm/plat-samsung/include/plat/regs-iis.h
- *
+/*
  * Copyright (c) 2003 Simtec Electronics <linux@simtec.co.uk>
  *		      http://www.simtec.co.uk/products/SWLINUX/
  *
@@ -10,8 +9,8 @@
  * S3C2410 IIS register definition
 */
 
-#ifndef __ASM_ARCH_REGS_IIS_H
-#define __ASM_ARCH_REGS_IIS_H
+#ifndef __SAMSUNG_REGS_IIS_H__
+#define __SAMSUNG_REGS_IIS_H__
 
 #define S3C2410_IISCON			(0x00)
 
@@ -67,4 +66,4 @@
 
 #define S3C2410_IISFIFO			(0x10)
 
-#endif /* __ASM_ARCH_REGS_IIS_H */
+#endif /* __SAMSUNG_REGS_IIS_H__ */
-- 
cgit v1.2.2


From e5d0fa9c3ec59a40e0285d96b65b7f62875acd42 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Wed, 25 Dec 2013 12:40:04 +0800
Subject: ASoC: fsl_sai: Add disable operation for the corresponding data
 channel.

Enables/Disables the corresponding data channel for tx/rx operation.
A channel must be enabled before its FIFO is accessed, and then disable
it when tx/rx is stopped or idle.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 21 +++++++++++++--------
 1 file changed, 13 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 596aabbf9037..af802465456e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -124,20 +124,17 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 				unsigned int fmt, int fsl_dir)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
-	u32 val_cr2, val_cr3, val_cr4, reg_cr2, reg_cr3, reg_cr4;
+	u32 val_cr2, val_cr4, reg_cr2, reg_cr4;
 
 	if (fsl_dir == FSL_FMT_TRANSMITTER) {
 		reg_cr2 = FSL_SAI_TCR2;
-		reg_cr3 = FSL_SAI_TCR3;
 		reg_cr4 = FSL_SAI_TCR4;
 	} else {
 		reg_cr2 = FSL_SAI_RCR2;
-		reg_cr3 = FSL_SAI_RCR3;
 		reg_cr4 = FSL_SAI_RCR4;
 	}
 
 	val_cr2 = sai_readl(sai, sai->base + reg_cr2);
-	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
 	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
 
 	if (sai->big_endian_data)
@@ -188,13 +185,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	val_cr3 |= FSL_SAI_CR3_TRCE;
-
 	if (fsl_dir == FSL_FMT_RECEIVER)
 		val_cr2 |= FSL_SAI_CR2_SYNC;
 
 	sai_writel(sai, val_cr2, sai->base + reg_cr2);
-	sai_writel(sai, val_cr3, sai->base + reg_cr3);
 	sai_writel(sai, val_cr4, sai->base + reg_cr4);
 
 	return 0;
@@ -278,7 +272,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
-	unsigned int tcsr, rcsr;
+	u32 tcsr, rcsr, val_cr3, reg_cr3;
 
 	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
 	rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR);
@@ -286,17 +280,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		tcsr |= FSL_SAI_CSR_FRDE;
 		rcsr &= ~FSL_SAI_CSR_FRDE;
+		reg_cr3 = FSL_SAI_TCR3;
 	} else {
 		rcsr |= FSL_SAI_CSR_FRDE;
 		tcsr &= ~FSL_SAI_CSR_FRDE;
+		reg_cr3 = FSL_SAI_RCR3;
 	}
 
+	val_cr3 = sai_readl(sai, sai->base + reg_cr3);
+
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		tcsr |= FSL_SAI_CSR_TERE;
 		rcsr |= FSL_SAI_CSR_TERE;
+		val_cr3 |= FSL_SAI_CR3_TRCE;
+
+		sai_writel(sai, val_cr3, sai->base + reg_cr3);
 		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
 		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
 		break;
@@ -308,8 +309,12 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 			tcsr &= ~FSL_SAI_CSR_TERE;
 			rcsr &= ~FSL_SAI_CSR_TERE;
 		}
+
+		val_cr3 &= ~FSL_SAI_CR3_TRCE;
+
 		sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
 		sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
+		sai_writel(sai, val_cr3, sai->base + reg_cr3);
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.2


From 95ab1297e7b1a57c2b777f210b74f3a2bd2ac269 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:25 +0100
Subject: ASoC: soc-utils: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-utils.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 5e633659c1b3..6ebdfd9a1a1d 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -59,10 +59,6 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
 EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
 
 static const struct snd_pcm_hardware dummy_dma_hardware = {
-	.formats		= 0xffffffff,
-	.channels_min		= 1,
-	.channels_max		= UINT_MAX,
-
 	/* Random values to keep userspace happy when checking constraints */
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_BLOCK_TRANSFER,
-- 
cgit v1.2.2


From 328089a47112a4fc06071e2003ecd75cc6d31029 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:20 +0100
Subject: ASoC: samsung: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/dma.c  | 6 ------
 sound/soc/samsung/idma.c | 8 --------
 2 files changed, 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index ee23194f7ab8..dc09b71b7d9f 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -35,12 +35,6 @@ static const struct snd_pcm_hardware dma_hardware = {
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				    SNDRV_PCM_INFO_MMAP |
 				    SNDRV_PCM_INFO_MMAP_VALID,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				    SNDRV_PCM_FMTBIT_U16_LE |
-				    SNDRV_PCM_FMTBIT_U8 |
-				    SNDRV_PCM_FMTBIT_S8,
-	.channels_min		= 2,
-	.channels_max		= 2,
 	.buffer_bytes_max	= 128*1024,
 	.period_bytes_min	= PAGE_SIZE,
 	.period_bytes_max	= PAGE_SIZE*2,
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index e4f318fc2f82..3d5cf1530b6f 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -35,14 +35,6 @@ static const struct snd_pcm_hardware idma_hardware = {
 		    SNDRV_PCM_INFO_MMAP_VALID |
 		    SNDRV_PCM_INFO_PAUSE |
 		    SNDRV_PCM_INFO_RESUME,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE |
-		    SNDRV_PCM_FMTBIT_U16_LE |
-		    SNDRV_PCM_FMTBIT_S24_LE |
-		    SNDRV_PCM_FMTBIT_U24_LE |
-		    SNDRV_PCM_FMTBIT_U8 |
-		    SNDRV_PCM_FMTBIT_S8,
-	.channels_min = 2,
-	.channels_max = 2,
 	.buffer_bytes_max = MAX_IDMA_BUFFER,
 	.period_bytes_min = 128,
 	.period_bytes_max = MAX_IDMA_PERIOD,
-- 
cgit v1.2.2


From 38136bde7691bdafa91c2320e014913aec6dbe6b Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:23 +0100
Subject: ASoC: txx9: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/txx9/txx9aclc.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index fbd077f4de72..f0829de28708 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -40,11 +40,6 @@ static const struct snd_pcm_hardware txx9aclc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_INTERLEAVED |
 			    SNDRV_PCM_INFO_BATCH |
 			    SNDRV_PCM_INFO_PAUSE,
-#ifdef __BIG_ENDIAN
-	.formats	  = SNDRV_PCM_FMTBIT_S16_BE,
-#else
-	.formats	  = SNDRV_PCM_FMTBIT_S16_LE,
-#endif
 	.period_bytes_min = 1024,
 	.period_bytes_max = 8 * 1024,
 	.periods_min	  = 2,
-- 
cgit v1.2.2


From 323702b4e06dfd1a4ee6cee5834a889b9663cccf Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:18 +0100
Subject: ASoC: mmp: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/pxa/mmp-pcm.c | 6 ------
 1 file changed, 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 682ee52942bf..5e8d81330173 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -36,14 +36,9 @@ struct mmp_dma_data {
 		SNDRV_PCM_INFO_PAUSE |		\
 		SNDRV_PCM_INFO_RESUME)
 
-#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
-			 SNDRV_PCM_FMTBIT_S24_LE | \
-			 SNDRV_PCM_FMTBIT_S32_LE)
-
 static struct snd_pcm_hardware mmp_pcm_hardware[] = {
 	{
 		.info			= MMP_PCM_INFO,
-		.formats		= MMP_PCM_FORMATS,
 		.period_bytes_min	= 1024,
 		.period_bytes_max	= 2048,
 		.periods_min		= 2,
@@ -53,7 +48,6 @@ static struct snd_pcm_hardware mmp_pcm_hardware[] = {
 	},
 	{
 		.info			= MMP_PCM_INFO,
-		.formats		= MMP_PCM_FORMATS,
 		.period_bytes_min	= 1024,
 		.period_bytes_max	= 2048,
 		.periods_min		= 2,
-- 
cgit v1.2.2


From 111bd7b18e13c66552e9a672000eeacacd414a65 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:11 +0100
Subject: ASoC: ep93xx: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/cirrus/ep93xx-pcm.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c
index ca6698d75736..5f664471d99e 100644
--- a/sound/soc/cirrus/ep93xx-pcm.c
+++ b/sound/soc/cirrus/ep93xx-pcm.c
@@ -30,15 +30,6 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
 				   SNDRV_PCM_INFO_MMAP_VALID	|
 				   SNDRV_PCM_INFO_INTERLEAVED	|
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER),
-				   
-	.rates			= SNDRV_PCM_RATE_8000_192000,
-	.rate_min		= SNDRV_PCM_RATE_8000,
-	.rate_max		= SNDRV_PCM_RATE_192000,
-	
-	.formats		= (SNDRV_PCM_FMTBIT_S16_LE |
-				   SNDRV_PCM_FMTBIT_S24_LE |
-				   SNDRV_PCM_FMTBIT_S32_LE),
-	
 	.buffer_bytes_max	= 131072,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 32768,
-- 
cgit v1.2.2


From 496a39d9ec238569fac6daceac8f5420c5edc2f1 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 31 Dec 2013 15:33:21 +0800
Subject: ASoC: fsl_sai: Fix one bug for hardware limitation.

This is maybe one bug or a limitation of the hardware that the {T,R}CR2's
Synchronous Mode bits must be set as late as possible, or the SAI device
maybe hanged up, and there has not any explaination about this limitation
in the SAI Data Sheet.

And the {T,R}CR2's Synchronous Mode bits must be set at the same time whether
for Tx or Rx stream.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 15 ++++++++++-----
 1 file changed, 10 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index af802465456e..2ece14716c67 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -145,7 +145,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		val_cr4 |= FSL_SAI_CR4_FSE;
-		val_cr4 |= FSL_SAI_CR4_FSP;
 		break;
 	default:
 		return -EINVAL;
@@ -185,9 +184,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	if (fsl_dir == FSL_FMT_RECEIVER)
-		val_cr2 |= FSL_SAI_CR2_SYNC;
-
 	sai_writel(sai, val_cr2, sai->base + reg_cr2);
 	sai_writel(sai, val_cr4, sai->base + reg_cr4);
 
@@ -253,6 +249,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
 	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
 
+	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
 	if (sai->big_endian_data)
 		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
 	else
@@ -272,7 +269,15 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
-	u32 tcsr, rcsr, val_cr3, reg_cr3;
+	u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3;
+
+	val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2);
+	val_cr2 &= ~FSL_SAI_CR2_SYNC;
+	sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2);
+
+	val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2);
+	val_cr2 |= FSL_SAI_CR2_SYNC;
+	sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2);
 
 	tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
 	rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR);
-- 
cgit v1.2.2


From 72aa62bed3ea30635156fad95f123a0b665072bf Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 31 Dec 2013 15:33:22 +0800
Subject: ASoC: fsl_sai: fix the endianess for SAI fifo data.

Revert the SAI's endianess for fifo data to/from DMA engine.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 2ece14716c67..5d38a6749b9f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -138,9 +138,9 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	val_cr4 = sai_readl(sai, sai->base + reg_cr4);
 
 	if (sai->big_endian_data)
-		val_cr4 |= FSL_SAI_CR4_MF;
-	else
 		val_cr4 &= ~FSL_SAI_CR4_MF;
+	else
+		val_cr4 |= FSL_SAI_CR4_MF;
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
@@ -251,9 +251,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 	val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
 	if (sai->big_endian_data)
-		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
-	else
 		val_cr5 |= FSL_SAI_CR5_FBT(0);
+	else
+		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
 
 	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
 	val_mr = ~0UL - ((1 << channels) - 1);
-- 
cgit v1.2.2


From 192043cf608909eb5728a5fd68f5234b90d9415b Mon Sep 17 00:00:00 2001
From: Bo Shen <voice.shen@atmel.com>
Date: Wed, 27 Nov 2013 18:05:10 +0800
Subject: ASoC: atmel: sam9x5_wm8731: remove platform_set_drvdata

When call snd_soc_register_card, it will set driver data to this
device through dev_set_drvdata, then in driver, no need to call
platform_set_drvdata again, so remove it.

Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/atmel/sam9x5_wm8731.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..6f4e812d6e61 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -153,8 +153,6 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	of_node_put(codec_np);
 	of_node_put(cpu_np);
 
-	platform_set_drvdata(pdev, card);
-
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(&pdev->dev,
-- 
cgit v1.2.2


From d4c22094b256a7327346738721b89a78d4680b08 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 23 Dec 2013 12:57:01 +0800
Subject: ASoC: simple-card: Add DAPM routes parse from device tree

Parses a simple DAPM route table from device tree.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 11 +++++++++--
 1 file changed, 9 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3d190d05ad4a..6230efb05fd7 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -116,12 +116,18 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 {
 	struct device_node *np;
 	char *name;
-	int ret = 0;
+	int ret;
 
 	/* get CPU/CODEC common format via simple-audio-card,format */
 	info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
 		(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
 
+	/* DAPM routes */
+	ret = snd_soc_of_parse_audio_routing(&info->snd_card,
+					"simple-audio-routing");
+	if (ret)
+		return ret;
+
 	/* CPU sub-node */
 	ret = -EINVAL;
 	np = of_get_child_by_name(node, "simple-audio-card,cpu");
@@ -182,6 +188,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
 		if (cinfo) {
 			int ret;
+			cinfo->snd_card.dev = &pdev->dev;
 			ret = asoc_simple_card_parse_of(np, cinfo, dev,
 							&of_cpu,
 							&of_codec,
@@ -193,6 +200,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 			}
 		}
 	} else {
+		cinfo->snd_card.dev = &pdev->dev;
 		cinfo = pdev->dev.platform_data;
 	}
 
@@ -232,7 +240,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	cinfo->snd_card.owner		= THIS_MODULE;
 	cinfo->snd_card.dai_link	= &cinfo->snd_link;
 	cinfo->snd_card.num_links	= 1;
-	cinfo->snd_card.dev		= &pdev->dev;
 
 	return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card);
 }
-- 
cgit v1.2.2


From e874ddead38996ec40c6a6be2347a69fac520126 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 23 Dec 2013 13:24:59 +0800
Subject: ASoC: simple-card: Cleanup __asoc_simple_card_dai_init() ret check

The ret parameter is always equal to zero till here.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6230efb05fd7..3ba65bb63439 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -25,7 +25,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 
 	daifmt |= set->fmt;
 
-	if (!ret && daifmt)
+	if (daifmt)
 		ret = snd_soc_dai_set_fmt(dai, daifmt);
 
 	if (ret == -ENOTSUPP) {
-- 
cgit v1.2.2


From 0f7f3d1f17c2e4d73e449e6acb2007b13813c58e Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Fri, 20 Dec 2013 12:40:16 +0200
Subject: ASoC: hdmi-codec: Add devicetree binding with documentation

Signed-off-by: Jyri Sarha <jsarha@ti.com>
cc: bcousson@baylibre.com
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/hdmi.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
index 32797a8e4ee9..9cb1c7d3e1dc 100644
--- a/sound/soc/codecs/hdmi.c
+++ b/sound/soc/codecs/hdmi.c
@@ -20,6 +20,7 @@
  */
 #include <linux/module.h>
 #include <sound/soc.h>
+#include <linux/of_device.h>
 
 #define DRV_NAME "hdmi-audio-codec"
 
@@ -60,6 +61,14 @@ static struct snd_soc_dai_driver hdmi_codec_dai = {
 
 };
 
+#ifdef CONFIG_OF
+static const struct of_device_id hdmi_audio_codec_ids[] = {
+	{ .compatible = "linux,hdmi-audio", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
+#endif
+
 static struct snd_soc_codec_driver hdmi_codec = {
 	.dapm_widgets = hdmi_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
@@ -83,6 +92,7 @@ static struct platform_driver hdmi_codec_driver = {
 	.driver		= {
 		.name	= DRV_NAME,
 		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(hdmi_audio_codec_ids),
 	},
 
 	.probe		= hdmi_codec_probe,
-- 
cgit v1.2.2


From e337853ebb46d012c069ca47ba3ce9f4744305ea Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:26:31 -0800
Subject: ASoC: rsnd: add rsnd_adg_set_ssi_clk() and cleanup adg

This patch adds rsnd_adg_set_ssi_clk() to access to
AUDIO_CLK_SEL0/1/2, and removes last user of
rsnd_write/read/bset which is very low level function.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/adg.c | 58 ++++++++++++++++++++-----------------------------
 1 file changed, 24 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 9430097979a5..55d0394955b2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -30,41 +30,41 @@ struct rsnd_adg {
 	     i++, (pos) = adg->clk[i])
 #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
 
-static enum rsnd_reg rsnd_adg_ssi_reg_get(int id)
+static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
 {
-	enum rsnd_reg reg;
+	int id = rsnd_mod_id(mod);
+	int shift = (id % 4) * 8;
+	u32 mask = 0xFF << shift;
+
+	val = val << shift;
 
 	/*
 	 * SSI 8 is not connected to ADG.
 	 * it works with SSI 7
 	 */
 	if (id == 8)
-		return RSND_REG_MAX;
-
-	if (0 <= id && id <= 3)
-		reg = RSND_REG_AUDIO_CLK_SEL0;
-	else if (4 <= id && id <= 7)
-		reg = RSND_REG_AUDIO_CLK_SEL1;
-	else
-		reg = RSND_REG_AUDIO_CLK_SEL2;
-
-	return reg;
+		return;
+
+	switch (id / 4) {
+	case 0:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL0, mask, val);
+		break;
+	case 1:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL1, mask, val);
+		break;
+	case 2:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL2, mask, val);
+		break;
+	}
 }
 
 int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod)
 {
-	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
-	enum rsnd_reg reg;
-	int id;
-
 	/*
 	 * "mod" = "ssi" here.
 	 * we can get "ssi id" from mod
 	 */
-	id  = rsnd_mod_id(mod);
-	reg = rsnd_adg_ssi_reg_get(id);
-
-	rsnd_write(priv, mod, reg, 0);
+	rsnd_adg_set_ssi_clk(mod, 0);
 
 	return 0;
 }
@@ -75,8 +75,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct clk *clk;
-	enum rsnd_reg reg;
-	int id, shift, i;
+	int i;
 	u32 data;
 	int sel_table[] = {
 		[CLKA] = 0x1,
@@ -125,19 +124,10 @@ found_clock:
 	 * This "mod" = "ssi" here.
 	 * we can get "ssi id" from mod
 	 */
-	id  = rsnd_mod_id(mod);
-	reg = rsnd_adg_ssi_reg_get(id);
-
-	dev_dbg(dev, "ADG: ssi%d selects clk%d = %d", id, i, rate);
-
-	/*
-	 * Enable SSIx clock
-	 */
-	shift = (id % 4) * 8;
+	rsnd_adg_set_ssi_clk(mod, data);
 
-	rsnd_bset(priv, mod, reg,
-		   0xFF << shift,
-		   data << shift);
+	dev_dbg(dev, "ADG: ssi%d selects clk%d = %d",
+		rsnd_mod_id(mod), i, rate);
 
 	return 0;
 }
-- 
cgit v1.2.2


From 729aca51a19f2e2b3404c29b82df61d714150a49 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:26:44 -0800
Subject: ASoC: rsnd: tidyup ssi comment

we can check rsnd_ssi_init(), not, rsnd_ssi_start()

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/ssi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 82b04c6f352d..aff5b76d1fd5 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -209,7 +209,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 			/*
 			 * this driver is assuming that
 			 * system word is 64fs (= 2 x 32bit)
-			 * see rsnd_ssi_start()
+			 * see rsnd_ssi_init()
 			 */
 			main_rate = rate / adg_clk_div_table[i]
 				* 32 * 2 * ssi_clk_mul_table[j];
-- 
cgit v1.2.2


From 690ef81ebe02a43991b0fcb418d77b8420346cfd Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:27:03 -0800
Subject: ASoC: rsnd: tidyup register naming

Use correct register name which appears in the datasheet

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/gen.c  | 4 ++--
 sound/soc/sh/rcar/rsnd.h | 4 ++--
 sound/soc/sh/rcar/scu.c  | 4 ++--
 3 files changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index d0ab203836de..862758d3ec06 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -314,11 +314,11 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL0,	0x08),
 		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL1,	0x0c),
 		RSND_GEN1_S_REG(gen, SRU,	SRC_TMG_SEL2,	0x10),
-		RSND_GEN1_S_REG(gen, SRU,	SRC_CTRL,	0xc0),
+		RSND_GEN1_S_REG(gen, SRU,	SRC_ROUTE_CTRL,	0xc0),
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE0,	0xD0),
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE1,	0xD4),
 		RSND_GEN1_M_REG(gen, SRU,	BUSIF_MODE,	0x20,	0x4),
-		RSND_GEN1_M_REG(gen, SRU,	BUSIF_ADINR,	0x214,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_ADINR,	0x214,	0x40),
 
 		RSND_GEN1_S_REG(gen, ADG,	BRRA,		0x00),
 		RSND_GEN1_S_REG(gen, ADG,	BRRB,		0x04),
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index bff7b9e53066..d5c0182c8e50 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -36,12 +36,12 @@ enum rsnd_reg {
 	RSND_REG_SRC_TMG_SEL0,
 	RSND_REG_SRC_TMG_SEL1,
 	RSND_REG_SRC_TMG_SEL2,
-	RSND_REG_SRC_CTRL,
+	RSND_REG_SRC_ROUTE_CTRL,
 	RSND_REG_SSI_MODE0,
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
-	RSND_REG_BUSIF_ADINR,
 	RSND_REG_INT_ENABLE,
+	RSND_REG_SRC_ADINR,
 
 	/* ADG */
 	RSND_REG_BRRA,
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 1ab1bce6be7f..187f7dc30369 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -115,7 +115,7 @@ static int rsnd_scu_set_mode(struct rsnd_priv *priv,
 
 	if (rsnd_is_gen1(priv)) {
 		val = (1 << id);
-		rsnd_mod_bset(mod, SRC_CTRL, val, val);
+		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
 	}
 
 	return 0;
@@ -141,7 +141,7 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
 	}
 
 	rsnd_mod_write(mod, BUSIF_MODE, 1);
-	rsnd_mod_write(mod, BUSIF_ADINR, adinr);
+	rsnd_mod_write(mod, SRC_ADINR, adinr);
 
 	return 0;
 }
-- 
cgit v1.2.2


From 7808aa30d6cf366e5f627dcbf7c84f9dc6e602ab Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:27:19 -0800
Subject: ASoC: rsnd: make sure variable name for 44.1kHz/48kHz

This driver is assuming that
RBGA is used as source clock of 44.1kHz category, and
RBGB is used as source clock of 48kHz category.
This patch clarifies the variable name.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/adg.c | 20 ++++++++++----------
 1 file changed, 10 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 55d0394955b2..2e71a7bda4c2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -19,8 +19,8 @@
 struct rsnd_adg {
 	struct clk *clk[CLKMAX];
 
-	int rate_of_441khz_div_6;
-	int rate_of_48khz_div_6;
+	int rbga_rate_for_441khz_div_6;	/* RBGA */
+	int rbgb_rate_for_48khz_div_6;	/* RBGB */
 	u32 ckr;
 };
 
@@ -101,12 +101,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate)
 	/*
 	 * find 1/6 clock from BRGA/BRGB
 	 */
-	if (rate == adg->rate_of_441khz_div_6) {
+	if (rate == adg->rbga_rate_for_441khz_div_6) {
 		data = 0x10;
 		goto found_clock;
 	}
 
-	if (rate == adg->rate_of_48khz_div_6) {
+	if (rate == adg->rbgb_rate_for_48khz_div_6) {
 		data = 0x20;
 		goto found_clock;
 	}
@@ -156,8 +156,8 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
 	 *	rsnd_adg_ssi_clk_try_start()
 	 */
 	ckr = 0;
-	adg->rate_of_441khz_div_6 = 0;
-	adg->rate_of_48khz_div_6  = 0;
+	adg->rbga_rate_for_441khz_div_6 = 0;
+	adg->rbgb_rate_for_48khz_div_6  = 0;
 	for_each_rsnd_clk(clk, adg, i) {
 		rate = clk_get_rate(clk);
 
@@ -165,14 +165,14 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg)
 			continue;
 
 		/* RBGA */
-		if (!adg->rate_of_441khz_div_6 && (0 == rate % 44100)) {
-			adg->rate_of_441khz_div_6 = rate / 6;
+		if (!adg->rbga_rate_for_441khz_div_6 && (0 == rate % 44100)) {
+			adg->rbga_rate_for_441khz_div_6 = rate / 6;
 			ckr |= brg_table[i] << 20;
 		}
 
 		/* RBGB */
-		if (!adg->rate_of_48khz_div_6 && (0 == rate % 48000)) {
-			adg->rate_of_48khz_div_6 = rate / 6;
+		if (!adg->rbgb_rate_for_48khz_div_6 && (0 == rate % 48000)) {
+			adg->rbgb_rate_for_48khz_div_6 = rate / 6;
 			ckr |= brg_table[i] << 16;
 		}
 	}
-- 
cgit v1.2.2


From 2582718cb6bd620d37a54db885c75bfe4822db45 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:27:37 -0800
Subject: ASoC: rsnd: route setting is needed only Gen1

Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.

This patch cleanup it with focusing about SRC and Gen2 part.

rsnd_scu_set_route() is needed only Gen1.

This patch renames it to rsnd_scu_set_route_if_gen1()
and it adds comment to rsnd_reg member
in order to clarify it is used for Gen1.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/rsnd.h | 10 +++++-----
 sound/soc/sh/rcar/scu.c  |  6 ++++--
 2 files changed, 9 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index d5c0182c8e50..a14bc9274b3f 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -32,11 +32,11 @@
  */
 enum rsnd_reg {
 	/* SRU/SCU/SSIU */
-	RSND_REG_SRC_ROUTE_SEL,
-	RSND_REG_SRC_TMG_SEL0,
-	RSND_REG_SRC_TMG_SEL1,
-	RSND_REG_SRC_TMG_SEL2,
-	RSND_REG_SRC_ROUTE_CTRL,
+	RSND_REG_SRC_ROUTE_SEL,		/* for Gen1 */
+	RSND_REG_SRC_TMG_SEL0,		/* for Gen1 */
+	RSND_REG_SRC_TMG_SEL1,		/* for Gen1 */
+	RSND_REG_SRC_TMG_SEL2,		/* for Gen1 */
+	RSND_REG_SRC_ROUTE_CTRL,	/* for Gen1 */
 	RSND_REG_SSI_MODE0,
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 187f7dc30369..7642ec52b590 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -36,7 +36,8 @@ struct rsnd_scu {
 		     ((pos) = (struct rsnd_scu *)(priv)->scu + i);	\
 	     i++)
 
-static int rsnd_scu_set_route(struct rsnd_priv *priv,
+/* Gen1 only */
+static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 			      struct rsnd_mod *mod,
 			      struct rsnd_dai *rdai,
 			      struct rsnd_dai_stream *io)
@@ -174,7 +175,8 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 	}
 
 	/* it use DMA transter */
-	ret = rsnd_scu_set_route(priv, mod, rdai, io);
+
+	ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-- 
cgit v1.2.2


From af8a478821345fd264fd2294e80f5b0a28a518bc Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:04 -0800
Subject: ASoC: rsnd: add rsnd_scu_transfer_start()

Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.

This patch cleanup it with focusing about SRC and Gen2 part.

SRC_CTRL/BUSIF_MODE are used for transfer start.
This patch adds rsnd_scu_transfer_start() and merge these

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/scu.c | 39 ++++++++++++++++++++-------------------
 1 file changed, 20 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 7642ec52b590..3d8b57b1630b 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -106,22 +106,6 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	return 0;
 }
 
-static int rsnd_scu_set_mode(struct rsnd_priv *priv,
-			     struct rsnd_mod *mod,
-			     struct rsnd_dai *rdai,
-			     struct rsnd_dai_stream *io)
-{
-	int id = rsnd_mod_id(mod);
-	u32 val;
-
-	if (rsnd_is_gen1(priv)) {
-		val = (1 << id);
-		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
-	}
-
-	return 0;
-}
-
 static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
 			      struct rsnd_mod *mod,
 			      struct rsnd_dai *rdai,
@@ -141,12 +125,29 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
 		return -EIO;
 	}
 
-	rsnd_mod_write(mod, BUSIF_MODE, 1);
 	rsnd_mod_write(mod, SRC_ADINR, adinr);
 
 	return 0;
 }
 
+static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
+				   struct rsnd_mod *mod,
+				   struct rsnd_dai *rdai,
+				   struct rsnd_dai_stream *io)
+{
+	int id = rsnd_mod_id(mod);
+	u32 val;
+
+	if (rsnd_is_gen1(priv)) {
+		val = (1 << id);
+		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
+	}
+
+	rsnd_mod_write(mod, BUSIF_MODE, 1);
+
+	return 0;
+}
+
 bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod)
 {
 	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
@@ -180,11 +181,11 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
-	ret = rsnd_scu_set_mode(priv, mod, rdai, io);
+	ret = rsnd_scu_set_hpbif(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-	ret = rsnd_scu_set_hpbif(priv, mod, rdai, io);
+	ret = rsnd_scu_transfer_start(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-- 
cgit v1.2.2


From 52ea2a79f440740b57925e729a59337414b4c300 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:19 -0800
Subject: ASoC: rsnd: INT_ENABLE is needed only Gen2

INT_ENABLE is needed only Gen2.
rsnd_mod_write() do nothing on Gen1, but it is confusable.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/rsnd.h | 2 +-
 sound/soc/sh/rcar/ssi.c  | 5 +++--
 2 files changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index a14bc9274b3f..3774dfcfaf0f 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -40,7 +40,7 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE0,
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
-	RSND_REG_INT_ENABLE,
+	RSND_REG_INT_ENABLE,		/* for Gen2 */
 	RSND_REG_SRC_ADINR,
 
 	/* ADG */
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index aff5b76d1fd5..01b5cf945d61 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -457,8 +457,9 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
 	/* enable PIO IRQ */
 	ssi->cr_etc = UIEN | OIEN | DIEN;
 
-	/* enable PIO interrupt */
-	rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000);
+	/* enable PIO interrupt if gen2 */
+	if (rsnd_is_gen2(priv))
+		rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000);
 
 	rsnd_ssi_hw_start(ssi, rdai, io);
 
-- 
cgit v1.2.2


From 99feec32f26a3c267f89ce48db4bd36650a95f7f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:31 -0800
Subject: ASoC: rsnd: scu cleanup: add rsnd_scu_rate_ctrl()

Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.

This patch cleanup it with focusing about SRC and Gen2 part.

rsnd_scu_set_hpbif() is renamed to rsnd_scu_rate_ctrl(),
since its naming doesn't indicate the function meaning.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/scu.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 3d8b57b1630b..5f4f57206faf 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -106,7 +106,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	return 0;
 }
 
-static int rsnd_scu_set_hpbif(struct rsnd_priv *priv,
+static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv,
 			      struct rsnd_mod *mod,
 			      struct rsnd_dai *rdai,
 			      struct rsnd_dai_stream *io)
@@ -181,7 +181,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 	if (ret < 0)
 		return ret;
 
-	ret = rsnd_scu_set_hpbif(priv, mod, rdai, io);
+	ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-- 
cgit v1.2.2


From adcf7d5e7605e8134a99d415b7afd13f03c4bf23 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:39 -0800
Subject: ASoC: rsnd: tidyup rsnd_ssi_master_clk_start() parameter

Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.

This patch cleanup it with focusing about SRC and Gen2 part.

ssi clock which is calculated from rsnd_ssi_master_clk_start()
should have flexibility since Renesas sound has
SRC (= Sampling Rate Converter).
But current implementation is using runtime->rate directly.
This patch tidyup rsnd_ssi_master_clk_start() parameter
as preparation of future SRC support

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/ssi.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 01b5cf945d61..2db9711549f5 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -187,9 +187,10 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod,
 }
 
 static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
-				     unsigned int rate)
+				     struct rsnd_dai_stream *io)
 {
 	struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod);
+	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int i, j, ret;
 	int adg_clk_div_table[] = {
@@ -199,6 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 		1, 2, 4, 8, 16, 6, 12,
 	};
 	unsigned int main_rate;
+	unsigned int rate = runtime->rate;
 
 	/*
 	 * Find best clock, and try to start ADG
@@ -251,14 +253,10 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi,
 		clk_enable(ssi->clk);
 
 		if (rsnd_rdai_is_clk_master(rdai)) {
-			struct snd_pcm_runtime *runtime;
-
-			runtime = rsnd_io_to_runtime(io);
-
 			if (rsnd_ssi_clk_from_parent(ssi))
 				rsnd_ssi_hw_start(ssi->parent, rdai, io);
 			else
-				rsnd_ssi_master_clk_start(ssi, runtime->rate);
+				rsnd_ssi_master_clk_start(ssi, io);
 		}
 	}
 
-- 
cgit v1.2.2


From ef749400434cefd14fe02fe3de9e9f0125b2256d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 19 Dec 2013 19:28:51 -0800
Subject: ASoC: rsnd: add SRC (Sampling Rate Converter) support

This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/rcar/adg.c  |  73 +++++++++++++++++++++++
 sound/soc/sh/rcar/gen.c  |  10 ++++
 sound/soc/sh/rcar/rsnd.h |  18 ++++++
 sound/soc/sh/rcar/scu.c  | 152 ++++++++++++++++++++++++++++++++++++++++++++---
 sound/soc/sh/rcar/ssi.c  |   2 +-
 5 files changed, 247 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index 2e71a7bda4c2..a53235c4d1b0 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -30,6 +30,79 @@ struct rsnd_adg {
 	     i++, (pos) = adg->clk[i])
 #define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
 
+static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+					 struct rsnd_mod *mod,
+					 unsigned int src_rate,
+					 unsigned int dst_rate)
+{
+	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+	struct device *dev = rsnd_priv_to_dev(priv);
+	int idx, sel, div, shift;
+	u32 mask, val;
+	int id = rsnd_mod_id(mod);
+	unsigned int sel_rate [] = {
+		clk_get_rate(adg->clk[CLKA]),	/* 000: CLKA */
+		clk_get_rate(adg->clk[CLKB]),	/* 001: CLKB */
+		clk_get_rate(adg->clk[CLKC]),	/* 010: CLKC */
+		0,				/* 011: MLBCLK (not used) */
+		adg->rbga_rate_for_441khz_div_6,/* 100: RBGA */
+		adg->rbgb_rate_for_48khz_div_6,	/* 101: RBGB */
+	};
+
+	/* find div (= 1/128, 1/256, 1/512, 1/1024, 1/2048 */
+	for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) {
+		for (div  = 128,	idx = 0;
+		     div <= 2048;
+		     div *= 2,		idx++) {
+			if (src_rate == sel_rate[sel] / div) {
+				val = (idx << 4) | sel;
+				goto find_rate;
+			}
+		}
+	}
+	dev_err(dev, "can't find convert src clk\n");
+	return -EINVAL;
+
+find_rate:
+	shift	= (id % 4) * 8;
+	mask	= 0xFF << shift;
+	val	= val << shift;
+
+	dev_dbg(dev, "adg convert src clk = %02x\n", val);
+
+	switch (id / 4) {
+	case 0:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL3, mask, val);
+		break;
+	case 1:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL4, mask, val);
+		break;
+	case 2:
+		rsnd_mod_bset(mod, AUDIO_CLK_SEL5, mask, val);
+		break;
+	}
+
+	/*
+	 * Gen1 doesn't need dst_rate settings,
+	 * since it uses SSI WS pin.
+	 * see also rsnd_src_set_route_if_gen1()
+	 */
+
+	return 0;
+}
+
+int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
+			     struct rsnd_mod *mod,
+			     unsigned int src_rate,
+			     unsigned int dst_rate)
+{
+	if (rsnd_is_gen1(priv))
+		return rsnd_adg_set_convert_clk_gen1(priv, mod,
+						     src_rate, dst_rate);
+
+	return -EINVAL;
+}
+
 static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
 {
 	int id = rsnd_mod_id(mod);
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 862758d3ec06..add088bd4b2a 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -318,13 +318,23 @@ static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen)
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE0,	0xD0),
 		RSND_GEN1_S_REG(gen, SRU,	SSI_MODE1,	0xD4),
 		RSND_GEN1_M_REG(gen, SRU,	BUSIF_MODE,	0x20,	0x4),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_ROUTE_MODE0,0x50,	0x8),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SWRSR,	0x200,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SRCIR,	0x204,	0x40),
 		RSND_GEN1_M_REG(gen, SRU,	SRC_ADINR,	0x214,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_IFSCR,	0x21c,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_IFSVR,	0x220,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_SRCCR,	0x224,	0x40),
+		RSND_GEN1_M_REG(gen, SRU,	SRC_MNFSR,	0x228,	0x40),
 
 		RSND_GEN1_S_REG(gen, ADG,	BRRA,		0x00),
 		RSND_GEN1_S_REG(gen, ADG,	BRRB,		0x04),
 		RSND_GEN1_S_REG(gen, ADG,	SSICKR,		0x08),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL0,	0x0c),
 		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL1,	0x10),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL3,	0x18),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL4,	0x1c),
+		RSND_GEN1_S_REG(gen, ADG,	AUDIO_CLK_SEL5,	0x20),
 
 		RSND_GEN1_M_REG(gen, SSI,	SSICR,		0x00,	0x40),
 		RSND_GEN1_M_REG(gen, SSI,	SSISR,		0x04,	0x40),
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 3774dfcfaf0f..4ca66cd899c8 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -41,7 +41,14 @@ enum rsnd_reg {
 	RSND_REG_SSI_MODE1,
 	RSND_REG_BUSIF_MODE,
 	RSND_REG_INT_ENABLE,		/* for Gen2 */
+	RSND_REG_SRC_ROUTE_MODE0,
+	RSND_REG_SRC_SWRSR,
+	RSND_REG_SRC_SRCIR,
 	RSND_REG_SRC_ADINR,
+	RSND_REG_SRC_IFSCR,
+	RSND_REG_SRC_IFSVR,
+	RSND_REG_SRC_SRCCR,
+	RSND_REG_SRC_MNFSR,
 
 	/* ADG */
 	RSND_REG_BRRA,
@@ -50,6 +57,9 @@ enum rsnd_reg {
 	RSND_REG_AUDIO_CLK_SEL0,
 	RSND_REG_AUDIO_CLK_SEL1,
 	RSND_REG_AUDIO_CLK_SEL2,
+	RSND_REG_AUDIO_CLK_SEL3,	/* for Gen1 */
+	RSND_REG_AUDIO_CLK_SEL4,	/* for Gen1 */
+	RSND_REG_AUDIO_CLK_SEL5,	/* for Gen1 */
 
 	/* SSI */
 	RSND_REG_SSICR,
@@ -227,6 +237,10 @@ int rsnd_adg_probe(struct platform_device *pdev,
 		   struct rsnd_priv *priv);
 void rsnd_adg_remove(struct platform_device *pdev,
 		   struct rsnd_priv *priv);
+int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
+			     struct rsnd_mod *mod,
+			     unsigned int src_rate,
+			     unsigned int dst_rate);
 
 /*
  *	R-Car sound priv
@@ -280,6 +294,10 @@ void rsnd_scu_remove(struct platform_device *pdev,
 		     struct rsnd_priv *priv);
 struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
 bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod);
+unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
+				   struct rsnd_mod *ssi_mod,
+				   struct snd_pcm_runtime *runtime);
+
 #define rsnd_scu_nr(priv) ((priv)->scu_nr)
 
 /*
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
index 5f4f57206faf..1406dd8d9ed2 100644
--- a/sound/soc/sh/rcar/scu.c
+++ b/sound/soc/sh/rcar/scu.c
@@ -13,9 +13,13 @@
 struct rsnd_scu {
 	struct rsnd_scu_platform_info *info; /* rcar_snd.h */
 	struct rsnd_mod mod;
+	struct clk *clk;
 };
 
 #define rsnd_scu_mode_flags(p) ((p)->info->flags)
+#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate)
+
+#define RSND_SCU_NAME_SIZE 16
 
 /*
  * ADINR
@@ -26,6 +30,15 @@ struct rsnd_scu {
 #define OTBL_18		(6 << 16)
 #define OTBL_16		(8 << 16)
 
+/*
+ *		image of SRC (Sampling Rate Converter)
+ *
+ * 96kHz   <-> +-----+	48kHz	+-----+	 48kHz	+-------+
+ * 48kHz   <-> | SRC | <------>	| SSI |	<----->	| codec |
+ * 44.1kHz <-> +-----+		+-----+		+-------+
+ * ...
+ *
+ */
 
 #define rsnd_mod_to_scu(_mod)	\
 	container_of((_mod), struct rsnd_scu, mod)
@@ -56,7 +69,7 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 		{ 0x3, 28, }, /* 7 */
 		{ 0x3, 30, }, /* 8 */
 	};
-
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	u32 mask;
 	u32 val;
 	int shift;
@@ -86,9 +99,18 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	 */
 	shift	= (id % 4) * 8;
 	mask	= 0x1F << shift;
-	if (8 == id) /* SRU8 is very special */
+
+	/*
+	 * ADG is used as source clock if SRC was used,
+	 * then, SSI WS is used as destination clock.
+	 * SSI WS is used as source clock if SRC is not used
+	 * (when playback, source/destination become reverse when capture)
+	 */
+	if (rsnd_scu_convert_rate(scu))	/* use ADG */
+		val = 0;
+	else if (8 == id)		/* use SSI WS, but SRU8 is special */
 		val = id << shift;
-	else
+	else				/* use SSI WS */
 		val = (id + 1) << shift;
 
 	switch (id / 4) {
@@ -106,14 +128,45 @@ static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
 	return 0;
 }
 
-static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv,
+unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
+				   struct rsnd_mod *ssi_mod,
+				   struct snd_pcm_runtime *runtime)
+{
+	struct rsnd_scu *scu;
+	unsigned int rate;
+
+	/* this function is assuming SSI id = SCU id here */
+	scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod)));
+
+	/*
+	 * return convert rate if SRC is used,
+	 * otherwise, return runtime->rate as usual
+	 */
+	rate = rsnd_scu_convert_rate(scu);
+	if (!rate)
+		rate = runtime->rate;
+
+	return rate;
+}
+
+static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv,
 			      struct rsnd_mod *mod,
 			      struct rsnd_dai *rdai,
 			      struct rsnd_dai_stream *io)
 {
 	struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	u32 convert_rate = rsnd_scu_convert_rate(scu);
 	u32 adinr = runtime->channels;
 
+	/* set/clear soft reset */
+	rsnd_mod_write(mod, SRC_SWRSR, 0);
+	rsnd_mod_write(mod, SRC_SWRSR, 1);
+
+	/* Initialize the operation of the SRC internal circuits */
+	rsnd_mod_write(mod, SRC_SRCIR, 1);
+
+	/* Set channel number and output bit length */
 	switch (runtime->sample_bits) {
 	case 16:
 		adinr |= OTBL_16;
@@ -124,9 +177,42 @@ static int rsnd_scu_rate_ctrl(struct rsnd_priv *priv,
 	default:
 		return -EIO;
 	}
-
 	rsnd_mod_write(mod, SRC_ADINR, adinr);
 
+	if (convert_rate) {
+		u32 fsrate = 0x0400000 / convert_rate * runtime->rate;
+		int ret;
+
+		/* Enable the initial value of IFS */
+		rsnd_mod_write(mod, SRC_IFSCR, 1);
+
+		/* Set initial value of IFS */
+		rsnd_mod_write(mod, SRC_IFSVR, fsrate);
+
+		/* Select SRC mode (fixed value) */
+		rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
+
+		/* Set the restriction value of the FS ratio (98%) */
+		rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98);
+
+		if (rsnd_is_gen1(priv)) {
+			/* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
+		}
+
+		/* set convert clock */
+		ret = rsnd_adg_set_convert_clk(priv, mod,
+					       runtime->rate,
+					       convert_rate);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* Cancel the initialization and operate the SRC function */
+	rsnd_mod_write(mod, SRC_SRCIR, 0);
+
+	/* use DMA transfer */
+	rsnd_mod_write(mod, BUSIF_MODE, 1);
+
 	return 0;
 }
 
@@ -135,6 +221,7 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
 				   struct rsnd_dai *rdai,
 				   struct rsnd_dai_stream *io)
 {
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	int id = rsnd_mod_id(mod);
 	u32 val;
 
@@ -143,7 +230,28 @@ static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
 		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
 	}
 
-	rsnd_mod_write(mod, BUSIF_MODE, 1);
+	if (rsnd_scu_convert_rate(scu))
+		rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
+
+	return 0;
+}
+
+static int rsnd_scu_transfer_stop(struct rsnd_priv *priv,
+				  struct rsnd_mod *mod,
+				  struct rsnd_dai *rdai,
+				  struct rsnd_dai_stream *io)
+{
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+	int id = rsnd_mod_id(mod);
+	u32 mask;
+
+	if (rsnd_is_gen1(priv)) {
+		mask = (1 << id);
+		rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0);
+	}
+
+	if (rsnd_scu_convert_rate(scu))
+		rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
 
 	return 0;
 }
@@ -161,6 +269,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 			  struct rsnd_dai_stream *io)
 {
 	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	int ret;
 
@@ -175,13 +284,15 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 		return 0;
 	}
 
+	clk_enable(scu->clk);
+
 	/* it use DMA transter */
 
 	ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
-	ret = rsnd_scu_rate_ctrl(priv, mod, rdai, io);
+	ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io);
 	if (ret < 0)
 		return ret;
 
@@ -194,9 +305,27 @@ static int rsnd_scu_start(struct rsnd_mod *mod,
 	return 0;
 }
 
+static int rsnd_scu_stop(struct rsnd_mod *mod,
+			  struct rsnd_dai *rdai,
+			  struct rsnd_dai_stream *io)
+{
+	struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+	struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
+
+	if (!rsnd_scu_hpbif_is_enable(mod))
+		return 0;
+
+	rsnd_scu_transfer_stop(priv, mod, rdai, io);
+
+	clk_disable(scu->clk);
+
+	return 0;
+}
+
 static struct rsnd_mod_ops rsnd_scu_ops = {
 	.name	= "scu",
 	.start	= rsnd_scu_start,
+	.stop	= rsnd_scu_stop,
 };
 
 struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
@@ -212,6 +341,8 @@ int rsnd_scu_probe(struct platform_device *pdev,
 {
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct rsnd_scu *scu;
+	struct clk *clk;
+	char name[RSND_SCU_NAME_SIZE];
 	int i, nr;
 
 	/*
@@ -228,9 +359,16 @@ int rsnd_scu_probe(struct platform_device *pdev,
 	priv->scu	= scu;
 
 	for_each_rsnd_scu(scu, priv, i) {
+		snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i);
+
+		clk = devm_clk_get(dev, name);
+		if (IS_ERR(clk))
+			return PTR_ERR(clk);
+
 		rsnd_mod_init(priv, &scu->mod,
 			      &rsnd_scu_ops, i);
 		scu->info = &info->scu_info[i];
+		scu->clk = clk;
 
 		dev_dbg(dev, "SCU%d probed\n", i);
 	}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 2db9711549f5..b7cd06be9436 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -200,7 +200,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi,
 		1, 2, 4, 8, 16, 6, 12,
 	};
 	unsigned int main_rate;
-	unsigned int rate = runtime->rate;
+	unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime);
 
 	/*
 	 * Find best clock, and try to start ADG
-- 
cgit v1.2.2


From e41975edc73d2c16d0784e5fa87a6162e2fcab80 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Fri, 20 Dec 2013 14:39:51 +0800
Subject: ASoC: core: Fix the DAI name getting.

From "ASoC: make snd_soc_dai_link more symmetrical", can we see that
the name of CPU DAI maybe omitted. If the DAI name is omitted, try to
use the component name instead.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-core.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..03c779ebd729 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -4617,10 +4617,14 @@ int snd_soc_of_get_dai_name(struct device_node *of_node,
 
 			if (id < 0 || id >= pos->num_dai) {
 				ret = -EINVAL;
-			} else {
-				*dai_name = pos->dai_drv[id].name;
-				ret = 0;
+				break;
 			}
+
+			ret = 0;
+
+			*dai_name = pos->dai_drv[id].name;
+			if (!*dai_name)
+				*dai_name = pos->name;
 		}
 
 		break;
-- 
cgit v1.2.2


From 14c3aa98397b3b4e67b042f388f4ad0f9b18250e Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Mon, 23 Dec 2013 17:28:53 +0800
Subject: ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card

This is a quick fix for the below two issues when building spdif as modules.

1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2)
snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create
imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe
it again due to the execution platform_driver_unregister() in probe() when
meeting -EPROBE_DEFER at Step 2.

2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok",
'rmmod snd-soc-imx-spdif' would cause kernel dump with warning:
WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90()
sysfs: can not remove 'dapm_widget', no directory
This should be caused by disordered resourse releasing of the whole link.
And trying to unregister the card and then CODEC dev can't fix this issue.

Thus this patch just provides a simple fix to these two bugs by using the
snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that
there's no need to handle the registering and unregistering of CODEC or
CODEC dai any more.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig     |  1 -
 sound/soc/fsl/imx-spdif.c | 84 ++++++++++++-----------------------------------
 2 files changed, 21 insertions(+), 64 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index b7ab71f2ccc1..8c10b8f80ace 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -197,7 +197,6 @@ config SND_SOC_IMX_SPDIF
 	tristate "SoC Audio support for i.MX boards with S/PDIF"
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_FSL_SPDIF
-	select SND_SOC_SPDIF
 	select REGMAP_MMIO
 	help
 	  SoC Audio support for i.MX boards with S/PDIF
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
index 8499d5292f08..e656245d24c9 100644
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@@ -14,17 +14,15 @@
 #include <sound/soc.h>
 
 struct imx_spdif_data {
-	struct snd_soc_dai_link dai[2];
+	struct snd_soc_dai_link dai;
 	struct snd_soc_card card;
-	struct platform_device *txdev;
-	struct platform_device *rxdev;
 };
 
 static int imx_spdif_audio_probe(struct platform_device *pdev)
 {
 	struct device_node *spdif_np, *np = pdev->dev.of_node;
 	struct imx_spdif_data *data;
-	int ret = 0, num_links = 0;
+	int ret = 0;
 
 	spdif_np = of_parse_phandle(np, "spdif-controller", 0);
 	if (!spdif_np) {
@@ -40,69 +38,42 @@ static int imx_spdif_audio_probe(struct platform_device *pdev)
 		goto end;
 	}
 
-	if (of_property_read_bool(np, "spdif-out")) {
-		data->dai[num_links].name = "S/PDIF TX";
-		data->dai[num_links].stream_name = "S/PDIF PCM Playback";
-		data->dai[num_links].codec_dai_name = "dit-hifi";
-		data->dai[num_links].codec_name = "spdif-dit";
-		data->dai[num_links].cpu_of_node = spdif_np;
-		data->dai[num_links].platform_of_node = spdif_np;
-		num_links++;
-
-		data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
-		if (IS_ERR(data->txdev)) {
-			ret = PTR_ERR(data->txdev);
-			dev_err(&pdev->dev, "register dit failed: %d\n", ret);
-			goto end;
-		}
-	}
+	data->dai.name = "S/PDIF PCM";
+	data->dai.stream_name = "S/PDIF PCM";
+	data->dai.codec_dai_name = "snd-soc-dummy-dai";
+	data->dai.codec_name = "snd-soc-dummy";
+	data->dai.cpu_of_node = spdif_np;
+	data->dai.platform_of_node = spdif_np;
+	data->dai.playback_only = true;
+	data->dai.capture_only = true;
 
-	if (of_property_read_bool(np, "spdif-in")) {
-		data->dai[num_links].name = "S/PDIF RX";
-		data->dai[num_links].stream_name = "S/PDIF PCM Capture";
-		data->dai[num_links].codec_dai_name = "dir-hifi";
-		data->dai[num_links].codec_name = "spdif-dir";
-		data->dai[num_links].cpu_of_node = spdif_np;
-		data->dai[num_links].platform_of_node = spdif_np;
-		num_links++;
-
-		data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
-		if (IS_ERR(data->rxdev)) {
-			ret = PTR_ERR(data->rxdev);
-			dev_err(&pdev->dev, "register dir failed: %d\n", ret);
-			goto error_dit;
-		}
-	}
+	if (of_property_read_bool(np, "spdif-out"))
+		data->dai.capture_only = false;
+
+	if (of_property_read_bool(np, "spdif-in"))
+		data->dai.playback_only = false;
 
-	if (!num_links) {
+	if (data->dai.playback_only && data->dai.capture_only) {
 		dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
-		goto error_dir;
+		goto end;
 	}
 
 	data->card.dev = &pdev->dev;
-	data->card.num_links = num_links;
-	data->card.dai_link = data->dai;
+	data->card.dai_link = &data->dai;
+	data->card.num_links = 1;
 
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)
-		goto error_dir;
+		goto end;
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
-		goto error_dir;
+		goto end;
 	}
 
 	platform_set_drvdata(pdev, data);
 
-	goto end;
-
-error_dir:
-	if (data->rxdev)
-		platform_device_unregister(data->rxdev);
-error_dit:
-	if (data->txdev)
-		platform_device_unregister(data->txdev);
 end:
 	if (spdif_np)
 		of_node_put(spdif_np);
@@ -110,18 +81,6 @@ end:
 	return ret;
 }
 
-static int imx_spdif_audio_remove(struct platform_device *pdev)
-{
-	struct imx_spdif_data *data = platform_get_drvdata(pdev);
-
-	if (data->rxdev)
-		platform_device_unregister(data->rxdev);
-	if (data->txdev)
-		platform_device_unregister(data->txdev);
-
-	return 0;
-}
-
 static const struct of_device_id imx_spdif_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-spdif", },
 	{ /* sentinel */ }
@@ -135,7 +94,6 @@ static struct platform_driver imx_spdif_driver = {
 		.of_match_table = imx_spdif_dt_ids,
 	},
 	.probe = imx_spdif_audio_probe,
-	.remove = imx_spdif_audio_remove,
 };
 
 module_platform_driver(imx_spdif_driver);
-- 
cgit v1.2.2


From 71467e46414d3bab220de77d3d085be0c0aa03e1 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 23 Dec 2013 15:25:38 +0800
Subject: ASoC: simple-card: Add device's module clock selection.

Try to get the device's module clock if the dt has no clocks and
system-clock-frequency properties.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 21 ++++++++++++++++++---
 1 file changed, 18 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 3ba65bb63439..58c217e403ae 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 	 * dai->sysclk come from
 	 *  "clocks = <&xxx>" (if system has common clock)
 	 *  or "system-clock-frequency = <xxx>"
+	 *  or device's module clock.
 	 */
-	clk = of_clk_get(np, 0);
-	if (IS_ERR(clk))
+	if (of_property_read_bool(np, "clocks")) {
+		clk = of_clk_get(np, 0);
+		if (IS_ERR(clk)) {
+			ret = PTR_ERR(clk);
+			goto parse_error;
+		}
+
+		dai->sysclk = clk_get_rate(clk);
+	} else if (of_property_read_bool(np, "system-clock-frequency")) {
 		of_property_read_u32(np,
 				     "system-clock-frequency",
 				     &dai->sysclk);
-	else
+	} else {
+		clk = of_clk_get(*node, 0);
+		if (IS_ERR(clk)) {
+			ret = PTR_ERR(clk);
+			goto parse_error;
+		}
+
 		dai->sysclk = clk_get_rate(clk);
+	}
 
 	ret = 0;
 
-- 
cgit v1.2.2


From f60e5473e6788f93849a61198bec4e02fea31e51 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 23 Dec 2013 11:42:55 +0100
Subject: ASoC: ssm2518: Fix off-by-one error by ffs()

ffs() returns the bit position from 1, while the ssm2158 driver code
assumes it being 0-based.  Also, the bit mask computation of the two
channel slots are incorrect; it must have worked just casually.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ssm2518.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 95aed552139a..cc8debce752f 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -549,13 +549,13 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		right_slot = 0;
 	} else {
 		/* We assume the left channel < right channel */
-		left_slot = ffs(tx_mask);
-		tx_mask &= ~(1 << tx_mask);
+		left_slot = __ffs(tx_mask);
+		tx_mask &= ~(1 << left_slot);
 		if (tx_mask == 0) {
 			right_slot = left_slot;
 		} else {
-			right_slot = ffs(tx_mask);
-			tx_mask &= ~(1 << tx_mask);
+			right_slot = __ffs(tx_mask);
+			tx_mask &= ~(1 << right_slot);
 		}
 	}
 
-- 
cgit v1.2.2


From c097d5fdf3b51cdb2521c3cffab0a8cf03c68cc6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Mon, 23 Dec 2013 12:41:39 +0000
Subject: ASoC: ad1836: Reject unsupported bit sizes

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ad1836.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 9a92b7962f41..d7c983862cf0 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -179,6 +179,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S32_LE:
 		word_len = AD1836_WORD_LEN_24;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1,
-- 
cgit v1.2.2


From b2c53e206967d01fd4fb6dd525f89ae738beb2e6 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 1 Jan 2014 14:01:34 +0100
Subject: ALSA: hda - Disable Front HP jack detection on Gigabyte Z87X-UD3H

This motherboard seems to have a flaky jack detection - when the
front HP is not present, the jack state quickly switches on and off.

This has been reported by three people in the bug, so I doubt it's
a user error this time.

BugLink: https://bugs.launchpad.net/bugs/1248116
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 16c0caa7f014..6a705a81de35 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1810,6 +1810,7 @@ enum {
 	ALC882_FIXUP_ACER_ASPIRE_7736,
 	ALC882_FIXUP_ASUS_W90V,
 	ALC889_FIXUP_CD,
+	ALC889_FIXUP_FRONT_HP_NO_PRESENCE,
 	ALC889_FIXUP_VAIO_TT,
 	ALC888_FIXUP_EEE1601,
 	ALC882_FIXUP_EAPD,
@@ -2027,6 +2028,15 @@ static const struct hda_fixup alc882_fixups[] = {
 			{ }
 		}
 	},
+	[ALC889_FIXUP_FRONT_HP_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x02214120 }, /* Front HP jack is flaky, disable jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC889_FIXUP_CD,
+	},
 	[ALC889_FIXUP_VAIO_TT] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -2258,7 +2268,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
+	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
-- 
cgit v1.2.2


From ec06b2bea73ed0db1e400ff5143c14e31836a77c Mon Sep 17 00:00:00 2001
From: Julia Lawall <Julia.Lawall@lip6.fr>
Date: Fri, 3 Jan 2014 00:40:30 +0100
Subject: ALSA: cs5535audio: use named constants for pci_power_t values

The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression pdev;
@@

pci_set_power_state(pdev,
- 3
+ PCI_D3hot
 )
// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/cs5535audio/cs5535audio.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 902bebd3b3fb..c0d2835344da 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -253,7 +253,7 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
 static int snd_cs5535audio_free(struct cs5535audio *cs5535au)
 {
 	synchronize_irq(cs5535au->irq);
-	pci_set_power_state(cs5535au->pci, 3);
+	pci_set_power_state(cs5535au->pci, PCI_D3hot);
 
 	if (cs5535au->irq >= 0)
 		free_irq(cs5535au->irq, cs5535au);
-- 
cgit v1.2.2


From 770bd4bf2e664939a9dacd3d26ec9ff7a3933210 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 27 Dec 2013 00:13:08 +0200
Subject: ALSA: rme9652: fix a missing comma in channel_map_9636_ds[]

The lack of comma leads to the wrong channel for an SPDIF channel.
Unfortunately this wasn't caught by compiler because it's still a
valid expression.

Reported-by: Alexander Aristov <aristov.alexander@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/rme9652.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index b96d9e1adf6d..3717f2dd30be 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -285,7 +285,7 @@ static char channel_map_9636_ds[26] = {
 	/* ADAT channels are remapped */
 	1, 3, 5, 7, 9, 11, 13, 15,
 	/* channels 8 and 9 are S/PDIF */
-	24, 25
+	24, 25,
 	/* others don't exist */
 	-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1
 };
-- 
cgit v1.2.2


From 2aac06f787940543fb37bfdb982eb99431bc6094 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Sat, 4 Jan 2014 16:59:11 +0530
Subject: ALSA: compress: remove the sample rate check

commit f0e9c080 - "ALSA: compress: change the way sample rates are sent to
kernel" changed the way sample rates are sent. So now we don't need to check for
PCM_RATE_xxx in kernel

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 9d518ac73eea..7a20897d33db 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -501,9 +501,6 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 	if (params->codec.ch_in == 0 || params->codec.ch_out == 0)
 		return -EINVAL;
 
-	if (!(params->codec.sample_rate & SNDRV_PCM_RATE_8000_192000))
-		return -EINVAL;
-
 	return 0;
 }
 
-- 
cgit v1.2.2


From 6873ee464a9fd23f0b7c2ab38e4ab8cea02cb50d Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Sun, 5 Jan 2014 10:21:16 +0400
Subject: ASoC: fsl_ssi: Fix printing return code on clk error

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index b2ebaf811599..6e3d38a85280 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1176,7 +1176,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		 */
 		ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud");
 		if (IS_ERR(ssi_private->baudclk))
-			dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret);
+			dev_warn(&pdev->dev, "could not get baud clock: %d\n",
+				 PTR_ERR(ssi_private->baudclk));
 		else
 			clk_prepare_enable(ssi_private->baudclk);
 
-- 
cgit v1.2.2


From e2a19ac6c5b27ac93fe744c0ff0823cde52c9cbb Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 6 Jan 2014 12:34:36 +0800
Subject: ASoC: simple-card: Fix the sysclk selection.

For spdif there is no need to do the sysclk setting.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 8 ++------
 1 file changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 58c217e403ae..d4402fb57253 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -106,12 +106,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 				     &dai->sysclk);
 	} else {
 		clk = of_clk_get(*node, 0);
-		if (IS_ERR(clk)) {
-			ret = PTR_ERR(clk);
-			goto parse_error;
-		}
-
-		dai->sysclk = clk_get_rate(clk);
+		if (!IS_ERR(clk))
+			dai->sysclk = clk_get_rate(clk);
 	}
 
 	ret = 0;
-- 
cgit v1.2.2


From a5d3f6abbf0f8be882d752da33b3e204c2d76f59 Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Sun, 5 Jan 2014 11:38:31 +0400
Subject: ASoC: mc13783: Use module_platform_driver_probe()

mc13783-codec is probed only by MC13XXX MFD core driver so
use module_platform_driver_probe().

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/mc13783.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index bae60164c7b7..8ab966860224 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -750,7 +750,7 @@ static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
 	.num_dapm_routes = ARRAY_SIZE(mc13783_routes),
 };
 
-static int mc13783_codec_probe(struct platform_device *pdev)
+static int __init mc13783_codec_probe(struct platform_device *pdev)
 {
 	struct mc13xxx *mc13xxx;
 	struct mc13783_priv *priv;
@@ -804,11 +804,9 @@ static struct platform_driver mc13783_codec_driver = {
 		   .name = "mc13783-codec",
 		   .owner = THIS_MODULE,
 		   },
-	.probe = mc13783_codec_probe,
 	.remove = mc13783_codec_remove,
 };
-
-module_platform_driver(mc13783_codec_driver);
+module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe);
 
 MODULE_DESCRIPTION("ASoC MC13783 driver");
 MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>");
-- 
cgit v1.2.2


From 0acb26a6c716ef2f8ab550475c5da4d187995cca Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Sun, 5 Jan 2014 11:38:32 +0400
Subject: ASoC: mc13783: Use core error messages if registration fails

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/mc13783.c | 8 --------
 1 file changed, 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 8ab966860224..c2def5d188ee 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -781,14 +781,6 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
 			mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
 
-	if (ret)
-		goto err_register_codec;
-
-	return 0;
-
-err_register_codec:
-	dev_err(&pdev->dev, "register codec failed with %d\n", ret);
-
 	return ret;
 }
 
-- 
cgit v1.2.2


From 295b84237b4ec2e1f148c8f6d7f59a7d06fda624 Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Sun, 5 Jan 2014 11:38:33 +0400
Subject: ASoC: mc13783: Drop fixed ADC & DAC ports usage

There are no users of this driver without pdata, so stop using
constant assignment of ADC and DAC ports.

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/mc13783.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index c2def5d188ee..997f708afc79 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -770,8 +770,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 		priv->adc_ssi_port = pdata->adc_ssi_port;
 		priv->dac_ssi_port = pdata->dac_ssi_port;
 	} else {
-		priv->adc_ssi_port = MC13783_SSI1_PORT;
-		priv->dac_ssi_port = MC13783_SSI2_PORT;
+		return -ENOSYS;
 	}
 
 	if (priv->adc_ssi_port == priv->dac_ssi_port)
-- 
cgit v1.2.2


From 2b32098f74ad6e8e3c0dbc714aa0f14c2f7df20a Mon Sep 17 00:00:00 2001
From: Alexander Shiyan <shc_work@mail.ru>
Date: Sun, 5 Jan 2014 11:38:34 +0400
Subject: ASoC: mc13783: trivial: Cleanup module

This is a trivial cleanup: remove useless variable mc13xxx and
extra spaces. No functional changes.

Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/mc13783.c | 17 +++++++----------
 1 file changed, 7 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 997f708afc79..582c2bbd42cb 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -752,20 +752,14 @@ static struct snd_soc_codec_driver soc_codec_dev_mc13783 = {
 
 static int __init mc13783_codec_probe(struct platform_device *pdev)
 {
-	struct mc13xxx *mc13xxx;
 	struct mc13783_priv *priv;
 	struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data;
 	int ret;
 
-	mc13xxx = dev_get_drvdata(pdev->dev.parent);
-
-
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
-	if (priv == NULL)
+	if (!priv)
 		return -ENOMEM;
 
-	dev_set_drvdata(&pdev->dev, priv);
-	priv->mc13xxx = mc13xxx;
 	if (pdata) {
 		priv->adc_ssi_port = pdata->adc_ssi_port;
 		priv->dac_ssi_port = pdata->dac_ssi_port;
@@ -773,6 +767,9 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
 		return -ENOSYS;
 	}
 
+	dev_set_drvdata(&pdev->dev, priv);
+	priv->mc13xxx = dev_get_drvdata(pdev->dev.parent);
+
 	if (priv->adc_ssi_port == priv->dac_ssi_port)
 		ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
 			mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync));
@@ -792,9 +789,9 @@ static int mc13783_codec_remove(struct platform_device *pdev)
 
 static struct platform_driver mc13783_codec_driver = {
 	.driver = {
-		   .name = "mc13783-codec",
-		   .owner = THIS_MODULE,
-		   },
+		.name	= "mc13783-codec",
+		.owner	= THIS_MODULE,
+	},
 	.remove = mc13783_codec_remove,
 };
 module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe);
-- 
cgit v1.2.2


From 6ed54f08bab0a93d53fddcd37b69d6b15fbef500 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 20 Dec 2013 14:20:07 +0100
Subject: ASoC: atmel: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/atmel/atmel-pcm-dma.c | 1 -
 sound/soc/atmel/atmel-pcm-pdc.c | 1 -
 2 files changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 06082e5e5dcb..b79a2a864513 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -50,7 +50,6 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = {
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_PAUSE,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 256,		/* lighting DMA overhead */
 	.period_bytes_max	= 2 * 0xffff,	/* if 2 bytes format */
 	.periods_min		= 8,
diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c
index 054ea4d9326a..33ec592ecd75 100644
--- a/sound/soc/atmel/atmel-pcm-pdc.c
+++ b/sound/soc/atmel/atmel-pcm-pdc.c
@@ -58,7 +58,6 @@ static const struct snd_pcm_hardware atmel_pcm_hardware = {
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_PAUSE,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
 	.periods_min		= 2,
-- 
cgit v1.2.2


From 08ae9b456d393dfd1bbe7619b994189be6a26449 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:06 +0100
Subject: ASoC: dpcm: Add helper function for initializing runtime pcm

We have the same code for initializing the runtime pcm on both the playback and
the capture path. Factor this out into a common helper function.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 30 +++++++++++++++---------------
 1 file changed, 15 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 10f29a0ad5a6..b649e32791df 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1228,6 +1228,17 @@ unwind:
 	return err;
 }
 
+static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
+	struct snd_soc_pcm_stream *stream)
+{
+	runtime->hw.rate_min = stream->rate_min;
+	runtime->hw.rate_max = stream->rate_max;
+	runtime->hw.channels_min = stream->channels_min;
+	runtime->hw.channels_max = stream->channels_max;
+	runtime->hw.formats &= stream->formats;
+	runtime->hw.rates = stream->rates;
+}
+
 static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -1235,21 +1246,10 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		runtime->hw.rate_min = cpu_dai_drv->playback.rate_min;
-		runtime->hw.rate_max = cpu_dai_drv->playback.rate_max;
-		runtime->hw.channels_min = cpu_dai_drv->playback.channels_min;
-		runtime->hw.channels_max = cpu_dai_drv->playback.channels_max;
-		runtime->hw.formats &= cpu_dai_drv->playback.formats;
-		runtime->hw.rates = cpu_dai_drv->playback.rates;
-	} else {
-		runtime->hw.rate_min = cpu_dai_drv->capture.rate_min;
-		runtime->hw.rate_max = cpu_dai_drv->capture.rate_max;
-		runtime->hw.channels_min = cpu_dai_drv->capture.channels_min;
-		runtime->hw.channels_max = cpu_dai_drv->capture.channels_max;
-		runtime->hw.formats &= cpu_dai_drv->capture.formats;
-		runtime->hw.rates = cpu_dai_drv->capture.rates;
-	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
+	else
+		dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
 }
 
 static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
-- 
cgit v1.2.2


From 002220a90db8ab9a6313887934dec25b54404cbd Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:07 +0100
Subject: ASoC: dpcm: Allow PCMs to omit the set of supported formats

Allow PCMs that do not impose any restrictions on the supported formats to set
the formats field to 0, Instead of assuming that this means that the PCM does
not support any formats (which doesn't make much sense), assume that it supports
all formats. This brings the behavior of DPCM closer to that of non-DPCM.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b649e32791df..feb0f2843026 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1235,7 +1235,10 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 	runtime->hw.rate_max = stream->rate_max;
 	runtime->hw.channels_min = stream->channels_min;
 	runtime->hw.channels_max = stream->channels_max;
-	runtime->hw.formats &= stream->formats;
+	if (runtime->hw.formats)
+		runtime->hw.formats &= stream->formats;
+	else
+		runtime->hw.formats = stream->formats;
 	runtime->hw.rates = stream->rates;
 }
 
-- 
cgit v1.2.2


From ff1b15acb44398f1a23e804fc0e178c952ee7fde Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Tue, 7 Jan 2014 08:00:13 -0200
Subject: ASoC: fsl: fsl_ssi: Use '%ld' to print 'long int'

Commit 6873ee464a (ASoC: fsl_ssi: Fix printing return code on clk error) caused
the following build warning:

sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe':
sound/soc/fsl/fsl_ssi.c:1196:6: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat]

Fix it by using '%ld' to print the 'long int' format.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6e3d38a85280..816ae4b28a53 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1176,7 +1176,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		 */
 		ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud");
 		if (IS_ERR(ssi_private->baudclk))
-			dev_warn(&pdev->dev, "could not get baud clock: %d\n",
+			dev_warn(&pdev->dev, "could not get baud clock: %ld\n",
 				 PTR_ERR(ssi_private->baudclk));
 		else
 			clk_prepare_enable(ssi_private->baudclk);
-- 
cgit v1.2.2


From b0a23b8b36e1fb754dcbdfe622e5ca5ded2f188b Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:09 +0100
Subject: ASoC: fsl: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_dma.c     | 7 -------
 sound/soc/fsl/imx-pcm-dma.c | 3 ---
 sound/soc/fsl/imx-pcm-fiq.c | 3 ---
 sound/soc/fsl/mpc5200_dma.c | 4 ----
 4 files changed, 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d570f8c81dc6..6bb0ea59284f 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -55,10 +55,6 @@
 			    SNDRV_PCM_FMTBIT_S32_BE     | \
 			    SNDRV_PCM_FMTBIT_U32_LE     | \
 			    SNDRV_PCM_FMTBIT_U32_BE)
-
-#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
-			  SNDRV_PCM_RATE_CONTINUOUS)
-
 struct dma_object {
 	struct snd_soc_platform_driver dai;
 	dma_addr_t ssi_stx_phys;
@@ -140,9 +136,6 @@ static const struct snd_pcm_hardware fsl_dma_hardware = {
 				  SNDRV_PCM_INFO_JOINT_DUPLEX |
 				  SNDRV_PCM_INFO_PAUSE,
 	.formats		= FSLDMA_PCM_FORMATS,
-	.rates  		= FSLDMA_PCM_RATES,
-	.rate_min       	= 5512,
-	.rate_max       	= 192000,
 	.period_bytes_min       = 512,  	/* A reasonable limit */
 	.period_bytes_max       = (u32) -1,
 	.periods_min    	= NUM_DMA_LINKS,
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index c5e47f866b4b..2585ae44e634 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -41,9 +41,6 @@ static const struct snd_pcm_hardware imx_pcm_hardware = {
 		SNDRV_PCM_INFO_PAUSE |
 		SNDRV_PCM_INFO_RESUME,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	.rate_min = 8000,
-	.channels_min = 2,
-	.channels_max = 2,
 	.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
 	.period_bytes_min = 128,
 	.period_bytes_max = 65535, /* Limited by SDMA engine */
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index c75d43bb2e92..6553202dd48c 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -162,9 +162,6 @@ static struct snd_pcm_hardware snd_imx_hardware = {
 		SNDRV_PCM_INFO_PAUSE |
 		SNDRV_PCM_INFO_RESUME,
 	.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	.rate_min = 8000,
-	.channels_min = 2,
-	.channels_max = 2,
 	.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
 	.period_bytes_min = 128,
 	.period_bytes_max = 16 * 1024,
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 71bf2f248cd4..f2b5d756b1f3 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -200,10 +200,6 @@ static const struct snd_pcm_hardware psc_dma_hardware = {
 		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
 		SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
-	.rate_min = 8000,
-	.rate_max = 48000,
-	.channels_min = 1,
-	.channels_max = 2,
 	.period_bytes_max	= 1024 * 1024,
 	.period_bytes_min	= 32,
 	.periods_min		= 2,
-- 
cgit v1.2.2


From f3b6079683e371eff8772882448020c29913cab1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:12 +0100
Subject: ASoC: mxs: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/mxs/mxs-pcm.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 04a6b0d60944..2e0863a70da3 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -36,11 +36,6 @@ static const struct snd_pcm_hardware snd_mxs_hardware = {
 				  SNDRV_PCM_INFO_RESUME |
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_HALF_DUPLEX,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
-				  SNDRV_PCM_FMTBIT_S20_3LE |
-				  SNDRV_PCM_FMTBIT_S24_LE,
-	.channels_min		= 2,
-	.channels_max		= 2,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
 	.periods_min		= 1,
-- 
cgit v1.2.2


From 96ae0f08ac574f3dac17cff9afdeee5562f61cbb Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 17:19:22 +0100
Subject: ASoC: mxs: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag

Since commit 7b11304 ("dma: mxs-dma: Report correct residue for cyclic DMA")
the mxs dmaengine driver has support for residue reporting. So there is no need
to specify the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag anymore. This allows a
finer grained resolution for the PCM pointer as well as avoids the race
condition that can occur with the period counting that is used when the
dmaengine driver does not support residue reporting.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/mxs/mxs-pcm.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 2e0863a70da3..a371b4f91c53 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -52,7 +52,6 @@ static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = {
 int mxs_pcm_platform_register(struct device *dev)
 {
 	return devm_snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config,
-		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX);
 }
 EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
-- 
cgit v1.2.2


From 0475680b5c2ef4bbdc3af1f6cfd014ea08c8d981 Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:54:58 +0000
Subject: ARM: ux500: Don't use enums for MSP IDs - for easy DT conversion

Signed-off-by: Lee Jones <lee.jones@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_i2s.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
index 258d0bcee0bd..875de0f68b85 100644
--- a/sound/soc/ux500/ux500_msp_i2s.h
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -475,7 +475,7 @@ struct ux500_msp_dma_params {
 };
 
 struct ux500_msp {
-	enum msp_i2s_id id;
+	int id;
 	void __iomem *registers;
 	struct device *dev;
 	struct ux500_msp_dma_params playback_dma_data;
-- 
cgit v1.2.2


From a61f9e314ad8ab9434ddd989b857ed93fdc725e2 Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:00 +0000
Subject: ASoC: ux500: Provide better checking for Device Tree and/or Platform
 Data

These drivers will not work without platform specific data, which is
passed in via Device Tree or Platform Data. To avoid the chance of
NULL pointer dereferencing and alike, let's ensure we have at least
one of the methods in play before continuing.

Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_dai.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index c6fb5cce980e..bc042cce115f 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -771,10 +771,14 @@ static const struct snd_soc_component_driver ux500_msp_component = {
 static int ux500_msp_drv_probe(struct platform_device *pdev)
 {
 	struct ux500_msp_i2s_drvdata *drvdata;
+	struct msp_i2s_platform_data *pdata = pdev->dev.platform_data;
+	struct device_node *np = pdev->dev.of_node;
 	int ret = 0;
 
-	dev_dbg(&pdev->dev, "%s: Enter (pdev->name = %s).\n", __func__,
-		pdev->name);
+	if (!pdata && !np) {
+		dev_err(&pdev->dev, "No platform data or Device Tree found\n");
+		return -ENODEV;
+	}
 
 	drvdata = devm_kzalloc(&pdev->dev,
 				sizeof(struct ux500_msp_i2s_drvdata),
-- 
cgit v1.2.2


From ae276e93b8ccb933c8cfca368427d1eafd07128d Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:01 +0000
Subject: ASoC: Ux500: Match platform by device node when booting Device Tree

We're getting closer to fully enabling the Ux500 ASoC driver for Device
Tree. When we switch over from using AUXDATA we'll need to match platform
by only Device Tree nodes. In this patch we NULL out the platform_name,
and supply nodes for each platform device.

Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/mop500.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 178d1bad6259..b3b66aa98dce 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -91,6 +91,8 @@ static int mop500_of_probe(struct platform_device *pdev,
 	for (i = 0; i < 2; i++) {
 		mop500_dai_links[i].cpu_of_node = msp_np[i];
 		mop500_dai_links[i].cpu_dai_name = NULL;
+		mop500_dai_links[i].platform_of_node = msp_np[i];
+		mop500_dai_links[i].platform_name = NULL;
 		mop500_dai_links[i].codec_of_node = codec_np;
 		mop500_dai_links[i].codec_name = NULL;
 	}
-- 
cgit v1.2.2


From 609a3050b8a516d12cf6dc0e8beb5875ededad3d Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:02 +0000
Subject: ASoC: ux500_pcm: Stop pretending that we support varying address
 widths

The Slave Config's addr_width attribute is populated by data_width of
dma_cfg, which in turn is derived from dma_params' data_size attribute
and that comes from the slot_width which is always 16 bits (2 Bytes).
We're cutting out the middle man here and just setting the DMA Slave
Config directly.

Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_pcm.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index ce554de5d9dc..32d457232110 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -109,20 +109,19 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ux500_msp_dma_params *dma_params;
-	struct stedma40_chan_cfg *dma_cfg;
 	int ret;
 
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-	dma_cfg = dma_params->dma_cfg;
 
 	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
 	if (ret)
 		return ret;
 
 	slave_config->dst_maxburst = 4;
-	slave_config->dst_addr_width = dma_cfg->dst_info.data_width;
 	slave_config->src_maxburst = 4;
-	slave_config->src_addr_width = dma_cfg->src_info.data_width;
+
+	slave_config->src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		slave_config->dst_addr = dma_params->tx_rx_addr;
-- 
cgit v1.2.2


From f6c377520c26297cc870173df3cd0acdef08bc1c Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:03 +0000
Subject: ASoC: ux500_pcm: Expect different saved DMA data when obtaining from
 DAI store

In preparation for full Device Tree enablement we must differentiate
between the two varying ways DMA data can be held in the DAI store. If
we're booting with Device Tree the provided 'snd_dmaengine_dai_dma_data'
data structure shall be used, whereas in order to avoid breaking legacy
platform data we also need to be able to translate DMA data stored using
the UX500 specific 'ux500_msp_dma_params' method.

Once we move over to solely use Device Tree, we can enforce the use of
'snd_dmaengine_dai_dma_data' and this code can be removed altogether.

Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_pcm.c | 19 +++++++++++++++----
 1 file changed, 15 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 32d457232110..8b53f22edcaf 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -108,10 +108,21 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 		struct dma_slave_config *slave_config)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct ux500_msp_dma_params *dma_params;
+	struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data;
+	struct snd_dmaengine_dai_dma_data *snd_dma_params;
+	struct ux500_msp_dma_params *ste_dma_params;
+	dma_addr_t dma_addr;
 	int ret;
 
-	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (pdata) {
+		ste_dma_params =
+			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+		dma_addr = ste_dma_params->tx_rx_addr;
+	} else {
+		snd_dma_params =
+			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+		dma_addr = snd_dma_params->addr;
+	}
 
 	ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
 	if (ret)
@@ -124,9 +135,9 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 	slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		slave_config->dst_addr = dma_params->tx_rx_addr;
+		slave_config->dst_addr = dma_addr;
 	else
-		slave_config->src_addr = dma_params->tx_rx_addr;
+		slave_config->src_addr = dma_addr;
 
 	return 0;
 }
-- 
cgit v1.2.2


From 05c56c24137273de3460872b6121a2bd762d11e8 Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:04 +0000
Subject: ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data

Soon we will strip out pdata support from the Ux500 set of ASoC drivers.
When this happens it will have to supply a DMA slave_config to the
dmaengine. At the moment a great deal of this comes from pdata via
AUXDATA. We need to become independent of this soon. This patch starts
the process by allocating memory for the associated data structures and
fetches the MSP id used for const struct indexing.

Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_i2s.c | 56 ++++++++++++++++++++++++++++++-----------
 1 file changed, 41 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 1ca8b08ae993..7f2a4acddcd7 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -646,6 +646,34 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir)
 
 }
 
+int ux500_msp_i2s_of_init_msp(struct platform_device *pdev,
+			      struct ux500_msp *msp,
+			      struct msp_i2s_platform_data **platform_data)
+{
+	struct msp_i2s_platform_data *pdata;
+
+	*platform_data = devm_kzalloc(&pdev->dev,
+				     sizeof(struct msp_i2s_platform_data),
+				     GFP_KERNEL);
+	pdata = *platform_data;
+	if (!pdata)
+		return -ENOMEM;
+
+	msp->playback_dma_data.dma_cfg = devm_kzalloc(&pdev->dev,
+					sizeof(struct stedma40_chan_cfg),
+					GFP_KERNEL);
+	if (!msp->playback_dma_data.dma_cfg)
+		return -ENOMEM;
+
+	msp->capture_dma_data.dma_cfg = devm_kzalloc(&pdev->dev,
+					sizeof(struct stedma40_chan_cfg),
+					GFP_KERNEL);
+	if (!msp->capture_dma_data.dma_cfg)
+		return -ENOMEM;
+
+	return 0;
+}
+
 int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 			struct ux500_msp **msp_p,
 			struct msp_i2s_platform_data *platform_data)
@@ -653,30 +681,28 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 	struct resource *res = NULL;
 	struct device_node *np = pdev->dev.of_node;
 	struct ux500_msp *msp;
+	int ret;
 
 	*msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL);
 	msp = *msp_p;
 	if (!msp)
 		return -ENOMEM;
 
-	if (np) {
-		if (!platform_data) {
-			platform_data = devm_kzalloc(&pdev->dev,
-				sizeof(struct msp_i2s_platform_data), GFP_KERNEL);
-			if (!platform_data)
-				return -ENOMEM;
-		}
-	} else
-		if (!platform_data)
+	if (!platform_data) {
+		if (np) {
+			ret = ux500_msp_i2s_of_init_msp(pdev, msp,
+							&platform_data);
+			if (ret)
+				return ret;
+		} else
 			return -EINVAL;
+	} else {
+		msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx;
+		msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx;
+		msp->id = platform_data->id;
+	}
 
-	dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__,
-		pdev->name, platform_data->id);
-
-	msp->id = platform_data->id;
 	msp->dev = &pdev->dev;
-	msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx;
-	msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx;
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (res == NULL) {
-- 
cgit v1.2.2


From f382acbe163a6faebd7cafe57800306970e241d4 Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:05 +0000
Subject: ASoC: ux500: Store DMA data in the DAI differently in the pdata and
 DT case

In this patch we do two things. Firstly, instead of open coding the
store of DMA data in to the DAI for later use, we use the API provided.
Secondly we create and store similar DMA data for the DT case, only
this time we use 'struct snd_dmaengine_dai_dma_data' which is provided
by the core for this very reason.

Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_dai.c | 42 +++++++++++++++++++++++++++++++++++++++--
 1 file changed, 40 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index bc042cce115f..f4d607a72668 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -17,12 +17,14 @@
 #include <linux/bitops.h>
 #include <linux/platform_device.h>
 #include <linux/clk.h>
+#include <linux/of.h>
 #include <linux/regulator/consumer.h>
 #include <linux/mfd/dbx500-prcmu.h>
 #include <linux/platform_data/asoc-ux500-msp.h>
 
 #include <sound/soc.h>
 #include <sound/soc-dai.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "ux500_msp_i2s.h"
 #include "ux500_msp_dai.h"
@@ -654,16 +656,52 @@ static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int ux500_msp_dai_of_probe(struct snd_soc_dai *dai)
+{
+	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	struct snd_dmaengine_dai_dma_data *playback_dma_data;
+	struct snd_dmaengine_dai_dma_data *capture_dma_data;
+
+	playback_dma_data = devm_kzalloc(dai->dev,
+					 sizeof(*playback_dma_data),
+					 GFP_KERNEL);
+	if (!playback_dma_data)
+		return -ENOMEM;
+
+	capture_dma_data = devm_kzalloc(dai->dev,
+					sizeof(*capture_dma_data),
+					GFP_KERNEL);
+	if (!capture_dma_data)
+		return -ENOMEM;
+
+	playback_dma_data->addr = drvdata->msp->playback_dma_data.tx_rx_addr;
+	capture_dma_data->addr = drvdata->msp->capture_dma_data.tx_rx_addr;
+
+	playback_dma_data->maxburst = 4;
+	capture_dma_data->maxburst = 4;
+
+	snd_soc_dai_init_dma_data(dai, playback_dma_data, capture_dma_data);
+
+	return 0;
+}
+
 static int ux500_msp_dai_probe(struct snd_soc_dai *dai)
 {
 	struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev);
+	struct msp_i2s_platform_data *pdata = dai->dev->platform_data;
+	int ret;
 
-	dai->playback_dma_data = &drvdata->msp->playback_dma_data;
-	dai->capture_dma_data = &drvdata->msp->capture_dma_data;
+	if (!pdata) {
+		ret = ux500_msp_dai_of_probe(dai);
+		return ret;
+	}
 
 	drvdata->msp->playback_dma_data.data_size = drvdata->slot_width;
 	drvdata->msp->capture_dma_data.data_size = drvdata->slot_width;
 
+	snd_soc_dai_init_dma_data(dai,
+				  &drvdata->msp->playback_dma_data,
+				  &drvdata->msp->capture_dma_data);
 	return 0;
 }
 
-- 
cgit v1.2.2


From ead20611a212db8ab4392cfc28092c9c849c69a4 Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:06 +0000
Subject: ASoC: ux500_pcm: Take out pointless dev_dbg() call

Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_pcm.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 8b53f22edcaf..3d1c342245f0 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -65,14 +65,10 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd,
 	struct snd_pcm_substream *substream)
 {
 	struct snd_soc_dai *dai = rtd->cpu_dai;
-	struct device *dev = dai->dev;
 	u16 per_data_width, mem_data_width;
 	struct stedma40_chan_cfg *dma_cfg;
 	struct ux500_msp_dma_params *dma_params;
 
-	dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
-		snd_pcm_stream_str(substream));
-
 	dma_params = snd_soc_dai_get_dma_data(dai, substream);
 	dma_cfg = dma_params->dma_cfg;
 
-- 
cgit v1.2.2


From 86a3fdfc63402ffbcee226c4a2503eee14a41afe Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:07 +0000
Subject: ASoC: ux500_pcm: Differentiate between pdata and DT initialisation

If booting with full DT support (i.e. DMA too, the last piece of the
puzzle), then we don't need to use the compatible_request_channel call
back or require some of the historical bumph which probably isn't
required by a platform data start-up now either. This will also be
ripped out in upcoming commits.

Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_pcm.c | 20 +++++++++++++++-----
 1 file changed, 15 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 3d1c342245f0..55a8634cc3da 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -145,15 +145,25 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = {
 	.prepare_slave_config = ux500_pcm_prepare_slave_config,
 };
 
+static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = {
+	.compat_request_channel = ux500_pcm_request_chan,
+	.prepare_slave_config = ux500_pcm_prepare_slave_config,
+};
+
 int ux500_pcm_register_platform(struct platform_device *pdev)
 {
+	const struct snd_dmaengine_pcm_config *pcm_config;
+	struct device_node *np = pdev->dev.of_node;
 	int ret;
 
-	ret = snd_dmaengine_pcm_register(&pdev->dev,
-			&ux500_dmaengine_pcm_config,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
-			SND_DMAENGINE_PCM_FLAG_COMPAT |
-			SND_DMAENGINE_PCM_FLAG_NO_DT);
+	if (np)
+		pcm_config = &ux500_dmaengine_of_pcm_config;
+	else
+		pcm_config = &ux500_dmaengine_pcm_config;
+
+	ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config,
+					 SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
+					 SND_DMAENGINE_PCM_FLAG_COMPAT);
 	if (ret < 0) {
 		dev_err(&pdev->dev,
 			"%s: ERROR: Failed to register platform '%s' (%d)!\n",
-- 
cgit v1.2.2


From 33899b19851db3d5baf1bcde49fe90cd5f68c82c Mon Sep 17 00:00:00 2001
From: Lee Jones <lee.jones@linaro.org>
Date: Thu, 19 Dec 2013 15:55:08 +0000
Subject: ASoC: ux500: Dramatically reduce the size of the DAI driver data
 struct

We no longer have a means to differentiate between MSP devices at probe
time, mainly because we don't really have to. So rather than have an over-
sized static data structure in place, where the only difference between
devices is the ID and name (which are unused), we'll just create one
succinct, statically assigned and shared one instead.

Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_dai.c | 96 ++++++-----------------------------------
 1 file changed, 14 insertions(+), 82 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index f4d607a72668..5f4807b2c007 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -718,87 +718,19 @@ static struct snd_soc_dai_ops ux500_msp_dai_ops[] = {
 	}
 };
 
-static struct snd_soc_dai_driver ux500_msp_dai_drv[UX500_NBR_OF_DAI] = {
-	{
-		.name = "ux500-msp-i2s.0",
-		.probe = ux500_msp_dai_probe,
-		.id = 0,
-		.suspend = NULL,
-		.resume = NULL,
-		.playback = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.capture = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.ops = ux500_msp_dai_ops,
-	},
-	{
-		.name = "ux500-msp-i2s.1",
-		.probe = ux500_msp_dai_probe,
-		.id = 1,
-		.suspend = NULL,
-		.resume = NULL,
-		.playback = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.capture = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.ops = ux500_msp_dai_ops,
-	},
-	{
-		.name = "ux500-msp-i2s.2",
-		.id = 2,
-		.probe = ux500_msp_dai_probe,
-		.suspend = NULL,
-		.resume = NULL,
-		.playback = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.capture = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.ops = ux500_msp_dai_ops,
-	},
-	{
-		.name = "ux500-msp-i2s.3",
-		.probe = ux500_msp_dai_probe,
-		.id = 3,
-		.suspend = NULL,
-		.resume = NULL,
-		.playback = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.capture = {
-			.channels_min = UX500_MSP_MIN_CHANNELS,
-			.channels_max = UX500_MSP_MAX_CHANNELS,
-			.rates = UX500_I2S_RATES,
-			.formats = UX500_I2S_FORMATS,
-		},
-		.ops = ux500_msp_dai_ops,
-	},
+static struct snd_soc_dai_driver ux500_msp_dai_drv = {
+	.probe                 = ux500_msp_dai_probe,
+	.suspend               = NULL,
+	.resume                = NULL,
+	.playback.channels_min = UX500_MSP_MIN_CHANNELS,
+	.playback.channels_max = UX500_MSP_MAX_CHANNELS,
+	.playback.rates        = UX500_I2S_RATES,
+	.playback.formats      = UX500_I2S_FORMATS,
+	.capture.channels_min  = UX500_MSP_MIN_CHANNELS,
+	.capture.channels_max  = UX500_MSP_MAX_CHANNELS,
+	.capture.rates         = UX500_I2S_RATES,
+	.capture.formats       = UX500_I2S_FORMATS,
+	.ops                   = ux500_msp_dai_ops,
 };
 
 static const struct snd_soc_component_driver ux500_msp_component = {
@@ -868,7 +800,7 @@ static int ux500_msp_drv_probe(struct platform_device *pdev)
 	dev_set_drvdata(&pdev->dev, drvdata);
 
 	ret = snd_soc_register_component(&pdev->dev, &ux500_msp_component,
-					 &ux500_msp_dai_drv[drvdata->msp->id], 1);
+					 &ux500_msp_dai_drv, 1);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "Error: %s: Failed to register MSP%d!\n",
 			__func__, drvdata->msp->id);
-- 
cgit v1.2.2


From f87a3e825cb0f7d4d51556ece147f1a6299ac1af Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 7 Jan 2014 09:13:42 +0800
Subject: ASoC: simple-card: fix the DAPM routes map parsing

The simple-card's DAPM route maping is optional.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index d4402fb57253..eb95beb25d43 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -134,10 +134,12 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 		(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
 
 	/* DAPM routes */
-	ret = snd_soc_of_parse_audio_routing(&info->snd_card,
+	if (of_property_read_bool(node, "simple-audio-routing")) {
+		ret = snd_soc_of_parse_audio_routing(&info->snd_card,
 					"simple-audio-routing");
-	if (ret)
-		return ret;
+		if (ret)
+			return ret;
+	}
 
 	/* CPU sub-node */
 	ret = -EINVAL;
-- 
cgit v1.2.2


From 8c0b8230b2d9708eed5b50f9f8442aaa879a3c57 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 7 Jan 2014 09:15:16 +0800
Subject: ASoC: simple-card: keep the property's name the same pattern

Even though we might not have rigor rule for the simple card property names,
according to the existing ones, they are all in a same pattern:

[simple-audio-card,]XXX;

Rename simple-audio-routing to simple-audio-card,routing, and make the simple
card's properties has one unified name.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index eb95beb25d43..0430be85f23c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -134,9 +134,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 		(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
 
 	/* DAPM routes */
-	if (of_property_read_bool(node, "simple-audio-routing")) {
+	if (of_property_read_bool(node, "simple-audio-card,routing")) {
 		ret = snd_soc_of_parse_audio_routing(&info->snd_card,
-					"simple-audio-routing");
+					"simple-audio-card,routing");
 		if (ret)
 			return ret;
 	}
-- 
cgit v1.2.2


From dcf0fa27a56025793a700e81edd261ee3369e294 Mon Sep 17 00:00:00 2001
From: Jean-Francois Moine <moinejf@free.fr>
Date: Fri, 3 Jan 2014 09:19:18 +0100
Subject: ASoC: pcm: Fix lack of platform bespoke_trigger() call

When the platform driver has no ops, the platform function
bespoke_trigger() is no more called.

The problem was introduced by the commit c5914b0aaea6494aaa9e415cbd32f8b7eb604af0
	"ASoC: pcm: Check for ops before deferencing them"

Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index feb0f2843026..d70eecd9e168 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -769,7 +769,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 			return ret;
 	}
 
-	if (platform->driver->ops && platform->driver->bespoke_trigger) {
+	if (platform->driver->bespoke_trigger) {
 		ret = platform->driver->bespoke_trigger(substream, cmd);
 		if (ret < 0)
 			return ret;
-- 
cgit v1.2.2


From 1e9de42f4324b91ce2e9da0939cab8fc6ae93893 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Tue, 7 Jan 2014 17:51:42 +0000
Subject: ASoC: dpcm: Explicitly set BE DAI link supported stream directions

Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).

Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..141a302e4e77 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2026,10 +2026,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	int ret = 0, playback = 0, capture = 0;
 
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		if (cpu_dai->driver->playback.channels_min)
-			playback = 1;
-		if (cpu_dai->driver->capture.channels_min)
-			capture = 1;
+		playback = rtd->dai_link->dpcm_playback;
+		capture = rtd->dai_link->dpcm_capture;
 	} else {
 		if (codec_dai->driver->playback.channels_min &&
 		    cpu_dai->driver->playback.channels_min)
-- 
cgit v1.2.2


From 58c57cfa0d0a2ff9191bf6210666ce81e1b2f6c2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jan 2014 18:22:49 +0100
Subject: ALSA: hda - Minor code optimization for patch_realtek.c

No functional change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 15 +++------------
 1 file changed, 3 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6a705a81de35..74b496b0d179 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -469,9 +469,7 @@ static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
  *	7  ~ 0	:	Assembly ID
  *	port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
  */
-static int alc_subsystem_id(struct hda_codec *codec,
-			    hda_nid_t porta, hda_nid_t porte,
-			    hda_nid_t portd, hda_nid_t porti)
+static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports)
 {
 	unsigned int ass, tmp, i;
 	unsigned nid;
@@ -561,14 +559,7 @@ do_sku:
 	      spec->gen.autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
-		if (tmp == 0)
-			nid = porta;
-		else if (tmp == 1)
-			nid = porte;
-		else if (tmp == 2)
-			nid = portd;
-		else if (tmp == 3)
-			nid = porti;
+		nid = ports[tmp];
 		if (found_in_nid_list(nid, spec->gen.autocfg.line_out_pins,
 				      spec->gen.autocfg.line_outs))
 			return 1;
@@ -581,7 +572,7 @@ do_sku:
  * ports contains an array of 4 pin NIDs for port-A, E, D and I */
 static void alc_ssid_check(struct hda_codec *codec, const hda_nid_t *ports)
 {
-	if (!alc_subsystem_id(codec, ports[0], ports[1], ports[2], ports[3])) {
+	if (!alc_subsystem_id(codec, ports)) {
 		struct alc_spec *spec = codec->spec;
 		snd_printd("realtek: "
 			   "Enable default setup for auto mode as fallback\n");
-- 
cgit v1.2.2


From 7546abfb8e1f9933b549f05898377e9444ee4cb2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 9 Dec 2013 10:18:09 +0100
Subject: ALSA: hda - Increment default stream numbers for AMD HDMI controllers

It turned out that some AMD HDMI controllers still don't provide
proper values in GCAP register (all zero), and the driver assigns only
one stream in that case, although the connected codec chip supports
more than one stream.

In this patch, the default max number of streams for AMD HDMI
controllers is increased to 8, which  should suffice for most use
cases.  The overhead by this increase is more azx_dev struct and BDL
allocations, so it's negligible.  Of course, if the controller
provides a proper GCAP register, the register value would be used.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 440c35546e57..53937dede9fa 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -297,9 +297,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define ULI_NUM_CAPTURE		5
 #define ULI_NUM_PLAYBACK	6
 
-/* ATI HDMI has 1 playback and 0 capture */
+/* ATI HDMI may have up to 8 playbacks and 0 capture */
 #define ATIHDMI_NUM_CAPTURE	0
-#define ATIHDMI_NUM_PLAYBACK	1
+#define ATIHDMI_NUM_PLAYBACK	8
 
 /* TERA has 4 playback and 3 capture */
 #define TERA_NUM_CAPTURE	3
-- 
cgit v1.2.2


From d81e397c65279847e473a2284927ee5b4f712693 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jan 2014 15:23:44 +0100
Subject: ALSA: hda - firmware patch code cleanup

Just a code refactoring: the need_codec flag in hda_patch_item struct
can be removed by checking the current mode instead.  No functional
change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 21 ++++++++++-----------
 1 file changed, 10 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 0ec6fb1e9846..79b3dd0d4dd8 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -763,19 +763,18 @@ DEFINE_PARSE_ID_MODE(revision_id);
 struct hda_patch_item {
 	const char *tag;
 	void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc);
-	int need_codec;
 };
 
 static struct hda_patch_item patch_items[NUM_LINE_MODES] = {
-	[LINE_MODE_CODEC] = { "[codec]", parse_codec_mode, 0 },
-	[LINE_MODE_MODEL] = { "[model]", parse_model_mode, 1 },
-	[LINE_MODE_VERB] = { "[verb]", parse_verb_mode, 1 },
-	[LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode, 1 },
-	[LINE_MODE_HINT] = { "[hint]", parse_hint_mode, 1 },
-	[LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode, 1 },
-	[LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode, 1 },
-	[LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode, 1 },
-	[LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode, 1 },
+	[LINE_MODE_CODEC] = { "[codec]", parse_codec_mode },
+	[LINE_MODE_MODEL] = { "[model]", parse_model_mode },
+	[LINE_MODE_VERB] = { "[verb]", parse_verb_mode },
+	[LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode },
+	[LINE_MODE_HINT] = { "[hint]", parse_hint_mode },
+	[LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode },
+	[LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode },
+	[LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode },
+	[LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode },
 };
 
 /* check the line starting with '[' -- change the parser mode accodingly */
@@ -846,7 +845,7 @@ int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf)
 		if (*buf == '[')
 			line_mode = parse_line_mode(buf, bus);
 		else if (patch_items[line_mode].parser &&
-			 (codec || !patch_items[line_mode].need_codec))
+			 (codec || line_mode <= LINE_MODE_CODEC))
 			patch_items[line_mode].parser(buf, bus, &codec);
 	}
 	return 0;
-- 
cgit v1.2.2


From 8e3ae6f7ad825129ce3691934bc7238f9b70029b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jan 2014 15:28:51 +0100
Subject: ALSA: hda - Add consistent tag names for firmware patch

Some tags used in the firmware patch file are inconsistent with hwdep
sysfs file names, such as, the firmware patch takes [hint] tag while
sysfs file is */hints.  This makes even me referring back to the
document often.  Let's provide the same tag names as sysfs for
reducing confusions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 51 ++++++++++++++++++++++++++++++++++++++---------
 1 file changed, 42 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 79b3dd0d4dd8..72d8389fb399 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -762,19 +762,50 @@ DEFINE_PARSE_ID_MODE(revision_id);
 
 struct hda_patch_item {
 	const char *tag;
+	const char *alias;
 	void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc);
 };
 
 static struct hda_patch_item patch_items[NUM_LINE_MODES] = {
-	[LINE_MODE_CODEC] = { "[codec]", parse_codec_mode },
-	[LINE_MODE_MODEL] = { "[model]", parse_model_mode },
-	[LINE_MODE_VERB] = { "[verb]", parse_verb_mode },
-	[LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode },
-	[LINE_MODE_HINT] = { "[hint]", parse_hint_mode },
-	[LINE_MODE_VENDOR_ID] = { "[vendor_id]", parse_vendor_id_mode },
-	[LINE_MODE_SUBSYSTEM_ID] = { "[subsystem_id]", parse_subsystem_id_mode },
-	[LINE_MODE_REVISION_ID] = { "[revision_id]", parse_revision_id_mode },
-	[LINE_MODE_CHIP_NAME] = { "[chip_name]", parse_chip_name_mode },
+	[LINE_MODE_CODEC] = {
+		.tag = "[codec]",
+		.parser = parse_codec_mode,
+	},
+	[LINE_MODE_MODEL] = {
+		.tag = "[model]",
+		.parser = parse_model_mode,
+	},
+	[LINE_MODE_VERB] = {
+		.tag = "[verb]",
+		.alias = "[init_verbs]",
+		.parser = parse_verb_mode,
+	},
+	[LINE_MODE_PINCFG] = {
+		.tag = "[pincfg]",
+		.alias = "[user_pin_configs]",
+		.parser = parse_pincfg_mode,
+	},
+	[LINE_MODE_HINT] = {
+		.tag = "[hint]",
+		.alias = "[hints]",
+		.parser = parse_hint_mode
+	},
+	[LINE_MODE_VENDOR_ID] = {
+		.tag = "[vendor_id]",
+		.parser = parse_vendor_id_mode,
+	},
+	[LINE_MODE_SUBSYSTEM_ID] = {
+		.tag = "[subsystem_id]",
+		.parser = parse_subsystem_id_mode,
+	},
+	[LINE_MODE_REVISION_ID] = {
+		.tag = "[revision_id]",
+		.parser = parse_revision_id_mode,
+	},
+	[LINE_MODE_CHIP_NAME] = {
+		.tag = "[chip_name]",
+		.parser = parse_chip_name_mode,
+	},
 };
 
 /* check the line starting with '[' -- change the parser mode accodingly */
@@ -786,6 +817,8 @@ static int parse_line_mode(char *buf, struct hda_bus *bus)
 			continue;
 		if (strmatch(buf, patch_items[i].tag))
 			return i;
+		if (patch_items[i].alias && strmatch(buf, patch_items[i].alias))
+			return i;
 	}
 	return LINE_MODE_NONE;
 }
-- 
cgit v1.2.2


From ed0e0d0617a8dc3d8b82c6e54827f269f2247b07 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jan 2014 17:48:11 +0100
Subject: ALSA: hda - Correct AD1986A 3stack pin configs

The 3stack pin configs for AD1986A codec had incorrect values that
resulted in broken mic and line-in.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index b174eb1567b8..819d4a98abaf 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -281,11 +281,11 @@ static const struct hda_fixup ad1986a_fixups[] = {
 		.v.pins = (const struct hda_pintbl[]) {
 			{ 0x1a, 0x02214021 }, /* headphone */
 			{ 0x1b, 0x01014011 }, /* front */
-			{ 0x1c, 0x01013012 }, /* surround */
-			{ 0x1d, 0x01019015 }, /* clfe */
+			{ 0x1c, 0x01813030 }, /* line-in */
+			{ 0x1d, 0x01a19020 }, /* rear mic */
 			{ 0x1e, 0x411111f0 }, /* N/A */
 			{ 0x1f, 0x02a190f0 }, /* mic */
-			{ 0x20, 0x018130f0 }, /* line-in */
+			{ 0x20, 0x411111f0 }, /* N/A */
 			{}
 		},
 	},
-- 
cgit v1.2.2


From 43a8e50a46a4e1dd1451e4a4ffa1f7695fb7d287 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jan 2014 18:11:44 +0100
Subject: ALSA: hda - Don't create duplicated ctls for loopback paths

AD1986A mic pins (0x1d and 0x1f) share the same widget for controlling
the loopback volume/mute, but the generic parser didn't check it.
This ended up with the duplicated controls for the same effect.

This patch adds the check of the duplication for avoiding it.

After this fix, there will be only one control although it affects
both paths; this remaining issue should be fixed later in a different
patch.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ce004326d54b..7513089c8985 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -2858,9 +2858,11 @@ static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid,
 	if (num_conns < idx)
 		return false;
 	nid = list[idx];
-	if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT))
+	if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT) &&
+	    !is_ctl_associated(codec, nid, HDA_OUTPUT, 0, NID_PATH_VOL_CTL))
 		*mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
-	if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT))
+	if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT) &&
+	    !is_ctl_associated(codec, nid, HDA_OUTPUT, 0, NID_PATH_MUTE_CTL))
 		*mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
 
 	return *mix_val || *mute_val;
-- 
cgit v1.2.2


From bece9e957cbfb37f12488b24166364307e39f5b0 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Wed, 8 Jan 2014 10:40:18 +0000
Subject: ASoC: utils: Add internal call to determine if DAI is dummy.

Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI.
This is for internal DAPM usage only and is used to determine whether to
insert a DAI link connection into the DAPM graph.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-utils.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 5e633659c1b3..d14bdb3c52df 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -123,6 +123,13 @@ static struct snd_soc_dai_driver dummy_dai = {
 	 },
 };
 
+int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
+{
+	if (dai->driver == &dummy_dai)
+		return 1;
+	return 0;
+}
+
 static int snd_soc_dummy_probe(struct platform_device *pdev)
 {
 	int ret;
-- 
cgit v1.2.2


From b893ea5f1cd1adbbd7e0794d16d47bbb46f80733 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Wed, 8 Jan 2014 10:40:19 +0000
Subject: ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.

Connect the DAPM graph through each BE DAI link to the componnent(s) on the
other side of the BE DAI link. This allows the graph to be walked on
both sides of the link when graph changes are made.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-core.c |  1 +
 sound/soc/soc-dapm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 50 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..7d9c0660ab24 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1728,6 +1728,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	snd_soc_dapm_link_dai_widgets(card);
+	snd_soc_dapm_connect_dai_link_widgets(card);
 
 	if (card->controls)
 		snd_soc_add_card_controls(card, card->controls, card->num_controls);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 67e63ab1f11e..51b4c192f41a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3634,6 +3634,55 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 	return 0;
 }
 
+void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd = card->rtd;
+	struct snd_soc_dai *cpu_dai, *codec_dai;
+	struct snd_soc_dapm_route r;
+	int i;
+
+	memset(&r, 0, sizeof(r));
+
+	/* for each BE DAI link... */
+	for (i = 0; i < card->num_rtd; i++) {
+		rtd = &card->rtd[i];
+		cpu_dai = rtd->cpu_dai;
+		codec_dai = rtd->codec_dai;
+
+		/* dynamic FE links have no fixed DAI mapping */
+		if (rtd->dai_link->dynamic)
+			continue;
+
+		/* there is no point in connecting BE DAI links with dummies */
+		if (snd_soc_dai_is_dummy(codec_dai) ||
+			snd_soc_dai_is_dummy(cpu_dai))
+			continue;
+
+		/* connect BE DAI playback if widgets are valid */
+		if (codec_dai->playback_widget && cpu_dai->playback_widget) {
+			r.source = cpu_dai->playback_widget->name;
+			r.sink = codec_dai->playback_widget->name;
+			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+				cpu_dai->codec->name, r.source,
+				codec_dai->platform->name, r.sink);
+
+			snd_soc_dapm_add_route(&card->dapm, &r);
+		}
+
+		/* connect BE DAI capture if widgets are valid */
+		if (codec_dai->capture_widget && cpu_dai->capture_widget) {
+			r.source = codec_dai->capture_widget->name;
+			r.sink = cpu_dai->capture_widget->name;
+			dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+				codec_dai->codec->name, r.source,
+				cpu_dai->platform->name, r.sink);
+
+			snd_soc_dapm_add_route(&card->dapm, &r);
+		}
+
+	}
+}
+
 static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 	int event)
 {
-- 
cgit v1.2.2


From e20970ada3f699c113fe64b04492af083d11a7d8 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Wed, 8 Jan 2014 11:22:25 +0100
Subject: ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant

The driver defines ADAU1701_SEROCTL_WORD_LEN_16 as 0x10 while it should be b10,
so 0x2. This patch fixes it.

Reported-by: Magnus Reftel <magnus.reftel@lockless.no>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
---
 sound/soc/codecs/adau1701.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index ebff1128be59..adee866f463f 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -71,7 +71,7 @@
 
 #define ADAU1701_SEROCTL_WORD_LEN_24	0x0000
 #define ADAU1701_SEROCTL_WORD_LEN_20	0x0001
-#define ADAU1701_SEROCTL_WORD_LEN_16	0x0010
+#define ADAU1701_SEROCTL_WORD_LEN_16	0x0002
 #define ADAU1701_SEROCTL_WORD_LEN_MASK	0x0003
 
 #define ADAU1701_AUXNPOW_VBPD		0x40
-- 
cgit v1.2.2


From 7ee4518ab75164533e282eb8f2827a74920a2a19 Mon Sep 17 00:00:00 2001
From: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Date: Wed, 8 Jan 2014 18:13:35 +0800
Subject: ASoC: ux500: Fix sparse non static symbol warning

Fixes the following sparse warning:

sound/soc/ux500/ux500_msp_i2s.c:649:5: warning:
 symbol 'ux500_msp_i2s_of_init_msp' was not declared. Should it be static?

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_msp_i2s.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 7f2a4acddcd7..959d7b4edf56 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -646,9 +646,9 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir)
 
 }
 
-int ux500_msp_i2s_of_init_msp(struct platform_device *pdev,
-			      struct ux500_msp *msp,
-			      struct msp_i2s_platform_data **platform_data)
+static int ux500_msp_i2s_of_init_msp(struct platform_device *pdev,
+				struct ux500_msp *msp,
+				struct msp_i2s_platform_data **platform_data)
 {
 	struct msp_i2s_platform_data *pdata;
 
-- 
cgit v1.2.2


From 633ff8f8a4393b4a13b94eddd2613198c32035e6 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Wed, 8 Jan 2014 16:13:05 +0800
Subject: ASoC: fsl-sai: Clean up the code

Makes the code slightly shorter.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_sai.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 5d38a6749b9f..cdd3fa830704 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -62,26 +62,25 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		reg_cr2 = FSL_SAI_RCR2;
 
 	val_cr2 = sai_readl(sai, sai->base + reg_cr2);
+	val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
+
 	switch (clk_id) {
 	case FSL_SAI_CLK_BUS:
-		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
 		val_cr2 |= FSL_SAI_CR2_MSEL_BUS;
 		break;
 	case FSL_SAI_CLK_MAST1:
-		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
 		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK1;
 		break;
 	case FSL_SAI_CLK_MAST2:
-		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
 		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK2;
 		break;
 	case FSL_SAI_CLK_MAST3:
-		val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
 		val_cr2 |= FSL_SAI_CR2_MSEL_MCLK3;
 		break;
 	default:
 		return -EINVAL;
 	}
+
 	sai_writel(sai, val_cr2, sai->base + reg_cr2);
 
 	return 0;
-- 
cgit v1.2.2


From a8fc415c29a62e4f0a7a932110ee9d8423e2cc52 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:49 +0200
Subject: ASoC: twl4030: Separate write condition checking from I/O function

Simplifies the code a bit and prepares it to the removal of local caching.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 40 +++++++++++++++++++++++-----------------
 1 file changed, 23 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index dfc51bb425da..419108ae31de 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -181,50 +181,56 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
 	cache[reg] = value;
 }
 
-/*
- * write to the twl4030 register space
- */
-static int twl4030_write(struct snd_soc_codec *codec,
-			unsigned int reg, unsigned int value)
+static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
+				      unsigned int reg)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	int write_to_reg = 0;
+	bool write_to_reg = false;
 
-	twl4030_write_reg_cache(codec, reg, value);
 	/* Decide if the given register can be written */
 	switch (reg) {
 	case TWL4030_REG_EAR_CTL:
 		if (twl4030->earpiece_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	case TWL4030_REG_PREDL_CTL:
 		if (twl4030->predrivel_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	case TWL4030_REG_PREDR_CTL:
 		if (twl4030->predriver_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	case TWL4030_REG_PRECKL_CTL:
 		if (twl4030->carkitl_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	case TWL4030_REG_PRECKR_CTL:
 		if (twl4030->carkitr_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	case TWL4030_REG_HS_GAIN_SET:
 		if (twl4030->hsl_enabled || twl4030->hsr_enabled)
-			write_to_reg = 1;
+			write_to_reg = true;
 		break;
 	default:
 		/* All other register can be written */
-		write_to_reg = 1;
+		write_to_reg = true;
 		break;
 	}
-	if (write_to_reg)
-		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-					    value, reg);
+
+	return write_to_reg;
+}
+
+/*
+ * write to the twl4030 register space
+ */
+static int twl4030_write(struct snd_soc_codec *codec,
+			 unsigned int reg, unsigned int value)
+{
+	twl4030_write_reg_cache(codec, reg, value);
+	if (twl4030_can_write_to_chip(codec, reg))
+		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 
 	return 0;
 }
-- 
cgit v1.2.2


From 7bfbdfea576e3ae109fa182519b6f004c6024952 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:50 +0200
Subject: ASoC: twl4030: Remove check defaults functionality

No need to keep the check defaults functionality anymore.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 23 -----------------------
 1 file changed, 23 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 419108ae31de..7b732ab70d2c 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -268,25 +268,6 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 	udelay(10);
 }
 
-static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
-{
-	int i, difference = 0;
-	u8 val;
-
-	dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
-	for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
-		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
-		if (val != twl4030_reg[i]) {
-			difference++;
-			dev_dbg(codec->dev,
-				 "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
-				 i, val, twl4030_reg[i]);
-		}
-	}
-	dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
-		 difference, difference ? "Not OK" : "OK");
-}
-
 static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
 {
 	int i;
@@ -378,10 +359,6 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 		}
 	}
 
-	/* Check defaults, if instructed before anything else */
-	if (pdata && pdata->check_defaults)
-		twl4030_check_defaults(codec);
-
 	/* Reset registers, if no setup data or if instructed to do so */
 	if (!pdata || (pdata && pdata->reset_registers))
 		twl4030_reset_registers(codec);
-- 
cgit v1.2.2


From 0dc41562a44c9e1012bb810c2a84e81c425867b0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:51 +0200
Subject: ASoC: twl4030: Remove reset registers functionality

The register states now tracked by the regmap implementation in the core which
makes the reset registers functionality 'redundant' since we know the state
of the registers now all the time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 17 -----------------
 1 file changed, 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 7b732ab70d2c..ab2f22299db2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -268,17 +268,6 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 	udelay(10);
 }
 
-static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
-{
-	int i;
-
-	/* set all audio section registers to reasonable defaults */
-	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
-		if (i != TWL4030_REG_APLL_CTL)
-			twl4030_write(codec, i, twl4030_reg[i]);
-
-}
-
 static void twl4030_setup_pdata_of(struct twl4030_codec_data *pdata,
 				   struct device_node *node)
 {
@@ -359,10 +348,6 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 		}
 	}
 
-	/* Reset registers, if no setup data or if instructed to do so */
-	if (!pdata || (pdata && pdata->reset_registers))
-		twl4030_reset_registers(codec);
-
 	/* Refresh APLL_CTL register from HW */
 	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
 			    TWL4030_REG_APLL_CTL);
@@ -2293,8 +2278,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec)
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	struct twl4030_codec_data *pdata = twl4030->pdata;
 
-	/* Reset registers to their chip default before leaving */
-	twl4030_reset_registers(codec);
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio))
-- 
cgit v1.2.2


From 8b3bca2966985f559f9ace1effc98955006f2b05 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:52 +0200
Subject: ASoC: twl4030: Introduce local ctl register cache

Few registers need to be cached in the codec driver level. These registers
should only be written when the path is active to avoid pop noise on the
given path.
This patch adds an array which covers the range where the sensitive registers
are located and uppon loadinf the driver the ctl cache will be initialized.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index ab2f22299db2..f88207712d3d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -150,10 +150,22 @@ struct twl4030_priv {
 	u8 earpiece_enabled;
 	u8 predrivel_enabled, predriver_enabled;
 	u8 carkitl_enabled, carkitr_enabled;
+	u8 ctl_cache[TWL4030_REG_PRECKR_CTL - TWL4030_REG_EAR_CTL + 1];
 
 	struct twl4030_codec_data *pdata;
 };
 
+static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030)
+{
+	int i;
+	u8 byte;
+
+	for (i = TWL4030_REG_EAR_CTL; i <= TWL4030_REG_PRECKR_CTL; i++) {
+		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, i);
+		twl4030->ctl_cache[i - TWL4030_REG_EAR_CTL] = byte;
+	}
+}
+
 /*
  * read twl4030 register cache
  */
@@ -348,6 +360,9 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 		}
 	}
 
+	/* Initialize the local ctl register cache */
+	tw4030_init_ctl_cache(twl4030);
+
 	/* Refresh APLL_CTL register from HW */
 	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
 			    TWL4030_REG_APLL_CTL);
-- 
cgit v1.2.2


From efc8acff1ffe18b981d70da7ab2525e5b3e5de85 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:53 +0200
Subject: ASoC: twl4030: Remove local reg cache

Depend on the regmap reg cache implementation for register caching done in
the twl-core driver.
The local register cache can be removed and we can keep only shadow copies
of certain ctl registers for pop noise reduction.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 207 ++++++++++++++-------------------------------
 1 file changed, 63 insertions(+), 144 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index f88207712d3d..dda53e8c51e5 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -48,86 +48,6 @@
 
 #define TWL4030_CACHEREGNUM	(TWL4030_REG_MISC_SET_2 + 1)
 
-/*
- * twl4030 register cache & default register settings
- */
-static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
-	0x00, /* this register not used		*/
-	0x00, /* REG_CODEC_MODE		(0x1)	*/
-	0x00, /* REG_OPTION		(0x2)	*/
-	0x00, /* REG_UNKNOWN		(0x3)	*/
-	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
-	0x00, /* REG_ANAMICL		(0x5)	*/
-	0x00, /* REG_ANAMICR		(0x6)	*/
-	0x00, /* REG_AVADC_CTL		(0x7)	*/
-	0x00, /* REG_ADCMICSEL		(0x8)	*/
-	0x00, /* REG_DIGMIXING		(0x9)	*/
-	0x0f, /* REG_ATXL1PGA		(0xA)	*/
-	0x0f, /* REG_ATXR1PGA		(0xB)	*/
-	0x0f, /* REG_AVTXL2PGA		(0xC)	*/
-	0x0f, /* REG_AVTXR2PGA		(0xD)	*/
-	0x00, /* REG_AUDIO_IF		(0xE)	*/
-	0x00, /* REG_VOICE_IF		(0xF)	*/
-	0x3f, /* REG_ARXR1PGA		(0x10)	*/
-	0x3f, /* REG_ARXL1PGA		(0x11)	*/
-	0x3f, /* REG_ARXR2PGA		(0x12)	*/
-	0x3f, /* REG_ARXL2PGA		(0x13)	*/
-	0x25, /* REG_VRXPGA		(0x14)	*/
-	0x00, /* REG_VSTPGA		(0x15)	*/
-	0x00, /* REG_VRX2ARXPGA		(0x16)	*/
-	0x00, /* REG_AVDAC_CTL		(0x17)	*/
-	0x00, /* REG_ARX2VTXPGA		(0x18)	*/
-	0x32, /* REG_ARXL1_APGA_CTL	(0x19)	*/
-	0x32, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
-	0x32, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
-	0x32, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
-	0x00, /* REG_ATX2ARXPGA		(0x1D)	*/
-	0x00, /* REG_BT_IF		(0x1E)	*/
-	0x55, /* REG_BTPGA		(0x1F)	*/
-	0x00, /* REG_BTSTPGA		(0x20)	*/
-	0x00, /* REG_EAR_CTL		(0x21)	*/
-	0x00, /* REG_HS_SEL		(0x22)	*/
-	0x00, /* REG_HS_GAIN_SET	(0x23)	*/
-	0x00, /* REG_HS_POPN_SET	(0x24)	*/
-	0x00, /* REG_PREDL_CTL		(0x25)	*/
-	0x00, /* REG_PREDR_CTL		(0x26)	*/
-	0x00, /* REG_PRECKL_CTL		(0x27)	*/
-	0x00, /* REG_PRECKR_CTL		(0x28)	*/
-	0x00, /* REG_HFL_CTL		(0x29)	*/
-	0x00, /* REG_HFR_CTL		(0x2A)	*/
-	0x05, /* REG_ALC_CTL		(0x2B)	*/
-	0x00, /* REG_ALC_SET1		(0x2C)	*/
-	0x00, /* REG_ALC_SET2		(0x2D)	*/
-	0x00, /* REG_BOOST_CTL		(0x2E)	*/
-	0x00, /* REG_SOFTVOL_CTL	(0x2F)	*/
-	0x13, /* REG_DTMF_FREQSEL	(0x30)	*/
-	0x00, /* REG_DTMF_TONEXT1H	(0x31)	*/
-	0x00, /* REG_DTMF_TONEXT1L	(0x32)	*/
-	0x00, /* REG_DTMF_TONEXT2H	(0x33)	*/
-	0x00, /* REG_DTMF_TONEXT2L	(0x34)	*/
-	0x79, /* REG_DTMF_TONOFF	(0x35)	*/
-	0x11, /* REG_DTMF_WANONOFF	(0x36)	*/
-	0x00, /* REG_I2S_RX_SCRAMBLE_H	(0x37)	*/
-	0x00, /* REG_I2S_RX_SCRAMBLE_M	(0x38)	*/
-	0x00, /* REG_I2S_RX_SCRAMBLE_L	(0x39)	*/
-	0x06, /* REG_APLL_CTL		(0x3A)	*/
-	0x00, /* REG_DTMF_CTL		(0x3B)	*/
-	0x44, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
-	0x69, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
-	0x00, /* REG_MISC_SET_1		(0x3E)	*/
-	0x00, /* REG_PCMBTMUX		(0x3F)	*/
-	0x00, /* not used		(0x40)	*/
-	0x00, /* not used		(0x41)	*/
-	0x00, /* not used		(0x42)	*/
-	0x00, /* REG_RX_PATH_SEL	(0x43)	*/
-	0x32, /* REG_VDL_APGA_CTL	(0x44)	*/
-	0x00, /* REG_VIBRA_CTL		(0x45)	*/
-	0x00, /* REG_VIBRA_SET		(0x46)	*/
-	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
-	0x00, /* REG_ANAMIC_GAIN	(0x48)	*/
-	0x00, /* REG_MISC_SET_2		(0x49)	*/
-};
-
 /* codec private data */
 struct twl4030_priv {
 	unsigned int codec_powered;
@@ -166,31 +86,48 @@ static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030)
 	}
 }
 
-/*
- * read twl4030 register cache
- */
-static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
-	unsigned int reg)
+static void twl4030_update_ctl_cache(struct snd_soc_codec *codec,
+				     unsigned int reg, unsigned int value)
 {
-	u8 *cache = codec->reg_cache;
-
-	if (reg >= TWL4030_CACHEREGNUM)
-		return -EIO;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
-	return cache[reg];
+	switch (reg) {
+	case TWL4030_REG_EAR_CTL:
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_PRECKL_CTL:
+	case TWL4030_REG_PRECKR_CTL:
+	case TWL4030_REG_HS_GAIN_SET:
+		twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value;
+		break;
+	default:
+		break;
+	}
 }
 
-/*
- * write twl4030 register cache
- */
-static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
-						u8 reg, u8 value)
+static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg)
 {
-	u8 *cache = codec->reg_cache;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
+	u8 value = 0;
 
 	if (reg >= TWL4030_CACHEREGNUM)
-		return;
-	cache[reg] = value;
+		return -EIO;
+
+	switch (reg) {
+	case TWL4030_REG_EAR_CTL:
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_PRECKL_CTL:
+	case TWL4030_REG_PRECKR_CTL:
+	case TWL4030_REG_HS_GAIN_SET:
+		value = twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL];
+		break;
+	default:
+		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg);
+		break;
+	}
+
+	return value;
 }
 
 static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
@@ -234,13 +171,10 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
 	return write_to_reg;
 }
 
-/*
- * write to the twl4030 register space
- */
 static int twl4030_write(struct snd_soc_codec *codec,
 			 unsigned int reg, unsigned int value)
 {
-	twl4030_write_reg_cache(codec, reg, value);
+	twl4030_update_ctl_cache(codec, reg, value);
 	if (twl4030_can_write_to_chip(codec, reg))
 		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 
@@ -270,10 +204,8 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 	else
 		mode = twl4030_audio_disable_resource(TWL4030_AUDIO_RES_POWER);
 
-	if (mode >= 0) {
-		twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode);
+	if (mode >= 0)
 		twl4030->codec_powered = enable;
-	}
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -363,13 +295,8 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	/* Initialize the local ctl register cache */
 	tw4030_init_ctl_cache(twl4030);
 
-	/* Refresh APLL_CTL register from HW */
-	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
-			    TWL4030_REG_APLL_CTL);
-	twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
-
 	/* anti-pop when changing analog gain */
-	reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	reg = twl4030_read(codec, TWL4030_REG_MISC_SET_1);
 	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
 		reg | TWL4030_SMOOTH_ANAVOL_EN);
 
@@ -386,15 +313,15 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 	twl4030->pdata = pdata;
 
-	reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	reg = twl4030_read(codec, TWL4030_REG_HS_POPN_SET);
 	reg &= ~TWL4030_RAMP_DELAY;
 	reg |= (pdata->ramp_delay_value << 2);
-	twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, reg);
 
 	/* initiate offset cancellation */
 	twl4030_codec_enable(codec, 1);
 
-	reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	reg = twl4030_read(codec, TWL4030_REG_ANAMICL);
 	reg &= ~TWL4030_OFFSET_CNCL_SEL;
 	reg |= pdata->offset_cncl_path;
 	twl4030_write(codec, TWL4030_REG_ANAMICL,
@@ -408,15 +335,14 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	msleep(20);
 	do {
 		usleep_range(1000, 2000);
+		twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, true);
 		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
 				    TWL4030_REG_ANAMICL);
+		twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, false);
 	} while ((i++ < 100) &&
 		 ((byte & TWL4030_CNCL_OFFSET_START) ==
 		  TWL4030_CNCL_OFFSET_START));
 
-	/* Make sure that the reg_cache has the same value as the HW */
-	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
-
 	twl4030_codec_enable(codec, 0);
 }
 
@@ -436,9 +362,6 @@ static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 			status = twl4030_audio_disable_resource(
 							TWL4030_AUDIO_RES_APLL);
 	}
-
-	if (status >= 0)
-		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
 }
 
 /* Earpiece */
@@ -661,8 +584,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
 	switch (event) {						\
 	case SND_SOC_DAPM_POST_PMU:					\
 		twl4030->pin_name##_enabled = 1;			\
-		twl4030_write(w->codec, reg,				\
-			twl4030_read_reg_cache(w->codec, reg));		\
+		twl4030_write(w->codec, reg, twl4030_read(w->codec, reg)); \
 		break;							\
 	case SND_SOC_DAPM_POST_PMD:					\
 		twl4030->pin_name##_enabled = 0;			\
@@ -683,7 +605,7 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
 {
 	unsigned char hs_ctl;
 
-	hs_ctl = twl4030_read_reg_cache(codec, reg);
+	hs_ctl = twl4030_read(codec, reg);
 
 	if (ramp) {
 		/* HF ramp-up */
@@ -763,7 +685,7 @@ static int aif_event(struct snd_soc_dapm_widget *w,
 {
 	u8 audio_if;
 
-	audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF);
+	audio_if = twl4030_read(w->codec, TWL4030_REG_AUDIO_IF);
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		/* Enable AIF */
@@ -793,8 +715,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 				    8388608, 16777216, 33554432, 67108864};
 	unsigned int delay;
 
-	hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
-	hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	hs_gain = twl4030_read(codec, TWL4030_REG_HS_GAIN_SET);
+	hs_pop = twl4030_read(codec, TWL4030_REG_HS_POPN_SET);
 	delay = (ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
 		twl4030->sysclk) + 1;
 
@@ -1738,7 +1660,7 @@ static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
 {
 	u8 reg, mask;
 
-	reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+	reg = twl4030_read(codec, TWL4030_REG_OPTION);
 
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
 		mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
@@ -1767,7 +1689,7 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
 		if (twl4030->configured)
 			twl4030_constraints(twl4030, twl4030->master_substream);
 	} else {
-		if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
+		if (!(twl4030_read(codec, TWL4030_REG_CODEC_MODE) &
 			TWL4030_OPTION_1)) {
 			/* In option2 4 channel is not supported, set the
 			 * constraint for the first stream for channels, the
@@ -1815,8 +1737,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 
 	 /* If the substream has 4 channel, do the necessary setup */
 	if (params_channels(params) == 4) {
-		format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
-		mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+		format = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
+		mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE);
 
 		/* Safety check: are we in the correct operating mode and
 		 * the interface is in TDM mode? */
@@ -1832,8 +1754,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 
 	/* bit rate */
-	old_mode = twl4030_read_reg_cache(codec,
-			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
+	old_mode = twl4030_read(codec,
+				TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
 	mode = old_mode & ~TWL4030_APLL_RATE;
 
 	switch (params_rate(params)) {
@@ -1874,7 +1796,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* sample size */
-	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
 	format = old_format;
 	format &= ~TWL4030_DATA_WIDTH;
 	switch (params_format(params)) {
@@ -1957,7 +1879,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	u8 old_format, format;
 
 	/* get format */
-	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
 	format = old_format;
 
 	/* set master/slave audio interface */
@@ -2007,7 +1929,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	u8 reg = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
 
 	if (tristate)
 		reg |= TWL4030_AIF_TRI_EN;
@@ -2024,7 +1946,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
 {
 	u8 reg, mask;
 
-	reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+	reg = twl4030_read(codec, TWL4030_REG_OPTION);
 
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
 		mask = TWL4030_ARXL1_VRX_EN;
@@ -2059,7 +1981,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	/* If the codec mode is not option2, the voice PCM interface is not
 	 * available.
 	 */
-	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+	mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE)
 		& TWL4030_OPT_MODE;
 
 	if (mode != TWL4030_OPTION_2) {
@@ -2091,7 +2013,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
 	twl4030_voice_enable(codec, substream->stream, 1);
 
 	/* bit rate */
-	old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+	old_mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE)
 		& ~(TWL4030_CODECPDZ);
 	mode = old_mode;
 
@@ -2154,7 +2076,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	u8 old_format, format;
 
 	/* get format */
-	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+	old_format = twl4030_read(codec, TWL4030_REG_VOICE_IF);
 	format = old_format;
 
 	/* set master/slave audio interface */
@@ -2201,7 +2123,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+	u8 reg = twl4030_read(codec, TWL4030_REG_VOICE_IF);
 
 	if (tristate)
 		reg |= TWL4030_VIF_TRI_EN;
@@ -2304,13 +2226,10 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 	.probe = twl4030_soc_probe,
 	.remove = twl4030_soc_remove,
-	.read = twl4030_read_reg_cache,
+	.read = twl4030_read,
 	.write = twl4030_write,
 	.set_bias_level = twl4030_set_bias_level,
 	.idle_bias_off = true,
-	.reg_cache_size = sizeof(twl4030_reg),
-	.reg_word_size = sizeof(u8),
-	.reg_cache_default = twl4030_reg,
 
 	.controls = twl4030_snd_controls,
 	.num_controls = ARRAY_SIZE(twl4030_snd_controls),
-- 
cgit v1.2.2


From 7ded5fe020e670befeab6777e7b8bc4bec272a3f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:54 +0200
Subject: ASoC: twl4030: Parameter alignment fixes (for code consistency)

Over time the multi line alignment got messed up. Correct them in one go
so the code will look consistent.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 93 ++++++++++++++++++++++------------------------
 1 file changed, 45 insertions(+), 48 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index dda53e8c51e5..7a5b91e70f98 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -171,8 +171,8 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
 	return write_to_reg;
 }
 
-static int twl4030_write(struct snd_soc_codec *codec,
-			 unsigned int reg, unsigned int value)
+static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg,
+			 unsigned int value)
 {
 	twl4030_update_ctl_cache(codec, reg, value);
 	if (twl4030_can_write_to_chip(codec, reg))
@@ -298,11 +298,11 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	/* anti-pop when changing analog gain */
 	reg = twl4030_read(codec, TWL4030_REG_MISC_SET_1);
 	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-		reg | TWL4030_SMOOTH_ANAVOL_EN);
+		      reg | TWL4030_SMOOTH_ANAVOL_EN);
 
 	twl4030_write(codec, TWL4030_REG_OPTION,
-		TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
-		TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
+		      TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
+		      TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
 
 	/* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
 	twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
@@ -325,7 +325,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	reg &= ~TWL4030_OFFSET_CNCL_SEL;
 	reg |= pdata->offset_cncl_path;
 	twl4030_write(codec, TWL4030_REG_ANAMICL,
-		reg | TWL4030_CNCL_OFFSET_START);
+		      reg | TWL4030_CNCL_OFFSET_START);
 
 	/*
 	 * Wait for offset cancellation to complete.
@@ -337,7 +337,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 		usleep_range(1000, 2000);
 		twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, true);
 		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
-				    TWL4030_REG_ANAMICL);
+				TWL4030_REG_ANAMICL);
 		twl_set_regcache_bypass(TWL4030_MODULE_AUDIO_VOICE, false);
 	} while ((i++ < 100) &&
 		 ((byte & TWL4030_CNCL_OFFSET_START) ==
@@ -577,7 +577,7 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
  */
 #define TWL4030_OUTPUT_PGA(pin_name, reg, mask)				\
 static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
-		struct snd_kcontrol *kcontrol, int event)		\
+			       struct snd_kcontrol *kcontrol, int event) \
 {									\
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \
 									\
@@ -588,8 +588,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
 		break;							\
 	case SND_SOC_DAPM_POST_PMD:					\
 		twl4030->pin_name##_enabled = 0;			\
-		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,		\
-					0, reg);			\
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, 0, reg);	\
 		break;							\
 	}								\
 	return 0;							\
@@ -632,7 +631,7 @@ static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
 }
 
 static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			       struct snd_kcontrol *kcontrol, int event)
 {
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -646,7 +645,7 @@ static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
 }
 
 static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			       struct snd_kcontrol *kcontrol, int event)
 {
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -660,14 +659,14 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
 }
 
 static int vibramux_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			  struct snd_kcontrol *kcontrol, int event)
 {
 	twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff);
 	return 0;
 }
 
 static int apll_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+		      struct snd_kcontrol *kcontrol, int event)
 {
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -681,7 +680,7 @@ static int apll_event(struct snd_soc_dapm_widget *w,
 }
 
 static int aif_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+		     struct snd_kcontrol *kcontrol, int event)
 {
 	u8 audio_if;
 
@@ -693,12 +692,12 @@ static int aif_event(struct snd_soc_dapm_widget *w,
 		twl4030_apll_enable(w->codec, 1);
 
 		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
-						audio_if | TWL4030_AIF_EN);
+			      audio_if | TWL4030_AIF_EN);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
 		/* disable the DAI before we stop it's source PLL */
 		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
-						audio_if &  ~TWL4030_AIF_EN);
+			      audio_if &  ~TWL4030_AIF_EN);
 		twl4030_apll_enable(w->codec, 0);
 		break;
 	}
@@ -736,9 +735,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 		hs_pop |= TWL4030_VMID_EN;
 		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		/* Actually write to the register */
-		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-					hs_gain,
-					TWL4030_REG_HS_GAIN_SET);
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain,
+				 TWL4030_REG_HS_GAIN_SET);
 		hs_pop |= TWL4030_RAMP_EN;
 		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		/* Wait ramp delay time + 1, so the VMID can settle */
@@ -751,9 +749,8 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 		/* Wait ramp delay time + 1, so the VMID can settle */
 		twl4030_wait_ms(delay);
 		/* Bypass the reg_cache to mute the headset */
-		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-					hs_gain & (~0x0f),
-					TWL4030_REG_HS_GAIN_SET);
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f),
+				 TWL4030_REG_HS_GAIN_SET);
 
 		hs_pop &= ~TWL4030_VMID_EN;
 		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
@@ -771,7 +768,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 }
 
 static int headsetlpga_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			     struct snd_kcontrol *kcontrol, int event)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
 
@@ -795,7 +792,7 @@ static int headsetlpga_event(struct snd_soc_dapm_widget *w,
 }
 
 static int headsetrpga_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			     struct snd_kcontrol *kcontrol, int event)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
 
@@ -819,7 +816,7 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
 }
 
 static int digimic_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+			 struct snd_kcontrol *kcontrol, int event)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
 	struct twl4030_codec_data *pdata = twl4030->pdata;
@@ -840,7 +837,7 @@ static int digimic_event(struct snd_soc_dapm_widget *w,
  * Custom volsw and volsw_2r get/put functions to handle these gain bits.
  */
 static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
+				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -869,7 +866,7 @@ static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
 }
 
 static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
+				     struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -898,7 +895,7 @@ static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
 }
 
 static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -925,7 +922,7 @@ static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
 }
 
 static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
+					struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -1656,7 +1653,7 @@ static void twl4030_constraints(struct twl4030_priv *twl4030,
 /* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
  * capture has to be enabled/disabled. */
 static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
-				int enable)
+			       int enable)
 {
 	u8 reg, mask;
 
@@ -1695,8 +1692,8 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
 			 * constraint for the first stream for channels, the
 			 * second stream will 'inherit' this cosntraint */
 			snd_pcm_hw_constraint_minmax(substream->runtime,
-						SNDRV_PCM_HW_PARAM_CHANNELS,
-						2, 2);
+						     SNDRV_PCM_HW_PARAM_CHANNELS,
+						     2, 2);
 		}
 		twl4030->master_substream = substream;
 	}
@@ -1728,8 +1725,8 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream,
 }
 
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params,
-			   struct snd_soc_dai *dai)
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -1845,8 +1842,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
-		int clk_id, unsigned int freq, int dir)
+static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
+				  unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -1871,8 +1868,7 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
-			     unsigned int fmt)
+static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -1942,7 +1938,7 @@ static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate)
 /* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
  * (VTXL, VTXR) for uplink has to be enabled/disabled. */
 static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
-				int enable)
+				 int enable)
 {
 	u8 reg, mask;
 
@@ -1962,7 +1958,7 @@ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
 }
 
 static int twl4030_voice_startup(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *dai)
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -1994,7 +1990,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 }
 
 static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
+				   struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 
@@ -2003,7 +1999,8 @@ static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
 }
 
 static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+				   struct snd_pcm_hw_params *params,
+				   struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -2013,8 +2010,8 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
 	twl4030_voice_enable(codec, substream->stream, 1);
 
 	/* bit rate */
-	old_mode = twl4030_read(codec, TWL4030_REG_CODEC_MODE)
-		& ~(TWL4030_CODECPDZ);
+	old_mode = twl4030_read(codec,
+				TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
 	mode = old_mode;
 
 	switch (params_rate(params)) {
@@ -2048,7 +2045,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
-		int clk_id, unsigned int freq, int dir)
+					int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -2069,7 +2066,7 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 }
 
 static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
-		unsigned int fmt)
+				     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -2242,7 +2239,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 static int twl4030_codec_probe(struct platform_device *pdev)
 {
 	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_twl4030,
-			twl4030_dai, ARRAY_SIZE(twl4030_dai));
+				      twl4030_dai, ARRAY_SIZE(twl4030_dai));
 }
 
 static int twl4030_codec_remove(struct platform_device *pdev)
-- 
cgit v1.2.2


From a450aa6f507542d87ad237cb402b8b6e56329924 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:55 +0200
Subject: ASoC: twl4030: Move the ctl cache update local to twl4030_write()
 function

There's no other users of this functionality, the code can be moved inside
of twl4030_write.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 36 ++++++++++++++++--------------------
 1 file changed, 16 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 7a5b91e70f98..c3c15f891270 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -86,25 +86,6 @@ static void tw4030_init_ctl_cache(struct twl4030_priv *twl4030)
 	}
 }
 
-static void twl4030_update_ctl_cache(struct snd_soc_codec *codec,
-				     unsigned int reg, unsigned int value)
-{
-	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-
-	switch (reg) {
-	case TWL4030_REG_EAR_CTL:
-	case TWL4030_REG_PREDL_CTL:
-	case TWL4030_REG_PREDR_CTL:
-	case TWL4030_REG_PRECKL_CTL:
-	case TWL4030_REG_PRECKR_CTL:
-	case TWL4030_REG_HS_GAIN_SET:
-		twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value;
-		break;
-	default:
-		break;
-	}
-}
-
 static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
@@ -174,7 +155,22 @@ static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
 static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg,
 			 unsigned int value)
 {
-	twl4030_update_ctl_cache(codec, reg, value);
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
+
+	/* Update the ctl cache */
+	switch (reg) {
+	case TWL4030_REG_EAR_CTL:
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_PRECKL_CTL:
+	case TWL4030_REG_PRECKR_CTL:
+	case TWL4030_REG_HS_GAIN_SET:
+		twl4030->ctl_cache[reg - TWL4030_REG_EAR_CTL] = value;
+		break;
+	default:
+		break;
+	}
+
 	if (twl4030_can_write_to_chip(codec, reg))
 		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 
-- 
cgit v1.2.2


From b703b504856b9a9df6bace81e251d185dd72e958 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 3 Jan 2014 15:27:56 +0200
Subject: ASoC: twl4030: Pass the twl4030_priv directly to
 twl4030_can_write_to_chip()

To avoid another lookup for the twl4030_priv in there.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/twl4030.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index c3c15f891270..00665ada23e2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -111,10 +111,9 @@ static unsigned int twl4030_read(struct snd_soc_codec *codec, unsigned int reg)
 	return value;
 }
 
-static bool twl4030_can_write_to_chip(struct snd_soc_codec *codec,
+static bool twl4030_can_write_to_chip(struct twl4030_priv *twl4030,
 				      unsigned int reg)
 {
-	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	bool write_to_reg = false;
 
 	/* Decide if the given register can be written */
@@ -171,7 +170,7 @@ static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg,
 		break;
 	}
 
-	if (twl4030_can_write_to_chip(codec, reg))
+	if (twl4030_can_write_to_chip(twl4030, reg))
 		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 
 	return 0;
-- 
cgit v1.2.2


From 7ae2b55b0129ecb63d73129ddcba6dcda0d37332 Mon Sep 17 00:00:00 2001
From: Andreas Pretzsch <apr@cn-eng.de>
Date: Tue, 7 Jan 2014 22:34:41 +0100
Subject: ASoC: ssm2602: add 16kHz sampling rate support

SSM260x also supports 16kHz with external master clocks of 12.000MHz,
12.288MHz and 18.432MHz.
Add matching coefficients, update constraints and announced rates.

Signed-off-by: Andreas Pretzsch <apr@cn-eng.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/ssm2602.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index c6dd48561884..af76bbd1b24f 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -194,7 +194,7 @@ static const struct snd_soc_dapm_route ssm2604_routes[] = {
 };
 
 static const unsigned int ssm2602_rates_12288000[] = {
-	8000, 32000, 48000, 96000,
+	8000, 16000, 32000, 48000, 96000,
 };
 
 static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
@@ -231,6 +231,11 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = {
 	{18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)},
 	{12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)},
 
+	/* 16k */
+	{12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)},
+	{18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)},
+	{12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)},
+
 	/* 8k */
 	{12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)},
 	{18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)},
@@ -473,9 +478,10 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
-		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
-		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+		SNDRV_PCM_RATE_96000)
 
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-- 
cgit v1.2.2


From b317b032d2dcb5e518cc9630cc6f1c7c24afedfc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jan 2014 11:44:21 +0100
Subject: ALSA: hda - Split Thinkpad ACPI-related code

Both patch_realtek.c and patch_conexant.c contain the fairy same code
snippet for supporting Thinkpad ACPI LED controls.  Split them into
thinkpad_helper.c and include it from both places.  Although this
isn't the best approach from the code size POV, the probability for
coexistence of both Realtek and Conexant codecs on a single machine is
pretty low, thus it'll end up with less memory footprint than
splitting to yet another module.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c  | 100 ++--------------------------------------
 sound/pci/hda/patch_realtek.c   |  93 ++-----------------------------------
 sound/pci/hda/thinkpad_helper.c |  99 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 105 insertions(+), 187 deletions(-)
 create mode 100644 sound/pci/hda/thinkpad_helper.c

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6df4c35b48c5..4e0ec146553d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3240,102 +3240,8 @@ enum {
 	CXT_FIXUP_THINKPAD_ACPI,
 };
 
-#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
-
-#include <linux/thinkpad_acpi.h>
-#include <linux/acpi.h>
-
-static int (*led_set_func)(int, bool);
-
-static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
-				 void **rv)
-{
-	bool *found = context;
-	*found = true;
-	return AE_OK;
-}
-
-static bool is_thinkpad(struct hda_codec *codec)
-{
-	bool found = false;
-	if (codec->subsystem_id >> 16 != 0x17aa)
-		return false;
-	if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
-		return true;
-	found = false;
-	return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
-}
-
-static void update_tpacpi_mute_led(void *private_data, int enabled)
-{
-	struct hda_codec *codec = private_data;
-	struct conexant_spec *spec = codec->spec;
-
-	if (spec->dynamic_eapd)
-		cx_auto_vmaster_hook(private_data, enabled);
-
-	if (led_set_func)
-		led_set_func(TPACPI_LED_MUTE, !enabled);
-}
-
-static void update_tpacpi_micmute_led(struct hda_codec *codec,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	if (!ucontrol || !led_set_func)
-		return;
-	if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
-		/* TODO: How do I verify if it's a mono or stereo here? */
-		bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1];
-		led_set_func(TPACPI_LED_MICMUTE, !val);
-	}
-}
-
-static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec,
-				  const struct hda_fixup *fix, int action)
-{
-	struct conexant_spec *spec = codec->spec;
-
-	bool removefunc = false;
-
-	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (!is_thinkpad(codec))
-			return;
-		if (!led_set_func)
-			led_set_func = symbol_request(tpacpi_led_set);
-		if (!led_set_func) {
-			snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n");
-			return;
-		}
-
-		removefunc = true;
-		if (led_set_func(TPACPI_LED_MUTE, false) >= 0) {
-			spec->gen.vmaster_mute.hook = update_tpacpi_mute_led;
-			removefunc = false;
-		}
-		if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) {
-			if (spec->gen.num_adc_nids > 1)
-				snd_printdd("Skipping micmute LED control due to several ADCs");
-			else {
-				spec->gen.cap_sync_hook = update_tpacpi_micmute_led;
-				removefunc = false;
-			}
-		}
-	}
-
-	if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
-		symbol_put(tpacpi_led_set);
-		led_set_func = NULL;
-	}
-}
-
-#else
-
-static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec,
-				  const struct hda_fixup *fix, int action)
-{
-}
-
-#endif
+/* for hda_fixup_thinkpad_acpi() */
+#include "thinkpad_helper.c"
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
 				  const struct hda_fixup *fix, int action)
@@ -3492,7 +3398,7 @@ static const struct hda_fixup cxt_fixups[] = {
 	},
 	[CXT_FIXUP_THINKPAD_ACPI] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = cxt_fixup_thinkpad_acpi,
+		.v.func = hda_fixup_thinkpad_acpi,
 	},
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 74b496b0d179..d35cf3dc6711 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3781,95 +3781,8 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec,
 	}
 }
 
-#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
-
-#include <linux/thinkpad_acpi.h>
-#include <linux/acpi.h>
-
-static int (*led_set_func)(int, bool);
-
-static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
-				 void **rv)
-{
-	bool *found = context;
-	*found = true;
-	return AE_OK;
-}
-
-static bool is_thinkpad(struct hda_codec *codec)
-{
-	bool found = false;
-	if (codec->subsystem_id >> 16 != 0x17aa)
-		return false;
-	if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
-		return true;
-	found = false;
-	return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
-}
-
-static void update_tpacpi_mute_led(void *private_data, int enabled)
-{
-	if (led_set_func)
-		led_set_func(TPACPI_LED_MUTE, !enabled);
-}
-
-static void update_tpacpi_micmute_led(struct hda_codec *codec,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	if (!ucontrol || !led_set_func)
-		return;
-	if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
-		/* TODO: How do I verify if it's a mono or stereo here? */
-		bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1];
-		led_set_func(TPACPI_LED_MICMUTE, !val);
-	}
-}
-
-static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
-				  const struct hda_fixup *fix, int action)
-{
-	struct alc_spec *spec = codec->spec;
-	bool removefunc = false;
-
-	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (!is_thinkpad(codec))
-			return;
-		if (!led_set_func)
-			led_set_func = symbol_request(tpacpi_led_set);
-		if (!led_set_func) {
-			snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n");
-			return;
-		}
-
-		removefunc = true;
-		if (led_set_func(TPACPI_LED_MUTE, false) >= 0) {
-			spec->gen.vmaster_mute.hook = update_tpacpi_mute_led;
-			removefunc = false;
-		}
-		if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) {
-			if (spec->gen.num_adc_nids > 1)
-				snd_printdd("Skipping micmute LED control due to several ADCs");
-			else {
-				spec->gen.cap_sync_hook = update_tpacpi_micmute_led;
-				removefunc = false;
-			}
-		}
-	}
-
-	if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
-		symbol_put(tpacpi_led_set);
-		led_set_func = NULL;
-	}
-}
-
-#else
-
-static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
-				  const struct hda_fixup *fix, int action)
-{
-}
-
-#endif
+/* for hda_fixup_thinkpad_acpi() */
+#include "thinkpad_helper.c"
 
 enum {
 	ALC269_FIXUP_SONY_VAIO,
@@ -4273,7 +4186,7 @@ static const struct hda_fixup alc269_fixups[] = {
 	},
 	[ALC269_FIXUP_THINKPAD_ACPI] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc_fixup_thinkpad_acpi,
+		.v.func = hda_fixup_thinkpad_acpi,
 	},
 	[ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
new file mode 100644
index 000000000000..8492b8a3ce47
--- /dev/null
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -0,0 +1,99 @@
+/* Helper functions for Thinkpad LED control;
+ * to be included from codec driver
+ */
+
+#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
+
+#include <linux/acpi.h>
+#include <linux/thinkpad_acpi.h>
+
+static int (*led_set_func)(int, bool);
+
+static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
+				 void **rv)
+{
+	bool *found = context;
+	*found = true;
+	return AE_OK;
+}
+
+static bool is_thinkpad(struct hda_codec *codec)
+{
+	bool found = false;
+	if (codec->subsystem_id >> 16 != 0x17aa)
+		return false;
+	if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
+		return true;
+	found = false;
+	return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
+}
+
+static void update_tpacpi_mute_led(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct hda_gen_spec *spec = codec->spec;
+
+	if (spec->vmaster_mute.hook)
+		spec->vmaster_mute.hook(private_data, enabled);
+
+	if (led_set_func)
+		led_set_func(TPACPI_LED_MUTE, !enabled);
+}
+
+static void update_tpacpi_micmute_led(struct hda_codec *codec,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	if (!ucontrol || !led_set_func)
+		return;
+	if (strcmp("Capture Switch", ucontrol->id.name) == 0 && ucontrol->id.index == 0) {
+		/* TODO: How do I verify if it's a mono or stereo here? */
+		bool val = ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1];
+		led_set_func(TPACPI_LED_MICMUTE, !val);
+	}
+}
+
+static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	bool removefunc = false;
+
+	if (action == HDA_FIXUP_ACT_PROBE) {
+		if (!is_thinkpad(codec))
+			return;
+		if (!led_set_func)
+			led_set_func = symbol_request(tpacpi_led_set);
+		if (!led_set_func) {
+			snd_printk(KERN_WARNING "Failed to find thinkpad-acpi symbol tpacpi_led_set\n");
+			return;
+		}
+
+		removefunc = true;
+		if (led_set_func(TPACPI_LED_MUTE, false) >= 0) {
+			spec->vmaster_mute.hook = update_tpacpi_mute_led;
+			removefunc = false;
+		}
+		if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) {
+			if (spec->num_adc_nids > 1)
+				snd_printdd("Skipping micmute LED control due to several ADCs");
+			else {
+				spec->cap_sync_hook = update_tpacpi_micmute_led;
+				removefunc = false;
+			}
+		}
+	}
+
+	if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
+		symbol_put(tpacpi_led_set);
+		led_set_func = NULL;
+	}
+}
+
+#else /* CONFIG_THINKPAD_ACPI */
+
+static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+}
+
+#endif /* CONFIG_THINKPAD_ACPI */
-- 
cgit v1.2.2


From 2841be9afa6c9d37d41386af30cd8813acd06739 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:28 +0100
Subject: ASoC: fsl-ssi: Fix probe error handling

This patch fixes the error handling in the fsl-ssi probe function.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 16 ++++++++++++----
 1 file changed, 12 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 816ae4b28a53..19891f2a5de4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -141,6 +141,7 @@ struct fsl_ssi_private {
 	bool imx_ac97;
 	bool use_dma;
 	bool baudclk_locked;
+	bool irq_stats;
 	u8 i2s_mode;
 	spinlock_t baudclk_lock;
 	struct clk *baudclk;
@@ -1224,6 +1225,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		ret = devm_request_irq(&pdev->dev, ssi_private->irq,
 					fsl_ssi_isr, 0, ssi_private->name,
 					ssi_private);
+		ssi_private->irq_stats = true;
 		if (ret < 0) {
 			dev_err(&pdev->dev, "could not claim irq %u\n",
 					ssi_private->irq);
@@ -1274,11 +1276,11 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 			ret = imx_pcm_fiq_init(pdev, &ssi_private->fiq_params);
 			if (ret)
-				goto error_dev;
+				goto error_pcm;
 		} else {
 			ret = imx_pcm_dma_init(pdev);
 			if (ret)
-				goto error_dev;
+				goto error_pcm;
 		}
 	}
 
@@ -1320,6 +1322,10 @@ done:
 	return 0;
 
 error_dai:
+	if (ssi_private->ssi_on_imx && !ssi_private->use_dma)
+		imx_pcm_fiq_exit(pdev);
+
+error_pcm:
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
@@ -1333,7 +1339,8 @@ error_clk:
 	}
 
 error_irqmap:
-	irq_dispose_mapping(ssi_private->irq);
+	if (ssi_private->irq_stats)
+		irq_dispose_mapping(ssi_private->irq);
 
 	return ret;
 }
@@ -1351,7 +1358,8 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 			clk_disable_unprepare(ssi_private->baudclk);
 		clk_disable_unprepare(ssi_private->clk);
 	}
-	irq_dispose_mapping(ssi_private->irq);
+	if (ssi_private->irq_stats)
+		irq_dispose_mapping(ssi_private->irq);
 
 	return 0;
 }
-- 
cgit v1.2.2


From 9368acc4383bd8cca671fdc49c5f7e241b6909b3 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:29 +0100
Subject: ASoC: fsl-ssi: Move sysfs stats to debugfs

Interrupt statistics of fsl_ssi are mainly for debugging purpose. Most
of those interrupts show error states, e.g. under/overflow.
The stats should be exposed via debugfs instead of sysfs.

This patch moves the statistics file to debugfs.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 184 ++++++++++++++++++++++++++++++------------------
 1 file changed, 117 insertions(+), 67 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 19891f2a5de4..e483e9d84f8b 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -35,6 +35,7 @@
 #include <linux/module.h>
 #include <linux/interrupt.h>
 #include <linux/clk.h>
+#include <linux/debugfs.h>
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
@@ -114,6 +115,14 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
 		    CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
 		    CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
 
+#define FSLSSI_SIER_DBG_RX_FLAGS (CCSR_SSI_SIER_RFF0_EN | \
+		CCSR_SSI_SIER_RLS_EN | CCSR_SSI_SIER_RFS_EN | \
+		CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_RFRC_EN)
+#define FSLSSI_SIER_DBG_TX_FLAGS (CCSR_SSI_SIER_TFE0_EN | \
+		CCSR_SSI_SIER_TLS_EN | CCSR_SSI_SIER_TFS_EN | \
+		CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN)
+#define FSLSSI_SISR_MASK (FSLSSI_SIER_DBG_RX_FLAGS | FSLSSI_SIER_DBG_TX_FLAGS)
+
 /**
  * fsl_ssi_private: per-SSI private data
  *
@@ -133,7 +142,6 @@ struct fsl_ssi_private {
 	unsigned int irq;
 	unsigned int fifo_depth;
 	struct snd_soc_dai_driver cpu_dai_drv;
-	struct device_attribute dev_attr;
 	struct platform_device *pdev;
 
 	bool new_binding;
@@ -175,6 +183,8 @@ struct fsl_ssi_private {
 		unsigned int tfe1;
 		unsigned int tfe0;
 	} stats;
+	struct dentry *dbg_dir;
+	struct dentry *dbg_stats;
 
 	char name[1];
 };
@@ -203,7 +213,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	   were interrupted for.  We mask it with the Interrupt Enable register
 	   so that we only check for events that we're interested in.
 	 */
-	sisr = read_ssi(&ssi->sisr) & SIER_FLAGS;
+	sisr = read_ssi(&ssi->sisr) & FSLSSI_SISR_MASK;
 
 	if (sisr & CCSR_SSI_SISR_RFRC) {
 		ssi_private->stats.rfrc++;
@@ -323,6 +333,102 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	return ret;
 }
 
+#if IS_ENABLED(CONFIG_DEBUG_FS)
+/* Show the statistics of a flag only if its interrupt is enabled.  The
+ * compiler will optimze this code to a no-op if the interrupt is not
+ * enabled.
+ */
+#define SIER_SHOW(flag, name) \
+	do { \
+		if (FSLSSI_SISR_MASK & CCSR_SSI_SIER_##flag) \
+			seq_printf(s, #name "=%u\n", ssi_private->stats.name); \
+	} while (0)
+
+
+/**
+ * fsl_sysfs_ssi_show: display SSI statistics
+ *
+ * Display the statistics for the current SSI device.  To avoid confusion,
+ * we only show those counts that are enabled.
+ */
+static ssize_t fsl_ssi_stats_show(struct seq_file *s, void *unused)
+{
+	struct fsl_ssi_private *ssi_private = s->private;
+
+	SIER_SHOW(RFRC_EN, rfrc);
+	SIER_SHOW(TFRC_EN, tfrc);
+	SIER_SHOW(CMDAU_EN, cmdau);
+	SIER_SHOW(CMDDU_EN, cmddu);
+	SIER_SHOW(RXT_EN, rxt);
+	SIER_SHOW(RDR1_EN, rdr1);
+	SIER_SHOW(RDR0_EN, rdr0);
+	SIER_SHOW(TDE1_EN, tde1);
+	SIER_SHOW(TDE0_EN, tde0);
+	SIER_SHOW(ROE1_EN, roe1);
+	SIER_SHOW(ROE0_EN, roe0);
+	SIER_SHOW(TUE1_EN, tue1);
+	SIER_SHOW(TUE0_EN, tue0);
+	SIER_SHOW(TFS_EN, tfs);
+	SIER_SHOW(RFS_EN, rfs);
+	SIER_SHOW(TLS_EN, tls);
+	SIER_SHOW(RLS_EN, rls);
+	SIER_SHOW(RFF1_EN, rff1);
+	SIER_SHOW(RFF0_EN, rff0);
+	SIER_SHOW(TFE1_EN, tfe1);
+	SIER_SHOW(TFE0_EN, tfe0);
+
+	return 0;
+}
+
+static int fsl_ssi_stats_open(struct inode *inode, struct file *file)
+{
+	return single_open(file, fsl_ssi_stats_show, inode->i_private);
+}
+
+static const struct file_operations fsl_ssi_stats_ops = {
+	.open = fsl_ssi_stats_open,
+	.read = seq_read,
+	.llseek = seq_lseek,
+	.release = single_release,
+};
+
+static int fsl_ssi_debugfs_create(struct fsl_ssi_private *ssi_private,
+		struct device *dev)
+{
+	ssi_private->dbg_dir = debugfs_create_dir(dev_name(dev), NULL);
+	if (!ssi_private->dbg_dir)
+		return -ENOMEM;
+
+	ssi_private->dbg_stats = debugfs_create_file("stats", S_IRUGO,
+			ssi_private->dbg_dir, ssi_private, &fsl_ssi_stats_ops);
+	if (!ssi_private->dbg_stats) {
+		debugfs_remove(ssi_private->dbg_dir);
+		return -ENOMEM;
+	}
+
+	return 0;
+}
+
+static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private)
+{
+	debugfs_remove(ssi_private->dbg_stats);
+	debugfs_remove(ssi_private->dbg_dir);
+}
+
+#else
+
+static int fsl_ssi_debugfs_create(struct fsl_ssi_private *ssi_private,
+		struct device *dev)
+{
+	return 0;
+}
+
+static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private)
+{
+}
+
+#endif /* IS_ENABLED(CONFIG_DEBUG_FS) */
+
 static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
 {
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
@@ -991,56 +1097,6 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
 	.write		= fsl_ssi_ac97_write,
 };
 
-/* Show the statistics of a flag only if its interrupt is enabled.  The
- * compiler will optimze this code to a no-op if the interrupt is not
- * enabled.
- */
-#define SIER_SHOW(flag, name) \
-	do { \
-		if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \
-			length += sprintf(buf + length, #name "=%u\n", \
-				ssi_private->stats.name); \
-	} while (0)
-
-
-/**
- * fsl_sysfs_ssi_show: display SSI statistics
- *
- * Display the statistics for the current SSI device.  To avoid confusion,
- * we only show those counts that are enabled.
- */
-static ssize_t fsl_sysfs_ssi_show(struct device *dev,
-	struct device_attribute *attr, char *buf)
-{
-	struct fsl_ssi_private *ssi_private =
-		container_of(attr, struct fsl_ssi_private, dev_attr);
-	ssize_t length = 0;
-
-	SIER_SHOW(RFRC_EN, rfrc);
-	SIER_SHOW(TFRC_EN, tfrc);
-	SIER_SHOW(CMDAU_EN, cmdau);
-	SIER_SHOW(CMDDU_EN, cmddu);
-	SIER_SHOW(RXT_EN, rxt);
-	SIER_SHOW(RDR1_EN, rdr1);
-	SIER_SHOW(RDR0_EN, rdr0);
-	SIER_SHOW(TDE1_EN, tde1);
-	SIER_SHOW(TDE0_EN, tde0);
-	SIER_SHOW(ROE1_EN, roe1);
-	SIER_SHOW(ROE0_EN, roe0);
-	SIER_SHOW(TUE1_EN, tue1);
-	SIER_SHOW(TUE0_EN, tue0);
-	SIER_SHOW(TFS_EN, tfs);
-	SIER_SHOW(RFS_EN, rfs);
-	SIER_SHOW(TLS_EN, tls);
-	SIER_SHOW(RLS_EN, rls);
-	SIER_SHOW(RFF1_EN, rff1);
-	SIER_SHOW(RFF0_EN, rff0);
-	SIER_SHOW(TFE1_EN, tfe1);
-	SIER_SHOW(TFE0_EN, tfe0);
-
-	return length;
-}
-
 /**
  * Make every character in a string lower-case
  */
@@ -1233,20 +1289,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		}
 	}
 
-	/* Initialize the the device_attribute structure */
-	dev_attr = &ssi_private->dev_attr;
-	sysfs_attr_init(&dev_attr->attr);
-	dev_attr->attr.name = "statistics";
-	dev_attr->attr.mode = S_IRUGO;
-	dev_attr->show = fsl_sysfs_ssi_show;
-
-	ret = device_create_file(&pdev->dev, dev_attr);
-	if (ret) {
-		dev_err(&pdev->dev, "could not create sysfs %s file\n",
-			ssi_private->dev_attr.attr.name);
-		goto error_clk;
-	}
-
 	/* Register with ASoC */
 	dev_set_drvdata(&pdev->dev, ssi_private);
 
@@ -1257,6 +1299,10 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		goto error_dev;
 	}
 
+	ret = fsl_ssi_debugfs_create(ssi_private, &pdev->dev);
+	if (ret)
+		goto error_dbgfs;
+
 	if (ssi_private->ssi_on_imx) {
 		if (!ssi_private->use_dma) {
 
@@ -1326,6 +1372,9 @@ error_dai:
 		imx_pcm_fiq_exit(pdev);
 
 error_pcm:
+	fsl_ssi_debugfs_remove(ssi_private);
+
+error_dbgfs:
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
@@ -1349,10 +1398,11 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 {
 	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
 
+	fsl_ssi_debugfs_remove(ssi_private);
+
 	if (!ssi_private->new_binding)
 		platform_device_unregister(ssi_private->pdev);
 	snd_soc_unregister_component(&pdev->dev);
-	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 	if (ssi_private->ssi_on_imx) {
 		if (!IS_ERR(ssi_private->baudclk))
 			clk_disable_unprepare(ssi_private->baudclk);
-- 
cgit v1.2.2


From c1953bfe1329eeb16991d430d574c4280697ad17 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:30 +0100
Subject: ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching

There is a small but significant difference between imx21-ssi and
imx51-ssi and above. imx21-ssi does not allow online reconfiguration of
some registers. They have to be configured at the beginning.

imx51-ssi has to reconfigure the SSI unit while it is running. Otherwise
it would not be possible to have two streams in parallel. The new SDMA
unit in imx51 and above has to be configured before the first DMA
request arrives. Therefor we need to setup TDMAE/RDMAE just before
starting the stream.

This patch introduces distinct imx51-ssi as compatible and adds
of_device_id matching in the probe function.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 32 ++++++++++++++++++++++++--------
 1 file changed, 24 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index e483e9d84f8b..671be33aa9d2 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -123,6 +123,13 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
 		CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TFRC_EN)
 #define FSLSSI_SISR_MASK (FSLSSI_SIER_DBG_RX_FLAGS | FSLSSI_SIER_DBG_TX_FLAGS)
 
+
+enum fsl_ssi_type {
+	FSL_SSI_MCP8610,
+	FSL_SSI_MX21,
+	FSL_SSI_MX51,
+};
+
 /**
  * fsl_ssi_private: per-SSI private data
  *
@@ -189,6 +196,14 @@ struct fsl_ssi_private {
 	char name[1];
 };
 
+static const struct of_device_id fsl_ssi_ids[] = {
+	{ .compatible = "fsl,mpc8610-ssi", .data = (void *) FSL_SSI_MCP8610},
+	{ .compatible = "fsl,imx51-ssi", .data = (void *) FSL_SSI_MX51},
+	{ .compatible = "fsl,imx21-ssi", .data = (void *) FSL_SSI_MX21},
+	{}
+};
+MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
+
 /**
  * fsl_ssi_isr: SSI interrupt handler
  *
@@ -1118,6 +1133,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	int ret = 0;
 	struct device_attribute *dev_attr = NULL;
 	struct device_node *np = pdev->dev.of_node;
+	const struct of_device_id *of_id;
+	enum fsl_ssi_type hw_type;
 	const char *p, *sprop;
 	const uint32_t *iprop;
 	struct resource res;
@@ -1132,6 +1149,11 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	if (!of_device_is_available(np))
 		return -ENODEV;
 
+	of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
+	if (!of_id)
+		return -EINVAL;
+	hw_type = (enum fsl_ssi_type) of_id->data;
+
 	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (!sprop) {
@@ -1211,7 +1233,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	ssi_private->baudclk_locked = false;
 	spin_lock_init(&ssi_private->baudclk_lock);
 
-	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) {
+	if (hw_type == FSL_SSI_MX21 || hw_type == FSL_SSI_MX51 ||
+			hw_type == FSL_SSI_MX35) {
 		u32 dma_events[2];
 		ssi_private->ssi_on_imx = true;
 
@@ -1414,13 +1437,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 	return 0;
 }
 
-static const struct of_device_id fsl_ssi_ids[] = {
-	{ .compatible = "fsl,mpc8610-ssi", },
-	{ .compatible = "fsl,imx21-ssi", },
-	{}
-};
-MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
-
 static struct platform_driver fsl_ssi_driver = {
 	.driver = {
 		.name = "fsl-ssi-dai",
-- 
cgit v1.2.2


From 0888efd166fa99b733b0b68e70d2fb3c3c7684ec Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:31 +0100
Subject: ASoC: fsl-ssi: Fix interrupt stats for imx

irqs should only be requested/released with enabled DMA. Previously
interrupt statistics where disabled for IMX Processors because of
different writeable SISR bits.

This patch introduces support for irqstats on imx processors again by
creating a sisr writeback mask and introducing a imx35-ssi compatible.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 40 ++++++++++++++++++++++++++++++++--------
 1 file changed, 32 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 671be33aa9d2..bc904696d820 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -127,6 +127,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
 enum fsl_ssi_type {
 	FSL_SSI_MCP8610,
 	FSL_SSI_MX21,
+	FSL_SSI_MX35,
 	FSL_SSI_MX51,
 };
 
@@ -151,6 +152,7 @@ struct fsl_ssi_private {
 	struct snd_soc_dai_driver cpu_dai_drv;
 	struct platform_device *pdev;
 
+	enum fsl_ssi_type hw_type;
 	bool new_binding;
 	bool ssi_on_imx;
 	bool imx_ac97;
@@ -199,6 +201,7 @@ struct fsl_ssi_private {
 static const struct of_device_id fsl_ssi_ids[] = {
 	{ .compatible = "fsl,mpc8610-ssi", .data = (void *) FSL_SSI_MCP8610},
 	{ .compatible = "fsl,imx51-ssi", .data = (void *) FSL_SSI_MX51},
+	{ .compatible = "fsl,imx35-ssi", .data = (void *) FSL_SSI_MX35},
 	{ .compatible = "fsl,imx21-ssi", .data = (void *) FSL_SSI_MX21},
 	{}
 };
@@ -222,7 +225,26 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
 	irqreturn_t ret = IRQ_NONE;
 	__be32 sisr;
-	__be32 sisr2 = 0;
+	__be32 sisr2;
+	__be32 sisr_write_mask = 0;
+
+	switch (ssi_private->hw_type) {
+	case FSL_SSI_MX21:
+		sisr_write_mask = 0;
+		break;
+
+	case FSL_SSI_MCP8610:
+	case FSL_SSI_MX35:
+		sisr_write_mask = CCSR_SSI_SISR_RFRC | CCSR_SSI_SISR_TFRC |
+			CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 |
+			CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1;
+		break;
+
+	case FSL_SSI_MX51:
+		sisr_write_mask = CCSR_SSI_SISR_ROE0 | CCSR_SSI_SISR_ROE1 |
+			CCSR_SSI_SISR_TUE0 | CCSR_SSI_SISR_TUE1;
+		break;
+	}
 
 	/* We got an interrupt, so read the status register to see what we
 	   were interrupted for.  We mask it with the Interrupt Enable register
@@ -232,13 +254,11 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 
 	if (sisr & CCSR_SSI_SISR_RFRC) {
 		ssi_private->stats.rfrc++;
-		sisr2 |= CCSR_SSI_SISR_RFRC;
 		ret = IRQ_HANDLED;
 	}
 
 	if (sisr & CCSR_SSI_SISR_TFRC) {
 		ssi_private->stats.tfrc++;
-		sisr2 |= CCSR_SSI_SISR_TFRC;
 		ret = IRQ_HANDLED;
 	}
 
@@ -279,25 +299,21 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 
 	if (sisr & CCSR_SSI_SISR_ROE1) {
 		ssi_private->stats.roe1++;
-		sisr2 |= CCSR_SSI_SISR_ROE1;
 		ret = IRQ_HANDLED;
 	}
 
 	if (sisr & CCSR_SSI_SISR_ROE0) {
 		ssi_private->stats.roe0++;
-		sisr2 |= CCSR_SSI_SISR_ROE0;
 		ret = IRQ_HANDLED;
 	}
 
 	if (sisr & CCSR_SSI_SISR_TUE1) {
 		ssi_private->stats.tue1++;
-		sisr2 |= CCSR_SSI_SISR_TUE1;
 		ret = IRQ_HANDLED;
 	}
 
 	if (sisr & CCSR_SSI_SISR_TUE0) {
 		ssi_private->stats.tue0++;
-		sisr2 |= CCSR_SSI_SISR_TUE0;
 		ret = IRQ_HANDLED;
 	}
 
@@ -341,6 +357,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 		ret = IRQ_HANDLED;
 	}
 
+	sisr2 = sisr & sisr_write_mask;
 	/* Clear the bits that we set */
 	if (sisr2)
 		write_ssi(sisr2, &ssi->sisr);
@@ -1180,6 +1197,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 
 	ssi_private->use_dma = !of_property_read_bool(np,
 			"fsl,fiq-stream-filter");
+	ssi_private->hw_type = hw_type;
 
 	if (ac97) {
 		memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_ac97_dai,
@@ -1299,7 +1317,13 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 			dma_events[0], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
 		imx_pcm_dma_params_init_data(&ssi_private->filter_data_rx,
 			dma_events[1], shared ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI);
-	} else if (ssi_private->use_dma) {
+	}
+
+	/*
+	 * Enable interrupts only for MCP8610 and MX51. The other MXs have
+	 * different writeable interrupt status registers.
+	 */
+	if (ssi_private->use_dma) {
 		/* The 'name' should not have any slashes in it. */
 		ret = devm_request_irq(&pdev->dev, ssi_private->irq,
 					fsl_ssi_isr, 0, ssi_private->name,
-- 
cgit v1.2.2


From bd3ca7d1b8ee0dcd502c8c15d1cf741bc165722f Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:32 +0100
Subject: ASoC: fsl-ssi: Add offline_config flag

imx50-ssi and later versions of this IP support online reconfiguration
of all registers. The reference manual does not list any registers that
can only be configured while the SSI unit is disabled. This patch
introduces the flag for later use.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 27 +++++++++++++++++++++++++++
 1 file changed, 27 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index bc904696d820..d0b9fe31f49a 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -159,6 +159,7 @@ struct fsl_ssi_private {
 	bool use_dma;
 	bool baudclk_locked;
 	bool irq_stats;
+	bool offline_config;
 	u8 i2s_mode;
 	spinlock_t baudclk_lock;
 	struct clk *baudclk;
@@ -1251,6 +1252,32 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	ssi_private->baudclk_locked = false;
 	spin_lock_init(&ssi_private->baudclk_lock);
 
+	/*
+	 * imx51 and later SoCs have a slightly different IP that allows the
+	 * SSI configuration while the SSI unit is running.
+	 *
+	 * More important, it is necessary on those SoCs to configure the
+	 * sperate TX/RX DMA bits just before starting the stream
+	 * (fsl_ssi_trigger). The SDMA unit has to be configured before fsl_ssi
+	 * sends any DMA requests to the SDMA unit, otherwise it is not defined
+	 * how the SDMA unit handles the DMA request.
+	 *
+	 * SDMA units are present on devices starting at imx35 but the imx35
+	 * reference manual states that the DMA bits should not be changed
+	 * while the SSI unit is running (SSIEN). So we support the necessary
+	 * online configuration of fsl-ssi starting at imx51.
+	 */
+	switch (hw_type) {
+	case FSL_SSI_MCP8610:
+	case FSL_SSI_MX21:
+	case FSL_SSI_MX35:
+		ssi_private->offline_config = true;
+		break;
+	case FSL_SSI_MX51:
+		ssi_private->offline_config = false;
+		break;
+	}
+
 	if (hw_type == FSL_SSI_MX21 || hw_type == FSL_SSI_MX51 ||
 			hw_type == FSL_SSI_MX35) {
 		u32 dma_events[2];
-- 
cgit v1.2.2


From 4e6ec0d98c045cb2c0c6550c65c4afae208872e9 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:33 +0100
Subject: ASoC: fsl-ssi: Add configuration helper functions

This patch adds a struct 'fsl_ssi_rxtx_reg_val' which holds register
values necessary to enable rx/tx. Based on those preset register values,
the added configuration functions will cleanly enable/disable different
parts of the SSI IP while supporting online/offline configuration.
Different operating modes can be setup directly as different register
values in fsl_ssi_reg_val.

These functions and structs will help to cleanup and simplify the
trigger function to support many different IP versions (online/offline
configuration) and different operating modes.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 122 ++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 122 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d0b9fe31f49a..a85268bb4507 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -131,6 +131,18 @@ enum fsl_ssi_type {
 	FSL_SSI_MX51,
 };
 
+struct fsl_ssi_reg_val {
+	u32 sier;
+	u32 srcr;
+	u32 stcr;
+	u32 scr;
+};
+
+struct fsl_ssi_rxtx_reg_val {
+	struct fsl_ssi_reg_val rx;
+	struct fsl_ssi_reg_val tx;
+};
+
 /**
  * fsl_ssi_private: per-SSI private data
  *
@@ -169,6 +181,8 @@ struct fsl_ssi_private {
 	struct imx_dma_data filter_data_tx;
 	struct imx_dma_data filter_data_rx;
 	struct imx_pcm_fiq_params fiq_params;
+	/* Register values for rx/tx configuration */
+	struct fsl_ssi_rxtx_reg_val rxtx_reg_val;
 
 	struct {
 		unsigned int rfrc;
@@ -462,6 +476,114 @@ static void fsl_ssi_debugfs_remove(struct fsl_ssi_private *ssi_private)
 
 #endif /* IS_ENABLED(CONFIG_DEBUG_FS) */
 
+/*
+ * Enable/Disable all rx/tx config flags at once.
+ */
+static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
+		bool enable)
+{
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	struct fsl_ssi_rxtx_reg_val *vals = &ssi_private->rxtx_reg_val;
+
+	if (enable) {
+		write_ssi_mask(&ssi->sier, 0, vals->rx.sier | vals->tx.sier);
+		write_ssi_mask(&ssi->srcr, 0, vals->rx.srcr | vals->tx.srcr);
+		write_ssi_mask(&ssi->stcr, 0, vals->rx.stcr | vals->tx.stcr);
+	} else {
+		write_ssi_mask(&ssi->srcr, vals->rx.srcr | vals->tx.srcr, 0);
+		write_ssi_mask(&ssi->stcr, vals->rx.stcr | vals->tx.stcr, 0);
+		write_ssi_mask(&ssi->sier, vals->rx.sier | vals->tx.sier, 0);
+	}
+}
+
+/*
+ * Enable/Disable a ssi configuration. You have to pass either
+ * ssi_private->rxtx_reg_val.rx or tx as vals parameter.
+ */
+static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
+		struct fsl_ssi_reg_val *vals)
+{
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	struct fsl_ssi_reg_val *avals;
+	u32 scr_val = read_ssi(&ssi->scr);
+	int nr_active_streams = !!(scr_val & CCSR_SSI_SCR_TE) +
+				!!(scr_val & CCSR_SSI_SCR_RE);
+
+	/* Find the other direction values rx or tx which we do not want to
+	 * modify */
+	if (&ssi_private->rxtx_reg_val.rx == vals)
+		avals = &ssi_private->rxtx_reg_val.tx;
+	else
+		avals = &ssi_private->rxtx_reg_val.rx;
+
+	/* If vals should be disabled, start with disabling the unit */
+	if (!enable) {
+		u32 scr = vals->scr & (vals->scr ^ avals->scr);
+		write_ssi_mask(&ssi->scr, scr, 0);
+	}
+
+	/*
+	 * We are running on a SoC which does not support online SSI
+	 * reconfiguration, so we have to enable all necessary flags at once
+	 * even if we do not use them later (capture and playback configuration)
+	 */
+	if (ssi_private->offline_config) {
+		if ((enable && !nr_active_streams) ||
+				(!enable && nr_active_streams == 1))
+			fsl_ssi_rxtx_config(ssi_private, enable);
+
+		goto config_done;
+	}
+
+	/*
+	 * Configure single direction units while the SSI unit is running
+	 * (online configuration)
+	 */
+	if (enable) {
+		write_ssi_mask(&ssi->sier, 0, vals->sier);
+		write_ssi_mask(&ssi->srcr, 0, vals->srcr);
+		write_ssi_mask(&ssi->stcr, 0, vals->stcr);
+	} else {
+		u32 sier;
+		u32 srcr;
+		u32 stcr;
+
+		/*
+		 * Disabling the necessary flags for one of rx/tx while the
+		 * other stream is active is a little bit more difficult. We
+		 * have to disable only those flags that differ between both
+		 * streams (rx XOR tx) and that are set in the stream that is
+		 * disabled now. Otherwise we could alter flags of the other
+		 * stream
+		 */
+
+		/* These assignments are simply vals without bits set in avals*/
+		sier = vals->sier & (vals->sier ^ avals->sier);
+		srcr = vals->srcr & (vals->srcr ^ avals->srcr);
+		stcr = vals->stcr & (vals->stcr ^ avals->stcr);
+
+		write_ssi_mask(&ssi->srcr, srcr, 0);
+		write_ssi_mask(&ssi->stcr, stcr, 0);
+		write_ssi_mask(&ssi->sier, sier, 0);
+	}
+
+config_done:
+	/* Enabling of subunits is done after configuration */
+	if (enable)
+		write_ssi_mask(&ssi->scr, 0, vals->scr);
+}
+
+
+static void fsl_ssi_rx_config(struct fsl_ssi_private *ssi_private, bool enable)
+{
+	fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.rx);
+}
+
+static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable)
+{
+	fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx);
+}
+
 static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
 {
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-- 
cgit v1.2.2


From 6de8387905a69568489284b4660737eebb0db8cf Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:34 +0100
Subject: ASoC: fsl-ssi: Move RX/TX configuration to seperate functions

This patch defines the appropriate register values for different
oparation modes and IP versions. We have to handle DMA/FIQ, AC97,
DEBUG-IRQs and offline/online configuration support.

With this patch we cleanup some driver code that was not reference
manual conform and try to cleanup the whole trigger function to seperate
the actual register values from the enable/disable logic, which is now
hidden in fsl_ssi_config helpers.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 89 +++++++++++++++++++++++++------------------------
 1 file changed, 46 insertions(+), 43 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a85268bb4507..a96ab4e60652 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -108,13 +108,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
 #endif
 
-/* SIER bitflag of interrupts to enable */
-#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \
-		    CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \
-		    CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \
-		    CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \
-		    CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN)
-
 #define FSLSSI_SIER_DBG_RX_FLAGS (CCSR_SSI_SIER_RFF0_EN | \
 		CCSR_SSI_SIER_RLS_EN | CCSR_SSI_SIER_RFS_EN | \
 		CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_RFRC_EN)
@@ -584,6 +577,41 @@ static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable)
 	fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx);
 }
 
+/*
+ * Setup rx/tx register values used to enable/disable the streams. These will
+ * be used later in fsl_ssi_config to setup the streams without the need to
+ * check for all different SSI modes.
+ */
+static void fsl_ssi_setup_reg_vals(struct fsl_ssi_private *ssi_private)
+{
+	struct fsl_ssi_rxtx_reg_val *reg = &ssi_private->rxtx_reg_val;
+
+	reg->rx.sier = CCSR_SSI_SIER_RFF0_EN;
+	reg->rx.srcr = CCSR_SSI_SRCR_RFEN0;
+	reg->rx.scr = 0;
+	reg->tx.sier = CCSR_SSI_SIER_TFE0_EN;
+	reg->tx.stcr = CCSR_SSI_STCR_TFEN0;
+	reg->tx.scr = 0;
+
+	if (!ssi_private->imx_ac97) {
+		reg->rx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE;
+		reg->rx.sier |= CCSR_SSI_SIER_RFF0_EN;
+		reg->tx.scr = CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE;
+		reg->tx.sier |= CCSR_SSI_SIER_TFE0_EN;
+	}
+
+	if (ssi_private->use_dma) {
+		reg->rx.sier |= CCSR_SSI_SIER_RDMAE;
+		reg->tx.sier |= CCSR_SSI_SIER_TDMAE;
+	} else {
+		reg->rx.sier |= CCSR_SSI_SIER_RIE;
+		reg->tx.sier |= CCSR_SSI_SIER_TIE;
+	}
+
+	reg->rx.sier |= FSLSSI_SIER_DBG_RX_FLAGS;
+	reg->tx.sier |= FSLSSI_SIER_DBG_TX_FLAGS;
+}
+
 static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private)
 {
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
@@ -620,6 +648,8 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 	u8 wm;
 	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
 
+	fsl_ssi_setup_reg_vals(ssi_private);
+
 	if (ssi_private->imx_ac97)
 		ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET;
 	else
@@ -643,13 +673,12 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 		ssi_private->i2s_mode |
 		(synchronous ? CCSR_SSI_SCR_SYN : 0));
 
-	write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
-		 CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
-		 CCSR_SSI_STCR_TSCKP, &ssi->stcr);
+	write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFSI |
+			CCSR_SSI_STCR_TEFS | CCSR_SSI_STCR_TSCKP, &ssi->stcr);
+
+	write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFSI |
+			CCSR_SSI_SRCR_REFS | CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
 
-	write_ssi(CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
-		 CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
-		 CCSR_SSI_SRCR_RSCKP, &ssi->srcr);
 	/*
 	 * The DC and PM bits are only used if the SSI is the clock master.
 	 */
@@ -1023,51 +1052,26 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
-	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-	unsigned int sier_bits;
 	unsigned long flags;
 
-	/*
-	 *  Enable only the interrupts and DMA requests
-	 *  that are needed for the channel. As the fiq
-	 *  is polling for this bits, we have to ensure
-	 *  that this are aligned with the preallocated
-	 *  buffers
-	 */
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (ssi_private->use_dma)
-			sier_bits = SIER_FLAGS;
-		else
-			sier_bits = CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TFE0_EN;
-	} else {
-		if (ssi_private->use_dma)
-			sier_bits = SIER_FLAGS;
-		else
-			sier_bits = CCSR_SSI_SIER_RIE | CCSR_SSI_SIER_RFF0_EN;
-	}
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			write_ssi_mask(&ssi->scr, 0,
-				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
+			fsl_ssi_tx_config(ssi_private, true);
 		else
-			write_ssi_mask(&ssi->scr, 0,
-				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
+			fsl_ssi_rx_config(ssi_private, true);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0);
+			fsl_ssi_tx_config(ssi_private, false);
 		else
-			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0);
+			fsl_ssi_rx_config(ssi_private, false);
 
 		if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) &
 					(CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) {
-			write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0);
 			spin_lock_irqsave(&ssi_private->baudclk_lock, flags);
 			ssi_private->baudclk_locked = false;
 			spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags);
@@ -1078,7 +1082,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 		return -EINVAL;
 	}
 
-	write_ssi(sier_bits, &ssi->sier);
 
 	return 0;
 }
-- 
cgit v1.2.2


From a5a7ee7c98bc2a7d0324de661778783ab2c29343 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Fri, 20 Dec 2013 14:11:35 +0100
Subject: ASoC: fsl-ssi: Drop ac97 specific trigger function

The normal trigger function can now be used for AC97. Drop AC97 trigger
function.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 61 +++++++------------------------------------------
 1 file changed, 8 insertions(+), 53 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a96ab4e60652..94dedcb0868d 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1052,6 +1052,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
 	unsigned long flags;
 
 	switch (cmd) {
@@ -1082,6 +1083,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 		return -EINVAL;
 	}
 
+	if (ssi_private->imx_ac97) {
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor);
+		else
+			write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor);
+	}
 
 	return 0;
 }
@@ -1129,58 +1136,6 @@ static const struct snd_soc_component_driver fsl_ssi_component = {
 	.name		= "fsl-ssi",
 };
 
-/**
- * fsl_ssi_ac97_trigger: start and stop the AC97 receive/transmit.
- *
- * This function is called by ALSA to start, stop, pause, and resume the
- * transfer of data.
- */
-static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(
-			rtd->cpu_dai);
-	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TIE |
-					CCSR_SSI_SIER_TFE0_EN);
-		else
-			write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RIE |
-					CCSR_SSI_SIER_RFF0_EN);
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TIE |
-					CCSR_SSI_SIER_TFE0_EN, 0);
-		else
-			write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RIE |
-					CCSR_SSI_SIER_RFF0_EN, 0);
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		write_ssi(CCSR_SSI_SOR_TX_CLR, &ssi->sor);
-	else
-		write_ssi(CCSR_SSI_SOR_RX_CLR, &ssi->sor);
-
-	return 0;
-}
-
-static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = {
-	.startup	= fsl_ssi_startup,
-	.trigger	= fsl_ssi_ac97_trigger,
-};
-
 static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
 	.ac97_control = 1,
 	.playback = {
@@ -1197,7 +1152,7 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = &fsl_ssi_ac97_dai_ops,
+	.ops = &fsl_ssi_dai_ops,
 };
 
 
-- 
cgit v1.2.2


From d8a64d6ade6a27dec2b8b37e4d9630c40a373bba Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Wed, 8 Jan 2014 17:42:18 +0000
Subject: ASoC: wm_adsp: Factor out ADSP2 boot proceedure

Move the ADSP2 boot proceedure into a work structure in preparation for
running it asynchronously with the reset of the audio path bring up.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm_adsp.c | 181 ++++++++++++++++++++++++++-------------------
 sound/soc/codecs/wm_adsp.h |   2 +
 2 files changed, 105 insertions(+), 78 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 8f720ded27c4..2087ae2eb323 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1492,107 +1492,131 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 	return 0;
 }
 
-int wm_adsp2_event(struct snd_soc_dapm_widget *w,
-		   struct snd_kcontrol *kcontrol, int event)
+void wm_adsp2_boot_work(struct work_struct *work)
 {
-	struct snd_soc_codec *codec = w->codec;
-	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
-	struct wm_adsp *dsp = &dsps[w->shift];
-	struct wm_adsp_alg_region *alg_region;
-	struct wm_coeff_ctl *ctl;
-	unsigned int val;
+	struct wm_adsp *dsp = container_of(work,
+					   struct wm_adsp,
+					   boot_work);
 	int ret;
+	unsigned int val;
 
-	dsp->card = codec->card;
+	/*
+	 * For simplicity set the DSP clock rate to be the
+	 * SYSCLK rate rather than making it configurable.
+	 */
+	ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val);
+	if (ret != 0) {
+		adsp_err(dsp, "Failed to read SYSCLK state: %d\n", ret);
+		return;
+	}
+	val = (val & ARIZONA_SYSCLK_FREQ_MASK)
+		>> ARIZONA_SYSCLK_FREQ_SHIFT;
 
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		/*
-		 * For simplicity set the DSP clock rate to be the
-		 * SYSCLK rate rather than making it configurable.
-		 */
-		ret = regmap_read(dsp->regmap, ARIZONA_SYSTEM_CLOCK_1, &val);
-		if (ret != 0) {
-			adsp_err(dsp, "Failed to read SYSCLK state: %d\n",
-				 ret);
-			return ret;
-		}
-		val = (val & ARIZONA_SYSCLK_FREQ_MASK)
-			>> ARIZONA_SYSCLK_FREQ_SHIFT;
+	ret = regmap_update_bits_async(dsp->regmap,
+				       dsp->base + ADSP2_CLOCKING,
+				       ADSP2_CLK_SEL_MASK, val);
+	if (ret != 0) {
+		adsp_err(dsp, "Failed to set clock rate: %d\n", ret);
+		return;
+	}
 
-		ret = regmap_update_bits_async(dsp->regmap,
-					       dsp->base + ADSP2_CLOCKING,
-					       ADSP2_CLK_SEL_MASK, val);
+	if (dsp->dvfs) {
+		ret = regmap_read(dsp->regmap,
+				  dsp->base + ADSP2_CLOCKING, &val);
 		if (ret != 0) {
-			adsp_err(dsp, "Failed to set clock rate: %d\n",
-				 ret);
-			return ret;
+			dev_err(dsp->dev, "Failed to read clocking: %d\n", ret);
+			return;
 		}
 
-		if (dsp->dvfs) {
-			ret = regmap_read(dsp->regmap,
-					  dsp->base + ADSP2_CLOCKING, &val);
+		if ((val & ADSP2_CLK_SEL_MASK) >= 3) {
+			ret = regulator_enable(dsp->dvfs);
 			if (ret != 0) {
 				dev_err(dsp->dev,
-					"Failed to read clocking: %d\n", ret);
-				return ret;
+					"Failed to enable supply: %d\n",
+					ret);
+				return;
 			}
 
-			if ((val & ADSP2_CLK_SEL_MASK) >= 3) {
-				ret = regulator_enable(dsp->dvfs);
-				if (ret != 0) {
-					dev_err(dsp->dev,
-						"Failed to enable supply: %d\n",
-						ret);
-					return ret;
-				}
-
-				ret = regulator_set_voltage(dsp->dvfs,
-							    1800000,
-							    1800000);
-				if (ret != 0) {
-					dev_err(dsp->dev,
-						"Failed to raise supply: %d\n",
-						ret);
-					return ret;
-				}
+			ret = regulator_set_voltage(dsp->dvfs,
+						    1800000,
+						    1800000);
+			if (ret != 0) {
+				dev_err(dsp->dev,
+					"Failed to raise supply: %d\n",
+					ret);
+				return;
 			}
 		}
+	}
 
-		ret = wm_adsp2_ena(dsp);
-		if (ret != 0)
-			return ret;
+	ret = wm_adsp2_ena(dsp);
+	if (ret != 0)
+		return;
 
-		ret = wm_adsp_load(dsp);
-		if (ret != 0)
-			goto err;
+	ret = wm_adsp_load(dsp);
+	if (ret != 0)
+		goto err;
 
-		ret = wm_adsp_setup_algs(dsp);
-		if (ret != 0)
-			goto err;
+	ret = wm_adsp_setup_algs(dsp);
+	if (ret != 0)
+		goto err;
 
-		ret = wm_adsp_load_coeff(dsp);
-		if (ret != 0)
-			goto err;
+	ret = wm_adsp_load_coeff(dsp);
+	if (ret != 0)
+		goto err;
 
-		/* Initialize caches for enabled and unset controls */
-		ret = wm_coeff_init_control_caches(dsp);
-		if (ret != 0)
-			goto err;
+	/* Initialize caches for enabled and unset controls */
+	ret = wm_coeff_init_control_caches(dsp);
+	if (ret != 0)
+		goto err;
 
-		/* Sync set controls */
-		ret = wm_coeff_sync_controls(dsp);
-		if (ret != 0)
-			goto err;
+	/* Sync set controls */
+	ret = wm_coeff_sync_controls(dsp);
+	if (ret != 0)
+		goto err;
+
+	ret = regmap_update_bits_async(dsp->regmap,
+				       dsp->base + ADSP2_CONTROL,
+				       ADSP2_CORE_ENA,
+				       ADSP2_CORE_ENA);
+	if (ret != 0)
+		goto err;
+
+	dsp->running = true;
+
+	return;
 
-		ret = regmap_update_bits_async(dsp->regmap,
-					       dsp->base + ADSP2_CONTROL,
-					       ADSP2_CORE_ENA | ADSP2_START,
-					       ADSP2_CORE_ENA | ADSP2_START);
+err:
+	regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL,
+			   ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0);
+}
+
+int wm_adsp2_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
+	struct wm_adsp *dsp = &dsps[w->shift];
+	struct wm_adsp_alg_region *alg_region;
+	struct wm_coeff_ctl *ctl;
+	int ret;
+
+	dsp->card = codec->card;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		queue_work(system_unbound_wq, &dsp->boot_work);
+		flush_work(&dsp->boot_work);
+
+		if (!dsp->running)
+			return -EIO;
+
+		ret = regmap_update_bits(dsp->regmap,
+					 dsp->base + ADSP2_CONTROL,
+					 ADSP2_START,
+					 ADSP2_START);
 		if (ret != 0)
 			goto err;
-
-		dsp->running = true;
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
@@ -1663,6 +1687,7 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
 
 	INIT_LIST_HEAD(&adsp->alg_regions);
 	INIT_LIST_HEAD(&adsp->ctl_list);
+	INIT_WORK(&adsp->boot_work, wm_adsp2_boot_work);
 
 	if (dvfs) {
 		adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD");
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index d018dea6254d..b172c1df9159 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -59,6 +59,8 @@ struct wm_adsp {
 	struct regulator *dvfs;
 
 	struct list_head ctl_list;
+
+	struct work_struct boot_work;
 };
 
 #define WM_ADSP1(wname, num) \
-- 
cgit v1.2.2


From 12db5edd6986a8358b92eb3fa6f8d2ee4fe1173b Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Wed, 8 Jan 2014 17:42:19 +0000
Subject: ASoC: wm_adsp: Start DSP booting earlier in the DAPM process

Move the start of booting the DSP to earlier in the DAPM process, and
move the final starting of the DSP to later in the DAPM process. This
allows us to overlap some of the processing with other components of the
system being brought up.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/arizona.h | 17 +++++++++--------
 sound/soc/codecs/wm_adsp.c | 24 +++++++++++++++++++++---
 sound/soc/codecs/wm_adsp.h | 10 ++++++++--
 3 files changed, 38 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 9e81b6392692..256548a5230e 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -166,20 +166,21 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 	ARIZONA_MIXER_INPUT_ROUTES(name " Input 4")
 
 #define ARIZONA_DSP_ROUTES(name) \
-	{ name, NULL, name " Aux 1" }, \
-	{ name, NULL, name " Aux 2" }, \
-	{ name, NULL, name " Aux 3" }, \
-	{ name, NULL, name " Aux 4" }, \
-	{ name, NULL, name " Aux 5" }, \
-	{ name, NULL, name " Aux 6" }, \
+	{ name, NULL, name " Preloader"}, \
+	{ name " Preloader", NULL, name " Aux 1" }, \
+	{ name " Preloader", NULL, name " Aux 2" }, \
+	{ name " Preloader", NULL, name " Aux 3" }, \
+	{ name " Preloader", NULL, name " Aux 4" }, \
+	{ name " Preloader", NULL, name " Aux 5" }, \
+	{ name " Preloader", NULL, name " Aux 6" }, \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 1"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 2"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 3"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 4"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 5"), \
 	ARIZONA_MIXER_INPUT_ROUTES(name " Aux 6"), \
-	ARIZONA_MIXER_ROUTES(name, name "L"), \
-	ARIZONA_MIXER_ROUTES(name, name "R")
+	ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \
+	ARIZONA_MIXER_ROUTES(name " Preloader", name "R")
 
 #define ARIZONA_RATE_ENUM_SIZE 4
 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 2087ae2eb323..a061183add67 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1591,6 +1591,27 @@ err:
 			   ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0);
 }
 
+int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec);
+	struct wm_adsp *dsp = &dsps[w->shift];
+
+	dsp->card = codec->card;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		queue_work(system_unbound_wq, &dsp->boot_work);
+		break;
+	default:
+		break;
+	};
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm_adsp2_early_event);
+
 int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event)
 {
@@ -1601,11 +1622,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
-	dsp->card = codec->card;
-
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		queue_work(system_unbound_wq, &dsp->boot_work);
 		flush_work(&dsp->boot_work);
 
 		if (!dsp->running)
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index b172c1df9159..a4f6b64deb61 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -68,8 +68,12 @@ struct wm_adsp {
 		wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
 
 #define WM_ADSP2(wname, num) \
-	SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \
-		wm_adsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD)
+{	.id = snd_soc_dapm_dai_link, .name = wname " Preloader", \
+	.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_early_event, \
+	.event_flags = SND_SOC_DAPM_PRE_PMU }, \
+{	.id = snd_soc_dapm_out_drv, .name = wname, \
+	.reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \
+	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD }
 
 extern const struct snd_kcontrol_new wm_adsp1_fw_controls[];
 extern const struct snd_kcontrol_new wm_adsp2_fw_controls[];
@@ -78,6 +82,8 @@ int wm_adsp1_init(struct wm_adsp *adsp);
 int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs);
 int wm_adsp1_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
+int wm_adsp2_early_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event);
 int wm_adsp2_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
 
-- 
cgit v1.2.2


From 053ad6a057d168f9f09006c84a4be73f35b21da9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jan 2014 10:37:07 +0100
Subject: ASoC: bcm: Remove obsoleted Kconfig dependency

CONFIG_SND_SOC_DMAENGINE_PCM was renamed to CONFIG_SND_DMAENGINE_PCM
recently.  And yet we don't have to select it since
CONFIG_SND_GENERIC_DMAENGINE_PCM selects the dependency by itself, so
just rip it off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/bcm/Kconfig | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
index 3d82a29ce3a8..6a834e109f1d 100644
--- a/sound/soc/bcm/Kconfig
+++ b/sound/soc/bcm/Kconfig
@@ -1,7 +1,6 @@
 config SND_BCM2835_SOC_I2S
 	tristate "SoC Audio support for the Broadcom BCM2835 I2S module"
 	depends on ARCH_BCM2835 || COMPILE_TEST
-	select SND_SOC_DMAENGINE_PCM
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select REGMAP_MMIO
 	help
-- 
cgit v1.2.2


From 862d761818ba2cb785a0a57557f97a43bd1bc922 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Wed, 8 Jan 2014 15:55:14 -0500
Subject: ALSA: hda - add device ID for Broadwell display audio controller

This patch adds the device ID for Intel Broadwell display HD-Audio controller,
and applies Haswell properties to this device.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 53937dede9fa..fa2879a21a50 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -4009,6 +4009,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	  .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
 	{ PCI_DEVICE(0x8086, 0x0d0c),
 	  .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
+	/* Broadwell */
+	{ PCI_DEVICE(0x8086, 0x160c),
+	  .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL },
 	/* 5 Series/3400 */
 	{ PCI_DEVICE(0x8086, 0x3b56),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM },
-- 
cgit v1.2.2


From 3adadd280a5a78efabdec394c1e745a3f5a1cc18 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Wed, 8 Jan 2014 15:55:24 -0500
Subject: ALSA: hda - add codec ID for Broadwell display audio codec

This patch adds codec ID (0x80862808) and module alias for Broadwell
display codec.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 59b83f4085a8..670da3eaf882 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3273,6 +3273,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x80862805, .name = "CougarPoint HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
 { .id = 0x80862807, .name = "Haswell HDMI",	.patch = patch_generic_hdmi },
+{ .id = 0x80862808, .name = "Broadwell HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862880, .name = "CedarTrail HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862882, .name = "Valleyview2 HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x808629fb, .name = "Crestline HDMI",	.patch = patch_generic_hdmi },
@@ -3328,6 +3329,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:80862805");
 MODULE_ALIAS("snd-hda-codec-id:80862806");
 MODULE_ALIAS("snd-hda-codec-id:80862807");
+MODULE_ALIAS("snd-hda-codec-id:80862808");
 MODULE_ALIAS("snd-hda-codec-id:80862880");
 MODULE_ALIAS("snd-hda-codec-id:80862882");
 MODULE_ALIAS("snd-hda-codec-id:808629fb");
-- 
cgit v1.2.2


From 75dcbe4dc251ebc28cdf0797b85774cdf53a4d29 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Wed, 8 Jan 2014 15:55:32 -0500
Subject: ALSA: hda/hdmi - apply all Haswell fix-ups to Broadwell display codec

Broadwell and Haswell have the same behavior on display audio. So this patch
defines is_haswell_plus() to include codecs for both Haswell and its successor
Broadwell, and apply all Haswell fix-ups to Broadwell.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 15 +++++++++------
 1 file changed, 9 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 670da3eaf882..64f0a5e73a25 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -46,6 +46,9 @@ module_param(static_hdmi_pcm, bool, 0644);
 MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 
 #define is_haswell(codec)  ((codec)->vendor_id == 0x80862807)
+#define is_broadwell(codec)    ((codec)->vendor_id == 0x80862808)
+#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec))
+
 #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882)
 
 struct hdmi_spec_per_cvt {
@@ -1101,7 +1104,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	if (!channels)
 		return;
 
-	if (is_haswell(codec))
+	if (is_haswell_plus(codec))
 		snd_hda_codec_write(codec, pin_nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
@@ -1280,7 +1283,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid,
 	struct hdmi_spec *spec = codec->spec;
 	int err;
 
-	if (is_haswell(codec))
+	if (is_haswell_plus(codec))
 		haswell_verify_D0(codec, cvt_nid, pin_nid);
 
 	err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format));
@@ -1421,7 +1424,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 			    mux_idx);
 
 	/* configure unused pins to choose other converters */
-	if (is_haswell(codec) || is_valleyview(codec))
+	if (is_haswell_plus(codec) || is_valleyview(codec))
 		intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx);
 
 	snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
@@ -1611,7 +1614,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 	if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
 		return 0;
 
-	if (is_haswell(codec))
+	if (is_haswell_plus(codec))
 		intel_haswell_fixup_connect_list(codec, pin_nid);
 
 	pin_idx = spec->num_pins;
@@ -2249,7 +2252,7 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 	codec->spec = spec;
 	hdmi_array_init(spec, 4);
 
-	if (is_haswell(codec)) {
+	if (is_haswell_plus(codec)) {
 		intel_haswell_enable_all_pins(codec, true);
 		intel_haswell_fixup_enable_dp12(codec);
 	}
@@ -2264,7 +2267,7 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
-	if (is_haswell(codec)) {
+	if (is_haswell_plus(codec)) {
 		codec->patch_ops.set_power_state = haswell_set_power_state;
 		codec->dp_mst = true;
 	}
-- 
cgit v1.2.2


From d7b135410e1b2f5ad64e56df09d133a23aa9fbcb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jan 2014 16:09:31 +0100
Subject: ALSA: Remove memory accounting in memalloc helper

It's almost superfluous, and doesn't help much for real uses.
Let's reduce the layer size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/memalloc.c | 31 ++-----------------------------
 1 file changed, 2 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 5e1c7bc73b29..bfaf4f04d91d 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -63,18 +63,6 @@ struct snd_mem_list {
  *
  */
 
-static long snd_allocated_pages; /* holding the number of allocated pages */
-
-static inline void inc_snd_pages(int order)
-{
-	snd_allocated_pages += 1 << order;
-}
-
-static inline void dec_snd_pages(int order)
-{
-	snd_allocated_pages -= 1 << order;
-}
-
 /**
  * snd_malloc_pages - allocate pages with the given size
  * @size: the size to allocate in bytes
@@ -87,7 +75,6 @@ static inline void dec_snd_pages(int order)
 void *snd_malloc_pages(size_t size, gfp_t gfp_flags)
 {
 	int pg;
-	void *res;
 
 	if (WARN_ON(!size))
 		return NULL;
@@ -95,9 +82,7 @@ void *snd_malloc_pages(size_t size, gfp_t gfp_flags)
 		return NULL;
 	gfp_flags |= __GFP_COMP;	/* compound page lets parts be mapped */
 	pg = get_order(size);
-	if ((res = (void *) __get_free_pages(gfp_flags, pg)) != NULL)
-		inc_snd_pages(pg);
-	return res;
+	return (void *) __get_free_pages(gfp_flags, pg);
 }
 
 /**
@@ -114,7 +99,6 @@ void snd_free_pages(void *ptr, size_t size)
 	if (ptr == NULL)
 		return;
 	pg = get_order(size);
-	dec_snd_pages(pg);
 	free_pages((unsigned long) ptr, pg);
 }
 
@@ -129,7 +113,6 @@ void snd_free_pages(void *ptr, size_t size)
 static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
 {
 	int pg;
-	void *res;
 	gfp_t gfp_flags;
 
 	if (WARN_ON(!dma))
@@ -139,11 +122,7 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d
 		| __GFP_COMP	/* compound page lets parts be mapped */
 		| __GFP_NORETRY /* don't trigger OOM-killer */
 		| __GFP_NOWARN; /* no stack trace print - this call is non-critical */
-	res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
-	if (res != NULL)
-		inc_snd_pages(pg);
-
-	return res;
+	return dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
 }
 
 /* free the coherent DMA pages */
@@ -155,7 +134,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
 	if (ptr == NULL)
 		return;
 	pg = get_order(size);
-	dec_snd_pages(pg);
 	dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
 }
 
@@ -432,14 +410,11 @@ static struct proc_dir_entry *snd_mem_proc;
 
 static int snd_mem_proc_read(struct seq_file *seq, void *offset)
 {
-	long pages = snd_allocated_pages >> (PAGE_SHIFT-12);
 	struct snd_mem_list *mem;
 	int devno;
 	static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" };
 
 	mutex_lock(&list_mutex);
-	seq_printf(seq, "pages  : %li bytes (%li pages per %likB)\n",
-		   pages * PAGE_SIZE, pages, PAGE_SIZE / 1024);
 	devno = 0;
 	list_for_each_entry(mem, &mem_list_head, list) {
 		devno++;
@@ -581,8 +556,6 @@ static void __exit snd_mem_exit(void)
 {
 	remove_proc_entry(SND_MEM_PROC_FILE, NULL);
 	free_all_reserved_pages();
-	if (snd_allocated_pages > 0)
-		printk(KERN_ERR "snd-malloc: Memory leak?  pages not freed = %li\n", snd_allocated_pages);
 }
 
 
-- 
cgit v1.2.2


From 47d98c026ef9a9888c36f3c4f26b81f548a0ca86 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jan 2014 16:12:25 +0100
Subject: ALSA: Remove memory reservation code from memalloc helper

Nowadays we have CMA for obtaining the contiguous memory pages
efficiently.  Let's kill the old kludge for reserving the memory pages
for large buffers.  It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/memalloc.c       | 264 --------------------------------------------
 sound/core/pcm_memory.c     |  19 +---
 sound/pci/es1968.c          |  28 +++--
 sound/pci/rme9652/hdsp.c    |  10 +-
 sound/pci/rme9652/rme9652.c |  10 +-
 5 files changed, 18 insertions(+), 313 deletions(-)

(limited to 'sound')

diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index bfaf4f04d91d..be1544ac4613 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -41,22 +41,6 @@ MODULE_DESCRIPTION("Memory allocator for ALSA system.");
 MODULE_LICENSE("GPL");
 
 
-/*
- */
-
-static DEFINE_MUTEX(list_mutex);
-static LIST_HEAD(mem_list_head);
-
-/* buffer preservation list */
-struct snd_mem_list {
-	struct snd_dma_buffer buffer;
-	unsigned int id;
-	struct list_head list;
-};
-
-/* id for pre-allocated buffers */
-#define SNDRV_DMA_DEVICE_UNUSED (unsigned int)-1
-
 /*
  *
  *  Generic memory allocators
@@ -318,251 +302,6 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab)
 	}
 }
 
-
-/**
- * snd_dma_get_reserved - get the reserved buffer for the given device
- * @dmab: the buffer allocation record to store
- * @id: the buffer id
- *
- * Looks for the reserved-buffer list and re-uses if the same buffer
- * is found in the list.  When the buffer is found, it's removed from the free list.
- *
- * Return: The size of buffer if the buffer is found, or zero if not found.
- */
-size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id)
-{
-	struct snd_mem_list *mem;
-
-	if (WARN_ON(!dmab))
-		return 0;
-
-	mutex_lock(&list_mutex);
-	list_for_each_entry(mem, &mem_list_head, list) {
-		if (mem->id == id &&
-		    (mem->buffer.dev.dev == NULL || dmab->dev.dev == NULL ||
-		     ! memcmp(&mem->buffer.dev, &dmab->dev, sizeof(dmab->dev)))) {
-			struct device *dev = dmab->dev.dev;
-			list_del(&mem->list);
-			*dmab = mem->buffer;
-			if (dmab->dev.dev == NULL)
-				dmab->dev.dev = dev;
-			kfree(mem);
-			mutex_unlock(&list_mutex);
-			return dmab->bytes;
-		}
-	}
-	mutex_unlock(&list_mutex);
-	return 0;
-}
-
-/**
- * snd_dma_reserve_buf - reserve the buffer
- * @dmab: the buffer to reserve
- * @id: the buffer id
- *
- * Reserves the given buffer as a reserved buffer.
- *
- * Return: Zero if successful, or a negative code on error.
- */
-int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id)
-{
-	struct snd_mem_list *mem;
-
-	if (WARN_ON(!dmab))
-		return -EINVAL;
-	mem = kmalloc(sizeof(*mem), GFP_KERNEL);
-	if (! mem)
-		return -ENOMEM;
-	mutex_lock(&list_mutex);
-	mem->buffer = *dmab;
-	mem->id = id;
-	list_add_tail(&mem->list, &mem_list_head);
-	mutex_unlock(&list_mutex);
-	return 0;
-}
-
-/*
- * purge all reserved buffers
- */
-static void free_all_reserved_pages(void)
-{
-	struct list_head *p;
-	struct snd_mem_list *mem;
-
-	mutex_lock(&list_mutex);
-	while (! list_empty(&mem_list_head)) {
-		p = mem_list_head.next;
-		mem = list_entry(p, struct snd_mem_list, list);
-		list_del(p);
-		snd_dma_free_pages(&mem->buffer);
-		kfree(mem);
-	}
-	mutex_unlock(&list_mutex);
-}
-
-
-#ifdef CONFIG_PROC_FS
-/*
- * proc file interface
- */
-#define SND_MEM_PROC_FILE	"driver/snd-page-alloc"
-static struct proc_dir_entry *snd_mem_proc;
-
-static int snd_mem_proc_read(struct seq_file *seq, void *offset)
-{
-	struct snd_mem_list *mem;
-	int devno;
-	static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" };
-
-	mutex_lock(&list_mutex);
-	devno = 0;
-	list_for_each_entry(mem, &mem_list_head, list) {
-		devno++;
-		seq_printf(seq, "buffer %d : ID %08x : type %s\n",
-			   devno, mem->id, types[mem->buffer.dev.type]);
-		seq_printf(seq, "  addr = 0x%lx, size = %d bytes\n",
-			   (unsigned long)mem->buffer.addr,
-			   (int)mem->buffer.bytes);
-	}
-	mutex_unlock(&list_mutex);
-	return 0;
-}
-
-static int snd_mem_proc_open(struct inode *inode, struct file *file)
-{
-	return single_open(file, snd_mem_proc_read, NULL);
-}
-
-/* FIXME: for pci only - other bus? */
-#ifdef CONFIG_PCI
-#define gettoken(bufp) strsep(bufp, " \t\n")
-
-static ssize_t snd_mem_proc_write(struct file *file, const char __user * buffer,
-				  size_t count, loff_t * ppos)
-{
-	char buf[128];
-	char *token, *p;
-
-	if (count > sizeof(buf) - 1)
-		return -EINVAL;
-	if (copy_from_user(buf, buffer, count))
-		return -EFAULT;
-	buf[count] = '\0';
-
-	p = buf;
-	token = gettoken(&p);
-	if (! token || *token == '#')
-		return count;
-	if (strcmp(token, "add") == 0) {
-		char *endp;
-		int vendor, device, size, buffers;
-		long mask;
-		int i, alloced;
-		struct pci_dev *pci;
-
-		if ((token = gettoken(&p)) == NULL ||
-		    (vendor = simple_strtol(token, NULL, 0)) <= 0 ||
-		    (token = gettoken(&p)) == NULL ||
-		    (device = simple_strtol(token, NULL, 0)) <= 0 ||
-		    (token = gettoken(&p)) == NULL ||
-		    (mask = simple_strtol(token, NULL, 0)) < 0 ||
-		    (token = gettoken(&p)) == NULL ||
-		    (size = memparse(token, &endp)) < 64*1024 ||
-		    size > 16*1024*1024 /* too big */ ||
-		    (token = gettoken(&p)) == NULL ||
-		    (buffers = simple_strtol(token, NULL, 0)) <= 0 ||
-		    buffers > 4) {
-			printk(KERN_ERR "snd-page-alloc: invalid proc write format\n");
-			return count;
-		}
-		vendor &= 0xffff;
-		device &= 0xffff;
-
-		alloced = 0;
-		pci = NULL;
-		while ((pci = pci_get_device(vendor, device, pci)) != NULL) {
-			if (mask > 0 && mask < 0xffffffff) {
-				if (pci_set_dma_mask(pci, mask) < 0 ||
-				    pci_set_consistent_dma_mask(pci, mask) < 0) {
-					printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", mask, vendor, device);
-					pci_dev_put(pci);
-					return count;
-				}
-			}
-			for (i = 0; i < buffers; i++) {
-				struct snd_dma_buffer dmab;
-				memset(&dmab, 0, sizeof(dmab));
-				if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
-							size, &dmab) < 0) {
-					printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
-					pci_dev_put(pci);
-					return count;
-				}
-				snd_dma_reserve_buf(&dmab, snd_dma_pci_buf_id(pci));
-			}
-			alloced++;
-		}
-		if (! alloced) {
-			for (i = 0; i < buffers; i++) {
-				struct snd_dma_buffer dmab;
-				memset(&dmab, 0, sizeof(dmab));
-				/* FIXME: We can allocate only in ZONE_DMA
-				 * without a device pointer!
-				 */
-				if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, NULL,
-							size, &dmab) < 0) {
-					printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
-					break;
-				}
-				snd_dma_reserve_buf(&dmab, (unsigned int)((vendor << 16) | device));
-			}
-		}
-	} else if (strcmp(token, "erase") == 0)
-		/* FIXME: need for releasing each buffer chunk? */
-		free_all_reserved_pages();
-	else
-		printk(KERN_ERR "snd-page-alloc: invalid proc cmd\n");
-	return count;
-}
-#endif /* CONFIG_PCI */
-
-static const struct file_operations snd_mem_proc_fops = {
-	.owner		= THIS_MODULE,
-	.open		= snd_mem_proc_open,
-	.read		= seq_read,
-#ifdef CONFIG_PCI
-	.write		= snd_mem_proc_write,
-#endif
-	.llseek		= seq_lseek,
-	.release	= single_release,
-};
-
-#endif /* CONFIG_PROC_FS */
-
-/*
- * module entry
- */
-
-static int __init snd_mem_init(void)
-{
-#ifdef CONFIG_PROC_FS
-	snd_mem_proc = proc_create(SND_MEM_PROC_FILE, 0644, NULL,
-				   &snd_mem_proc_fops);
-#endif
-	return 0;
-}
-
-static void __exit snd_mem_exit(void)
-{
-	remove_proc_entry(SND_MEM_PROC_FILE, NULL);
-	free_all_reserved_pages();
-}
-
-
-module_init(snd_mem_init)
-module_exit(snd_mem_exit)
-
-
 /*
  * exports
  */
@@ -570,8 +309,5 @@ EXPORT_SYMBOL(snd_dma_alloc_pages);
 EXPORT_SYMBOL(snd_dma_alloc_pages_fallback);
 EXPORT_SYMBOL(snd_dma_free_pages);
 
-EXPORT_SYMBOL(snd_dma_get_reserved_buf);
-EXPORT_SYMBOL(snd_dma_reserve_buf);
-
 EXPORT_SYMBOL(snd_malloc_pages);
 EXPORT_SYMBOL(snd_free_pages);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 0af622c34e19..01f8eafebda6 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -53,15 +53,6 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz
 	struct snd_dma_buffer *dmab = &substream->dma_buffer;
 	int err;
 
-	/* already reserved? */
-	if (snd_dma_get_reserved_buf(dmab, substream->dma_buf_id) > 0) {
-		if (dmab->bytes >= size)
-			return 0; /* yes */
-		/* no, free the reserved block */
-		snd_dma_free_pages(dmab);
-		dmab->bytes = 0;
-	}
-
 	do {
 		if ((err = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev,
 					       size, dmab)) < 0) {
@@ -82,10 +73,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream
 {
 	if (substream->dma_buffer.area == NULL)
 		return;
-	if (substream->dma_buf_id)
-		snd_dma_reserve_buf(&substream->dma_buffer, substream->dma_buf_id);
-	else
-		snd_dma_free_pages(&substream->dma_buffer);
+	snd_dma_free_pages(&substream->dma_buffer);
 	substream->dma_buffer.area = NULL;
 }
 
@@ -260,11 +248,6 @@ static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream,
  *
  * Do pre-allocation for the given DMA buffer type.
  *
- * When substream->dma_buf_id is set, the function tries to look for
- * the reserved buffer, and the buffer is not freed but reserved at
- * destruction time.  The dma_buf_id must be unique for all systems
- * (in the same DMA buffer type) e.g. using snd_dma_pci_buf_id().
- *
  * Return: Zero if successful, or a negative error code on failure.
  */
 int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index b0e3d92c4656..772cc36f951d 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -1422,7 +1422,7 @@ static void snd_es1968_free_dmabuf(struct es1968 *chip)
 
 	if (! chip->dma.area)
 		return;
-	snd_dma_reserve_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci));
+	snd_dma_free_pages(&chip->dma);
 	while ((p = chip->buf_list.next) != &chip->buf_list) {
 		struct esm_memory *chunk = list_entry(p, struct esm_memory, list);
 		list_del(p);
@@ -1438,20 +1438,18 @@ snd_es1968_init_dmabuf(struct es1968 *chip)
 
 	chip->dma.dev.type = SNDRV_DMA_TYPE_DEV;
 	chip->dma.dev.dev = snd_dma_pci_data(chip->pci);
-	if (! snd_dma_get_reserved_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci))) {
-		err = snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV,
-						   snd_dma_pci_data(chip->pci),
-						   chip->total_bufsize, &chip->dma);
-		if (err < 0 || ! chip->dma.area) {
-			snd_printk(KERN_ERR "es1968: can't allocate dma pages for size %d\n",
-				   chip->total_bufsize);
-			return -ENOMEM;
-		}
-		if ((chip->dma.addr + chip->dma.bytes - 1) & ~((1 << 28) - 1)) {
-			snd_dma_free_pages(&chip->dma);
-			snd_printk(KERN_ERR "es1968: DMA buffer beyond 256MB.\n");
-			return -ENOMEM;
-		}
+	err = snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV,
+					   snd_dma_pci_data(chip->pci),
+					   chip->total_bufsize, &chip->dma);
+	if (err < 0 || ! chip->dma.area) {
+		snd_printk(KERN_ERR "es1968: can't allocate dma pages for size %d\n",
+			   chip->total_bufsize);
+		return -ENOMEM;
+	}
+	if ((chip->dma.addr + chip->dma.bytes - 1) & ~((1 << 28) - 1)) {
+		snd_dma_free_pages(&chip->dma);
+		snd_printk(KERN_ERR "es1968: DMA buffer beyond 256MB.\n");
+		return -ENOMEM;
 	}
 
 	INIT_LIST_HEAD(&chip->buf_list);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index f59a321a6d6a..bd90c80bb494 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -584,10 +584,6 @@ static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer
 {
 	dmab->dev.type = SNDRV_DMA_TYPE_DEV;
 	dmab->dev.dev = snd_dma_pci_data(pci);
-	if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
-		if (dmab->bytes >= size)
-			return 0;
-	}
 	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
 				size, dmab) < 0)
 		return -ENOMEM;
@@ -596,10 +592,8 @@ static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer
 
 static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
 {
-	if (dmab->area) {
-		dmab->dev.dev = NULL; /* make it anonymous */
-		snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
-	}
+	if (dmab->area)
+		snd_dma_free_pages(dmab);
 }
 
 
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 3717f2dd30be..1503ee3585fd 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -294,10 +294,6 @@ static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer
 {
 	dmab->dev.type = SNDRV_DMA_TYPE_DEV;
 	dmab->dev.dev = snd_dma_pci_data(pci);
-	if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
-		if (dmab->bytes >= size)
-			return 0;
-	}
 	if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
 				size, dmab) < 0)
 		return -ENOMEM;
@@ -306,10 +302,8 @@ static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer
 
 static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
 {
-	if (dmab->area) {
-		dmab->dev.dev = NULL; /* make it anonymous */
-		snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
-	}
+	if (dmab->area)
+		snd_dma_free_pages(dmab);
 }
 
 
-- 
cgit v1.2.2


From f5d8e6df4bfa749b4d029ba43ac8a027b1003c2a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 9 Jan 2014 07:44:11 +0100
Subject: ALSA: Remove superfluous header inclusions in memalloc.c

After cutting off the proc and page reservation codes, we don't need
many headers any longer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/memalloc.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index be1544ac4613..e568c14ca88d 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -22,17 +22,13 @@
  */
 
 #include <linux/module.h>
-#include <linux/proc_fs.h>
 #include <linux/init.h>
-#include <linux/pci.h>
 #include <linux/slab.h>
 #include <linux/mm.h>
-#include <linux/seq_file.h>
 #include <asm/uaccess.h>
 #include <linux/dma-mapping.h>
 #include <linux/genalloc.h>
 #include <linux/moduleparam.h>
-#include <linux/mutex.h>
 #include <sound/memalloc.h>
 
 
-- 
cgit v1.2.2


From 2d82ea2005624a80b18916e37bce921d6e868b59 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 9 Jan 2014 07:48:12 +0100
Subject: ALSA: Merge memalloc code into snd-pcm module

Instead of keeping a separate snd-page-alloc module, merge into the
core snd-pcm module, as we don't need to keep it as an individual
module due to the drop of page reservation.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/Makefile   | 10 ++++------
 sound/core/memalloc.c | 10 ----------
 2 files changed, 4 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/core/Makefile b/sound/core/Makefile
index 5e890cfed423..394a38909f6b 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -10,14 +10,12 @@ snd-$(CONFIG_SND_VMASTER) += vmaster.o
 snd-$(CONFIG_SND_KCTL_JACK) += ctljack.o
 snd-$(CONFIG_SND_JACK)	  += jack.o
 
-snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
-		pcm_memory.o
+snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
+		pcm_memory.o memalloc.o
+snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
 
 snd-pcm-dmaengine-objs := pcm_dmaengine.o
 
-snd-page-alloc-y := memalloc.o
-snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
-
 snd-rawmidi-objs  := rawmidi.o
 snd-timer-objs    := timer.o
 snd-hrtimer-objs  := hrtimer.o
@@ -31,7 +29,7 @@ obj-$(CONFIG_SND_HWDEP)		+= snd-hwdep.o
 obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
 obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
-obj-$(CONFIG_SND_PCM)		+= snd-pcm.o snd-page-alloc.o
+obj-$(CONFIG_SND_PCM)		+= snd-pcm.o
 obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
 
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index e568c14ca88d..4595f93d151e 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -21,22 +21,12 @@
  *
  */
 
-#include <linux/module.h>
-#include <linux/init.h>
 #include <linux/slab.h>
 #include <linux/mm.h>
-#include <asm/uaccess.h>
 #include <linux/dma-mapping.h>
 #include <linux/genalloc.h>
-#include <linux/moduleparam.h>
 #include <sound/memalloc.h>
 
-
-MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
-MODULE_DESCRIPTION("Memory allocator for ALSA system.");
-MODULE_LICENSE("GPL");
-
-
 /*
  *
  *  Generic memory allocators
-- 
cgit v1.2.2


From 18b1a902ad55610b161bfc8fb905c372bb8372df Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 9 Jan 2014 09:06:54 +0000
Subject: ASoC: wm_adsp: Mark wm_adsp2_boot_work as static

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm_adsp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index a061183add67..f6e317c78459 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1492,7 +1492,7 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 	return 0;
 }
 
-void wm_adsp2_boot_work(struct work_struct *work)
+static void wm_adsp2_boot_work(struct work_struct *work)
 {
 	struct wm_adsp *dsp = container_of(work,
 					   struct wm_adsp,
-- 
cgit v1.2.2


From d7fa71042304fbc43cfc81d199b922759c67e013 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Thu, 9 Jan 2014 11:16:11 +0100
Subject: ASoC: fsl-ssi: Fix stats compile warning

single_open requires a function with signature
'int (*)(struct seq_file *, void *)'. This patch fixes the warning by
fixing the wrong return type of fsl_ssi_stats_show.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 94dedcb0868d..f662dddf2085 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -391,7 +391,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
  * Display the statistics for the current SSI device.  To avoid confusion,
  * we only show those counts that are enabled.
  */
-static ssize_t fsl_ssi_stats_show(struct seq_file *s, void *unused)
+static int fsl_ssi_stats_show(struct seq_file *s, void *unused)
 {
 	struct fsl_ssi_private *ssi_private = s->private;
 
-- 
cgit v1.2.2


From fa69b0f93e3e383dc50df9529db67c09a1db3787 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 18:37:22 +0000
Subject: ASoC: ad1836: Use params_width() rather than explicit memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ad1836.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 9a92b7962f41..af490bebd7f5 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -168,15 +168,15 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	int word_len = 0;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		word_len = AD1836_WORD_LEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		word_len = AD1836_WORD_LEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 24:
+	case 32:
 		word_len = AD1836_WORD_LEN_24;
 		break;
 	}
-- 
cgit v1.2.2


From d4dd1fdf9ee320ef7fde77922c853c74a8cd3c7d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 18:38:20 +0000
Subject: ASoC: ad193x: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ad193x.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index aea7e52cf714..d6cdb3bb1636 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -249,15 +249,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		word_len = 3;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		word_len = 1;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 24:
+	case 32:
 		word_len = 0;
 		break;
 	}
-- 
cgit v1.2.2


From 7c2aff6ab53a24d6a688fd7ae45fc14a97f48eda Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 18:49:58 +0000
Subject: ASoC: adau1373: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/adau1373.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 59654b1e7f3f..eb836ed5271f 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1078,17 +1078,17 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream,
 		ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
 		(div << 2) | ADAU1373_BCLKDIV_64);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		ctrl = ADAU1373_DAI_WLEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		ctrl = ADAU1373_DAI_WLEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		ctrl = ADAU1373_DAI_WLEN_24;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		ctrl = ADAU1373_DAI_WLEN_32;
 		break;
 	default:
-- 
cgit v1.2.2


From 9b58e7163407f75ec150dc2f91f561fcb681753e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 18:50:25 +0000
Subject: ASoC: adau1701: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/adau1701.c | 34 ++++++++++++++++------------------
 1 file changed, 16 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index ebff1128be59..52e3d83e26e6 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -299,20 +299,20 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv)
 }
 
 static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
-		snd_pcm_format_t format)
+					   struct snd_pcm_hw_params *params)
 {
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK;
 	unsigned int val;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAU1701_SEROCTL_WORD_LEN_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAU1701_SEROCTL_WORD_LEN_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAU1701_SEROCTL_WORD_LEN_24;
 		break;
 	default:
@@ -320,14 +320,14 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
 	}
 
 	if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) {
-		switch (format) {
-		case SNDRV_PCM_FORMAT_S16_LE:
+		switch (params_width(params)) {
+		case 16:
 			val |= ADAU1701_SEROCTL_MSB_DEALY16;
 			break;
-		case SNDRV_PCM_FORMAT_S20_3LE:
+		case 20:
 			val |= ADAU1701_SEROCTL_MSB_DEALY12;
 			break;
-		case SNDRV_PCM_FORMAT_S24_LE:
+		case 24:
 			val |= ADAU1701_SEROCTL_MSB_DEALY8;
 			break;
 		}
@@ -340,7 +340,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec,
 }
 
 static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
-			snd_pcm_format_t format)
+					    struct snd_pcm_hw_params *params)
 {
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
@@ -348,14 +348,14 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec,
 	if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
 		return 0;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAU1701_SERICTL_RIGHTJ_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAU1701_SERICTL_RIGHTJ_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAU1701_SERICTL_RIGHTJ_24;
 		break;
 	default:
@@ -374,7 +374,6 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
 	unsigned int clkdiv = adau1701->sysclk / params_rate(params);
-	snd_pcm_format_t format;
 	unsigned int val;
 	int ret;
 
@@ -406,11 +405,10 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream,
 	regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL,
 		ADAU1701_DSPCTRL_SR_MASK, val);
 
-	format = params_format(params);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return adau1701_set_playback_pcm_format(codec, format);
+		return adau1701_set_playback_pcm_format(codec, params);
 	else
-		return adau1701_set_capture_pcm_format(codec, format);
+		return adau1701_set_capture_pcm_format(codec, params);
 }
 
 static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai,
-- 
cgit v1.2.2


From cf7d8b274f152f289bf9ef821f656133cd3401e4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 18:50:40 +0000
Subject: ASoC: adav80x: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/adav80x.c | 30 ++++++++++++++----------------
 1 file changed, 14 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 14a7c169d004..371a0e9e1af6 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -453,22 +453,22 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
 }
 
 static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, snd_pcm_format_t format)
+		struct snd_soc_dai *dai, struct snd_pcm_hw_params *params)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAV80X_CAPTURE_WORD_LEN16;
 		break;
-	case SNDRV_PCM_FORMAT_S18_3LE:
+	case 18:
 		val = ADAV80X_CAPTRUE_WORD_LEN18;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAV80X_CAPTURE_WORD_LEN20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAV80X_CAPTURE_WORD_LEN24;
 		break;
 	default:
@@ -482,7 +482,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
 }
 
 static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, snd_pcm_format_t format)
+		struct snd_soc_dai *dai, struct snd_pcm_hw_params *params)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
@@ -490,17 +490,17 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
 	if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
 		return 0;
 
-	switch (format) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
 		break;
-	case SNDRV_PCM_FORMAT_S18_3LE:
+	case 18:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
 		break;
 	default:
@@ -524,12 +524,10 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		adav80x_set_playback_pcm_format(codec, dai,
-			params_format(params));
+		adav80x_set_playback_pcm_format(codec, dai, params);
 		adav80x_set_dac_clock(codec, rate);
 	} else {
-		adav80x_set_capture_pcm_format(codec, dai,
-			params_format(params));
+		adav80x_set_capture_pcm_format(codec, dai, params);
 		adav80x_set_adc_clock(codec, rate);
 	}
 	adav80x->rate = rate;
-- 
cgit v1.2.2


From c098284a4bfb81b86331aadd08b482233c21fa2c Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:10 +0100
Subject: ASoC: intel: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/intel/sst_platform.c | 10 ----------
 sound/soc/intel/sst_platform.h |  4 ----
 2 files changed, 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst_platform.c
index b6b5eb698d33..f465a8180863 100644
--- a/sound/soc/intel/sst_platform.c
+++ b/sound/soc/intel/sst_platform.c
@@ -89,16 +89,6 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = {
 			SNDRV_PCM_INFO_MMAP_VALID |
 			SNDRV_PCM_INFO_BLOCK_TRANSFER |
 			SNDRV_PCM_INFO_SYNC_START),
-	.formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 |
-			SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 |
-			SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32),
-	.rates = (SNDRV_PCM_RATE_8000|
-			SNDRV_PCM_RATE_44100 |
-			SNDRV_PCM_RATE_48000),
-	.rate_min = SST_MIN_RATE,
-	.rate_max = SST_MAX_RATE,
-	.channels_min =	SST_MIN_CHANNEL,
-	.channels_max =	SST_MAX_CHANNEL,
 	.buffer_bytes_max = SST_MAX_BUFFER,
 	.period_bytes_min = SST_MIN_PERIOD_BYTES,
 	.period_bytes_max = SST_MAX_PERIOD_BYTES,
diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst_platform.h
index cacc9066ec52..bee64fb7d2ef 100644
--- a/sound/soc/intel/sst_platform.h
+++ b/sound/soc/intel/sst_platform.h
@@ -33,10 +33,6 @@
 #define SST_STEREO		2
 #define SST_MAX_CAP		5
 
-#define SST_MIN_RATE		8000
-#define SST_MAX_RATE		48000
-#define SST_MIN_CHANNEL		1
-#define SST_MAX_CHANNEL		5
 #define SST_MAX_BUFFER		(800*1024)
 #define SST_MIN_BUFFER		(800*1024)
 #define SST_MIN_PERIOD_BYTES	32
-- 
cgit v1.2.2


From 3317208c8838479a1cfe1ef395ec895d160957f0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:11 +0100
Subject: ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/kirkwood/kirkwood-dma.c | 16 ----------------
 1 file changed, 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 4af1936cf0f4..aac22fccdcdc 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -21,16 +21,6 @@
 #include <sound/soc.h>
 #include "kirkwood.h"
 
-#define KIRKWOOD_RATES \
-	(SNDRV_PCM_RATE_8000_192000 |		\
-	 SNDRV_PCM_RATE_CONTINUOUS |		\
-	 SNDRV_PCM_RATE_KNOT)
-
-#define KIRKWOOD_FORMATS \
-	(SNDRV_PCM_FMTBIT_S16_LE | \
-	 SNDRV_PCM_FMTBIT_S24_LE | \
-	 SNDRV_PCM_FMTBIT_S32_LE)
-
 static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = subs->private_data;
@@ -43,12 +33,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
 		 SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_PAUSE),
-	.formats		= KIRKWOOD_FORMATS,
-	.rates			= KIRKWOOD_RATES,
-	.rate_min		= 8000,
-	.rate_max		= 384000,
-	.channels_min		= 1,
-	.channels_max		= 8,
 	.buffer_bytes_max	= KIRKWOOD_SND_MAX_BUFFER_BYTES,
 	.period_bytes_min	= KIRKWOOD_SND_MIN_PERIOD_BYTES,
 	.period_bytes_max	= KIRKWOOD_SND_MAX_PERIOD_BYTES,
-- 
cgit v1.2.2


From 115367713460fd375380f5dc663271f07c513b33 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:13 +0100
Subject: ASoC: nuc900: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/nuc900/nuc900-pcm.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index f588ee45b4fd..f434ed79d1b6 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -32,9 +32,6 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = {
 					SNDRV_PCM_INFO_MMAP_VALID |
 					SNDRV_PCM_INFO_PAUSE |
 					SNDRV_PCM_INFO_RESUME,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
-	.channels_min		= 1,
-	.channels_max		= 2,
 	.buffer_bytes_max	= 4*1024,
 	.period_bytes_min	= 1*1024,
 	.period_bytes_max	= 4*1024,
-- 
cgit v1.2.2


From a7ddf151b0eb12a8840d9d127f1679bb1c89a1ff Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:14 +0100
Subject: ASoC: sh: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/sh/dma-sh7760.c | 17 -----------------
 sound/soc/sh/fsi.c        |  6 ------
 sound/soc/sh/rcar/core.c  |  6 ------
 3 files changed, 29 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 1a8b03e4b41b..c85f8eb66c97 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -89,29 +89,12 @@ struct camelot_pcm {
 #define DMABRG_PREALLOC_BUFFER		32 * 1024
 #define DMABRG_PREALLOC_BUFFER_MAX	32 * 1024
 
-/* support everything the SSI supports */
-#define DMABRG_RATES	\
-	SNDRV_PCM_RATE_8000_192000
-
-#define DMABRG_FMTS	\
-	(SNDRV_PCM_FMTBIT_S8      | SNDRV_PCM_FMTBIT_U8      |	\
-	 SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_U16_LE  |	\
-	 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE |	\
-	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
-	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
-
 static struct snd_pcm_hardware camelot_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_BATCH),
-	.formats =	DMABRG_FMTS,
-	.rates =	DMABRG_RATES,
-	.rate_min =		8000,
-	.rate_max =		192000,
-	.channels_min =		2,
-	.channels_max =		8,		/* max of the SSI */
 	.buffer_bytes_max =	DMABRG_PERIOD_MAX,
 	.period_bytes_min =	DMABRG_PERIOD_MIN,
 	.period_bytes_max =	DMABRG_PERIOD_MAX / 2,
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index b33ca7cd085b..ef89fa8e4fc8 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1777,12 +1777,6 @@ static struct snd_pcm_hardware fsi_pcm_hardware = {
 			SNDRV_PCM_INFO_MMAP		|
 			SNDRV_PCM_INFO_MMAP_VALID	|
 			SNDRV_PCM_INFO_PAUSE,
-	.formats		= FSI_FMTS,
-	.rates			= FSI_RATES,
-	.rate_min		= 8000,
-	.rate_max		= 192000,
-	.channels_min		= 2,
-	.channels_max		= 2,
 	.buffer_bytes_max	= 64 * 1024,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index b3653d37f75f..743de5e3b1e1 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -628,12 +628,6 @@ static struct snd_pcm_hardware rsnd_pcm_hardware = {
 			SNDRV_PCM_INFO_MMAP		|
 			SNDRV_PCM_INFO_MMAP_VALID	|
 			SNDRV_PCM_INFO_PAUSE,
-	.formats		= RSND_FMTS,
-	.rates			= RSND_RATES,
-	.rate_min		= 8000,
-	.rate_max		= 192000,
-	.channels_min		= 2,
-	.channels_max		= 2,
 	.buffer_bytes_max	= 64 * 1024,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 8192,
-- 
cgit v1.2.2


From df021a72c92e8b9fe9b5d3f11105125484e8751f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:15 +0100
Subject: ASoC: ux500: Don't set unused struct snd_pcm_hardware fields

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/ux500/ux500_pcm.c | 15 ---------------
 1 file changed, 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 55a8634cc3da..51a66a87305a 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -28,12 +28,6 @@
 #include "ux500_msp_i2s.h"
 #include "ux500_pcm.h"
 
-#define UX500_PLATFORM_MIN_RATE 8000
-#define UX500_PLATFORM_MAX_RATE 48000
-
-#define UX500_PLATFORM_MIN_CHANNELS 1
-#define UX500_PLATFORM_MAX_CHANNELS 8
-
 #define UX500_PLATFORM_PERIODS_BYTES_MIN	128
 #define UX500_PLATFORM_PERIODS_BYTES_MAX	(64 * PAGE_SIZE)
 #define UX500_PLATFORM_PERIODS_MIN		2
@@ -45,15 +39,6 @@ static const struct snd_pcm_hardware ux500_pcm_hw = {
 		SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_RESUME |
 		SNDRV_PCM_INFO_PAUSE,
-	.formats = SNDRV_PCM_FMTBIT_S16_LE |
-		SNDRV_PCM_FMTBIT_U16_LE |
-		SNDRV_PCM_FMTBIT_S16_BE |
-		SNDRV_PCM_FMTBIT_U16_BE,
-	.rates = SNDRV_PCM_RATE_KNOT,
-	.rate_min = UX500_PLATFORM_MIN_RATE,
-	.rate_max = UX500_PLATFORM_MAX_RATE,
-	.channels_min = UX500_PLATFORM_MIN_CHANNELS,
-	.channels_max = UX500_PLATFORM_MAX_CHANNELS,
 	.buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
 	.period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
 	.period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
-- 
cgit v1.2.2


From 16d7ea9167839d0b971ab29030886280595dd5fc Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 6 Jan 2014 14:19:16 +0100
Subject: ASoC: Allow PCMs to restrict the supported formats

Some DMA cores might add additional restrictions on which in memory audio
formats can be supported by the compound sound card. If the PCM component
specifies a set of formats it supports (by setting the formats field to non 0)
take these into account when calculating the format set for the compound sound
card.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 141a302e4e77..e7f16b54a97d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -158,7 +158,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
 		cpu_stream->channels_min);
 	hw->channels_max = min(codec_stream->channels_max,
 		cpu_stream->channels_max);
-	hw->formats = codec_stream->formats & cpu_stream->formats;
+	if (hw->formats)
+		hw->formats &= codec_stream->formats & cpu_stream->formats;
+	else
+		hw->formats = codec_stream->formats & cpu_stream->formats;
 	hw->rates = codec_stream->rates & cpu_stream->rates;
 	if (codec_stream->rates
 		& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-- 
cgit v1.2.2


From e1cffe8c9f3a4f74b8b212c9fbe2873a8ee2f395 Mon Sep 17 00:00:00 2001
From: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Date: Thu, 9 Jan 2014 22:27:31 +0800
Subject: ASoC: fsl-ssi: Add missing clk_disable_unprepare() on error in
 fsl_ssi_probe()

Add the missing clk_disable_unprepare() before return from
fsl_ssi_probe() in the request irq error handling case.

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index f662dddf2085..6c2f040f49ae 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1439,7 +1439,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		if (ret < 0) {
 			dev_err(&pdev->dev, "could not claim irq %u\n",
 					ssi_private->irq);
-			goto error_irqmap;
+			goto error_clk;
 		}
 	}
 
-- 
cgit v1.2.2


From 2b56b5f02029531007c8601b23f282b840715401 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Thu, 9 Jan 2014 18:42:48 +0800
Subject: ASoC: fsl_ssi: Set default slot number for common cases

For those platforms using DAI master mode like I2S, it's better to pre-set
a default slot number so that there's no need for these common cases to set
the slot number from its machine driver any more.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6c2f040f49ae..7864ec5cf5f9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -711,6 +711,17 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private)
 	if (ssi_private->imx_ac97)
 		fsl_ssi_setup_ac97(ssi_private);
 
+	/*
+	 * Set a default slot number so that there is no need for those common
+	 * cases like I2S mode to call the extra set_tdm_slot() any more.
+	 */
+	if (!ssi_private->imx_ac97) {
+		write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK,
+				CCSR_SSI_SxCCR_DC(2));
+		write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK,
+				CCSR_SSI_SxCCR_DC(2));
+	}
+
 	return 0;
 }
 
-- 
cgit v1.2.2


From 708ec0241c56b85176937e79314430f4f71e40c6 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Thu, 9 Jan 2014 17:19:08 +0800
Subject: ASoC: simple-card: fix a bug where cinfo will be NULL before using it

If the dt is not used, the cinfo will be always NULL before using it.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 0430be85f23c..6c61b1758f78 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -213,8 +213,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 			}
 		}
 	} else {
-		cinfo->snd_card.dev = &pdev->dev;
 		cinfo = pdev->dev.platform_data;
+		cinfo->snd_card.dev = &pdev->dev;
 	}
 
 	if (!cinfo) {
-- 
cgit v1.2.2


From 34787d0a258ebb3686676fb37a9e8717cbdd835a Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Thu, 9 Jan 2014 17:49:40 +0800
Subject: ASoC: simple-card: fix the cinfo error check

If the dt is used and the cinfo is NULL, the -ENOMEM should be return.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6c61b1758f78..11030a63b811 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -211,15 +211,17 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 					dev_err(dev, "parse error %d\n", ret);
 				return ret;
 			}
+		} else {
+			return -ENOMEM;
 		}
 	} else {
 		cinfo = pdev->dev.platform_data;
-		cinfo->snd_card.dev = &pdev->dev;
-	}
+		if (!cinfo) {
+			dev_err(dev, "no info for asoc-simple-card\n");
+			return -EINVAL;
+		}
 
-	if (!cinfo) {
-		dev_err(dev, "no info for asoc-simple-card\n");
-		return -EINVAL;
+		cinfo->snd_card.dev = &pdev->dev;
 	}
 
 	if (!cinfo->name	||
-- 
cgit v1.2.2


From e240a46965dd14db6e6bdf8974ebb347f3db860b Mon Sep 17 00:00:00 2001
From: Sarah Bessmer <aotos@fastmail.fm>
Date: Thu, 9 Jan 2014 11:36:31 -0800
Subject: ALSA: ctxfi - Work around emu20k1 glitch to prevent buffered sound
 data loss

Occasionally, on playback stream ringbuffer wraparound, the EMU20K1
hardware will momentarily return 0 instead of the proper current(loop)
address. This patch handles that case, fixing the problem of playback
position corruption and subsequent loss of buffered sound data, that
occurs with some common buffering layout patterns(e.g. multiple
simultaneous output streams with differently-sized or
non-power-of-2-sized buffers).

An alternate means of fixing the problem would be to read the ca
register continuously, until two sequential reads return the same
value; however, that would be a more invasive change, has performance
implications, and isn't necessary unless there are also issues with the
value not being updated atomically in regards to individual bits or
something similar(which I have not encountered through light testing).

I have no EMU20K2 hardware to confirm if the issue is present there,
but even if it's not, this change shouldn't break anything that's not
already broken.

Signed-off-by: Sarah Bessmer <aotos@fastmail.fm>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctatc.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b5fa583a239a..eb86829529eb 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -435,6 +435,11 @@ atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm)
 		return 0;
 	position = src->ops->get_ca(src);
 
+	if (position < apcm->vm_block->addr) {
+		snd_printdd("ctxfi: bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", position, apcm->vm_block->addr, apcm->vm_block->size);
+		position = apcm->vm_block->addr;
+	}
+
 	size = apcm->vm_block->size;
 	max_cisz = src->multi * src->rsc.msr;
 	max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8);
-- 
cgit v1.2.2


From afb6d4ed3fd88bacf8b0abcbf053c79c604f509f Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Date: Thu, 9 Jan 2014 14:29:25 +0000
Subject: ASoC: wm5110: Add controls for headphone short circuit protection

Add controls to enable/disable the headphone short circuit protection of
the headphone outputs.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm5110.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index eee627b9bd13..618fea33b3af 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -313,6 +313,13 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
+SOC_SINGLE("HPOUT1 SC Protect Switch", ARIZONA_HP1_SHORT_CIRCUIT_CTRL,
+	   ARIZONA_HP1_SC_ENA_SHIFT, 1, 0),
+SOC_SINGLE("HPOUT2 SC Protect Switch", ARIZONA_HP2_SHORT_CIRCUIT_CTRL,
+	   ARIZONA_HP2_SC_ENA_SHIFT, 1, 0),
+SOC_SINGLE("HPOUT3 SC Protect Switch", ARIZONA_HP3_SHORT_CIRCUIT_CTRL,
+	   ARIZONA_HP3_SC_ENA_SHIFT, 1, 0),
+
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
-- 
cgit v1.2.2


From 43d24e76b69826ce32292f47060ad78cdd0197fa Mon Sep 17 00:00:00 2001
From: Nicolin Chen <Guangyu.Chen@freescale.com>
Date: Fri, 10 Jan 2014 17:54:06 +0800
Subject: ASoC: fsl_esai: Add ESAI CPU DAI driver

This patch implements a device-tree-only CPU DAI driver for Freescale ESAI
controller that supports:

 - 12 channels playback and 8 channels record.
   [ Some of the inner transmitters and receivers are sharing same group of
     pins. So the maxmium 12 output or 8 input channels are only valid if
     there is no pin conflict occurring to it. ]

 - Independent (asynchronous mode) or shared (synchronous mode) transmit and
   receive sections with separate or shared internal/external clocks and frame
   syncs, operating in Master or Slave mode.
   [ Current ALSA seems not to allow CPU DAI drivers to configure DAI format
     separately for PLAYBACK and CAPTURE. So this first version only supports
     the case that uses the same DAI format for both directions. ]

 - Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B.

 - Programmable word length (8, 16, 20 or 24bits)

 - Flexible selection between system clock or external oscillator as input
   clock source, programmable internal clock divider and frame sync generation.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig    |   3 +
 sound/soc/fsl/Makefile   |   2 +
 sound/soc/fsl/fsl_esai.c | 815 +++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/fsl_esai.h | 354 ++++++++++++++++++++
 4 files changed, 1174 insertions(+)
 create mode 100644 sound/soc/fsl/fsl_esai.c
 create mode 100644 sound/soc/fsl/fsl_esai.h

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index ac4fe4ea15a9..f2f39dd13bc7 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -8,6 +8,9 @@ config SND_SOC_FSL_SSI
 config SND_SOC_FSL_SPDIF
 	tristate
 
+config SND_SOC_FSL_ESAI
+	tristate
+
 config SND_SOC_FSL_UTILS
 	tristate
 
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index aaccbee17006..b12ad4b9b4da 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -14,11 +14,13 @@ obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-spdif-objs := fsl_spdif.o
+snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
 obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o
+obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o
 obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o
 obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
 
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
new file mode 100644
index 000000000000..d0c72ed261e7
--- /dev/null
+++ b/sound/soc/fsl/fsl_esai.c
@@ -0,0 +1,815 @@
+/*
+ * Freescale ESAI ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_irq.h>
+#include <linux/of_platform.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_esai.h"
+#include "imx-pcm.h"
+
+#define FSL_ESAI_RATES		SNDRV_PCM_RATE_8000_192000
+#define FSL_ESAI_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
+				SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_S20_3LE | \
+				SNDRV_PCM_FMTBIT_S24_LE)
+
+/**
+ * fsl_esai: ESAI private data
+ *
+ * @dma_params_rx: DMA parameters for receive channel
+ * @dma_params_tx: DMA parameters for transmit channel
+ * @pdev: platform device pointer
+ * @regmap: regmap handler
+ * @coreclk: clock source to access register
+ * @extalclk: esai clock source to derive HCK, SCK and FS
+ * @fsysclk: system clock source to derive HCK, SCK and FS
+ * @fifo_depth: depth of tx/rx FIFO
+ * @slot_width: width of each DAI slot
+ * @hck_rate: clock rate of desired HCKx clock
+ * @sck_div: if using PSR/PM dividers for SCKx clock
+ * @slave_mode: if fully using DAI slave mode
+ * @synchronous: if using tx/rx synchronous mode
+ * @name: driver name
+ */
+struct fsl_esai {
+	struct snd_dmaengine_dai_dma_data dma_params_rx;
+	struct snd_dmaengine_dai_dma_data dma_params_tx;
+	struct platform_device *pdev;
+	struct regmap *regmap;
+	struct clk *coreclk;
+	struct clk *extalclk;
+	struct clk *fsysclk;
+	u32 fifo_depth;
+	u32 slot_width;
+	u32 hck_rate[2];
+	bool sck_div[2];
+	bool slave_mode;
+	bool synchronous;
+	char name[32];
+};
+
+static irqreturn_t esai_isr(int irq, void *devid)
+{
+	struct fsl_esai *esai_priv = (struct fsl_esai *)devid;
+	struct platform_device *pdev = esai_priv->pdev;
+	u32 esr;
+
+	regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr);
+
+	if (esr & ESAI_ESR_TINIT_MASK)
+		dev_dbg(&pdev->dev, "isr: Transmition Initialized\n");
+
+	if (esr & ESAI_ESR_RFF_MASK)
+		dev_warn(&pdev->dev, "isr: Receiving overrun\n");
+
+	if (esr & ESAI_ESR_TFE_MASK)
+		dev_warn(&pdev->dev, "isr: Transmition underrun\n");
+
+	if (esr & ESAI_ESR_TLS_MASK)
+		dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n");
+
+	if (esr & ESAI_ESR_TDE_MASK)
+		dev_dbg(&pdev->dev, "isr: Transmition data exception\n");
+
+	if (esr & ESAI_ESR_TED_MASK)
+		dev_dbg(&pdev->dev, "isr: Transmitting even slots\n");
+
+	if (esr & ESAI_ESR_TD_MASK)
+		dev_dbg(&pdev->dev, "isr: Transmitting data\n");
+
+	if (esr & ESAI_ESR_RLS_MASK)
+		dev_dbg(&pdev->dev, "isr: Just received the last slot\n");
+
+	if (esr & ESAI_ESR_RDE_MASK)
+		dev_dbg(&pdev->dev, "isr: Receiving data exception\n");
+
+	if (esr & ESAI_ESR_RED_MASK)
+		dev_dbg(&pdev->dev, "isr: Receiving even slots\n");
+
+	if (esr & ESAI_ESR_RD_MASK)
+		dev_dbg(&pdev->dev, "isr: Receiving data\n");
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * This function is used to calculate the divisors of psr, pm, fp and it is
+ * supposed to be called in set_dai_sysclk() and set_bclk().
+ *
+ * @ratio: desired overall ratio for the paticipating dividers
+ * @usefp: for HCK setting, there is no need to set fp divider
+ * @fp: bypass other dividers by setting fp directly if fp != 0
+ * @tx: current setting is for playback or capture
+ */
+static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
+				bool usefp, u32 fp)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	u32 psr, pm = 999, maxfp, prod, sub, savesub, i, j;
+
+	maxfp = usefp ? 16 : 1;
+
+	if (usefp && fp)
+		goto out_fp;
+
+	if (ratio > 2 * 8 * 256 * maxfp || ratio < 2) {
+		dev_err(dai->dev, "the ratio is out of range (2 ~ %d)\n",
+				2 * 8 * 256 * maxfp);
+		return -EINVAL;
+	} else if (ratio % 2) {
+		dev_err(dai->dev, "the raio must be even if using upper divider\n");
+		return -EINVAL;
+	}
+
+	ratio /= 2;
+
+	psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8;
+
+	/* Set the max fluctuation -- 0.1% of the max devisor */
+	savesub = (psr ? 1 : 8)  * 256 * maxfp / 1000;
+
+	/* Find the best value for PM */
+	for (i = 1; i <= 256; i++) {
+		for (j = 1; j <= maxfp; j++) {
+			/* PSR (1 or 8) * PM (1 ~ 256) * FP (1 ~ 16) */
+			prod = (psr ? 1 : 8) * i * j;
+
+			if (prod == ratio)
+				sub = 0;
+			else if (prod / ratio == 1)
+				sub = prod - ratio;
+			else if (ratio / prod == 1)
+				sub = ratio - prod;
+			else
+				continue;
+
+			/* Calculate the fraction */
+			sub = sub * 1000 / ratio;
+			if (sub < savesub) {
+				savesub = sub;
+				pm = i;
+				fp = j;
+			}
+
+			/* We are lucky */
+			if (savesub == 0)
+				goto out;
+		}
+	}
+
+	if (pm == 999) {
+		dev_err(dai->dev, "failed to calculate proper divisors\n");
+		return -EINVAL;
+	}
+
+out:
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+			   ESAI_xCCR_xPSR_MASK | ESAI_xCCR_xPM_MASK,
+			   psr | ESAI_xCCR_xPM(pm));
+
+out_fp:
+	/* Bypass fp if not being required */
+	if (maxfp <= 1)
+		return 0;
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+			   ESAI_xCCR_xFP_MASK, ESAI_xCCR_xFP(fp));
+
+	return 0;
+}
+
+/**
+ * This function mainly configures the clock frequency of MCLK (HCKT/HCKR)
+ *
+ * @Parameters:
+ * clk_id: The clock source of HCKT/HCKR
+ *	  (Input from outside; output from inside, FSYS or EXTAL)
+ * freq: The required clock rate of HCKT/HCKR
+ * dir: The clock direction of HCKT/HCKR
+ *
+ * Note: If the direction is input, we do not care about clk_id.
+ */
+static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				   unsigned int freq, int dir)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	struct clk *clksrc = esai_priv->extalclk;
+	bool tx = clk_id <= ESAI_HCKT_EXTAL;
+	bool in = dir == SND_SOC_CLOCK_IN;
+	u32 ret, ratio, ecr = 0;
+	unsigned long clk_rate;
+
+	/* sck_div can be only bypassed if ETO/ERO=0 and SNC_SOC_CLOCK_OUT */
+	esai_priv->sck_div[tx] = true;
+
+	/* Set the direction of HCKT/HCKR pins */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCCR(tx),
+			   ESAI_xCCR_xHCKD, in ? 0 : ESAI_xCCR_xHCKD);
+
+	if (in)
+		goto out;
+
+	switch (clk_id) {
+	case ESAI_HCKT_FSYS:
+	case ESAI_HCKR_FSYS:
+		clksrc = esai_priv->fsysclk;
+		break;
+	case ESAI_HCKT_EXTAL:
+		ecr |= ESAI_ECR_ETI;
+	case ESAI_HCKR_EXTAL:
+		ecr |= ESAI_ECR_ERI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (IS_ERR(clksrc)) {
+		dev_err(dai->dev, "no assigned %s clock\n",
+				clk_id % 2 ? "extal" : "fsys");
+		return PTR_ERR(clksrc);
+	}
+	clk_rate = clk_get_rate(clksrc);
+
+	ratio = clk_rate / freq;
+	if (ratio * freq > clk_rate)
+		ret = ratio * freq - clk_rate;
+	else if (ratio * freq < clk_rate)
+		ret = clk_rate - ratio * freq;
+	else
+		ret = 0;
+
+	/* Block if clock source can not be divided into the required rate */
+	if (ret != 0 && clk_rate / ret < 1000) {
+		dev_err(dai->dev, "failed to derive required HCK%c rate\n",
+				tx ? 'T' : 'R');
+		return -EINVAL;
+	}
+
+	if (ratio == 1) {
+		/* Bypass all the dividers if not being needed */
+		ecr |= tx ? ESAI_ECR_ETO : ESAI_ECR_ERO;
+		goto out;
+	}
+
+	ret = fsl_esai_divisor_cal(dai, tx, ratio, false, 0);
+	if (ret)
+		return ret;
+
+	esai_priv->sck_div[tx] = false;
+
+out:
+	esai_priv->hck_rate[tx] = freq;
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR,
+			   tx ? ESAI_ECR_ETI | ESAI_ECR_ETO :
+			   ESAI_ECR_ERI | ESAI_ECR_ERO, ecr);
+
+	return 0;
+}
+
+/**
+ * This function configures the related dividers according to the bclk rate
+ */
+static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	u32 hck_rate = esai_priv->hck_rate[tx];
+	u32 sub, ratio = hck_rate / freq;
+
+	/* Don't apply for fully slave mode*/
+	if (esai_priv->slave_mode)
+		return 0;
+
+	if (ratio * freq > hck_rate)
+		sub = ratio * freq - hck_rate;
+	else if (ratio * freq < hck_rate)
+		sub = hck_rate - ratio * freq;
+	else
+		sub = 0;
+
+	/* Block if clock source can not be divided into the required rate */
+	if (sub != 0 && hck_rate / sub < 1000) {
+		dev_err(dai->dev, "failed to derive required SCK%c rate\n",
+				tx ? 'T' : 'R');
+		return -EINVAL;
+	}
+
+	if (esai_priv->sck_div[tx] && (ratio > 16 || ratio == 0)) {
+		dev_err(dai->dev, "the ratio is out of range (1 ~ 16)\n");
+		return -EINVAL;
+	}
+
+	return fsl_esai_divisor_cal(dai, tx, ratio, true,
+			esai_priv->sck_div[tx] ? 0 : ratio);
+}
+
+static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
+				     u32 rx_mask, int slots, int slot_width)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
+			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
+			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
+			   ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask));
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
+			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
+			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
+			   ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask));
+
+	esai_priv->slot_width = slot_width;
+
+	return 0;
+}
+
+static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	u32 xcr = 0, xccr = 0, mask;
+
+	/* DAI mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Data on rising edge of bclk, frame low, 1clk before data */
+		xcr |= ESAI_xCR_xFSR;
+		xccr |= ESAI_xCCR_xFSP | ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* Data on rising edge of bclk, frame high */
+		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		/* Data on rising edge of bclk, frame high, right aligned */
+		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* Data on rising edge of bclk, frame high, 1clk before data */
+		xcr |= ESAI_xCR_xFSL | ESAI_xCR_xFSR;
+		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* Data on rising edge of bclk, frame high */
+		xcr |= ESAI_xCR_xFSL;
+		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* Nothing to do for both normal cases */
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Invert bit clock */
+		xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/* Invert frame clock */
+		xccr ^= ESAI_xCCR_xFSP;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		/* Invert both clocks */
+		xccr ^= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	esai_priv->slave_mode = false;
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		esai_priv->slave_mode = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		xccr |= ESAI_xCCR_xCKD;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		xccr |= ESAI_xCCR_xFSD;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		xccr |= ESAI_xCCR_xFSD | ESAI_xCCR_xCKD;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR;
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr);
+
+	mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP |
+		ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA;
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr);
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr);
+
+	return 0;
+}
+
+static int fsl_esai_startup(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+	/*
+	 * Some platforms might use the same bit to gate all three or two of
+	 * clocks, so keep all clocks open/close at the same time for safety
+	 */
+	clk_prepare_enable(esai_priv->coreclk);
+	if (!IS_ERR(esai_priv->extalclk))
+		clk_prepare_enable(esai_priv->extalclk);
+	if (!IS_ERR(esai_priv->fsysclk))
+		clk_prepare_enable(esai_priv->fsysclk);
+
+	if (!dai->active) {
+		/* Reset Port C */
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
+				   ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
+				   ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
+
+		/* Set synchronous mode */
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_SAICR,
+				   ESAI_SAICR_SYNC, esai_priv->synchronous ?
+				   ESAI_SAICR_SYNC : 0);
+
+		/* Set a default slot number -- 2 */
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
+				   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
+				   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+	}
+
+	return 0;
+}
+
+static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      struct snd_soc_dai *dai)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	u32 width = snd_pcm_format_width(params_format(params));
+	u32 channels = params_channels(params);
+	u32 bclk, mask, val, ret;
+
+	bclk = params_rate(params) * esai_priv->slot_width * 2;
+
+	ret = fsl_esai_set_bclk(dai, tx, bclk);
+	if (ret)
+		return ret;
+
+	/* Use Normal mode to support monaural audio */
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+			   ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ?
+			   ESAI_xCR_xMOD_NETWORK : 0);
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+			   ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR);
+
+	mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
+	      (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
+	val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
+	     (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
+
+	mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
+	val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
+
+	regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
+
+	return 0;
+}
+
+static void fsl_esai_shutdown(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+	if (!IS_ERR(esai_priv->fsysclk))
+		clk_disable_unprepare(esai_priv->fsysclk);
+	if (!IS_ERR(esai_priv->extalclk))
+		clk_disable_unprepare(esai_priv->extalclk);
+	clk_disable_unprepare(esai_priv->coreclk);
+}
+
+static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
+			    struct snd_soc_dai *dai)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	u8 i, channels = substream->runtime->channels;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+				   ESAI_xFCR_xFEN_MASK, ESAI_xFCR_xFEN);
+
+		/* Write initial words reqiured by ESAI as normal procedure */
+		for (i = 0; tx && i < channels; i++)
+			regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0);
+
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
+				   tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
+				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0);
+
+		/* Disable and reset FIFO */
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+				   ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR);
+		regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
+				   ESAI_xFCR_xFR, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops fsl_esai_dai_ops = {
+	.startup = fsl_esai_startup,
+	.shutdown = fsl_esai_shutdown,
+	.trigger = fsl_esai_trigger,
+	.hw_params = fsl_esai_hw_params,
+	.set_sysclk = fsl_esai_set_dai_sysclk,
+	.set_fmt = fsl_esai_set_dai_fmt,
+	.set_tdm_slot = fsl_esai_set_dai_tdm_slot,
+};
+
+static int fsl_esai_dai_probe(struct snd_soc_dai *dai)
+{
+	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai, &esai_priv->dma_params_tx,
+				  &esai_priv->dma_params_rx);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver fsl_esai_dai = {
+	.probe = fsl_esai_dai_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 12,
+		.rates = FSL_ESAI_RATES,
+		.formats = FSL_ESAI_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+		.rates = FSL_ESAI_RATES,
+		.formats = FSL_ESAI_FORMATS,
+	},
+	.ops = &fsl_esai_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_esai_component = {
+	.name		= "fsl-esai",
+};
+
+static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_ESAI_ERDR:
+	case REG_ESAI_ECR:
+	case REG_ESAI_ESR:
+	case REG_ESAI_TFCR:
+	case REG_ESAI_TFSR:
+	case REG_ESAI_RFCR:
+	case REG_ESAI_RFSR:
+	case REG_ESAI_RX0:
+	case REG_ESAI_RX1:
+	case REG_ESAI_RX2:
+	case REG_ESAI_RX3:
+	case REG_ESAI_SAISR:
+	case REG_ESAI_SAICR:
+	case REG_ESAI_TCR:
+	case REG_ESAI_TCCR:
+	case REG_ESAI_RCR:
+	case REG_ESAI_RCCR:
+	case REG_ESAI_TSMA:
+	case REG_ESAI_TSMB:
+	case REG_ESAI_RSMA:
+	case REG_ESAI_RSMB:
+	case REG_ESAI_PRRC:
+	case REG_ESAI_PCRC:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case REG_ESAI_ETDR:
+	case REG_ESAI_ECR:
+	case REG_ESAI_TFCR:
+	case REG_ESAI_RFCR:
+	case REG_ESAI_TX0:
+	case REG_ESAI_TX1:
+	case REG_ESAI_TX2:
+	case REG_ESAI_TX3:
+	case REG_ESAI_TX4:
+	case REG_ESAI_TX5:
+	case REG_ESAI_TSR:
+	case REG_ESAI_SAICR:
+	case REG_ESAI_TCR:
+	case REG_ESAI_TCCR:
+	case REG_ESAI_RCR:
+	case REG_ESAI_RCCR:
+	case REG_ESAI_TSMA:
+	case REG_ESAI_TSMB:
+	case REG_ESAI_RSMA:
+	case REG_ESAI_RSMB:
+	case REG_ESAI_PRRC:
+	case REG_ESAI_PCRC:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static const struct regmap_config fsl_esai_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+
+	.max_register = REG_ESAI_PCRC,
+	.readable_reg = fsl_esai_readable_reg,
+	.writeable_reg = fsl_esai_writeable_reg,
+};
+
+static int fsl_esai_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct fsl_esai *esai_priv;
+	struct resource *res;
+	const uint32_t *iprop;
+	void __iomem *regs;
+	int irq, ret;
+
+	esai_priv = devm_kzalloc(&pdev->dev, sizeof(*esai_priv), GFP_KERNEL);
+	if (!esai_priv)
+		return -ENOMEM;
+
+	esai_priv->pdev = pdev;
+	strcpy(esai_priv->name, np->name);
+
+	/* Get the addresses and IRQ */
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	regs = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(regs))
+		return PTR_ERR(regs);
+
+	esai_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+			"core", regs, &fsl_esai_regmap_config);
+	if (IS_ERR(esai_priv->regmap)) {
+		dev_err(&pdev->dev, "failed to init regmap: %ld\n",
+				PTR_ERR(esai_priv->regmap));
+		return PTR_ERR(esai_priv->regmap);
+	}
+
+	esai_priv->coreclk = devm_clk_get(&pdev->dev, "core");
+	if (IS_ERR(esai_priv->coreclk)) {
+		dev_err(&pdev->dev, "failed to get core clock: %ld\n",
+				PTR_ERR(esai_priv->coreclk));
+		return PTR_ERR(esai_priv->coreclk);
+	}
+
+	esai_priv->extalclk = devm_clk_get(&pdev->dev, "extal");
+	if (IS_ERR(esai_priv->extalclk))
+		dev_warn(&pdev->dev, "failed to get extal clock: %ld\n",
+				PTR_ERR(esai_priv->extalclk));
+
+	esai_priv->fsysclk = devm_clk_get(&pdev->dev, "fsys");
+	if (IS_ERR(esai_priv->fsysclk))
+		dev_warn(&pdev->dev, "failed to get fsys clock: %ld\n",
+				PTR_ERR(esai_priv->fsysclk));
+
+	irq = platform_get_irq(pdev, 0);
+	if (irq < 0) {
+		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		return irq;
+	}
+
+	ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0,
+			       esai_priv->name, esai_priv);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to claim irq %u\n", irq);
+		return ret;
+	}
+
+	/* Set a default slot size */
+	esai_priv->slot_width = 32;
+
+	/* Set a default master/slave state */
+	esai_priv->slave_mode = true;
+
+	/* Determine the FIFO depth */
+	iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+	if (iprop)
+		esai_priv->fifo_depth = be32_to_cpup(iprop);
+	else
+		esai_priv->fifo_depth = 64;
+
+	esai_priv->dma_params_tx.maxburst = 16;
+	esai_priv->dma_params_rx.maxburst = 16;
+	esai_priv->dma_params_tx.addr = res->start + REG_ESAI_ETDR;
+	esai_priv->dma_params_rx.addr = res->start + REG_ESAI_ERDR;
+
+	esai_priv->synchronous =
+		of_property_read_bool(np, "fsl,esai-synchronous");
+
+	/* Implement full symmetry for synchronous mode */
+	if (esai_priv->synchronous) {
+		fsl_esai_dai.symmetric_rates = 1;
+		fsl_esai_dai.symmetric_channels = 1;
+		fsl_esai_dai.symmetric_samplebits = 1;
+	}
+
+	dev_set_drvdata(&pdev->dev, esai_priv);
+
+	/* Reset ESAI unit */
+	ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ERST);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to reset ESAI: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * We need to enable ESAI so as to access some of its registers.
+	 * Otherwise, we would fail to dump regmap from user space.
+	 */
+	ret = regmap_write(esai_priv->regmap, REG_ESAI_ECR, ESAI_ECR_ESAIEN);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to enable ESAI: %d\n", ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
+					      &fsl_esai_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+		return ret;
+	}
+
+	ret = imx_pcm_dma_init(pdev);
+	if (ret)
+		dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
+
+	return ret;
+}
+
+static const struct of_device_id fsl_esai_dt_ids[] = {
+	{ .compatible = "fsl,imx35-esai", },
+	{}
+};
+MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids);
+
+static struct platform_driver fsl_esai_driver = {
+	.probe = fsl_esai_probe,
+	.driver = {
+		.name = "fsl-esai-dai",
+		.owner = THIS_MODULE,
+		.of_match_table = fsl_esai_dt_ids,
+	},
+};
+
+module_platform_driver(fsl_esai_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale ESAI CPU DAI driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:fsl-esai-dai");
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
new file mode 100644
index 000000000000..9c9f957fcae1
--- /dev/null
+++ b/sound/soc/fsl/fsl_esai.h
@@ -0,0 +1,354 @@
+/*
+ * fsl_esai.h - ALSA ESAI interface for the Freescale i.MX SoC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <Guangyu.Chen@freescale.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#ifndef _FSL_ESAI_DAI_H
+#define _FSL_ESAI_DAI_H
+
+/* ESAI Register Map */
+#define REG_ESAI_ETDR		0x00
+#define REG_ESAI_ERDR		0x04
+#define REG_ESAI_ECR		0x08
+#define REG_ESAI_ESR		0x0C
+#define REG_ESAI_TFCR		0x10
+#define REG_ESAI_TFSR		0x14
+#define REG_ESAI_RFCR		0x18
+#define REG_ESAI_RFSR		0x1C
+#define REG_ESAI_xFCR(tx)	(tx ? REG_ESAI_TFCR : REG_ESAI_RFCR)
+#define REG_ESAI_xFSR(tx)	(tx ? REG_ESAI_TFSR : REG_ESAI_RFSR)
+#define REG_ESAI_TX0		0x80
+#define REG_ESAI_TX1		0x84
+#define REG_ESAI_TX2		0x88
+#define REG_ESAI_TX3		0x8C
+#define REG_ESAI_TX4		0x90
+#define REG_ESAI_TX5		0x94
+#define REG_ESAI_TSR		0x98
+#define REG_ESAI_RX0		0xA0
+#define REG_ESAI_RX1		0xA4
+#define REG_ESAI_RX2		0xA8
+#define REG_ESAI_RX3		0xAC
+#define REG_ESAI_SAISR		0xCC
+#define REG_ESAI_SAICR		0xD0
+#define REG_ESAI_TCR		0xD4
+#define REG_ESAI_TCCR		0xD8
+#define REG_ESAI_RCR		0xDC
+#define REG_ESAI_RCCR		0xE0
+#define REG_ESAI_xCR(tx)	(tx ? REG_ESAI_TCR : REG_ESAI_RCR)
+#define REG_ESAI_xCCR(tx)	(tx ? REG_ESAI_TCCR : REG_ESAI_RCCR)
+#define REG_ESAI_TSMA		0xE4
+#define REG_ESAI_TSMB		0xE8
+#define REG_ESAI_RSMA		0xEC
+#define REG_ESAI_RSMB		0xF0
+#define REG_ESAI_xSMA(tx)	(tx ? REG_ESAI_TSMA : REG_ESAI_RSMA)
+#define REG_ESAI_xSMB(tx)	(tx ? REG_ESAI_TSMB : REG_ESAI_RSMB)
+#define REG_ESAI_PRRC		0xF8
+#define REG_ESAI_PCRC		0xFC
+
+/* ESAI Control Register -- REG_ESAI_ECR 0x8 */
+#define ESAI_ECR_ETI_SHIFT	19
+#define ESAI_ECR_ETI_MASK	(1 << ESAI_ECR_ETI_SHIFT)
+#define ESAI_ECR_ETI		(1 << ESAI_ECR_ETI_SHIFT)
+#define ESAI_ECR_ETO_SHIFT	18
+#define ESAI_ECR_ETO_MASK	(1 << ESAI_ECR_ETO_SHIFT)
+#define ESAI_ECR_ETO		(1 << ESAI_ECR_ETO_SHIFT)
+#define ESAI_ECR_ERI_SHIFT	17
+#define ESAI_ECR_ERI_MASK	(1 << ESAI_ECR_ERI_SHIFT)
+#define ESAI_ECR_ERI		(1 << ESAI_ECR_ERI_SHIFT)
+#define ESAI_ECR_ERO_SHIFT	16
+#define ESAI_ECR_ERO_MASK	(1 << ESAI_ECR_ERO_SHIFT)
+#define ESAI_ECR_ERO		(1 << ESAI_ECR_ERO_SHIFT)
+#define ESAI_ECR_ERST_SHIFT	1
+#define ESAI_ECR_ERST_MASK	(1 << ESAI_ECR_ERST_SHIFT)
+#define ESAI_ECR_ERST		(1 << ESAI_ECR_ERST_SHIFT)
+#define ESAI_ECR_ESAIEN_SHIFT	0
+#define ESAI_ECR_ESAIEN_MASK	(1 << ESAI_ECR_ESAIEN_SHIFT)
+#define ESAI_ECR_ESAIEN		(1 << ESAI_ECR_ESAIEN_SHIFT)
+
+/* ESAI Status Register -- REG_ESAI_ESR 0xC */
+#define ESAI_ESR_TINIT_SHIFT	10
+#define ESAI_ESR_TINIT_MASK	(1 << ESAI_ESR_TINIT_SHIFT)
+#define ESAI_ESR_TINIT		(1 << ESAI_ESR_TINIT_SHIFT)
+#define ESAI_ESR_RFF_SHIFT	9
+#define ESAI_ESR_RFF_MASK	(1 << ESAI_ESR_RFF_SHIFT)
+#define ESAI_ESR_RFF		(1 << ESAI_ESR_RFF_SHIFT)
+#define ESAI_ESR_TFE_SHIFT	8
+#define ESAI_ESR_TFE_MASK	(1 << ESAI_ESR_TFE_SHIFT)
+#define ESAI_ESR_TFE		(1 << ESAI_ESR_TFE_SHIFT)
+#define ESAI_ESR_TLS_SHIFT	7
+#define ESAI_ESR_TLS_MASK	(1 << ESAI_ESR_TLS_SHIFT)
+#define ESAI_ESR_TLS		(1 << ESAI_ESR_TLS_SHIFT)
+#define ESAI_ESR_TDE_SHIFT	6
+#define ESAI_ESR_TDE_MASK	(1 << ESAI_ESR_TDE_SHIFT)
+#define ESAI_ESR_TDE		(1 << ESAI_ESR_TDE_SHIFT)
+#define ESAI_ESR_TED_SHIFT	5
+#define ESAI_ESR_TED_MASK	(1 << ESAI_ESR_TED_SHIFT)
+#define ESAI_ESR_TED		(1 << ESAI_ESR_TED_SHIFT)
+#define ESAI_ESR_TD_SHIFT	4
+#define ESAI_ESR_TD_MASK	(1 << ESAI_ESR_TD_SHIFT)
+#define ESAI_ESR_TD		(1 << ESAI_ESR_TD_SHIFT)
+#define ESAI_ESR_RLS_SHIFT	3
+#define ESAI_ESR_RLS_MASK	(1 << ESAI_ESR_RLS_SHIFT)
+#define ESAI_ESR_RLS		(1 << ESAI_ESR_RLS_SHIFT)
+#define ESAI_ESR_RDE_SHIFT	2
+#define ESAI_ESR_RDE_MASK	(1 << ESAI_ESR_RDE_SHIFT)
+#define ESAI_ESR_RDE		(1 << ESAI_ESR_RDE_SHIFT)
+#define ESAI_ESR_RED_SHIFT	1
+#define ESAI_ESR_RED_MASK	(1 << ESAI_ESR_RED_SHIFT)
+#define ESAI_ESR_RED		(1 << ESAI_ESR_RED_SHIFT)
+#define ESAI_ESR_RD_SHIFT	0
+#define ESAI_ESR_RD_MASK	(1 << ESAI_ESR_RD_SHIFT)
+#define ESAI_ESR_RD		(1 << ESAI_ESR_RD_SHIFT)
+
+/*
+ * Transmit FIFO Configuration Register -- REG_ESAI_TFCR 0x10
+ * Receive FIFO Configuration Register -- REG_ESAI_RFCR 0x18
+ */
+#define ESAI_xFCR_TIEN_SHIFT	19
+#define ESAI_xFCR_TIEN_MASK	(1 << ESAI_xFCR_TIEN_SHIFT)
+#define ESAI_xFCR_TIEN		(1 << ESAI_xFCR_TIEN_SHIFT)
+#define ESAI_xFCR_REXT_SHIFT	19
+#define ESAI_xFCR_REXT_MASK	(1 << ESAI_xFCR_REXT_SHIFT)
+#define ESAI_xFCR_REXT		(1 << ESAI_xFCR_REXT_SHIFT)
+#define ESAI_xFCR_xWA_SHIFT	16
+#define ESAI_xFCR_xWA_WIDTH	3
+#define ESAI_xFCR_xWA_MASK	(((1 << ESAI_xFCR_xWA_WIDTH) - 1) << ESAI_xFCR_xWA_SHIFT)
+#define ESAI_xFCR_xWA(v)	(((8 - ((v) >> 2)) << ESAI_xFCR_xWA_SHIFT) & ESAI_xFCR_xWA_MASK)
+#define ESAI_xFCR_xFWM_SHIFT	8
+#define ESAI_xFCR_xFWM_WIDTH	8
+#define ESAI_xFCR_xFWM_MASK	(((1 << ESAI_xFCR_xFWM_WIDTH) - 1) << ESAI_xFCR_xFWM_SHIFT)
+#define ESAI_xFCR_xFWM(v)	((((v) - 1) << ESAI_xFCR_xFWM_SHIFT) & ESAI_xFCR_xFWM_MASK)
+#define ESAI_xFCR_xE_SHIFT	2
+#define ESAI_xFCR_TE_WIDTH	6
+#define ESAI_xFCR_RE_WIDTH	4
+#define ESAI_xFCR_TE_MASK	(((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
+#define ESAI_xFCR_RE_MASK	(((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
+#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_xFR_SHIFT	1
+#define ESAI_xFCR_xFR_MASK	(1 << ESAI_xFCR_xFR_SHIFT)
+#define ESAI_xFCR_xFR		(1 << ESAI_xFCR_xFR_SHIFT)
+#define ESAI_xFCR_xFEN_SHIFT	0
+#define ESAI_xFCR_xFEN_MASK	(1 << ESAI_xFCR_xFEN_SHIFT)
+#define ESAI_xFCR_xFEN		(1 << ESAI_xFCR_xFEN_SHIFT)
+
+/*
+ * Transmit FIFO Status Register -- REG_ESAI_TFSR 0x14
+ * Receive FIFO Status Register --REG_ESAI_RFSR 0x1C
+ */
+#define ESAI_xFSR_NTFO_SHIFT	12
+#define ESAI_xFSR_NRFI_SHIFT	12
+#define ESAI_xFSR_NTFI_SHIFT	8
+#define ESAI_xFSR_NRFO_SHIFT	8
+#define ESAI_xFSR_NTFx_WIDTH	3
+#define ESAI_xFSR_NRFx_WIDTH	2
+#define ESAI_xFSR_NTFO_MASK	(((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFO_SHIFT)
+#define ESAI_xFSR_NTFI_MASK	(((1 << ESAI_xFSR_NTFx_WIDTH) - 1) << ESAI_xFSR_NTFI_SHIFT)
+#define ESAI_xFSR_NRFO_MASK	(((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFO_SHIFT)
+#define ESAI_xFSR_NRFI_MASK	(((1 << ESAI_xFSR_NRFx_WIDTH) - 1) << ESAI_xFSR_NRFI_SHIFT)
+#define ESAI_xFSR_xFCNT_SHIFT	0
+#define ESAI_xFSR_xFCNT_WIDTH	8
+#define ESAI_xFSR_xFCNT_MASK	(((1 << ESAI_xFSR_xFCNT_WIDTH) - 1) << ESAI_xFSR_xFCNT_SHIFT)
+
+/* ESAI Transmit Slot Register -- REG_ESAI_TSR 0x98 */
+#define ESAI_TSR_SHIFT		0
+#define ESAI_TSR_WIDTH		24
+#define ESAI_TSR_MASK		(((1 << ESAI_TSR_WIDTH) - 1) << ESAI_TSR_SHIFT)
+
+/* Serial Audio Interface Status Register -- REG_ESAI_SAISR 0xCC */
+#define ESAI_SAISR_TODFE_SHIFT	17
+#define ESAI_SAISR_TODFE_MASK	(1 << ESAI_SAISR_TODFE_SHIFT)
+#define ESAI_SAISR_TODFE	(1 << ESAI_SAISR_TODFE_SHIFT)
+#define ESAI_SAISR_TEDE_SHIFT	16
+#define ESAI_SAISR_TEDE_MASK	(1 << ESAI_SAISR_TEDE_SHIFT)
+#define ESAI_SAISR_TEDE		(1 << ESAI_SAISR_TEDE_SHIFT)
+#define ESAI_SAISR_TDE_SHIFT	15
+#define ESAI_SAISR_TDE_MASK	(1 << ESAI_SAISR_TDE_SHIFT)
+#define ESAI_SAISR_TDE		(1 << ESAI_SAISR_TDE_SHIFT)
+#define ESAI_SAISR_TUE_SHIFT	14
+#define ESAI_SAISR_TUE_MASK	(1 << ESAI_SAISR_TUE_SHIFT)
+#define ESAI_SAISR_TUE		(1 << ESAI_SAISR_TUE_SHIFT)
+#define ESAI_SAISR_TFS_SHIFT	13
+#define ESAI_SAISR_TFS_MASK	(1 << ESAI_SAISR_TFS_SHIFT)
+#define ESAI_SAISR_TFS		(1 << ESAI_SAISR_TFS_SHIFT)
+#define ESAI_SAISR_RODF_SHIFT	10
+#define ESAI_SAISR_RODF_MASK	(1 << ESAI_SAISR_RODF_SHIFT)
+#define ESAI_SAISR_RODF		(1 << ESAI_SAISR_RODF_SHIFT)
+#define ESAI_SAISR_REDF_SHIFT	9
+#define ESAI_SAISR_REDF_MASK	(1 << ESAI_SAISR_REDF_SHIFT)
+#define ESAI_SAISR_REDF		(1 << ESAI_SAISR_REDF_SHIFT)
+#define ESAI_SAISR_RDF_SHIFT	8
+#define ESAI_SAISR_RDF_MASK	(1 << ESAI_SAISR_RDF_SHIFT)
+#define ESAI_SAISR_RDF		(1 << ESAI_SAISR_RDF_SHIFT)
+#define ESAI_SAISR_ROE_SHIFT	7
+#define ESAI_SAISR_ROE_MASK	(1 << ESAI_SAISR_ROE_SHIFT)
+#define ESAI_SAISR_ROE		(1 << ESAI_SAISR_ROE_SHIFT)
+#define ESAI_SAISR_RFS_SHIFT	6
+#define ESAI_SAISR_RFS_MASK	(1 << ESAI_SAISR_RFS_SHIFT)
+#define ESAI_SAISR_RFS		(1 << ESAI_SAISR_RFS_SHIFT)
+#define ESAI_SAISR_IF2_SHIFT	2
+#define ESAI_SAISR_IF2_MASK	(1 << ESAI_SAISR_IF2_SHIFT)
+#define ESAI_SAISR_IF2		(1 << ESAI_SAISR_IF2_SHIFT)
+#define ESAI_SAISR_IF1_SHIFT	1
+#define ESAI_SAISR_IF1_MASK	(1 << ESAI_SAISR_IF1_SHIFT)
+#define ESAI_SAISR_IF1		(1 << ESAI_SAISR_IF1_SHIFT)
+#define ESAI_SAISR_IF0_SHIFT	0
+#define ESAI_SAISR_IF0_MASK	(1 << ESAI_SAISR_IF0_SHIFT)
+#define ESAI_SAISR_IF0		(1 << ESAI_SAISR_IF0_SHIFT)
+
+/* Serial Audio Interface Control Register -- REG_ESAI_SAICR 0xD0 */
+#define ESAI_SAICR_ALC_SHIFT	8
+#define ESAI_SAICR_ALC_MASK	(1 << ESAI_SAICR_ALC_SHIFT)
+#define ESAI_SAICR_ALC		(1 << ESAI_SAICR_ALC_SHIFT)
+#define ESAI_SAICR_TEBE_SHIFT	7
+#define ESAI_SAICR_TEBE_MASK	(1 << ESAI_SAICR_TEBE_SHIFT)
+#define ESAI_SAICR_TEBE		(1 << ESAI_SAICR_TEBE_SHIFT)
+#define ESAI_SAICR_SYNC_SHIFT	6
+#define ESAI_SAICR_SYNC_MASK	(1 << ESAI_SAICR_SYNC_SHIFT)
+#define ESAI_SAICR_SYNC		(1 << ESAI_SAICR_SYNC_SHIFT)
+#define ESAI_SAICR_OF2_SHIFT	2
+#define ESAI_SAICR_OF2_MASK	(1 << ESAI_SAICR_OF2_SHIFT)
+#define ESAI_SAICR_OF2		(1 << ESAI_SAICR_OF2_SHIFT)
+#define ESAI_SAICR_OF1_SHIFT	1
+#define ESAI_SAICR_OF1_MASK	(1 << ESAI_SAICR_OF1_SHIFT)
+#define ESAI_SAICR_OF1		(1 << ESAI_SAICR_OF1_SHIFT)
+#define ESAI_SAICR_OF0_SHIFT	0
+#define ESAI_SAICR_OF0_MASK	(1 << ESAI_SAICR_OF0_SHIFT)
+#define ESAI_SAICR_OF0		(1 << ESAI_SAICR_OF0_SHIFT)
+
+/*
+ * Transmit Control Register -- REG_ESAI_TCR 0xD4
+ * Receive Control Register -- REG_ESAI_RCR 0xDC
+ */
+#define ESAI_xCR_xLIE_SHIFT	23
+#define ESAI_xCR_xLIE_MASK	(1 << ESAI_xCR_xLIE_SHIFT)
+#define ESAI_xCR_xLIE		(1 << ESAI_xCR_xLIE_SHIFT)
+#define ESAI_xCR_xIE_SHIFT	22
+#define ESAI_xCR_xIE_MASK	(1 << ESAI_xCR_xIE_SHIFT)
+#define ESAI_xCR_xIE		(1 << ESAI_xCR_xIE_SHIFT)
+#define ESAI_xCR_xEDIE_SHIFT	21
+#define ESAI_xCR_xEDIE_MASK	(1 << ESAI_xCR_xEDIE_SHIFT)
+#define ESAI_xCR_xEDIE		(1 << ESAI_xCR_xEDIE_SHIFT)
+#define ESAI_xCR_xEIE_SHIFT	20
+#define ESAI_xCR_xEIE_MASK	(1 << ESAI_xCR_xEIE_SHIFT)
+#define ESAI_xCR_xEIE		(1 << ESAI_xCR_xEIE_SHIFT)
+#define ESAI_xCR_xPR_SHIFT	19
+#define ESAI_xCR_xPR_MASK	(1 << ESAI_xCR_xPR_SHIFT)
+#define ESAI_xCR_xPR		(1 << ESAI_xCR_xPR_SHIFT)
+#define ESAI_xCR_PADC_SHIFT	17
+#define ESAI_xCR_PADC_MASK	(1 << ESAI_xCR_PADC_SHIFT)
+#define ESAI_xCR_PADC		(1 << ESAI_xCR_PADC_SHIFT)
+#define ESAI_xCR_xFSR_SHIFT	16
+#define ESAI_xCR_xFSR_MASK	(1 << ESAI_xCR_xFSR_SHIFT)
+#define ESAI_xCR_xFSR		(1 << ESAI_xCR_xFSR_SHIFT)
+#define ESAI_xCR_xFSL_SHIFT	15
+#define ESAI_xCR_xFSL_MASK	(1 << ESAI_xCR_xFSL_SHIFT)
+#define ESAI_xCR_xFSL		(1 << ESAI_xCR_xFSL_SHIFT)
+#define ESAI_xCR_xSWS_SHIFT	10
+#define ESAI_xCR_xSWS_WIDTH	5
+#define ESAI_xCR_xSWS_MASK	(((1 << ESAI_xCR_xSWS_WIDTH) - 1) << ESAI_xCR_xSWS_SHIFT)
+#define ESAI_xCR_xSWS(s, w)	((w < 24 ? (s - w + ((w - 8) >> 2)) : (s < 32 ? 0x1e : 0x1f)) << ESAI_xCR_xSWS_SHIFT)
+#define ESAI_xCR_xMOD_SHIFT	8
+#define ESAI_xCR_xMOD_WIDTH	2
+#define ESAI_xCR_xMOD_MASK	(((1 << ESAI_xCR_xMOD_WIDTH) - 1) << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_ONDEMAND	(0x1 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_NETWORK	(0x1 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xMOD_AC97	(0x3 << ESAI_xCR_xMOD_SHIFT)
+#define ESAI_xCR_xWA_SHIFT	7
+#define ESAI_xCR_xWA_MASK	(1 << ESAI_xCR_xWA_SHIFT)
+#define ESAI_xCR_xWA		(1 << ESAI_xCR_xWA_SHIFT)
+#define ESAI_xCR_xSHFD_SHIFT	6
+#define ESAI_xCR_xSHFD_MASK	(1 << ESAI_xCR_xSHFD_SHIFT)
+#define ESAI_xCR_xSHFD		(1 << ESAI_xCR_xSHFD_SHIFT)
+#define ESAI_xCR_xE_SHIFT	0
+#define ESAI_xCR_TE_WIDTH	6
+#define ESAI_xCR_RE_WIDTH	4
+#define ESAI_xCR_TE_MASK	(((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
+#define ESAI_xCR_RE_MASK	(((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
+#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+
+/*
+ * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
+ * Receive Clock Control Register -- REG_ESAI_RCCR 0xE0
+ */
+#define ESAI_xCCR_xHCKD_SHIFT	23
+#define ESAI_xCCR_xHCKD_MASK	(1 << ESAI_xCCR_xHCKD_SHIFT)
+#define ESAI_xCCR_xHCKD		(1 << ESAI_xCCR_xHCKD_SHIFT)
+#define ESAI_xCCR_xFSD_SHIFT	22
+#define ESAI_xCCR_xFSD_MASK	(1 << ESAI_xCCR_xFSD_SHIFT)
+#define ESAI_xCCR_xFSD		(1 << ESAI_xCCR_xFSD_SHIFT)
+#define ESAI_xCCR_xCKD_SHIFT	21
+#define ESAI_xCCR_xCKD_MASK	(1 << ESAI_xCCR_xCKD_SHIFT)
+#define ESAI_xCCR_xCKD		(1 << ESAI_xCCR_xCKD_SHIFT)
+#define ESAI_xCCR_xHCKP_SHIFT	20
+#define ESAI_xCCR_xHCKP_MASK	(1 << ESAI_xCCR_xHCKP_SHIFT)
+#define ESAI_xCCR_xHCKP		(1 << ESAI_xCCR_xHCKP_SHIFT)
+#define ESAI_xCCR_xFSP_SHIFT	19
+#define ESAI_xCCR_xFSP_MASK	(1 << ESAI_xCCR_xFSP_SHIFT)
+#define ESAI_xCCR_xFSP		(1 << ESAI_xCCR_xFSP_SHIFT)
+#define ESAI_xCCR_xCKP_SHIFT	18
+#define ESAI_xCCR_xCKP_MASK	(1 << ESAI_xCCR_xCKP_SHIFT)
+#define ESAI_xCCR_xCKP		(1 << ESAI_xCCR_xCKP_SHIFT)
+#define ESAI_xCCR_xFP_SHIFT	14
+#define ESAI_xCCR_xFP_WIDTH	4
+#define ESAI_xCCR_xFP_MASK	(((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
+#define ESAI_xCCR_xFP(v)	((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
+#define ESAI_xCCR_xDC_SHIFT     9
+#define ESAI_xCCR_xDC_WIDTH	4
+#define ESAI_xCCR_xDC_MASK	(((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
+#define ESAI_xCCR_xDC(v)	((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
+#define ESAI_xCCR_xPSR_SHIFT	8
+#define ESAI_xCCR_xPSR_MASK	(1 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPSR_BYPASS	(1 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPSR_DIV8	(0 << ESAI_xCCR_xPSR_SHIFT)
+#define ESAI_xCCR_xPM_SHIFT     0
+#define ESAI_xCCR_xPM_WIDTH     8
+#define ESAI_xCCR_xPM_MASK	(((1 << ESAI_xCCR_xPM_WIDTH) - 1) << ESAI_xCCR_xPM_SHIFT)
+#define ESAI_xCCR_xPM(v)	((((v) - 1) << ESAI_xCCR_xPM_SHIFT) & ESAI_xCCR_xPM_MASK)
+
+/* Transmit Slot Mask Register A/B -- REG_ESAI_TSMA/B 0xE4 ~ 0xF0 */
+#define ESAI_xSMA_xS_SHIFT	0
+#define ESAI_xSMA_xS_WIDTH	16
+#define ESAI_xSMA_xS_MASK	(((1 << ESAI_xSMA_xS_WIDTH) - 1) << ESAI_xSMA_xS_SHIFT)
+#define ESAI_xSMA_xS(v)		((v) & ESAI_xSMA_xS_MASK)
+#define ESAI_xSMB_xS_SHIFT	0
+#define ESAI_xSMB_xS_WIDTH	16
+#define ESAI_xSMB_xS_MASK	(((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT)
+#define ESAI_xSMB_xS(v)		(((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK)
+
+/* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */
+#define ESAI_PRRC_PDC_SHIFT	0
+#define ESAI_PRRC_PDC_WIDTH	12
+#define ESAI_PRRC_PDC_MASK	(((1 << ESAI_PRRC_PDC_WIDTH) - 1) << ESAI_PRRC_PDC_SHIFT)
+#define ESAI_PRRC_PDC(v)	((v) & ESAI_PRRC_PDC_MASK)
+
+/* Port C Control Register -- REG_ESAI_PCRC 0xFC */
+#define ESAI_PCRC_PC_SHIFT	0
+#define ESAI_PCRC_PC_WIDTH	12
+#define ESAI_PCRC_PC_MASK	(((1 << ESAI_PCRC_PC_WIDTH) - 1) << ESAI_PCRC_PC_SHIFT)
+#define ESAI_PCRC_PC(v)		((v) & ESAI_PCRC_PC_MASK)
+
+#define ESAI_GPIO		0xfff
+
+/* ESAI clock source */
+#define ESAI_HCKT_FSYS		0
+#define ESAI_HCKT_EXTAL		1
+#define ESAI_HCKR_FSYS		2
+#define ESAI_HCKR_EXTAL		3
+
+/* ESAI clock divider */
+#define ESAI_TX_DIV_PSR		0
+#define ESAI_TX_DIV_PM		1
+#define ESAI_TX_DIV_FP		2
+#define ESAI_RX_DIV_PSR		3
+#define ESAI_RX_DIV_PM		4
+#define ESAI_RX_DIV_FP		5
+#endif /* _FSL_ESAI_DAI_H */
-- 
cgit v1.2.2


From 6ab08ced64a4d5ea321fd1d5ebbd2217789b040f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 Jan 2014 13:01:41 +0100
Subject: ALSA: PCM: Warn when buffer preallocation fails

The failures of buffer preallocations at driver initializations aren't
critical but it's still helpful to inform, so that user can know that
something doesn't work as expected.

For example, the recent page allocator change triggered regressions,
but developers didn't notice until recently because the driver didn't
complain.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_memory.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 01f8eafebda6..54debc07f5cb 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -51,6 +51,7 @@ static const size_t snd_minimum_buffer = 16384;
 static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t size)
 {
 	struct snd_dma_buffer *dmab = &substream->dma_buffer;
+	size_t orig_size = size;
 	int err;
 
 	do {
@@ -63,6 +64,10 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz
 		size >>= 1;
 	} while (size >= snd_minimum_buffer);
 	dmab->bytes = 0; /* tell error */
+	pr_warn("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n",
+		substream->pcm->card->number, substream->pcm->device,
+		substream->stream ? 'c' : 'p', substream->number,
+		substream->pcm->name, orig_size);
 	return 0;
 }
 
-- 
cgit v1.2.2


From 80ab8eae70e51d578ebbeb228e0f7a562471b8b7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 Jan 2014 14:20:42 +0100
Subject: ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masks

The PCI devices with DMA masks smaller than 32bit should enable
CONFIG_ZONE_DMA.  Since the recent change of page allocator, page
allocations via dma_alloc_coherent() with the limited DMA mask bits
may fail more frequently, ended up with no available buffers, when
CONFIG_ZONE_DMA isn't enabled.  With CONFIG_ZONE_DMA, the system has
much more chance to obtain such pages.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 3a0504dfeb8a..8756c8e32922 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -25,6 +25,7 @@ config SND_ALS300
 	select SND_PCM
 	select SND_AC97_CODEC
 	select SND_OPL3_LIB
+	select ZONE_DMA
 	help
 	  Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+
 
@@ -49,6 +50,7 @@ config SND_ALI5451
 	tristate "ALi M5451 PCI Audio Controller"
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for the integrated AC97 sound
 	  device on motherboards using the ALi M5451 Audio Controller
@@ -153,6 +155,7 @@ config SND_AZT3328
 	select SND_PCM
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for Aztech AZF3328 (PCI168)
 	  soundcards.
@@ -459,6 +462,7 @@ config SND_EMU10K1
 	select SND_HWDEP
 	select SND_RAWMIDI
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y to include support for Sound Blaster PCI 512, Live!,
 	  Audigy and E-mu APS (partially supported) soundcards.
@@ -474,6 +478,7 @@ config SND_EMU10K1X
 	tristate "Emu10k1X (Dell OEM Version)"
 	select SND_AC97_CODEC
 	select SND_RAWMIDI
+	select ZONE_DMA
 	help
 	  Say Y here to include support for the Dell OEM version of the
 	  Sound Blaster Live!.
@@ -507,6 +512,7 @@ config SND_ES1938
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on ESS Solo-1
 	  (ES1938, ES1946, ES1969) chips.
@@ -518,6 +524,7 @@ config SND_ES1968
 	tristate "ESS ES1968/1978 (Maestro-1/2/2E)"
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on ESS Maestro
 	  1/2/2E chips.
@@ -606,6 +613,7 @@ config SND_ICE1712
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
 	select BITREVERSE
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on the
 	  ICE1712 (Envy24) chip.
@@ -693,6 +701,7 @@ config SND_LX6464ES
 config SND_MAESTRO3
 	tristate "ESS Allegro/Maestro3"
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on ESS Maestro 3
 	  (Allegro) chips.
@@ -789,6 +798,7 @@ config SND_SIS7019
 	tristate "SiS 7019 Audio Accelerator"
 	depends on X86 && !X86_64
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for the SiS 7019 Audio Accelerator.
 
@@ -800,6 +810,7 @@ config SND_SONICVIBES
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on the S3
 	  SonicVibes chip.
@@ -811,6 +822,7 @@ config SND_TRIDENT
 	tristate "Trident 4D-Wave DX/NX; SiS 7018"
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select ZONE_DMA
 	help
 	  Say Y here to include support for soundcards based on Trident
 	  4D-Wave DX/NX or SiS 7018 chips.
-- 
cgit v1.2.2


From cbd209f41ea5f39394de5c1fe2dd9aa54a9c5744 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Jan 2014 12:40:07 +0100
Subject: ALSA: hda - Don't set indep_hp flag for old AD codecs

Some old AD codecs don't like the independent HP handling, either it
contains a single DAC (AD1981) or it mandates the mixer routing
(AD1986A).  This patch removes the indep_hp flag for such codecs.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 819d4a98abaf..7a426ed491f2 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -185,7 +185,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = {
 };
 
 
-static int ad198x_parse_auto_config(struct hda_codec *codec)
+static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp)
 {
 	struct ad198x_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
@@ -195,7 +195,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
 	codec->no_trigger_sense = 1;
 	codec->no_sticky_stream = 1;
 
-	spec->gen.indep_hp = 1;
+	spec->gen.indep_hp = indep_hp;
 	spec->gen.add_stereo_mix_input = 1;
 
 	err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0);
@@ -379,7 +379,7 @@ static int patch_ad1986a(struct hda_codec *codec)
 			   ad1986a_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, false);
 	if (err < 0) {
 		snd_hda_gen_free(codec);
 		return err;
@@ -481,7 +481,7 @@ static int patch_ad1983(struct hda_codec *codec)
 
 	spec->gen.beep_nid = 0x10;
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, false);
 	if (err < 0)
 		goto error;
 	err = ad1983_add_spdif_mux_ctl(codec);
@@ -568,7 +568,7 @@ static int patch_ad1981(struct hda_codec *codec)
 	snd_hda_pick_fixup(codec, NULL, ad1981_fixup_tbl, ad1981_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, false);
 	if (err < 0)
 		goto error;
 	err = ad1983_add_spdif_mux_ctl(codec);
@@ -894,7 +894,7 @@ static int patch_ad1988(struct hda_codec *codec)
 	snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, true);
 	if (err < 0)
 		goto error;
 	err = ad1988_add_spdif_mux_ctl(codec);
@@ -1071,7 +1071,7 @@ static int patch_ad1884(struct hda_codec *codec)
 	snd_hda_pick_fixup(codec, NULL, ad1884_fixup_tbl, ad1884_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, true);
 	if (err < 0)
 		goto error;
 	err = ad1983_add_spdif_mux_ctl(codec);
@@ -1113,7 +1113,7 @@ static int patch_ad1882(struct hda_codec *codec)
 	spec->gen.mixer_merge_nid = 0x21;
 	spec->gen.beep_nid = 0x10;
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-	err = ad198x_parse_auto_config(codec);
+	err = ad198x_parse_auto_config(codec, true);
 	if (err < 0)
 		goto error;
 	err = ad1988_add_spdif_mux_ctl(codec);
-- 
cgit v1.2.2


From 4a608b3af38c6a98d1a3269703292137156407f8 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:28:07 +0000
Subject: ASoC: alc5623: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/alc5623.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 256c364193a5..d3036283482a 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -714,17 +714,17 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
 	iface &= ~ALC5623_DAI_I2S_DL_MASK;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		iface |= ALC5623_DAI_I2S_DL_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		iface |= ALC5623_DAI_I2S_DL_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		iface |= ALC5623_DAI_I2S_DL_24;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		iface |= ALC5623_DAI_I2S_DL_32;
 		break;
 	default:
-- 
cgit v1.2.2


From 2dad2283c5c1aaef5467e4cda67110ee236d7726 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:28:27 +0000
Subject: ASoC: alc5632: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/alc5632.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 19e9f222d09c..fb001c56cf8d 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -869,14 +869,14 @@ static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream,
 	iface &= ~ALC5632_DAI_I2S_DL_MASK;
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		iface |= ALC5632_DAI_I2S_DL_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		iface |= ALC5632_DAI_I2S_DL_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		iface |= ALC5632_DAI_I2S_DL_24;
 		break;
 	default:
-- 
cgit v1.2.2


From 1b6b0dfac283635eebf92b3bbb62ae5be898cea0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:48:20 +0000
Subject: ASoC: cs42l51: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/cs42l51.c | 14 +++++---------
 1 file changed, 5 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 1e0fa3b5f79a..6e9ea8379a91 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -423,21 +423,17 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream,
 		intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		switch (params_format(params)) {
-		case SNDRV_PCM_FORMAT_S16_LE:
-		case SNDRV_PCM_FORMAT_S16_BE:
+		switch (params_width(params)) {
+		case 16:
 			fmt = CS42L51_DAC_DIF_RJ16;
 			break;
-		case SNDRV_PCM_FORMAT_S18_3LE:
-		case SNDRV_PCM_FORMAT_S18_3BE:
+		case 18:
 			fmt = CS42L51_DAC_DIF_RJ18;
 			break;
-		case SNDRV_PCM_FORMAT_S20_3LE:
-		case SNDRV_PCM_FORMAT_S20_3BE:
+		case 20:
 			fmt = CS42L51_DAC_DIF_RJ20;
 			break;
-		case SNDRV_PCM_FORMAT_S24_LE:
-		case SNDRV_PCM_FORMAT_S24_BE:
+		case 24:
 			fmt = CS42L51_DAC_DIF_RJ24;
 			break;
 		default:
-- 
cgit v1.2.2


From 0194c42a8f7e2b992558eb0bfd2274f850340782 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:49:37 +0000
Subject: ASoC: da7210: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/da7210.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 9c1231456502..85b307c24b91 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -778,17 +778,17 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 
 	dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		dai_cfg1 |= DA7210_DAI_WORD_S16_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		dai_cfg1 |= DA7210_DAI_WORD_S20_3LE;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		dai_cfg1 |= DA7210_DAI_WORD_S24_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		dai_cfg1 |= DA7210_DAI_WORD_S32_LE;
 		break;
 	default:
-- 
cgit v1.2.2


From e7610743d4f4d54c2de32ae8f28fbd50922463d3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:49:51 +0000
Subject: ASoC: da7213: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/da7213.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 4a6f1daf911f..0c77e7ad7423 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1067,17 +1067,17 @@ static int da7213_hw_params(struct snd_pcm_substream *substream,
 	u8 fs;
 
 	/* Set DAI format */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S24_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		dai_ctrl |= DA7213_DAI_WORD_LENGTH_S32_LE;
 		break;
 	default:
-- 
cgit v1.2.2


From abf82ae6a6a875ff04e882aaf4dade40d5b0a794 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:50:07 +0000
Subject: ASoC: da732x: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/da732x.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index dc0284dc9e6f..f295b6569910 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -973,17 +973,17 @@ static int da732x_hw_params(struct snd_pcm_substream *substream,
 
 	reg_aif = dai->driver->base;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		aif |= DA732X_AIF_WORD_16;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		aif |= DA732X_AIF_WORD_20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		aif |= DA732X_AIF_WORD_24;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		aif |= DA732X_AIF_WORD_32;
 		break;
 	default:
-- 
cgit v1.2.2


From 2822a9d01cc8132c82c8fad81df084f45af6a8e2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:50:21 +0000
Subject: ASoC: da9055: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/da9055.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index fc9802d1281d..52b79a487ac7 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1058,17 +1058,17 @@ static int da9055_hw_params(struct snd_pcm_substream *substream,
 	u8 aif_ctrl, fs;
 	u32 sysclk;
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		aif_ctrl = DA9055_AIF_WORD_S16_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		aif_ctrl = DA9055_AIF_WORD_S20_3LE;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		aif_ctrl = DA9055_AIF_WORD_S24_LE;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		aif_ctrl = DA9055_AIF_WORD_S32_LE;
 		break;
 	default:
-- 
cgit v1.2.2


From 359e2ae8974550c65d0d85711c9f86aa6ed215d8 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 19:50:38 +0000
Subject: ASoC: isabelle: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/isabelle.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 53b455b8c07a..5839048ec467 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -951,11 +951,11 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream,
 			ISABELLE_FS_RATE_MASK, fs_val);
 
 	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	switch (params_width(params)) {
+	case 20:
 		aif |= ISABELLE_AIF_LENGTH_20;
 		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
+	case 32:
 		aif |= ISABELLE_AIF_LENGTH_32;
 		break;
 	default:
-- 
cgit v1.2.2


From 793f77036d5ca91d0dcfff16c7ae05d9116ce34a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 20:39:22 +0000
Subject: ASoC: max98088: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/max98088.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 53d7dab4e054..ee660e2d3df3 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1233,12 +1233,12 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream,
 
        rate = params_rate(params);
 
-       switch (params_format(params)) {
-       case SNDRV_PCM_FORMAT_S16_LE:
+       switch (params_width(params)) {
+       case 16:
                snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT,
                        M98088_DAI_WS, 0);
                break;
-       case SNDRV_PCM_FORMAT_S24_LE:
+       case 24:
                snd_soc_update_bits(codec, M98088_REG_14_DAI1_FORMAT,
                        M98088_DAI_WS, M98088_DAI_WS);
                break;
-- 
cgit v1.2.2


From 7821afc4865e976c55403bdb13d798a133efc815 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 20:39:30 +0000
Subject: ASoC: max98090: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/max98090.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 0569a4c3ae00..51f9b3d16b41 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1840,8 +1840,8 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 
 	max98090->lrclk = params_rate(params);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT,
 			M98090_WS_MASK, 0);
 		break;
-- 
cgit v1.2.2


From 580ce08d5c1a96aeb0e3434bb5144defb6a334a2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 20:39:37 +0000
Subject: ASoC: max98095: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/max98095.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 67244315c721..3ba1170ebb53 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1213,12 +1213,12 @@ static int max98095_dai1_hw_params(struct snd_pcm_substream *substream,
 
 	rate = params_rate(params);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT,
 			M98095_DAI_WS, 0);
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT,
 			M98095_DAI_WS, M98095_DAI_WS);
 		break;
-- 
cgit v1.2.2


From 0058e459600c87b03aad1842474b68a7cf6211ca Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 8 Jan 2014 20:39:44 +0000
Subject: ASoC: max9850: Use params_width() rather than memory format

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/max9850.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index c5dd61785f8d..82757ebf0301 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -149,14 +149,14 @@ static int max9850_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
 	snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
 
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (params_width(params)) {
+	case 16:
 		da = 0;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
+	case 20:
 		da = 0x2;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
+	case 24:
 		da = 0x3;
 		break;
 	default:
-- 
cgit v1.2.2


From 6d0d5103bdc45242b8d02e4130fbe5a3ea9f668a Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Sat, 11 Jan 2014 14:48:30 +0100
Subject: ASoC: codec: tlv320aic32x4: Fix regmap range config

This codec driver fails to probe because it has a higher regmap
range_max value than max_register. This patch sets the range_max to the
max_register value as described in the for struct regmap_range_cfg:
"@range_max: Address of the highest register in virtual range."

Fixes: 4d208ca429ad (ASoC: tlv320aic32x4: Convert to direct regmap API usage)
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org (v3.13 if the fix misses -final)
---
 sound/soc/codecs/tlv320aic32x4.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 18cdcca9014c..6941fa9baf6a 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -268,7 +268,7 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
 		.window_start = 0,
 		.window_len = 128,
 		.range_min = AIC32X4_PAGE1,
-		.range_max = AIC32X4_PAGE1 + 127,
+		.range_max = AIC32X4_RMICPGAVOL,
 	},
 };
 
-- 
cgit v1.2.2


From dfc6e469b6d1ee5e8c71fa398b7b58fbb600dad8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Jan 2014 16:09:57 +0100
Subject: ALSA: hda - Apply codec power_filter to FG nodes

Apply the codec->power_filter to the FG nodes in general for reducing
hackish set_power_state ops override in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 12 +++++++++---
 sound/pci/hda/hda_generic.c    |  5 +++--
 sound/pci/hda/hda_generic.h    |  3 +++
 sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++------------------------
 4 files changed, 27 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ab2a991d916f..ec4536c8d8d4 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3927,6 +3927,8 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
 					     hda_nid_t nid,
 					     unsigned int power_state)
 {
+	if (nid == codec->afg || nid == codec->mfg)
+		return power_state;
 	if (power_state == AC_PWRST_D3 &&
 	    get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_PIN &&
 	    (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) {
@@ -3965,9 +3967,13 @@ static unsigned int hda_set_power_state(struct hda_codec *codec,
 			codec->patch_ops.set_power_state(codec, fg,
 							 power_state);
 		else {
-			snd_hda_codec_read(codec, fg, flags,
-					   AC_VERB_SET_POWER_STATE,
-					   power_state);
+			state = power_state;
+			if (codec->power_filter)
+				state = codec->power_filter(codec, fg, state);
+			if (state == power_state || power_state != AC_PWRST_D3)
+				snd_hda_codec_read(codec, fg, flags,
+						   AC_VERB_SET_POWER_STATE,
+						   state);
 			snd_hda_codec_set_power_to_all(codec, fg, power_state);
 		}
 		state = hda_sync_power_state(codec, fg, power_state);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 7513089c8985..8321a97d5c05 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -4312,11 +4312,11 @@ static int check_auto_mic_availability(struct hda_codec *codec)
 }
 
 /* power_filter hook; make inactive widgets into power down */
-static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
+unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 						  hda_nid_t nid,
 						  unsigned int power_state)
 {
-	if (power_state != AC_PWRST_D0)
+	if (power_state != AC_PWRST_D0 || nid == codec->afg)
 		return power_state;
 	if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER)
 		return power_state;
@@ -4324,6 +4324,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 		return power_state;
 	return AC_PWRST_D3;
 }
+EXPORT_SYMBOL_GPL(snd_hda_gen_path_power_filter);
 
 /* mute all aamix inputs initially; parse up to the first leaves */
 static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 0929a06df812..07f767231c9f 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -335,5 +335,8 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec);
 #ifdef CONFIG_PM
 int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid);
 #endif
+unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
+					   hda_nid_t nid,
+					   unsigned int power_state);
 
 #endif /* __SOUND_HDA_GENERIC_H */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 088a5afbd1b9..6998cf29b9bc 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -368,6 +368,17 @@ static int stac_vrefout_set(struct hda_codec *codec,
 	return 1;
 }
 
+/* prevent codec AFG to D3 state when vref-out pin is used for mute LED */
+/* this hook is set in stac_setup_gpio() */
+static unsigned int stac_vref_led_power_filter(struct hda_codec *codec,
+					       hda_nid_t nid,
+					       unsigned int power_state)
+{
+	if (nid == codec->afg && power_state == AC_PWRST_D3)
+		return AC_PWRST_D1;
+	return snd_hda_gen_path_power_filter(codec, nid, power_state);
+}
+
 /* update mute-LED accoring to the master switch */
 static void stac_update_led_status(struct hda_codec *codec, int enabled)
 {
@@ -4260,30 +4271,8 @@ static int stac_suspend(struct hda_codec *codec)
 	stac_shutup(codec);
 	return 0;
 }
-
-static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg,
-				 unsigned int power_state)
-{
-	unsigned int afg_power_state = power_state;
-	struct sigmatel_spec *spec = codec->spec;
-
-	if (power_state == AC_PWRST_D3) {
-		if (spec->vref_mute_led_nid) {
-			/* with vref-out pin used for mute led control
-			 * codec AFG is prevented from D3 state
-			 */
-			afg_power_state = AC_PWRST_D1;
-		}
-		/* this delay seems necessary to avoid click noise at power-down */
-		msleep(100);
-	}
-	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
-			afg_power_state);
-	snd_hda_codec_set_power_to_all(codec, fg, power_state);
-}
 #else
 #define stac_suspend		NULL
-#define stac_set_power_state	NULL
 #endif /* CONFIG_PM */
 
 static const struct hda_codec_ops stac_patch_ops = {
@@ -4466,8 +4455,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
 			spec->gpio_dir |= spec->gpio_led;
 			spec->gpio_data |= spec->gpio_led;
 		} else {
-			codec->patch_ops.set_power_state =
-					stac_set_power_state;
+			codec->power_filter = stac_vref_led_power_filter;
 		}
 	}
 
-- 
cgit v1.2.2


From ba194a4de5c81ee200b6af88743b26f7b69aa659 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 13 Jan 2014 17:08:08 +0800
Subject: ASoC: simple-card: use snd_soc_card_set/get_drvdata

Remove asoc_simple_get_card_info macro and use snd_soc_card_set_drvdata
and snd_soc_card_get_drvdata instead.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 11030a63b811..5528dd6a4e4e 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -14,9 +14,6 @@
 #include <linux/module.h>
 #include <sound/simple_card.h>
 
-#define asoc_simple_get_card_info(p) \
-	container_of(p->dai_link, struct asoc_simple_card_info, snd_link)
-
 static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 				       struct asoc_simple_dai *set,
 				       unsigned int daifmt)
@@ -41,7 +38,8 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
 
 static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd);
+	struct asoc_simple_card_info *info =
+				snd_soc_card_get_drvdata(rtd->card);
 	struct snd_soc_dai *codec = rtd->codec_dai;
 	struct snd_soc_dai *cpu = rtd->cpu_dai;
 	unsigned int daifmt = info->daifmt;
@@ -256,6 +254,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	cinfo->snd_card.dai_link	= &cinfo->snd_link;
 	cinfo->snd_card.num_links	= 1;
 
+	snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo);
+
 	return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card);
 }
 
-- 
cgit v1.2.2


From 493a52a9b6645f61954580c7d4bd52fa62110934 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Tue, 14 Jan 2014 14:07:36 +0800
Subject: ALSA: hda - automute via amp instead of pinctl on some AIO models

On some AIO (All In One) models with the codec alc668
(Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack,
the system will switch the output to headphone and set the speaker to
automute as well as change the speaker Pin-ctls from 0x40 to 0x00,
this will bring loud noise to the headphone.

I tried to disable the corresponding EAPD, but it did not help to
eliminate the noise.

According to Takashi's suggestion, we use amp operation to replace the
pinctl modification for the automute, this really eliminate the noise.

BugLink: https://bugs.launchpad.net/bugs/1268468
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++---
 1 file changed, 19 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d35cf3dc6711..a1c16457880e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3580,6 +3580,15 @@ static void alc_fixup_headset_mode_alc255(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 }
 
+static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		struct alc_spec *spec = codec->spec;
+		spec->gen.auto_mute_via_amp = 1;
+	}
+}
+
 static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -4839,6 +4848,7 @@ enum {
 	ALC662_FIXUP_BASS_CHMAP,
 	ALC662_FIXUP_BASS_1A,
 	ALC662_FIXUP_BASS_1A_CHMAP,
+	ALC668_FIXUP_AUTO_MUTE,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -4999,6 +5009,12 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic_0x12,
 	},
+	[ALC668_FIXUP_AUTO_MUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_auto_mute_via_amp,
+		.chained = true,
+		.chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE
+	},
 	[ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5044,11 +5060,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE),
+	SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE),
 	SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
-- 
cgit v1.2.2


From 11e424e88bd493b5d55d73d96c82bd889002ef30 Mon Sep 17 00:00:00 2001
From: Eduard Gilmutdinov <edgilmutdinov@gmail.com>
Date: Fri, 20 Dec 2013 14:06:58 +0600
Subject: ALSA: usb-audio: Add support for Focusrite Saffire 6 USB

Signed-off-by: Eduard Gilmutdinov <edgilmutdinov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++
 sound/usb/quirks.c       |  9 +++++----
 2 files changed, 45 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index f5f0595ef9c7..34547800b0f7 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -2569,6 +2569,46 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.type = QUIRK_MIDI_NOVATION
 	}
 },
+{
+	USB_DEVICE(0x1235, 0x0010),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Focusrite",
+		.product_name = "Saffire 6 USB",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
+					.channels = 4,
+					.iface = 0,
+					.altsetting = 1,
+					.altset_idx = 1,
+					.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
+					.endpoint = 0x01,
+					.ep_attr = USB_ENDPOINT_XFER_ISOC,
+					.rates = SNDRV_PCM_RATE_44100 |
+						 SNDRV_PCM_RATE_48000,
+					.rate_min = 44100,
+					.rate_max = 48000,
+					.nr_rates = 2,
+					.rate_table = (unsigned int[]) {
+						44100, 48000
+					}
+				}
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_MIDI_RAW_BYTES
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Access Music devices */
 {
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index ede753685491..89730707614c 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -674,8 +674,9 @@ static int snd_usb_gamecon780_boot_quirk(struct usb_device *dev)
 
 /*
  * Novation Twitch DJ controller
+ * Focusrite Novation Saffire 6 USB audio card
  */
-static int snd_usb_twitch_boot_quirk(struct usb_device *dev)
+static int snd_usb_novation_boot_quirk(struct usb_device *dev)
 {
 	/* preemptively set up the device because otherwise the
 	 * raw MIDI endpoints are not active */
@@ -984,9 +985,9 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 		/* Digidesign Mbox 2 */
 		return snd_usb_mbox2_boot_quirk(dev);
 
-	case USB_ID(0x1235, 0x0018):
-		/* Focusrite Novation Twitch */
-		return snd_usb_twitch_boot_quirk(dev);
+	case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */
+	case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */
+		return snd_usb_novation_boot_quirk(dev);
 
 	case USB_ID(0x133e, 0x0815):
 		/* Access Music VirusTI Desktop */
-- 
cgit v1.2.2


From 8c4b79cf216f467faacdfa7c6c40b6b85cc05782 Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@ivitera.com>
Date: Tue, 14 Jan 2014 13:42:57 +0100
Subject: ALSA: usb-audio: Fix Creative VF0420 rate

Creative Live! Cam Vista IM (VF0420) reports rate of 16kHz while working
at 8kHz. The patch adds its USB ID to the existing quirk.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/format.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/format.c b/sound/usb/format.c
index 3525231c6b97..d244fd3703d8 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -189,8 +189,10 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 			     chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
-			/* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */
-			if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068))
+			/* Creative VF0420/VF0470 Live Cams report 16 kHz instead of 8kHz */
+			if (rate == 16000 &&
+			    (chip->usb_id == USB_ID(0x041e, 0x4064) ||
+			     chip->usb_id == USB_ID(0x041e, 0x4068)))
 				rate = 8000;
 
 			fp->rate_table[fp->nr_rates] = rate;
-- 
cgit v1.2.2


From 358b7dfa1c32dfb77ff3261d244991a7c7c6d2cb Mon Sep 17 00:00:00 2001
From: Daniel Mack <zonque@gmail.com>
Date: Tue, 14 Jan 2014 14:37:56 +0100
Subject: ALSA: snd-usb: re-order some quirk entries

No code change, just a cosmetic cleanup to keep entries ordered by the
device ID within a block of unique vendor IDs.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 170 +++++++++++++++++++++++------------------------
 1 file changed, 84 insertions(+), 86 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 34547800b0f7..f652b10ce905 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -72,22 +72,21 @@
 	}
 },
 
-/* Creative/Toshiba Multimedia Center SB-0500 */
+/* Creative/E-Mu devices */
 {
-	USB_DEVICE(0x041e, 0x3048),
+	USB_DEVICE(0x041e, 0x3010),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Toshiba",
-		.product_name = "SB-0500",
+		.vendor_name = "Creative Labs",
+		.product_name = "Sound Blaster MP3+",
 		.ifnum = QUIRK_NO_INTERFACE
 	}
 },
-
-/* Creative/E-Mu devices */
+/* Creative/Toshiba Multimedia Center SB-0500 */
 {
-	USB_DEVICE(0x041e, 0x3010),
+	USB_DEVICE(0x041e, 0x3048),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Creative Labs",
-		.product_name = "Sound Blaster MP3+",
+		.vendor_name = "Toshiba",
+		.product_name = "SB-0500",
 		.ifnum = QUIRK_NO_INTERFACE
 	}
 },
@@ -2521,17 +2520,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE(0x1235, 0x0018),
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Novation",
-		.product_name = "Twitch",
+	USB_DEVICE(0x1235, 0x0010),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Focusrite",
+		.product_name = "Saffire 6 USB",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_COMPOSITE,
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = & (const struct audioformat) {
+				.data = &(const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
 					.channels = 4,
 					.iface = 0,
@@ -2561,26 +2560,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661),
+	USB_DEVICE(0x1235, 0x0018),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		.vendor_name = "Novation",
-		.product_name = "ReMOTE25",
-		.ifnum = 0,
-		.type = QUIRK_MIDI_NOVATION
-	}
-},
-{
-	USB_DEVICE(0x1235, 0x0010),
-	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
-		.vendor_name = "Focusrite",
-		.product_name = "Saffire 6 USB",
+		.product_name = "Twitch",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_COMPOSITE,
 		.data = (const struct snd_usb_audio_quirk[]) {
 			{
 				.ifnum = 0,
 				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
-				.data = &(const struct audioformat) {
+				.data = & (const struct audioformat) {
 					.formats = SNDRV_PCM_FMTBIT_S24_3LE,
 					.channels = 4,
 					.iface = 0,
@@ -2609,6 +2599,66 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Novation",
+		.product_name = "ReMOTE25",
+		.ifnum = 0,
+		.type = QUIRK_MIDI_NOVATION
+	}
+},
+{
+	/*
+	 * Focusrite Scarlett 18i6
+	 *
+	 * Avoid mixer creation, which otherwise fails because some of
+	 * the interface descriptor subtypes for interface 0 are
+	 * unknown.  That should be fixed or worked-around but this at
+	 * least allows the device to be used successfully with a DAW
+	 * and an external mixer.  See comments below about other
+	 * ignored interfaces.
+	 */
+	USB_DEVICE(0x1235, 0x8004),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Focusrite",
+		.product_name = "Scarlett 18i6",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = & (const struct snd_usb_audio_quirk[]) {
+			{
+				/* InterfaceSubClass 1 (Control Device) */
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				/* InterfaceSubClass 1 (Control Device) */
+				.ifnum = 3,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 4,
+				.type = QUIRK_MIDI_STANDARD_INTERFACE
+			},
+			{
+				/* InterfaceSubClass 1 (Device Firmware Update) */
+				.ifnum = 5,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Access Music devices */
 {
@@ -2711,7 +2761,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2719,13 +2769,13 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "Hauppauge",
-		.product_name = "HVR-850",
+		.product_name = "HVR-950Q",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_AUDIO_ALIGN_TRANSFER,
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2739,7 +2789,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2753,7 +2803,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2767,7 +2817,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2781,7 +2831,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f),
+	USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240),
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
 		       USB_DEVICE_ID_MATCH_INT_CLASS |
 		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
@@ -2789,7 +2839,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "Hauppauge",
-		.product_name = "HVR-950Q",
+		.product_name = "HVR-850",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_AUDIO_ALIGN_TRANSFER,
 	}
@@ -3092,58 +3142,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
-{
-	/*
-	 * Focusrite Scarlett 18i6
-	 *
-	 * Avoid mixer creation, which otherwise fails because some of
-	 * the interface descriptor subtypes for interface 0 are
-	 * unknown.  That should be fixed or worked-around but this at
-	 * least allows the device to be used successfully with a DAW
-	 * and an external mixer.  See comments below about other
-	 * ignored interfaces.
-	 */
-	USB_DEVICE(0x1235, 0x8004),
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Focusrite",
-		.product_name = "Scarlett 18i6",
-		.ifnum = QUIRK_ANY_INTERFACE,
-		.type = QUIRK_COMPOSITE,
-		.data = & (const struct snd_usb_audio_quirk[]) {
-			{
-				/* InterfaceSubClass 1 (Control Device) */
-				.ifnum = 0,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = 1,
-				.type = QUIRK_AUDIO_STANDARD_INTERFACE
-			},
-			{
-				.ifnum = 2,
-				.type = QUIRK_AUDIO_STANDARD_INTERFACE
-			},
-			{
-				/* InterfaceSubClass 1 (Control Device) */
-				.ifnum = 3,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = 4,
-				.type = QUIRK_MIDI_STANDARD_INTERFACE
-			},
-			{
-				/* InterfaceSubClass 1 (Device Firmware Update) */
-				.ifnum = 5,
-				.type = QUIRK_IGNORE_INTERFACE
-			},
-			{
-				.ifnum = -1
-			}
-		}
-	}
-},
-
 {
 	/*
 	 * Some USB MIDI devices don't have an audio control interface,
-- 
cgit v1.2.2


From cf67c8e71bc15cf6b1843ab88c31cf732f2f2ff0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 14 Jan 2014 14:56:55 +0100
Subject: ALSA: hda - Fix endless vmaster hook call in thinkpad_helper.c

The new vmaster hook, update_tpacpi_mute_led(), calls the original
vmaster hook, but I forgot to save the original hook function but keep
calling the updated one, which of course results in a stupid endless
loop.  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/thinkpad_helper.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 8492b8a3ce47..5799fbc24c28 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -8,6 +8,7 @@
 #include <linux/thinkpad_acpi.h>
 
 static int (*led_set_func)(int, bool);
+static void (*old_vmaster_hook)(void *, int);
 
 static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
 				 void **rv)
@@ -30,11 +31,8 @@ static bool is_thinkpad(struct hda_codec *codec)
 
 static void update_tpacpi_mute_led(void *private_data, int enabled)
 {
-	struct hda_codec *codec = private_data;
-	struct hda_gen_spec *spec = codec->spec;
-
-	if (spec->vmaster_mute.hook)
-		spec->vmaster_mute.hook(private_data, enabled);
+	if (old_vmaster_hook)
+		old_vmaster_hook(private_data, enabled);
 
 	if (led_set_func)
 		led_set_func(TPACPI_LED_MUTE, !enabled);
@@ -70,6 +68,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
 
 		removefunc = true;
 		if (led_set_func(TPACPI_LED_MUTE, false) >= 0) {
+			old_vmaster_hook = spec->vmaster_mute.hook;
 			spec->vmaster_mute.hook = update_tpacpi_mute_led;
 			removefunc = false;
 		}
@@ -86,6 +85,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
 	if (led_set_func && (action == HDA_FIXUP_ACT_FREE || removefunc)) {
 		symbol_put(tpacpi_led_set);
 		led_set_func = NULL;
+		old_vmaster_hook = NULL;
 	}
 }
 
-- 
cgit v1.2.2


From 1a21576562507b2b10b9aa53555857868fe51c94 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 2 Jan 2014 13:07:47 +0100
Subject: sound: oss: msnd_pinnacle: avoid interruptible_sleep_on_timeout

We want to remove all sleep_on variants from the kernel because they are
racy. In case of the pinnacle driver, we can replace
interruptible_sleep_on_timeout with wait_event_interruptible_timeout
by changing the meaning of a few flags used in the driver so they
are cleared at wakeup time, which is a somewhat more appropriate
way to do the same, although probably still racy.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/msnd_pinnacle.c | 31 ++++++++++++++++++-------------
 1 file changed, 18 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c
index 11ff7c55240c..c23f9f95bfa5 100644
--- a/sound/oss/msnd_pinnacle.c
+++ b/sound/oss/msnd_pinnacle.c
@@ -664,12 +664,15 @@ static long dev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 
 static void dsp_write_flush(void)
 {
+	int timeout = get_play_delay_jiffies(dev.DAPF.len);
+
 	if (!(dev.mode & FMODE_WRITE) || !test_bit(F_WRITING, &dev.flags))
 		return;
 	set_bit(F_WRITEFLUSH, &dev.flags);
-	interruptible_sleep_on_timeout(
-		&dev.writeflush,
-		get_play_delay_jiffies(dev.DAPF.len));
+	wait_event_interruptible_timeout(
+		dev.writeflush,
+		!test_bit(F_WRITEFLUSH, &dev.flags),
+		timeout);
 	clear_bit(F_WRITEFLUSH, &dev.flags);
 	if (!signal_pending(current)) {
 		current->state = TASK_INTERRUPTIBLE;
@@ -897,6 +900,7 @@ static int dsp_read(char __user *buf, size_t len)
 {
 	int count = len;
 	char *page = (char *)__get_free_page(GFP_KERNEL);
+	int timeout = get_rec_delay_jiffies(DAR_BUFF_SIZE);
 
 	if (!page)
 		return -ENOMEM;
@@ -936,11 +940,11 @@ static int dsp_read(char __user *buf, size_t len)
 
 		if (count > 0) {
 			set_bit(F_READBLOCK, &dev.flags);
-			if (!interruptible_sleep_on_timeout(
-				&dev.readblock,
-				get_rec_delay_jiffies(DAR_BUFF_SIZE)))
+			if (wait_event_interruptible_timeout(
+					dev.readblock,
+					test_bit(F_READBLOCK, &dev.flags),
+					timeout) <= 0)
 				clear_bit(F_READING, &dev.flags);
-			clear_bit(F_READBLOCK, &dev.flags);
 			if (signal_pending(current)) {
 				free_page((unsigned long)page);
 				return -EINTR;
@@ -955,6 +959,7 @@ static int dsp_write(const char __user *buf, size_t len)
 {
 	int count = len;
 	char *page = (char *)__get_free_page(GFP_KERNEL);
+	int timeout = get_play_delay_jiffies(DAP_BUFF_SIZE);
 
 	if (!page)
 		return -ENOMEM;
@@ -995,10 +1000,10 @@ static int dsp_write(const char __user *buf, size_t len)
 
 		if (count > 0) {
 			set_bit(F_WRITEBLOCK, &dev.flags);
-			interruptible_sleep_on_timeout(
-				&dev.writeblock,
-				get_play_delay_jiffies(DAP_BUFF_SIZE));
-			clear_bit(F_WRITEBLOCK, &dev.flags);
+			wait_event_interruptible_timeout(
+				dev.writeblock,
+				test_bit(F_WRITEBLOCK, &dev.flags),
+				timeout);
 			if (signal_pending(current)) {
 				free_page((unsigned long)page);
 				return -EINTR;
@@ -1044,7 +1049,7 @@ static __inline__ void eval_dsp_msg(register WORD wMessage)
 			clear_bit(F_WRITING, &dev.flags);
 		}
 
-		if (test_bit(F_WRITEBLOCK, &dev.flags))
+		if (test_and_clear_bit(F_WRITEBLOCK, &dev.flags))
 			wake_up_interruptible(&dev.writeblock);
 		break;
 
@@ -1055,7 +1060,7 @@ static __inline__ void eval_dsp_msg(register WORD wMessage)
 
 		pack_DARQ_to_DARF(dev.last_recbank);
 
-		if (test_bit(F_READBLOCK, &dev.flags))
+		if (test_and_clear_bit(F_READBLOCK, &dev.flags))
 			wake_up_interruptible(&dev.readblock);
 		break;
 
-- 
cgit v1.2.2


From 7bd6972a921e148beea54919a00aa7b0bf046ff1 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 2 Jan 2014 13:07:49 +0100
Subject: sound: oss: vwsnd: avoid interruptible_sleep_on

Interruptible_sleep_on is racy and we want to remove it. This replaces
the use in the vwsnd driver with an open-coded prepare_to_wait
loop that fixes the race between concurrent open() and close() calls,
and also drops the global mutex while waiting here, which restores
the original behavior that was changed during the BKL removal.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/vwsnd.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
index 4bbcc0fcd4eb..a077e9c69a5e 100644
--- a/sound/oss/vwsnd.c
+++ b/sound/oss/vwsnd.c
@@ -2921,6 +2921,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file)
 	vwsnd_dev_t *devc;
 	int minor = iminor(inode);
 	int sw_samplefmt;
+	DEFINE_WAIT(wait);
 
 	DBGE("(inode=0x%p, file=0x%p)\n", inode, file);
 
@@ -2937,21 +2938,26 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file)
 	}
 
 	mutex_lock(&devc->open_mutex);
-	while (devc->open_mode & file->f_mode) {
+	while (1) {
+		prepare_to_wait(&devc->open_wait, &wait, TASK_INTERRUPTIBLE);
+		if (!(devc->open_mode & file->f_mode))
+			break;
+
 		mutex_unlock(&devc->open_mutex);
+		mutex_unlock(&vwsnd_mutex);
 		if (file->f_flags & O_NONBLOCK) {
 			DEC_USE_COUNT;
-			mutex_unlock(&vwsnd_mutex);
 			return -EBUSY;
 		}
-		interruptible_sleep_on(&devc->open_wait);
+		schedule();
 		if (signal_pending(current)) {
 			DEC_USE_COUNT;
-			mutex_unlock(&vwsnd_mutex);
 			return -ERESTARTSYS;
 		}
+		mutex_lock(&vwsnd_mutex);
 		mutex_lock(&devc->open_mutex);
 	}
+	finish_wait(&devc->open_wait, &wait);
 	devc->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE);
 	mutex_unlock(&devc->open_mutex);
 
-- 
cgit v1.2.2


From 76439c2ac686c547ca2f53bfe964c100e697ff4a Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 2 Jan 2014 13:07:48 +0100
Subject: sound: oss: midibuf: fix sleep_on races

sleep_on is known to be racy and going away because of this. All instances
of interruptible_sleep_on and interruptible_sleep_on_timeout in the midibuf
driver can trivially be replaced with wait_event_interruptible and
wait_event_interruptible_timeout.

[fixed coding style warnings by tiwai]

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/midibuf.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index 8cdb2cfe65c8..8f45cd999965 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -86,9 +86,8 @@ static void drain_midi_queue(int dev)
 	 */
 
 	if (midi_devs[dev]->buffer_status != NULL)
-		while (!signal_pending(current) && midi_devs[dev]->buffer_status(dev)) 
-			interruptible_sleep_on_timeout(&midi_sleeper[dev],
-						       HZ/10);
+		wait_event_interruptible_timeout(midi_sleeper[dev],
+				!midi_devs[dev]->buffer_status(dev), HZ/10);
 }
 
 static void midi_input_intr(int dev, unsigned char data)
@@ -233,8 +232,8 @@ void MIDIbuf_release(int dev, struct file *file)
 							   * devices
 							 */
 
-		while (!signal_pending(current) && DATA_AVAIL(midi_out_buf[dev]))
-			  interruptible_sleep_on(&midi_sleeper[dev]);
+		wait_event_interruptible(midi_sleeper[dev],
+					 !DATA_AVAIL(midi_out_buf[dev]));
 		/*
 		 *	Sync
 		 */
@@ -282,8 +281,8 @@ int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count)
 				goto out;
 			}
 
-			interruptible_sleep_on(&midi_sleeper[dev]);
-			if (signal_pending(current)) 
+			if (wait_event_interruptible(midi_sleeper[dev],
+						SPACE_AVAIL(midi_out_buf[dev])))
 			{
 				c = -EINTR;
 				goto out;
@@ -325,8 +324,9 @@ int MIDIbuf_read(int dev, struct file *file, char __user *buf, int count)
  			c = -EAGAIN;
 			goto out;
  		}
-		interruptible_sleep_on_timeout(&input_sleeper[dev],
-					       parms[dev].prech_timeout);
+		wait_event_interruptible_timeout(input_sleeper[dev],
+						 DATA_AVAIL(midi_in_buf[dev]),
+						 parms[dev].prech_timeout);
 
 		if (signal_pending(current))
 			c = -EINTR;	/* The user is getting restless */
-- 
cgit v1.2.2


From 1a1e0a80ceb766852e8abd5d4c3d9475611a7d85 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 2 Jan 2014 13:07:50 +0100
Subject: sound: oss: dmasound: kill SLEEP() macro to avoid race

The use of interruptible_sleep_on_timeout in the dmasound driver
is questionable and we want to kill off all sleep_on variants.
This replaces the calls with wait_event_interruptible_timeout
where possible, to wait for a particular event instead of blocking
in a racy way. In the sq_write function, the easiest solution is
an open-coded prepare_to_wait loop.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/dmasound/dmasound.h      |  1 -
 sound/oss/dmasound/dmasound_core.c | 28 +++++++++++++++++++---------
 2 files changed, 19 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h
index 1308d8d34186..01019f06fa91 100644
--- a/sound/oss/dmasound/dmasound.h
+++ b/sound/oss/dmasound/dmasound.h
@@ -239,7 +239,6 @@ struct sound_queue {
     int busy, syncing, xruns, died;
 };
 
-#define SLEEP(queue)		interruptible_sleep_on_timeout(&queue, HZ)
 #define WAKE_UP(queue)		(wake_up_interruptible(&queue))
 
 extern struct sound_queue dmasound_write_sq;
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index bac43b5b6e95..f4ee85a4c42f 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -619,15 +619,27 @@ static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft,
 	}
 
 	while (uLeft) {
+		DEFINE_WAIT(wait);
+
 		while (write_sq.count >= write_sq.max_active) {
+			prepare_to_wait(&write_sq.action_queue, &wait, TASK_INTERRUPTIBLE);
 			sq_play();
-			if (write_sq.non_blocking)
+			if (write_sq.non_blocking) {
+				finish_wait(&write_sq.action_queue, &wait);
 				return uWritten > 0 ? uWritten : -EAGAIN;
-			SLEEP(write_sq.action_queue);
-			if (signal_pending(current))
+			}
+			if (write_sq.count < write_sq.max_active)
+				break;
+
+			schedule_timeout(HZ);
+			if (signal_pending(current)) {
+				finish_wait(&write_sq.action_queue, &wait);
 				return uWritten > 0 ? uWritten : -EINTR;
+			}
 		}
 
+		finish_wait(&write_sq.action_queue, &wait);
+
 		/* Here, we can avoid disabling the interrupt by first
 		 * copying and translating the data, and then updating
 		 * the write_sq variables. Until this is done, the interrupt
@@ -707,11 +719,8 @@ static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode,
 			if (file->f_flags & O_NONBLOCK)
 				return rc;
 			rc = -EINTR;
-			while (sq->busy) {
-				SLEEP(sq->open_queue);
-				if (signal_pending(current))
-					return rc;
-			}
+			if (wait_event_interruptible(sq->open_queue, !sq->busy))
+				return rc;
 			rc = 0;
 #else
 			/* OSS manual says we will return EBUSY regardless
@@ -844,7 +853,8 @@ static int sq_fsync(void)
 	sq_play();	/* there may be an incomplete frame waiting */
 
 	while (write_sq.active) {
-		SLEEP(write_sq.sync_queue);
+		wait_event_interruptible_timeout(write_sq.sync_queue,
+						 !write_sq.active, HZ);
 		if (signal_pending(current)) {
 			/* While waiting for audio output to drain, an
 			 * interrupt occurred.  Stop audio output immediately
-- 
cgit v1.2.2


From cdef2e5f358db5fd27eec4c72eb295c232287813 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 2 Jan 2014 13:07:51 +0100
Subject: sound: oss: remove last sleep_on users

There are three files in oss for which I could not find an easy way to
replace interruptible_sleep_on_timeout with a non-racy version. This
patch instead just adds a private implementation of the function, now
named oss_broken_sleep_on, and changes over the remaining users in
sound/oss/ so we can remove the global interface.

[fixed coding style warnings by tiwai]

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/dmabuf.c        | 14 ++++++--------
 sound/oss/sequencer.c     | 16 +++++++---------
 sound/oss/sleep.h         | 18 ++++++++++++++++++
 sound/oss/swarm_cs4297a.c | 14 ++++++++------
 4 files changed, 39 insertions(+), 23 deletions(-)
 create mode 100644 sound/oss/sleep.h

(limited to 'sound')

diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index 461d94cfecbe..e3f29132d3ac 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -28,6 +28,7 @@
 #include <linux/mm.h>
 #include <linux/gfp.h>
 #include "sound_config.h"
+#include "sleep.h"
 
 #define DMAP_FREE_ON_CLOSE      0
 #define DMAP_KEEP_ON_CLOSE      1
@@ -351,8 +352,7 @@ static void dma_reset_output(int dev)
 	if (!signal_pending(current) && adev->dmap_out->qlen && 
 	    adev->dmap_out->underrun_count == 0){
 		spin_unlock_irqrestore(&dmap->lock,flags);
-		interruptible_sleep_on_timeout(&adev->out_sleeper,
-					       dmabuf_timeout(dmap));
+		oss_broken_sleep_on(&adev->out_sleeper, dmabuf_timeout(dmap));
 		spin_lock_irqsave(&dmap->lock,flags);
 	}
 	adev->dmap_out->flags &= ~(DMA_SYNCING | DMA_ACTIVE);
@@ -446,7 +446,7 @@ int DMAbuf_sync(int dev)
 			long t = dmabuf_timeout(dmap);
 			spin_unlock_irqrestore(&dmap->lock,flags);
 			/* FIXME: not safe may miss events */
-			t = interruptible_sleep_on_timeout(&adev->out_sleeper, t);
+			t = oss_broken_sleep_on(&adev->out_sleeper, t);
 			spin_lock_irqsave(&dmap->lock,flags);
 			if (!t) {
 				adev->dmap_out->flags &= ~DMA_SYNCING;
@@ -466,7 +466,7 @@ int DMAbuf_sync(int dev)
 			while (!signal_pending(current) &&
 			       adev->d->local_qlen(dev)){
 				spin_unlock_irqrestore(&dmap->lock,flags);
-				interruptible_sleep_on_timeout(&adev->out_sleeper,
+				oss_broken_sleep_on(&adev->out_sleeper,
 							       dmabuf_timeout(dmap));
 				spin_lock_irqsave(&dmap->lock,flags);
 			}
@@ -587,8 +587,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 			timeout = dmabuf_timeout(dmap);
 
 		spin_unlock_irqrestore(&dmap->lock,flags);
-		timeout = interruptible_sleep_on_timeout(&adev->in_sleeper,
-							 timeout);
+		timeout = oss_broken_sleep_on(&adev->in_sleeper, timeout);
 		if (!timeout) {
 			/* FIXME: include device name */
 			err = -EIO;
@@ -768,8 +767,7 @@ static int output_sleep(int dev, int dontblock)
 		timeout_value = dmabuf_timeout(dmap);
 	else
 		timeout_value = MAX_SCHEDULE_TIMEOUT;
-	timeout_value = interruptible_sleep_on_timeout(&adev->out_sleeper,
-						       timeout_value);
+	timeout_value = oss_broken_sleep_on(&adev->out_sleeper, timeout_value);
 	if (timeout != MAX_SCHEDULE_TIMEOUT && !timeout_value) {
 		printk(KERN_WARNING "Sound: DMA (output) timed out - IRQ/DRQ config error?\n");
 		dma_reset_output(dev);
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 4ff60a6427d9..9b9f7d385134 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -19,6 +19,7 @@
 #include "sound_config.h"
 
 #include "midi_ctrl.h"
+#include "sleep.h"
 
 static int      sequencer_ok;
 static struct sound_timer_operations *tmr;
@@ -100,8 +101,7 @@ int sequencer_read(int dev, struct file *file, char __user *buf, int count)
   			return -EAGAIN;
   		}
 
- 		interruptible_sleep_on_timeout(&midi_sleeper,
-					       pre_event_timeout);
+		oss_broken_sleep_on(&midi_sleeper, pre_event_timeout);
 		spin_lock_irqsave(&lock,flags);
 		if (!iqlen)
 		{
@@ -343,7 +343,7 @@ static int seq_queue(unsigned char *note, char nonblock)
 		/*
 		 * Sleep until there is enough space on the queue
 		 */
-		interruptible_sleep_on(&seq_sleeper);
+		oss_broken_sleep_on(&seq_sleeper, MAX_SCHEDULE_TIMEOUT);
 	}
 	if (qlen >= SEQ_MAX_QUEUE)
 	{
@@ -1122,8 +1122,7 @@ static void seq_drain_midi_queues(void)
 		 */
 
  		if (n)
- 			interruptible_sleep_on_timeout(&seq_sleeper,
-						       HZ/10);
+			oss_broken_sleep_on(&seq_sleeper, HZ/10);
 	}
 }
 
@@ -1145,8 +1144,7 @@ void sequencer_release(int dev, struct file *file)
 		while (!signal_pending(current) && qlen > 0)
 		{
   			seq_sync();
- 			interruptible_sleep_on_timeout(&seq_sleeper,
-						       3*HZ);
+			oss_broken_sleep_on(&seq_sleeper, 3*HZ);
  			/* Extra delay */
 		}
 	}
@@ -1201,7 +1199,7 @@ static int seq_sync(void)
 		seq_startplay();
 
  	if (qlen > 0)
- 		interruptible_sleep_on_timeout(&seq_sleeper, HZ);
+		oss_broken_sleep_on(&seq_sleeper, HZ);
 	return qlen;
 }
 
@@ -1224,7 +1222,7 @@ static void midi_outc(int dev, unsigned char data)
 
 	spin_lock_irqsave(&lock,flags);
  	while (n && !midi_devs[dev]->outputc(dev, data)) {
- 		interruptible_sleep_on_timeout(&seq_sleeper, HZ/25);
+		oss_broken_sleep_on(&seq_sleeper, HZ/25);
   		n--;
   	}
 	spin_unlock_irqrestore(&lock,flags);
diff --git a/sound/oss/sleep.h b/sound/oss/sleep.h
new file mode 100644
index 000000000000..a20fc925a5ce
--- /dev/null
+++ b/sound/oss/sleep.h
@@ -0,0 +1,18 @@
+#include <linux/wait.h>
+
+/*
+ * Do not use. This is a replacement for the old
+ * "interruptible_sleep_on_timeout" function that has been
+ * deprecated for ages. All users should instead try to use
+ * wait_event_interruptible_timeout.
+ */
+
+static inline long
+oss_broken_sleep_on(wait_queue_head_t *q, long timeout)
+{
+	DEFINE_WAIT(wait);
+	prepare_to_wait(q, &wait, TASK_INTERRUPTIBLE);
+	timeout = schedule_timeout(timeout);
+	finish_wait(q, &wait);
+	return timeout;
+}
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 7d8803a00b79..f851fd0e199c 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -90,6 +90,8 @@
 #include <asm/sibyte/sb1250_mac.h>
 #include <asm/sibyte/sb1250.h>
 
+#include "sleep.h"
+
 struct cs4297a_state;
 
 static DEFINE_MUTEX(swarm_cs4297a_mutex);
@@ -748,7 +750,7 @@ static int serdma_reg_access(struct cs4297a_state *s, u64 data)
                 /* Since a writer has the DSP open, we have to mux the
                    request in */
                 s->reg_request = data;
-                interruptible_sleep_on(&s->dma_dac.reg_wait);
+		oss_broken_sleep_on(&s->dma_dac.reg_wait, MAX_SCHEDULE_TIMEOUT);
                 /* XXXKW how can I deal with the starvation case where
                    the opener isn't writing? */
         } else {
@@ -790,7 +792,7 @@ static int cs4297a_read_ac97(struct cs4297a_state *s, u32 offset,
         if (serdma_reg_access(s, (0xCLL << 60) | (1LL << 47) | ((u64)(offset & 0x7F) << 40)))
                 return -1;
 
-        interruptible_sleep_on(&s->dma_adc.reg_wait);
+	oss_broken_sleep_on(&s->dma_adc.reg_wait, MAX_SCHEDULE_TIMEOUT);
         *value = s->read_value;
         CS_DBGOUT(CS_AC97, 2,
                   printk(KERN_INFO "cs4297a: rdr reg %x -> %x\n", s->read_reg, s->read_value));
@@ -1740,7 +1742,7 @@ static ssize_t cs4297a_read(struct file *file, char *buffer, size_t count,
 			start_adc(s);
 			if (file->f_flags & O_NONBLOCK)
 				return ret ? ret : -EAGAIN;
-			interruptible_sleep_on(&s->dma_adc.wait);
+			oss_broken_sleep_on(&s->dma_adc.wait, MAX_SCHEDULE_TIMEOUT);
 			if (signal_pending(current))
 				return ret ? ret : -ERESTARTSYS;
 			continue;
@@ -1836,7 +1838,7 @@ static ssize_t cs4297a_write(struct file *file, const char *buffer,
 			start_dac(s);
 			if (file->f_flags & O_NONBLOCK)
 				return ret ? ret : -EAGAIN;
-			interruptible_sleep_on(&d->wait);
+			oss_broken_sleep_on(&d->wait, MAX_SCHEDULE_TIMEOUT);
 			if (signal_pending(current))
 				return ret ? ret : -ERESTARTSYS;
 			continue;
@@ -2452,7 +2454,7 @@ static int cs4297a_locked_open(struct inode *inode, struct file *file)
 				return -EBUSY;
 			}
 			mutex_unlock(&s->open_sem_dac);
-			interruptible_sleep_on(&s->open_wait_dac);
+			oss_broken_sleep_on(&s->open_wait_dac, MAX_SCHEDULE_TIMEOUT);
 
 			if (signal_pending(current)) {
                                 printk("open - sig pending\n");
@@ -2469,7 +2471,7 @@ static int cs4297a_locked_open(struct inode *inode, struct file *file)
 				return -EBUSY;
 			}
 			mutex_unlock(&s->open_sem_adc);
-			interruptible_sleep_on(&s->open_wait_adc);
+			oss_broken_sleep_on(&s->open_wait_adc, MAX_SCHEDULE_TIMEOUT);
 
 			if (signal_pending(current)) {
                                 printk("open - sig pending\n");
-- 
cgit v1.2.2


From 817873f4b155b22a24c48d6a38ee32007e2d856e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 10:24:40 +0100
Subject: ASoC: pcm: Properly initialize hw->rate_max

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll
end up with the rate_max field of the runtime hardware set to 0.  (Note that it
is still possible for the components to constrain the supported sample rates
using other methods, e.g. setting a list constraint) If rate_max is 0 this means
that the sound card doesn't support any rates at all, which is not the desired
result. So initialize rate_max to UINT_MAX. For symmetry reasons also set
rate_min to 0.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 1a617fde46e6..2b8949647e32 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -170,6 +170,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 		& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
 		hw->rates |= codec_stream->rates;
 
+	hw->rate_min = 0;
+	hw->rate_max = UINT_MAX;
+
 	snd_pcm_limit_hw_rates(runtime);
 
 	hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
-- 
cgit v1.2.2


From 24710c97960ac343c613786d250a1e0063555faa Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 10:24:41 +0100
Subject: ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates

SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain
interval) are supported. There is no need to manually set other rate bits.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c         | 3 +--
 sound/soc/fsl/mpc5200_psc_i2s.c | 3 +--
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 35e277379b86..dd5e6a76d29f 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -79,8 +79,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set)
  * ALSA that we support all rates and let the codec driver decide what rates
  * are really supported.
  */
-#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
-			  SNDRV_PCM_RATE_CONTINUOUS)
+#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
 
 /**
  * FSLSSI_I2S_FORMATS: audio formats supported by the SSI
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index f4efaadb80a2..5d07e8a74a21 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -26,8 +26,7 @@
  * ALSA that we support all rates and let the codec driver decide what rates
  * are really supported.
  */
-#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
-			SNDRV_PCM_RATE_CONTINUOUS)
+#define PSC_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
 
 /**
  * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
-- 
cgit v1.2.2


From bf103eb4af73596edbab5faab67e29ea1e87c769 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 10:24:42 +0100
Subject: ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain
interval) are supported. There is no need to manually set other rate bits.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Daniel Glöckner <daniel-gl@gmx.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/s6000/s6000-i2s.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 73bb99f0109a..7eba7979b9af 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -405,8 +405,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-#define S6000_I2S_RATES	(SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
-			 SNDRV_PCM_RATE_8000_192000)
+#define S6000_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
 #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops s6000_i2s_dai_ops = {
-- 
cgit v1.2.2


From e3a9269f874067fcefc5eb8037466161fb0fe3f4 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 10:24:43 +0100
Subject: ALSA: Add helper function for intersecting two rate masks

A bit of special care is necessary when creating the intersection of two rate
masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and
SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two
rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a
specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of
discrete rates specified by a list constraint. For all other cases the supported
rates are specified directly in the rate mask.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/core/pcm_misc.c | 39 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 39 insertions(+)

(limited to 'sound')

diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 43f24cce3dec..4560ca0e5651 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -514,3 +514,42 @@ unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
 	return 0;
 }
 EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate);
+
+static unsigned int snd_pcm_rate_mask_sanitize(unsigned int rates)
+{
+	if (rates & SNDRV_PCM_RATE_CONTINUOUS)
+		return SNDRV_PCM_RATE_CONTINUOUS;
+	else if (rates & SNDRV_PCM_RATE_KNOT)
+		return SNDRV_PCM_RATE_KNOT;
+	return rates;
+}
+
+/**
+ * snd_pcm_rate_mask_intersect - computes the intersection between two rate masks
+ * @rates_a: The first rate mask
+ * @rates_b: The second rate mask
+ *
+ * This function computes the rates that are supported by both rate masks passed
+ * to the function. It will take care of the special handling of
+ * SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT.
+ *
+ * Return: A rate mask containing the rates that are supported by both rates_a
+ * and rates_b.
+ */
+unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
+	unsigned int rates_b)
+{
+	rates_a = snd_pcm_rate_mask_sanitize(rates_a);
+	rates_b = snd_pcm_rate_mask_sanitize(rates_b);
+
+	if (rates_a & SNDRV_PCM_RATE_CONTINUOUS)
+		return rates_b;
+	else if (rates_b & SNDRV_PCM_RATE_CONTINUOUS)
+		return rates_a;
+	else if (rates_a & SNDRV_PCM_RATE_KNOT)
+		return rates_b;
+	else if (rates_b & SNDRV_PCM_RATE_KNOT)
+		return rates_a;
+	return rates_a & rates_b;
+}
+EXPORT_SYMBOL_GPL(snd_pcm_rate_mask_intersect);
-- 
cgit v1.2.2


From 55dcdb5051930dee75e9e2c0da90bc82ee3dcd77 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 10:24:44 +0100
Subject: ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper

Instead of open-coding the intersecting of two rate masks (and getting slightly
wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect()
helper function.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 9 ++-------
 1 file changed, 2 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2b8949647e32..4bbda0a4ee03 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -162,13 +162,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 		hw->formats &= codec_stream->formats & cpu_stream->formats;
 	else
 		hw->formats = codec_stream->formats & cpu_stream->formats;
-	hw->rates = codec_stream->rates & cpu_stream->rates;
-	if (codec_stream->rates
-		& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-		hw->rates |= cpu_stream->rates;
-	if (cpu_stream->rates
-		& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
-		hw->rates |= codec_stream->rates;
+	hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates,
+		cpu_stream->rates);
 
 	hw->rate_min = 0;
 	hw->rate_max = UINT_MAX;
-- 
cgit v1.2.2


From ca919fe4b972b9428ab42bead11b04a4ebf0f632 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Tue, 14 Jan 2014 12:35:32 +0800
Subject: ASoC: simple-card: fix one bug to writing to the platform data

It's a bug that writing to the platform data directly, for it should
be constant. So just copy it before writing.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++-------------------
 1 file changed, 21 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 5528dd6a4e4e..53395f54849a 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -9,9 +9,10 @@
  * published by the Free Software Foundation.
  */
 #include <linux/clk.h>
+#include <linux/module.h>
 #include <linux/of.h>
 #include <linux/platform_device.h>
-#include <linux/module.h>
+#include <linux/string.h>
 #include <sound/simple_card.h>
 
 static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
@@ -190,36 +191,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *of_cpu, *of_codec, *of_platform;
 	struct device *dev = &pdev->dev;
+	int ret;
 
 	cinfo		= NULL;
 	of_cpu		= NULL;
 	of_codec	= NULL;
 	of_platform	= NULL;
+
+	cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
+	if (!cinfo)
+		return -ENOMEM;
+
 	if (np && of_device_is_available(np)) {
-		cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
-		if (cinfo) {
-			int ret;
-			cinfo->snd_card.dev = &pdev->dev;
-			ret = asoc_simple_card_parse_of(np, cinfo, dev,
-							&of_cpu,
-							&of_codec,
-							&of_platform);
-			if (ret < 0) {
-				if (ret != -EPROBE_DEFER)
-					dev_err(dev, "parse error %d\n", ret);
-				return ret;
-			}
-		} else {
-			return -ENOMEM;
+		cinfo->snd_card.dev = dev;
+
+		ret = asoc_simple_card_parse_of(np, cinfo, dev,
+						&of_cpu,
+						&of_codec,
+						&of_platform);
+		if (ret < 0) {
+			if (ret != -EPROBE_DEFER)
+				dev_err(dev, "parse error %d\n", ret);
+			return ret;
 		}
 	} else {
-		cinfo = pdev->dev.platform_data;
-		if (!cinfo) {
+		if (!dev->platform_data) {
 			dev_err(dev, "no info for asoc-simple-card\n");
 			return -EINVAL;
 		}
 
-		cinfo->snd_card.dev = &pdev->dev;
+		memcpy(cinfo, dev->platform_data, sizeof(*cinfo));
+		cinfo->snd_card.dev = dev;
 	}
 
 	if (!cinfo->name	||
-- 
cgit v1.2.2


From 93b943edfc5e439f7b843535e0bb0f7d2371f67f Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 14:02:18 +0100
Subject: ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime

Currently we have two different snd_soc_platform_driver structs in the generic
dmaengine PCM driver. One for dmaengine drivers that support residue reporting
and one for those which do not. When registering the PCM component we check
whether the NO_RESIDUE flag is set or not and use the corresponding
snd_soc_platform_driver. This patch modifies the driver to only have one
snd_soc_platform_driver struct where the pointer() callback checks the
NO_RESIDUE flag at runtime. This allows us to set the NO_RESIDUE flag after the
PCM component has been registered. This becomes necessary when querying whether
the dmaengine driver supports residue reporting from the dmaengine driver itself
since the DMA channel might only be requested after the PCM component has been
registered.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 39 ++++++++++++++---------------------
 1 file changed, 15 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 2a6c569d991f..4e2bed89a4a4 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -248,6 +248,18 @@ err_free:
 	return ret;
 }
 
+static snd_pcm_uframes_t dmaengine_pcm_pointer(
+	struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
+
+	if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
+		return snd_dmaengine_pcm_pointer_no_residue(substream);
+	else
+		return snd_dmaengine_pcm_pointer(substream);
+}
+
 static const struct snd_pcm_ops dmaengine_pcm_ops = {
 	.open		= dmaengine_pcm_open,
 	.close		= snd_dmaengine_pcm_close,
@@ -255,7 +267,7 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = {
 	.hw_params	= dmaengine_pcm_hw_params,
 	.hw_free	= snd_pcm_lib_free_pages,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= dmaengine_pcm_pointer,
 };
 
 static const struct snd_soc_platform_driver dmaengine_pcm_platform = {
@@ -265,23 +277,6 @@ static const struct snd_soc_platform_driver dmaengine_pcm_platform = {
 	.probe_order	= SND_SOC_COMP_ORDER_LATE,
 };
 
-static const struct snd_pcm_ops dmaengine_no_residue_pcm_ops = {
-	.open		= dmaengine_pcm_open,
-	.close		= snd_dmaengine_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= dmaengine_pcm_hw_params,
-	.hw_free	= snd_pcm_lib_free_pages,
-	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
-};
-
-static const struct snd_soc_platform_driver dmaengine_no_residue_pcm_platform = {
-	.ops		= &dmaengine_no_residue_pcm_ops,
-	.pcm_new	= dmaengine_pcm_new,
-	.pcm_free	= dmaengine_pcm_free,
-	.probe_order	= SND_SOC_COMP_ORDER_LATE,
-};
-
 static const char * const dmaengine_pcm_dma_channel_names[] = {
 	[SNDRV_PCM_STREAM_PLAYBACK] = "tx",
 	[SNDRV_PCM_STREAM_CAPTURE] = "rx",
@@ -374,12 +369,8 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	if (ret)
 		goto err_free_dma;
 
-	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
-		ret = snd_soc_add_platform(dev, &pcm->platform,
-				&dmaengine_no_residue_pcm_platform);
-	else
-		ret = snd_soc_add_platform(dev, &pcm->platform,
-				&dmaengine_pcm_platform);
+	ret = snd_soc_add_platform(dev, &pcm->platform,
+		&dmaengine_pcm_platform);
 	if (ret)
 		goto err_free_dma;
 
-- 
cgit v1.2.2


From 478028e088d6a94666d8a776be2cd2291faf3bbd Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 14:02:19 +0100
Subject: ASoC: generic-dmaengine-pcm: Check DMA residue granularity

The dmaengine framework now exposes the granularity with which it is able to
report the transfer residue for a certain DMA channel. Check the granularity in
the generic dmaengine PCM driver and
	a) Set the SNDRV_PCM_INFO_BATCH if the granularity is per period or worse.
	b) Fallback to the (race condition prone) period counting if the driver does
	not support any residue reporting.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 27 +++++++++++++++++++++++++++
 1 file changed, 27 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 4e2bed89a4a4..560a7787d8a7 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -144,6 +144,8 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 	if (ret == 0) {
 		if (dma_caps.cmd_pause)
 			hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
+		if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
+			hw.info |= SNDRV_PCM_INFO_BATCH;
 	}
 
 	return snd_soc_set_runtime_hwparams(substream, &hw);
@@ -187,6 +189,21 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel(
 						 dma_data->filter_data);
 }
 
+static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan)
+{
+	struct dma_slave_caps dma_caps;
+	int ret;
+
+	ret = dma_get_slave_caps(chan, &dma_caps);
+	if (ret != 0)
+		return true;
+
+	if (dma_caps.residue_granularity == DMA_RESIDUE_GRANULARITY_DESCRIPTOR)
+		return false;
+
+	return true;
+}
+
 static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
@@ -239,6 +256,16 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd)
 				max_buffer_size);
 		if (ret)
 			goto err_free;
+
+		/*
+		 * This will only return false if we know for sure that at least
+		 * one channel does not support residue reporting. If the DMA
+		 * driver does not implement the slave_caps API we rely having
+		 * the NO_RESIDUE flag set manually in case residue reporting is
+		 * not supported.
+		 */
+		if (!dmaengine_pcm_can_report_residue(pcm->chan[i]))
+			pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE;
 	}
 
 	return 0;
-- 
cgit v1.2.2


From 153e66f5136bc5b33db253ad2db011177196626e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 14:02:20 +0100
Subject: ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag

The pl330 driver properly reports that it does not have residue reporting
support, which means the PCM dmanegine driver is able to figure this out on its
own. So there is no need to set the flag manually. Removing the flag has the
advantage that once the pl330 driver gains support for residue reporting it will
automatically be used by the generic dmaengine PCM driver.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/adi/axi-i2s.c   | 3 +--
 sound/soc/adi/axi-spdif.c | 3 +--
 2 files changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7f91a86dd734..6058c1fd5070 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -236,8 +236,7 @@ static int axi_i2s_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret)
 		goto err_clk_disable;
 
diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c
index 8db7a9920695..198e3a4640f6 100644
--- a/sound/soc/adi/axi-spdif.c
+++ b/sound/soc/adi/axi-spdif.c
@@ -229,8 +229,7 @@ static int axi_spdif_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 	if (ret)
 		goto err_clk_disable;
 
-- 
cgit v1.2.2


From d70e861a3154833467023123e218e9b1ba558809 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 11 Jan 2014 14:02:21 +0100
Subject: ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag

The Samsung dmaengine ASoC driver is used with two different dmaengine drivers.
The pl80x, which properly supports residue reporting and the pl330, which
reports that it does not support residue reporting. So there is no need to
manually set the NO_RESIDUE flag. This has the advantage that a proper (race
condition free) PCM pointer() implementation is used when the pl80x driver is
used. Also once the pl330 driver supports residue reporting the ASoC PCM driver
will automatically start using it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/dmaengine.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c
index 3be479d51b9b..750ce5808d9f 100644
--- a/sound/soc/samsung/dmaengine.c
+++ b/sound/soc/samsung/dmaengine.c
@@ -68,7 +68,6 @@ int samsung_asoc_dma_platform_register(struct device *dev)
 {
 	return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config,
 					  SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME |
-					  SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 					  SND_DMAENGINE_PCM_FLAG_COMPAT);
 }
 EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register);
-- 
cgit v1.2.2


From f7d3c17096f6cbca8f0113d5a092ffcc72c7bf41 Mon Sep 17 00:00:00 2001
From: Arun Shamanna Lakshmi <aruns@nvidia.com>
Date: Tue, 14 Jan 2014 15:31:54 -0800
Subject: ASoC: dapm: Change prototype of soc_widget_read

soc_widget_read API returns the register data and it is possible
that a register can contain 0xffffffff. Thus, change the prototype
of soc_widget_read to return only the error code and pass the reg
data through pointer argument.

Signed-off-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-dapm.c | 26 +++++++++++++++-----------
 1 file changed, 15 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 51b4c192f41a..2a44fe9122a2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -371,12 +371,16 @@ static void dapm_reset(struct snd_soc_card *card)
 	}
 }
 
-static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
+static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg,
+	unsigned int *value)
 {
-	if (w->codec)
-		return snd_soc_read(w->codec, reg);
-	else if (w->platform)
-		return snd_soc_platform_read(w->platform, reg);
+	if (w->codec) {
+		*value = snd_soc_read(w->codec, reg);
+		return 0;
+	} else if (w->platform) {
+		*value = snd_soc_platform_read(w->platform, reg);
+		return 0;
+	}
 
 	dev_err(w->dapm->dev, "ASoC: no valid widget read method\n");
 	return -1;
@@ -430,13 +434,12 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w,
 			return ret;
 	} else {
 		soc_widget_lock(w);
-		ret = soc_widget_read(w, reg);
+		ret = soc_widget_read(w, reg, &old);
 		if (ret < 0) {
 			soc_widget_unlock(w);
 			return ret;
 		}
 
-		old = ret;
 		new = (old & ~mask) | (value & mask);
 		change = old != new;
 		if (change) {
@@ -513,7 +516,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 		unsigned int invert = mc->invert;
 
 		if (reg != SND_SOC_NOPM) {
-			val = soc_widget_read(w, reg);
+			soc_widget_read(w, reg, &val);
 			val = (val >> shift) & mask;
 			if (invert)
 				val = max - val;
@@ -529,7 +532,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 			w->kcontrol_news[i].private_value;
 		int val, item;
 
-		val = soc_widget_read(w, e->reg);
+		soc_widget_read(w, e->reg, &val);
 		item = (val >> e->shift_l) & e->mask;
 
 		if (item < e->max && !strcmp(p->name, e->texts[item]))
@@ -558,7 +561,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 			w->kcontrol_news[i].private_value;
 		int val, item;
 
-		val = soc_widget_read(w, e->reg);
+		soc_widget_read(w, e->reg, &val);
 		val = (val >> e->shift_l) & e->mask;
 		for (item = 0; item < e->max; item++) {
 			if (val == e->values[item])
@@ -2782,7 +2785,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 
 		/* Read the initial power state from the device */
 		if (w->reg >= 0) {
-			val = soc_widget_read(w, w->reg) >> w->shift;
+			soc_widget_read(w, w->reg, &val);
+			val = val >> w->shift;
 			val &= w->mask;
 			if (val == w->on_val)
 				w->power = 1;
-- 
cgit v1.2.2


From 1104a9c822f0e9f5e57a236f20a142166dd8f91e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Wed, 15 Jan 2014 19:04:19 +0000
Subject: ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation
 provided

Make it easier for generic code to work with set_sysclk() by distinguishing
between the operation not being supported and an error as is done for
other operations like set_dai_fmt()

Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 03c779ebd729..0ebf1dac330d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3484,7 +3484,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 		return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0,
 						      freq, dir);
 	else
-		return -EINVAL;
+		return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 
@@ -3505,7 +3505,7 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 		return codec->driver->set_sysclk(codec, clk_id, source,
 						 freq, dir);
 	else
-		return -EINVAL;
+		return -ENOTSUPP;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk);
 
-- 
cgit v1.2.2


From e8e08c521dc101cf7e7e1caf4f487f9fe11a9a7a Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Wed, 15 Jan 2014 18:12:40 +0100
Subject: ASoC: tlv320aic32x4: Fix regmap range_min

range_min is the lowest address in the virtual register range. This is
the first register with address 0, not the first register of page 1.

Currently all writes to page 1 are mapped to page 0, so the codec fails
to operate.

Fixes: 4d208ca429ad (ASoC: tlv320aic32x4: Convert to direct regmap API usage)
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org (v3.13 if the fix misses -final)
---
 sound/soc/codecs/tlv320aic32x4.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 6941fa9baf6a..385dec16eb8a 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -267,7 +267,7 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = {
 		.selector_mask  = 0xff,
 		.window_start = 0,
 		.window_len = 128,
-		.range_min = AIC32X4_PAGE1,
+		.range_min = 0,
 		.range_max = AIC32X4_RMICPGAVOL,
 	},
 };
-- 
cgit v1.2.2


From c48ae0ab3790efba2dfb1a4709c0ef8da024de1a Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Thu, 16 Jan 2014 15:51:10 +0800
Subject: ALSA: hda - add headset mic detect quirks for some Dell machines

When we plug a 3-ring headset on some Dell machines, the headset
mic can't be detected, after apply this patch, the headset mic
can work well on all those machines.

On the machine with the Subsytem ID 0x10280610, if we use
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be
detected and work well, but the sound can't be outputed via
headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE
can fix this problem.

BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: David Chen <david.chen@canonical.com>
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Tested-by: Shawn Wang <shawn.wang@canonical.com>
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a1c16457880e..f9b22fb6dd0b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4253,7 +4253,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x060f, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK),
@@ -4264,6 +4265,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5065,6 +5071,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE),
+	SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
-- 
cgit v1.2.2


From ec4f2857cd41b7c5af6350e7ab2a8fe1874dae59 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Thu, 16 Jan 2014 16:08:04 +0800
Subject: ASoC: core: Fix possible NULL pointer dereference of pcm->config

Since the soc generic dmaengine pcm driver allows using the defualt settings,
so the pcm->config maybe NULL.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++++---
 1 file changed, 6 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 560a7787d8a7..5bace124ef43 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -176,17 +176,20 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel(
 {
 	struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform);
 	struct snd_dmaengine_dai_dma_data *dma_data;
+	dma_filter_fn fn = NULL;
 
 	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
 		return pcm->chan[0];
 
-	if (pcm->config->compat_request_channel)
+	if (pcm->config && pcm->config->compat_request_channel)
 		return pcm->config->compat_request_channel(rtd, substream);
 
-	return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn,
-						 dma_data->filter_data);
+	if (pcm->config)
+		fn = pcm->config->compat_filter_fn;
+
+	return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data);
 }
 
 static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan)
-- 
cgit v1.2.2


From 23607025303af6e84bc2cd4cabe89c21f6a22a3f Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Fri, 17 Jan 2014 17:03:55 +0000
Subject: ASoC: DPCM: make some DPCM API calls non static for compressed usage

The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-pcm.c | 29 ++++++++++++-----------------
 1 file changed, 12 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..64bf3f827aac 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
 
 /* DPCM stream event, send event to FE and all active BEs. */
-static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
+int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 	int event)
 {
 	struct snd_soc_dpcm *dpcm;
@@ -773,7 +773,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
 }
 
 /* disconnect a BE and FE */
-static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
+void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm, *d;
 
@@ -869,7 +869,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list,
 	return 0;
 }
 
-static int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
+int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
 	int stream, struct snd_soc_dapm_widget_list **list_)
 {
 	struct snd_soc_dai *cpu_dai = fe->cpu_dai;
@@ -891,11 +891,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
 	return paths;
 }
 
-static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list)
-{
-	kfree(*list);
-}
-
 static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream,
 	struct snd_soc_dapm_widget_list **list_)
 {
@@ -965,7 +960,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
 			continue;
 
 		/* don't connect if FE is not running */
-		if (!fe->dpcm[stream].runtime)
+		if (!fe->dpcm[stream].runtime && !fe->fe_compr)
 			continue;
 
 		/* newly connected FE and BE */
@@ -990,7 +985,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
  * Find the corresponding BE DAIs that source or sink audio to this
  * FE substream.
  */
-static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
+int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
 	int stream, struct snd_soc_dapm_widget_list **list, int new)
 {
 	if (new)
@@ -999,7 +994,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
 		return dpcm_prune_paths(fe, stream, list);
 }
 
-static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
+void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 
@@ -1037,7 +1032,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe,
 	}
 }
 
-static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
+int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 	int err, count = 0;
@@ -1186,7 +1181,7 @@ be_err:
 	return ret;
 }
 
-static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
+int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 
@@ -1247,7 +1242,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
+int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 
@@ -1312,7 +1307,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
+int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 	int ret;
@@ -1442,7 +1437,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm,
 	return ret;
 }
 
-static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
+int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 			       int cmd)
 {
 	struct snd_soc_dpcm *dpcm;
@@ -1610,7 +1605,7 @@ out:
 	return ret;
 }
 
-static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
+int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm;
 	int ret = 0;
-- 
cgit v1.2.2


From 2a99ef0fdb35a0f8d6b56ccc5d9d821e9ff100c1 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Date: Fri, 17 Jan 2014 17:03:56 +0000
Subject: ASoC: compress: Add suport for DPCM into compressed audio

Currently compressed audio streams are statically routed from the /dev
to the DAI link. Some DSPs can route compressed data to multiple BE DAIs
like they do for PCM data.

Add support to allow dynamically routed compressed streams using the existing
DPCM infrastructure. This patch adds special FE versions of the compressed ops
that work out the runtime routing.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-compress.c | 301 ++++++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 300 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 53c9ecdd119f..5e9690c85d8f 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -24,6 +24,7 @@
 #include <sound/compress_driver.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
+#include <sound/soc-dpcm.h>
 
 static int soc_compr_open(struct snd_compr_stream *cstream)
 {
@@ -75,6 +76,98 @@ out:
 	return ret;
 }
 
+static int soc_compr_open_fe(struct snd_compr_stream *cstream)
+{
+	struct snd_soc_pcm_runtime *fe = cstream->private_data;
+	struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream;
+	struct snd_soc_platform *platform = fe->platform;
+	struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+	struct snd_soc_dai *codec_dai = fe->codec_dai;
+	struct snd_soc_dpcm *dpcm;
+	struct snd_soc_dapm_widget_list *list;
+	int stream;
+	int ret = 0;
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+	else
+		stream = SNDRV_PCM_STREAM_CAPTURE;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+
+	if (platform->driver->compr_ops && platform->driver->compr_ops->open) {
+		ret = platform->driver->compr_ops->open(cstream);
+		if (ret < 0) {
+			pr_err("compress asoc: can't open platform %s\n", platform->name);
+			goto out;
+		}
+	}
+
+	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) {
+		ret = fe->dai_link->compr_ops->startup(cstream);
+		if (ret < 0) {
+			pr_err("compress asoc: %s startup failed\n", fe->dai_link->name);
+			goto machine_err;
+		}
+	}
+
+	fe->dpcm[stream].runtime = fe_substream->runtime;
+
+	if (dpcm_path_get(fe, stream, &list) <= 0) {
+		dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
+			fe->dai_link->name, stream ? "capture" : "playback");
+	}
+
+	/* calculate valid and active FE <-> BE dpcms */
+	dpcm_process_paths(fe, stream, &list, 1);
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_startup(fe, stream);
+	if (ret < 0) {
+		/* clean up all links */
+		list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+			dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+		dpcm_be_disconnect(fe, stream);
+		fe->dpcm[stream].runtime = NULL;
+		goto fe_err;
+	}
+
+	dpcm_clear_pending_state(fe, stream);
+	dpcm_path_put(&list);
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+		cpu_dai->playback_active++;
+		codec_dai->playback_active++;
+	} else {
+		cpu_dai->capture_active++;
+		codec_dai->capture_active++;
+	}
+
+	cpu_dai->active++;
+	codec_dai->active++;
+	fe->codec->active++;
+
+	mutex_unlock(&fe->card->mutex);
+
+	return 0;
+
+fe_err:
+	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
+		fe->dai_link->compr_ops->shutdown(cstream);
+machine_err:
+	if (platform->driver->compr_ops && platform->driver->compr_ops->free)
+		platform->driver->compr_ops->free(cstream);
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
 /*
  * Power down the audio subsystem pmdown_time msecs after close is called.
  * This is to ensure there are no pops or clicks in between any music tracks
@@ -164,6 +257,65 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
 	return 0;
 }
 
+static int soc_compr_free_fe(struct snd_compr_stream *cstream)
+{
+	struct snd_soc_pcm_runtime *fe = cstream->private_data;
+	struct snd_soc_platform *platform = fe->platform;
+	struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+	struct snd_soc_dai *codec_dai = fe->codec_dai;
+	struct snd_soc_dpcm *dpcm;
+	int stream, ret;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+		cpu_dai->playback_active--;
+		codec_dai->playback_active--;
+	} else {
+		stream = SNDRV_PCM_STREAM_CAPTURE;
+		cpu_dai->capture_active--;
+		codec_dai->capture_active--;
+	}
+
+	cpu_dai->active--;
+	codec_dai->active--;
+	fe->codec->active--;
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_hw_free(fe, stream);
+	if (ret < 0)
+		dev_err(fe->dev, "compressed hw_free failed %d\n", ret);
+
+	ret = dpcm_be_dai_shutdown(fe, stream);
+
+	/* mark FE's links ready to prune */
+	list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be)
+		dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
+	else
+		dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+	dpcm_be_disconnect(fe, stream);
+
+	fe->dpcm[stream].runtime = NULL;
+
+	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown)
+		fe->dai_link->compr_ops->shutdown(cstream);
+
+	if (platform->driver->compr_ops && platform->driver->compr_ops->free)
+		platform->driver->compr_ops->free(cstream);
+
+	mutex_unlock(&fe->card->mutex);
+	return 0;
+}
+
 static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
 {
 
@@ -194,6 +346,59 @@ out:
 	return ret;
 }
 
+static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd)
+{
+	struct snd_soc_pcm_runtime *fe = cstream->private_data;
+	struct snd_soc_platform *platform = fe->platform;
+	int ret = 0, stream;
+
+	if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN ||
+		cmd == SND_COMPR_TRIGGER_DRAIN) {
+
+		if (platform->driver->compr_ops &&
+			platform->driver->compr_ops->trigger)
+		return platform->driver->compr_ops->trigger(cstream, cmd);
+	}
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+	else
+		stream = SNDRV_PCM_STREAM_CAPTURE;
+
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+
+	if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) {
+		ret = platform->driver->compr_ops->trigger(cstream, cmd);
+		if (ret < 0)
+			goto out;
+	}
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_trigger(fe, stream, cmd);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED;
+		break;
+	}
+
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
 static int soc_compr_set_params(struct snd_compr_stream *cstream,
 					struct snd_compr_params *params)
 {
@@ -241,6 +446,64 @@ err:
 	return ret;
 }
 
+static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
+					struct snd_compr_params *params)
+{
+	struct snd_soc_pcm_runtime *fe = cstream->private_data;
+	struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream;
+	struct snd_soc_platform *platform = fe->platform;
+	int ret = 0, stream;
+
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+	else
+		stream = SNDRV_PCM_STREAM_CAPTURE;
+
+	mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+
+	if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) {
+		ret = platform->driver->compr_ops->set_params(cstream, params);
+		if (ret < 0)
+			goto out;
+	}
+
+	if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->set_params) {
+		ret = fe->dai_link->compr_ops->set_params(cstream);
+		if (ret < 0)
+			goto out;
+	}
+
+	/*
+	 * Create an empty hw_params for the BE as the machine driver must
+	 * fix this up to match DSP decoder and ASRC configuration.
+	 * I.e. machine driver fixup for compressed BE is mandatory.
+	 */
+	memset(&fe->dpcm[fe_substream->stream].hw_params, 0,
+		sizeof(struct snd_pcm_hw_params));
+
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+	ret = dpcm_be_dai_hw_params(fe, stream);
+	if (ret < 0)
+		goto out;
+
+	ret = dpcm_be_dai_prepare(fe, stream);
+	if (ret < 0)
+		goto out;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
+	else
+		dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START);
+
+	fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
+
+out:
+	fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+	mutex_unlock(&fe->card->mutex);
+	return ret;
+}
+
 static int soc_compr_get_params(struct snd_compr_stream *cstream,
 					struct snd_codec *params)
 {
@@ -360,6 +623,7 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream,
 
 	return ret;
 }
+
 /* ASoC Compress operations */
 static struct snd_compr_ops soc_compr_ops = {
 	.open		= soc_compr_open,
@@ -375,6 +639,21 @@ static struct snd_compr_ops soc_compr_ops = {
 	.get_codec_caps = soc_compr_get_codec_caps
 };
 
+/* ASoC Dynamic Compress operations */
+static struct snd_compr_ops soc_compr_dyn_ops = {
+	.open		= soc_compr_open_fe,
+	.free		= soc_compr_free_fe,
+	.set_params	= soc_compr_set_params_fe,
+	.get_params	= soc_compr_get_params,
+	.set_metadata   = soc_compr_set_metadata,
+	.get_metadata	= soc_compr_get_metadata,
+	.trigger	= soc_compr_trigger_fe,
+	.pointer	= soc_compr_pointer,
+	.ack		= soc_compr_ack,
+	.get_caps	= soc_compr_get_caps,
+	.get_codec_caps = soc_compr_get_codec_caps
+};
+
 /* create a new compress */
 int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 {
@@ -383,6 +662,7 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_compr *compr;
+	struct snd_pcm *be_pcm;
 	char new_name[64];
 	int ret = 0, direction = 0;
 
@@ -410,7 +690,26 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 		ret = -ENOMEM;
 		goto compr_err;
 	}
-	memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
+
+	if (rtd->dai_link->dynamic) {
+		snprintf(new_name, sizeof(new_name), "(%s)",
+			rtd->dai_link->stream_name);
+
+		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
+				1, 0, &be_pcm);
+		if (ret < 0) {
+			dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n",
+				rtd->dai_link->name);
+			goto compr_err;
+		}
+
+		rtd->pcm = be_pcm;
+		rtd->fe_compr = 1;
+		be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+		be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+		memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
+	} else
+		memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
 
 	/* Add copy callback for not memory mapped DSPs */
 	if (platform->driver->compr_ops && platform->driver->compr_ops->copy)
-- 
cgit v1.2.2


From bd23c5b661858446267f4d6b2fb4edd8eb710dda Mon Sep 17 00:00:00 2001
From: Arun Shamanna Lakshmi <aruns@nvidia.com>
Date: Wed, 15 Jan 2014 13:03:16 -0800
Subject: ASoC: dapm: Fix double prefix addition

The prefix for the codec driver can be used during dual identical
codec usecases. However, dapm adds prefix twice for codec DAI widget
in snd_soc_dapm_add_route API.

This change is to avoid double prefix addition for codec DAI widget
and is needed while using identical dual codecs.

Signed-off-by: Songhee Baek <sbaek@nvidia.com>
Signed-off-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/soc-dapm.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2a44fe9122a2..dc8ff13187f7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2476,7 +2476,8 @@ err:
 }
 
 static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
-				  const struct snd_soc_dapm_route *route)
+				  const struct snd_soc_dapm_route *route,
+				  unsigned int is_prefixed)
 {
 	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
 	struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL;
@@ -2486,7 +2487,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	char prefixed_source[80];
 	int ret;
 
-	if (dapm->codec && dapm->codec->name_prefix) {
+	if (dapm->codec && dapm->codec->name_prefix && !is_prefixed) {
 		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
 			 dapm->codec->name_prefix, route->sink);
 		sink = prefixed_sink;
@@ -2614,7 +2615,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 
 	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
-		r = snd_soc_dapm_add_route(dapm, route);
+		r = snd_soc_dapm_add_route(dapm, route, false);
 		if (r < 0) {
 			dev_err(dapm->dev, "ASoC: Failed to add route %s -> %s -> %s\n",
 				route->source,
@@ -3670,7 +3671,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 				cpu_dai->codec->name, r.source,
 				codec_dai->platform->name, r.sink);
 
-			snd_soc_dapm_add_route(&card->dapm, &r);
+			snd_soc_dapm_add_route(&card->dapm, &r, true);
 		}
 
 		/* connect BE DAI capture if widgets are valid */
@@ -3681,7 +3682,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 				codec_dai->codec->name, r.source,
 				cpu_dai->platform->name, r.sink);
 
-			snd_soc_dapm_add_route(&card->dapm, &r);
+			snd_soc_dapm_add_route(&card->dapm, &r, true);
 		}
 
 	}
-- 
cgit v1.2.2


From a23544a6cc448e3a90630814ddb79690eb9ce444 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@linaro.org>
Date: Mon, 20 Jan 2014 22:52:28 +0000
Subject: ASoC: omap: Make RX51 depend on GPIOLIB due to jack usage

Since the GPIO jacks are only supported if gpiolib is built and fail to
compile otherwise add a build depedency.  This is unlikely to have any
practical impact outside of coverage testing.

Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
---
 sound/soc/omap/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 4a07f7179690..22ad9c5654b5 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -30,6 +30,7 @@ config SND_OMAP_SOC_RX51
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_TPA6130A2
+	depends on GPIOLIB
 	help
 	  Say Y if you want to add support for SoC audio on Nokia RX-51
 	  hardware. This is also known as Nokia N900 product.
-- 
cgit v1.2.2


From ae1f8ce1925645b047394a59d4108e27fad9a9d8 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Mon, 20 Jan 2014 17:35:40 -0200
Subject: ASoC: fsl_ssi: We do support master mode now

Since commit aafa85e71a (ASoC: fsl_ssi: Add DAI master mode support for SSI on
i.MX series) master mode is supported, so update the comments and code to
reflect that.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 7 +------
 1 file changed, 1 insertion(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index f9090b167ad7..c5205690791e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1262,18 +1262,13 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		return -EINVAL;
 	hw_type = (enum fsl_ssi_type) of_id->data;
 
-	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (!sprop) {
 		dev_err(&pdev->dev, "fsl,mode property is necessary\n");
 		return -EINVAL;
 	}
-	if (!strcmp(sprop, "ac97-slave")) {
+	if (!strcmp(sprop, "ac97-slave"))
 		ac97 = true;
-	} else if (strcmp(sprop, "i2s-slave")) {
-		dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
-		return -ENODEV;
-	}
 
 	/* The DAI name is the last part of the full name of the node. */
 	p = strrchr(np->full_name, '/') + 1;
-- 
cgit v1.2.2


From efe2ab9b0274b60327ca148e101177ba9708b9ba Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Mon, 20 Jan 2014 17:35:39 -0200
Subject: ASoC: fsl_ssi: Do not print 'baud clock' error message all the time

Currently everytime we get the following message on boot:

fsl-ssi-dai 202c000.ssi: could not get baud clock: -2

This is not really useful information to get on every boot, so make it a debug
message instead.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_ssi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c5205690791e..1c791ddbf006 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1386,7 +1386,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 		 */
 		ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud");
 		if (IS_ERR(ssi_private->baudclk))
-			dev_warn(&pdev->dev, "could not get baud clock: %ld\n",
+			dev_dbg(&pdev->dev, "could not get baud clock: %ld\n",
 				 PTR_ERR(ssi_private->baudclk));
 		else
 			clk_prepare_enable(ssi_private->baudclk);
-- 
cgit v1.2.2


From abffae647e5db10133dc87ce7bb1fd30d7c0f737 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Wed, 22 Jan 2014 17:30:38 +0530
Subject: ASoC: samsung: Fix build regressions due to gpio re-org
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Recent changes through commits c67d0f29262b ("ARM: s3c24xx: get rid
of custom <mach/gpio.h>"), b0161caa72b6 ("ARM: S3C[24|64]xx: move includes
back under <mach/> scope"), 364374121b78 ("ARM: s3c24xx: explicit
dependency on <plat/gpio-cfg.h>") and 41c3548e6da6 ("ARM: s3c64xx: get rid
of custom <mach/gpio.h>") caused build regressions due to broken
dependencies. Fix the following errors by including the necessary header
files explicitly:

sound/soc/samsung/h1940_uda1380.c:56:3: error: implicit declaration of function ‘S3C2410_GPG’
sound/soc/samsung/h1940_uda1380.c:149:18: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/h1940_uda1380.c:234:21: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/h1940_uda1380.c:270:12: error: ‘S3C_GPIO_END’ undeclared (first use in this function)
sound/soc/samsung/neo1973_wm8753.c:239:2: error: implicit declaration of function ‘S3C2410_GPJ’
sound/soc/samsung/rx1950_uda1380.c:67:3: error: implicit declaration of function ‘S3C2410_GPG’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘s3c_gpio_cfgall_range’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘S3C2410_GPE’
sound/soc/samsung/s3c2412-i2s.c:86:2: error: implicit declaration of function ‘S3C_GPIO_SFN’
sound/soc/samsung/s3c2412-i2s.c:87:10: error: ‘S3C_GPIO_PULL_NONE’ undeclared
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘s3c_gpio_cfgall_range’
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘S3C2410_GPE’
sound/soc/samsung/s3c24xx-i2s.c:394:2: error: implicit declaration of function ‘S3C_GPIO_SFN’
sound/soc/samsung/s3c24xx-i2s.c:395:10: error: ‘S3C_GPIO_PULL_NONE’ undeclared
sound/soc/samsung/smartq_wm8987.c:112:3: error: implicit declaration of function ‘S3C64XX_GPL’

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/h1940_uda1380.c  | 1 +
 sound/soc/samsung/neo1973_wm8753.c | 1 +
 sound/soc/samsung/rx1950_uda1380.c | 1 +
 sound/soc/samsung/s3c2412-i2s.c    | 2 ++
 sound/soc/samsung/s3c24xx-i2s.c    | 2 ++
 sound/soc/samsung/smartq_wm8987.c  | 1 +
 6 files changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index fa91376e323d..fbced589d077 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -23,6 +23,7 @@
 #include "regs-iis.h"
 #include <asm/mach-types.h>
 
+#include <mach/gpio-samsung.h>
 #include "s3c24xx-i2s.h"
 
 static unsigned int rates[] = {
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 807db417d234..98a04c11202d 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -20,6 +20,7 @@
 
 #include <sound/soc.h>
 
+#include <mach/gpio-samsung.h>
 #include <asm/mach-types.h>
 #include "regs-iis.h"
 
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 704460a37005..06ebdc061770 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -24,6 +24,7 @@
 #include <sound/soc.h>
 #include <sound/jack.h>
 
+#include <mach/gpio-samsung.h>
 #include "regs-iis.h"
 #include <asm/mach-types.h>
 
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index ea885cb9f76c..d0794458963a 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -26,6 +26,8 @@
 #include <sound/pcm_params.h>
 
 #include <mach/dma.h>
+#include <mach/gpio-samsung.h>
+#include <plat/gpio-cfg.h>
 
 #include "dma.h"
 #include "regs-i2s-v2.h"
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 9c8ebd872fac..f31e916dd8c4 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -24,6 +24,8 @@
 #include <sound/pcm_params.h>
 
 #include <mach/dma.h>
+#include <mach/gpio-samsung.h>
+#include <plat/gpio-cfg.h>
 #include "regs-iis.h"
 
 #include "dma.h"
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 58ae3237ef69..c3b2adafb7b5 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -19,6 +19,7 @@
 #include <sound/soc.h>
 #include <sound/jack.h>
 
+#include <mach/gpio-samsung.h>
 #include <asm/mach-types.h>
 
 #include "i2s.h"
-- 
cgit v1.2.2


From 9eeb95727c1137217d61141096c308dc04c234d1 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Wed, 22 Jan 2014 17:30:39 +0530
Subject: ASoC: Samsung: s3c-i2s-v2: Fix build error
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fixes the following error introduced by commit eca3b01d0885
("ASoC: switch over to use snd_soc_register_component() on s3c i2s"):
sound/soc/samsung/s3c-i2s-v2.c:732:32: error: ‘drv’ undeclared (first use in this function)

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/s3c-i2s-v2.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index fefc56100349..79e7efb9283c 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -729,7 +729,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
 			   struct snd_soc_component_driver *cmp_drv,
 			   struct snd_soc_dai_driver *dai_drv)
 {
-	struct snd_soc_dai_ops *ops = drv->ops;
+	struct snd_soc_dai_ops *ops = dai_drv->ops;
 
 	ops->trigger = s3c2412_i2s_trigger;
 	if (!ops->hw_params)
@@ -742,8 +742,8 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
 	if (!ops->delay)
 		ops->delay = s3c2412_i2s_delay;
 
-	drv->suspend = s3c2412_i2s_suspend;
-	drv->resume = s3c2412_i2s_resume;
+	dai_drv->suspend = s3c2412_i2s_suspend;
+	dai_drv->resume = s3c2412_i2s_resume;
 
 	return snd_soc_register_component(dev, cmp_drv, dai_drv, 1);
 }
-- 
cgit v1.2.2


From f83183ca3c25275706427a9f88f83208dec647d1 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Wed, 22 Jan 2014 17:30:43 +0530
Subject: ASoC: samsung: smdk_wm8994: Fix build error
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fixes the following build error and warning when OF is not defined:
sound/soc/samsung/smdk_wm8994.c:191:23: error: ‘samsung_wm8994_of_match’ undeclared (first use in this function)
sound/soc/samsung/smdk_wm8994.c:47:32: warning: ‘smdk_board_data’ defined but not used [-Wunused-variable]

of_match_ptr() is used so that samsung_wm8994_of_match gets dropped (as unused)
by the compiler when OF is not defined.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/smdk_wm8994.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index b072bd107b31..d38ae98e2f32 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -152,13 +152,11 @@ static struct snd_soc_card smdk = {
 	.num_links = ARRAY_SIZE(smdk_dai),
 };
 
-#ifdef CONFIG_OF
 static const struct of_device_id samsung_wm8994_of_match[] = {
 	{ .compatible = "samsung,smdk-wm8994", .data = &smdk_board_data },
 	{},
 };
 MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match);
-#endif /* CONFIG_OF */
 
 static int smdk_audio_probe(struct platform_device *pdev)
 {
@@ -188,7 +186,7 @@ static int smdk_audio_probe(struct platform_device *pdev)
 		smdk_dai[0].platform_of_node = smdk_dai[0].cpu_of_node;
 	}
 
-	id = of_match_device(samsung_wm8994_of_match, &pdev->dev);
+	id = of_match_device(of_match_ptr(samsung_wm8994_of_match), &pdev->dev);
 	if (id)
 		*board = *((struct smdk_wm8994_data *)id->data);
 
-- 
cgit v1.2.2


From 0cf0f174400d6953df8c09ed5ed92b37ec7a675b Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Thu, 23 Jan 2014 16:23:26 +0530
Subject: ASoC: wm5100: Export wm5100_detect

Export the symbol so that it is accessible to modules. Fixes the
following error:

ERROR: "wm5100_detect" [sound/soc/samsung/snd-soc-lowland.ko] undefined!

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/wm5100.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 4cf91deabc02..4e3e31aaf509 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -14,6 +14,7 @@
 #include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/export.h>
 #include <linux/pm.h>
 #include <linux/gcd.h>
 #include <linux/gpio.h>
@@ -2141,6 +2142,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(wm5100_detect);
 
 static irqreturn_t wm5100_irq(int irq, void *data)
 {
-- 
cgit v1.2.2


From 4a9eee01b80b2e24d0d177df08334c0e4e4c3375 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Thu, 23 Jan 2014 16:22:06 +0530
Subject: ASoC: samsung: Fix Kconfig dependency

Select S3C24XX_DMA instead of S3C2410_DMA to avoid following dependency issues
and build errors:

warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))
warning: (CPU_S3C2410 && CPU_S3C2442 && SND_SOC_SAMSUNG && SND_S3C24XX_I2S && SND_S3C2412_SOC_I2S && SND_SOC_SAMSUNG_SMDK2443_WM9710 && SND_SOC_SAMSUNG_LN2440SBC_ALC650) selects S3C2410_DMA which has unmet direct dependencies (ARCH_S3C24XX && S3C24XX_DMA && (CPU_S3C2410 || CPU_S3C2442))

arch/arm/mach-s3c24xx/built-in.o: In function `s3c2410_dma_add':
arch/arm/mach-s3c24xx/dma-s3c2410.c:134: undefined reference to `s3c2410_dma_init'
arch/arm/mach-s3c24xx/dma-s3c2410.c:135: undefined reference to `s3c24xx_dma_order_set'
arch/arm/mach-s3c24xx/dma-s3c2410.c:136: undefined reference to `s3c24xx_dma_init_map'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/soc/samsung/s3c24xx-i2s.c:293: undefined reference to `s3c2410_dma_ctrl'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
arch/arm/plat-samsung/include/plat/dma-ops.h:60: undefined reference to `s3c_dma_get_ops'
sound/built-in.o: In function `s3c2412_i2s_trigger':
sound/soc/samsung/s3c-i2s-v2.c:432: undefined reference to `s3c2410_dma_ctrl'

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 27930fc432dc..e8fbf4e5732f 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -19,7 +19,7 @@ config SND_S3C_DMA_LEGACY
 
 config SND_S3C24XX_I2S
 	tristate
-	select S3C2410_DMA
+	select S3C24XX_DMA
 
 config SND_S3C_I2SV2_SOC
 	tristate
-- 
cgit v1.2.2


From 9b35a7b65f15ac8769619146ab12933ffa5dcaf8 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Fri, 24 Jan 2014 14:09:25 +0530
Subject: ASoC: Samsung: Fix build error due to missing dependency

Depend on MFD_ARIZONA to avoid the following build errors:

sound/soc/codecs/arizona.c:218: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:226: undefined reference to `arizona_request_irq'
sound/soc/codecs/arizona.c:1719: undefined reference to `arizona_request_irq'

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index e8fbf4e5732f..454f41cfc828 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -210,7 +210,7 @@ config SND_SOC_TOBERMORY
 
 config SND_SOC_BELLS
 	tristate "Audio support for Wolfson Bells"
-	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM5102
 	select SND_SOC_WM5110
-- 
cgit v1.2.2


From 57e33781ce5aaab86b4db7799f3505988b5226e2 Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Fri, 24 Jan 2014 16:23:22 +0530
Subject: ASoC: samsung: Add NULL check in i2s.c

'res' could be NULL from one of the operations above (line 1243). Thus
check 'res' for NULL before releasing the region to avoid null pointer
dereference.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/i2s.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 92f64363427d..a9da24f34834 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1268,7 +1268,8 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 
 	return 0;
 err:
-	release_mem_region(regs_base, resource_size(res));
+	if (res)
+		release_mem_region(regs_base, resource_size(res));
 
 	return ret;
 }
-- 
cgit v1.2.2


From 4c021cc5a49eeb423c2708374a5745b1acdc4f8b Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Fri, 24 Jan 2014 16:23:23 +0530
Subject: ASoC: samsung: Remove dma.h inclusion

mach/dma.h is not referenced by this file. Remove it.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/samsung/i2s.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index a9da24f34834..0a9b44c940ce 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -22,8 +22,6 @@
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
-#include <mach/dma.h>
-
 #include <linux/platform_data/asoc-s3c.h>
 
 #include "dma.h"
-- 
cgit v1.2.2


From b44aa40f875b06b1986a18251d601ee1832c4476 Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Mon, 27 Jan 2014 13:03:05 +0100
Subject: ASoC: tlv320aic32x4: Fix mono playback

Playback of a mono stream should output the same stream on both
channels. At the moment only the left analog signal is valid, the right
one is just noise.

This patch maps the left digital channel onto both DACs when receiving a
mono stream.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++-------
 sound/soc/codecs/tlv320aic32x4.h |  1 +
 2 files changed, 12 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 385dec16eb8a..ee9ad05f9137 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -450,6 +450,17 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream,
 	}
 	snd_soc_write(codec, AIC32X4_IFACE1, data);
 
+	if (params_channels(params) == 1) {
+		data = AIC32X4_RDAC2LCHN | AIC32X4_LDAC2LCHN;
+	} else {
+		if (aic32x4->swapdacs)
+			data = AIC32X4_RDAC2LCHN | AIC32X4_LDAC2RCHN;
+		else
+			data = AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN;
+	}
+	snd_soc_update_bits(codec, AIC32X4_DACSETUP, AIC32X4_DAC_CHAN_MASK,
+			data);
+
 	return 0;
 }
 
@@ -606,13 +617,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}
 	snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
 
-	/* Do DACs need to be swapped? */
-	if (aic32x4->swapdacs) {
-		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN);
-	} else {
-		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN);
-	}
-
 	/* Mic PGA routing */
 	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
 		snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 35774223fd91..83795af8c75f 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -138,6 +138,7 @@
 #define AIC32X4_LDAC2RCHN		(0x02 << 4)
 #define AIC32X4_LDAC2LCHN		(0x01 << 4)
 #define AIC32X4_RDAC2RCHN		(0x01 << 2)
+#define AIC32X4_DAC_CHAN_MASK		0x3c
 
 #define AIC32X4_SSTEP2WCLK		0x01
 #define AIC32X4_MUTEON			0x0C
-- 
cgit v1.2.2


From 609e6025b8ed347c5eba54eb12357193e46dd34f Mon Sep 17 00:00:00 2001
From: Markus Pargmann <mpa@pengutronix.de>
Date: Mon, 27 Jan 2014 13:03:06 +0100
Subject: ASoC: tlv320aic32x4: Fix MICPGA input configuration

Currently the Negative Terminal Input Routing Configuration is only set
when there is a special routing configuration. If we don't use one of
the inputs IN1 or IN2 as negative terminal input, the PGA and recording
does not work.

This patch adds a route from CM1L/CM1R to the PGA as negative input by
default. With this configuration the PGA can amplify all input signals
and line-in/mic works again.

Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/tlv320aic32x4.c | 10 ++++++----
 sound/soc/codecs/tlv320aic32x4.h |  2 ++
 2 files changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index ee9ad05f9137..688151ba309a 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -618,12 +618,14 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
 
 	/* Mic PGA routing */
-	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
+	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K)
 		snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
-	}
-	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) {
+	else
+		snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K);
+	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K)
 		snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
-	}
+	else
+		snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K);
 
 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
index 83795af8c75f..995f033a855d 100644
--- a/sound/soc/codecs/tlv320aic32x4.h
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -120,7 +120,9 @@
 #define AIC32X4_MICBIAS_2075V		0x60
 
 #define AIC32X4_LMICPGANIN_IN2R_10K	0x10
+#define AIC32X4_LMICPGANIN_CM1L_10K	0x40
 #define AIC32X4_RMICPGANIN_IN1L_10K	0x10
+#define AIC32X4_RMICPGANIN_CM1R_10K	0x40
 
 #define AIC32X4_LMICPGAVOL_NOGAIN	0x80
 #define AIC32X4_RMICPGAVOL_NOGAIN	0x80
-- 
cgit v1.2.2