Merge branch 'topic/asoc' into for-linus

6 files changed, 172 insertions, 17 deletions

diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig
index 57429041189c..91a28de94109 100644
--- a/sound/soc/ep93xx/Kconfig
+++ b/sound/soc/ep93xx/Kconfig
@@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE
         help
           Say Y or M here if you want to add support for AC97 audio on the
           Simplemachines Sim.One board.
+
+config SND_EP93XX_SOC_EDB93XX
+        tristate "SoC Audio support for Cirrus Logic EDB93xx boards"
+        depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A)
+        select SND_EP93XX_SOC_I2S
+        select SND_SOC_CS4271
+        help
+          Say Y or M here if you want to add support for I2S audio on the
+          Cirrus Logic EDB93xx boards.
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile
index 8e7977fb6b7d..5514146cbdf0 100644
--- a/sound/soc/ep93xx/Makefile
+++ b/sound/soc/ep93xx/Makefile
@@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97)		+= snd-soc-ep93xx-ac97.o
 # EP93XX Machine Support
 snd-soc-snappercl15-objs                        := snappercl15.o
 snd-soc-simone-objs                             := simone.o
+snd-soc-edb93xx-objs                            := edb93xx.o
 
 obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15)        += snd-soc-snappercl15.o
 obj-$(CONFIG_SND_EP93XX_SOC_SIMONE)             += snd-soc-simone.o
+obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX)            += snd-soc-edb93xx.o
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
new file mode 100644
index 000000000000..d3aa15119d26
--- /dev/null
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -0,0 +1,142 @@
+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * 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.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include "ep93xx-pcm.h"
+
+#define edb93xx_has_audio() (machine_is_edb9301() ||    \
+                             machine_is_edb9302() ||    \
+                             machine_is_edb9302a() ||   \
+                             machine_is_edb9307a() ||   \
+                             machine_is_edb9315a())
+
+static int edb93xx_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_dai *cpu_dai = rtd->cpu_dai;
+        int err;
+        unsigned int mclk_rate;
+        unsigned int rate = params_rate(params);
+
+        /*
+         * According to CS4271 datasheet we use MCLK/LRCK=256 for
+         * rates below 50kHz and 128 for higher sample rates
+         */
+        if (rate < 50000)
+                mclk_rate = rate * 64 * 4;
+        else
+                mclk_rate = rate * 64 * 2;
+
+        err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBS_CFS);
+        if (err)
+                return err;
+
+        err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBS_CFS);
+        if (err)
+                return err;
+
+        err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
+                                     SND_SOC_CLOCK_IN);
+        if (err)
+                return err;
+
+        return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
+                                      SND_SOC_CLOCK_OUT);
+}
+
+static struct snd_soc_ops edb93xx_ops = {
+        .hw_params      = edb93xx_hw_params,
+};
+
+static struct snd_soc_dai_link edb93xx_dai = {
+        .name           = "CS4271",
+        .stream_name    = "CS4271 HiFi",
+        .platform_name  = "ep93xx-pcm-audio",
+        .cpu_dai_name   = "ep93xx-i2s",
+        .codec_name     = "spi0.0",
+        .codec_dai_name = "cs4271-hifi",
+        .ops            = &edb93xx_ops,
+};
+
+static struct snd_soc_card snd_soc_edb93xx = {
+        .name           = "EDB93XX",
+        .dai_link       = &edb93xx_dai,
+        .num_links      = 1,
+};
+
+static struct platform_device *edb93xx_snd_device;
+
+static int __init edb93xx_init(void)
+{
+        int ret;
+
+        if (!edb93xx_has_audio())
+                return -ENODEV;
+
+        ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
+                                 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
+                                 EP93XX_SYSCON_I2SCLKDIV_SPOL);
+        if (ret)
+                return ret;
+
+        edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!edb93xx_snd_device) {
+                ret = -ENOMEM;
+                goto free_i2s;
+        }
+
+        platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
+        ret = platform_device_add(edb93xx_snd_device);
+        if (ret)
+                goto device_put;
+
+        return 0;
+
+device_put:
+        platform_device_put(edb93xx_snd_device);
+free_i2s:
+        ep93xx_i2s_release();
+        return ret;
+}
+module_init(edb93xx_init);
+
+static void __exit edb93xx_exit(void)
+{
+        platform_device_unregister(edb93xx_snd_device);
+        ep93xx_i2s_release();
+}
+module_exit(edb93xx_exit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 68a0bae1208a..104e95cda0ad 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
         struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
         unsigned v = 0;
 
-
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index fff579a1c134..042f4e93746f 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 {
         struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
         unsigned word_len, div, sdiv, lrdiv;
-        int found = 0, err;
+        int err;
 
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_S16_LE:
@@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
          * the codec uses.
          */
         div = clk_get_rate(info->mclk) / params_rate(params);
-        for (sdiv = 2; sdiv <= 4; sdiv += 2)
-                for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
-                        if (sdiv * lrdiv == div) {
-                                found = 1;
-                                goto out;
-                        }
-out:
-        if (!found)
-                return -EINVAL;
+        sdiv = 4;
+        if (div > (256 + 512) / 2) {
+                lrdiv = 128;
+        } else {
+                lrdiv = 64;
+                if (div < (128 + 256) / 2)
+                        sdiv = 2;
+        }
 
         err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
         if (err)
@@ -314,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai)
         struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
         if (!dai->active)
-                return;
+                return 0;
 
         ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
         ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+        return 0;
 }
 
 static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
@@ -325,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
         struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
         if (!dai->active)
-                return;
+                return 0;
 
         ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
         ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+        return 0;
 }
 #else
 #define ep93xx_i2s_suspend      NULL
@@ -352,13 +355,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = {
         .playback       = {
                 .channels_min   = 2,
                 .channels_max   = 2,
-                .rates          = SNDRV_PCM_RATE_8000_96000,
+                .rates          = SNDRV_PCM_RATE_8000_192000,
                 .formats        = EP93XX_I2S_FORMATS,
         },
         .capture        = {
                  .channels_min  = 2,
                  .channels_max  = 2,
-                 .rates         = SNDRV_PCM_RATE_8000_96000,
+                 .rates         = SNDRV_PCM_RATE_8000_192000,
                  .formats       = EP93XX_I2S_FORMATS,
         },
         .ops            = &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 06670776f649..a456e491155f 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
                                    SNDRV_PCM_INFO_INTERLEAVED   |
                                    SNDRV_PCM_INFO_BLOCK_TRANSFER),
                                    
-        .rates                  = SNDRV_PCM_RATE_8000_96000,
+        .rates                  = SNDRV_PCM_RATE_8000_192000,
         .rate_min               = SNDRV_PCM_RATE_8000,
-        .rate_max               = SNDRV_PCM_RATE_96000,
+        .rate_max               = SNDRV_PCM_RATE_192000,
         
         .formats                = (SNDRV_PCM_FMTBIT_S16_LE |
                                    SNDRV_PCM_FMTBIT_S24_LE |