From f724bd240adef304e222590826cb0c17d6168b68 Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Thu, 4 Nov 2010 20:08:12 -0700
Subject: sound/oss/dev_table.c: Use vzalloc

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/dev_table.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
index 727bdb9ba2dc..d8cf3e58dc76 100644
--- a/sound/oss/dev_table.c
+++ b/sound/oss/dev_table.c
@@ -71,7 +71,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
 
-	op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+	op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations)));
 	sound_nblocks++;
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -81,7 +81,6 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
 		sound_unload_audiodev(num);
 		return -(ENOMEM);
 	}
-	memset((char *) op, 0, sizeof(struct audio_operations));
 	init_waitqueue_head(&op->in_sleeper);
 	init_waitqueue_head(&op->out_sleeper);	
 	init_waitqueue_head(&op->poll_sleeper);
@@ -128,7 +127,7 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
 	/* FIXME: This leaks a mixer_operations struct every time its called
 	   until you unload sound! */
 	   
-	op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+	op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations)));
 	sound_nblocks++;
 	if (sound_nblocks >= MAX_MEM_BLOCKS)
 		sound_nblocks = MAX_MEM_BLOCKS - 1;
@@ -137,7 +136,6 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
 		printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
 		return -ENOMEM;
 	}
-	memset((char *) op, 0, sizeof(struct mixer_operations));
 	memcpy((char *) op, (char *) driver, driver_size);
 
 	strlcpy(op->name, name, sizeof(op->name));
-- 
cgit v1.2.2


From ea7dd251251a8d4694e9929104209dcc06220630 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Tue, 9 Nov 2010 00:11:03 +0100
Subject: sound/oss: Remove unnecessary casts of void ptr

The [vk][cmz]alloc(_node) family of functions return void pointers which
it's completely unnecessary/pointless to cast to other pointer types since
that happens implicitly.

This patch removes such casts from sound/oss/

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/midibuf.c   | 4 ++--
 sound/oss/pss.c       | 6 +++---
 sound/oss/sequencer.c | 4 ++--
 3 files changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
index 782b3b84dac6..ceedb1eff203 100644
--- a/sound/oss/midibuf.c
+++ b/sound/oss/midibuf.c
@@ -178,7 +178,7 @@ int MIDIbuf_open(int dev, struct file *file)
 		return err;
 
 	parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT;
-	midi_in_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+	midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf));
 
 	if (midi_in_buf[dev] == NULL)
 	{
@@ -188,7 +188,7 @@ int MIDIbuf_open(int dev, struct file *file)
 	}
 	midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0;
 
-	midi_out_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+	midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf));
 
 	if (midi_out_buf[dev] == NULL)
 	{
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index e19dd5dcc2de..9b800ce5100e 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -859,7 +859,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 			return 0;
 
 		case SNDCTL_COPR_LOAD:
-			buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
+			buf = vmalloc(sizeof(copr_buffer));
 			if (buf == NULL)
 				return -ENOSPC;
 			if (copy_from_user(buf, arg, sizeof(copr_buffer))) {
@@ -871,7 +871,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 			return err;
 		
 		case SNDCTL_COPR_SENDMSG:
-			mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+			mbuf = vmalloc(sizeof(copr_msg));
 			if (mbuf == NULL)
 				return -ENOSPC;
 			if (copy_from_user(mbuf, arg, sizeof(copr_msg))) {
@@ -895,7 +895,7 @@ static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg,
 
 		case SNDCTL_COPR_RCVMSG:
 			err = 0;
-			mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+			mbuf = vmalloc(sizeof(copr_msg));
 			if (mbuf == NULL)
 				return -ENOSPC;
 			data = (unsigned short *)mbuf->data;
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index e85789e53816..5ea1098ac427 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -1646,13 +1646,13 @@ void sequencer_init(void)
 {
 	if (sequencer_ok)
 		return;
-	queue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * EV_SZ);
+	queue = vmalloc(SEQ_MAX_QUEUE * EV_SZ);
 	if (queue == NULL)
 	{
 		printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n");
 		return;
 	}
-	iqueue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
+	iqueue = vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
 	if (iqueue == NULL)
 	{
 		printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n");
-- 
cgit v1.2.2


From 89feca1a16b05651d9c500e5572c0d6882873396 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 13 Oct 2010 15:48:24 +0200
Subject: ALSA: HDA: Enable digital mic on IDT 92HD87B

BugLink: http://launchpad.net/bugs/673075

According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.

Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++--
 1 file changed, 12 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 93fa59cc60ef..cfd73afad882 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -389,6 +389,11 @@ static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
 	0x11, 0x20, 0
 };
 
+#define STAC92HD87B_NUM_DMICS	 1
+static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = {
+	0x11, 0
+};
+
 #define STAC92HD83XXX_NUM_CAPS	2
 static unsigned long stac92hd83xxx_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
@@ -5452,12 +5457,17 @@ again:
 				stac92hd83xxx_brd_tbl[spec->board_config]);
 
 	switch (codec->vendor_id) {
+	case 0x111d76d1:
+	case 0x111d76d9:
+		spec->dmic_nids = stac92hd87b_dmic_nids;
+		spec->num_dmics = stac92xx_connected_ports(codec,
+				stac92hd87b_dmic_nids,
+				STAC92HD87B_NUM_DMICS);
+		/* Fall through */
 	case 0x111d7666:
 	case 0x111d7667:
 	case 0x111d7668:
 	case 0x111d7669:
-	case 0x111d76d1:
-	case 0x111d76d9:
 		spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids);
 		spec->pin_nids = stac92hd88xxx_pin_nids;
 		spec->mono_nid = 0;
-- 
cgit v1.2.2


From e9161512017f11050ef2b826cbb10be1673554c6 Mon Sep 17 00:00:00 2001
From: Florian Fainelli <florian@openwrt.org>
Date: Tue, 9 Nov 2010 18:29:08 +0100
Subject: ALSA: sound/mixart: avoid redefining {readl,write}_{le,be} accessors

If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.

Signed-off-by: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart_hwdep.h | 10 ++++++++++
 1 file changed, 10 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h
index a46f5083db99..812e288ef2e7 100644
--- a/sound/pci/mixart/mixart_hwdep.h
+++ b/sound/pci/mixart/mixart_hwdep.h
@@ -25,11 +25,21 @@
 
 #include <sound/hwdep.h>
 
+#ifndef readl_be
 #define readl_be(x) be32_to_cpu(__raw_readl(x))
+#endif
+
+#ifndef writel_be
 #define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+#endif
 
+#ifndef readl_le
 #define readl_le(x) le32_to_cpu(__raw_readl(x))
+#endif
+
+#ifndef writel_le
 #define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr)
+#endif
 
 #define MIXART_MEM(mgr,x)	((mgr)->mem[0].virt + (x))
 #define MIXART_REG(mgr,x)	((mgr)->mem[1].virt + (x))
-- 
cgit v1.2.2


From fa2b30af84e84129b8d4cf955890ad167cc20cf0 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Tue, 9 Nov 2010 23:00:41 +0100
Subject: ALSA: sound/pci/ctxfi/ctpcm.c: Remove potential for use after free

In each function, the value apcm is stored in the private_data field of
runtime.  At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure.  ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field.  But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free.  On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c).  To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm.  This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.

In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.

The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@

*e->f = a
... when != e->f = e1
    when any
if (...) {
  ... when != free1(...,e,...)
      when != e->f = e2
* kfree(a)
  ... when != free2(...,e,...)
      when != e->f = e3
}
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctpcm.c | 16 ++++++++++------
 1 file changed, 10 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 85ab43e89212..457d21189b0d 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -129,8 +129,6 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
 
 	apcm->substream = substream;
 	apcm->interrupt = ct_atc_pcm_interrupt;
-	runtime->private_data = apcm;
-	runtime->private_free = ct_atc_pcm_free_substream;
 	if (IEC958 == substream->pcm->device) {
 		runtime->hw = ct_spdif_passthru_playback_hw;
 		atc->spdif_out_passthru(atc, 1);
@@ -155,8 +153,12 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
 	}
 
 	apcm->timer = ct_timer_instance_new(atc->timer, apcm);
-	if (!apcm->timer)
+	if (!apcm->timer) {
+		kfree(apcm);
 		return -ENOMEM;
+	}
+	runtime->private_data = apcm;
+	runtime->private_free = ct_atc_pcm_free_substream;
 
 	return 0;
 }
@@ -278,8 +280,6 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
 	apcm->started = 0;
 	apcm->substream = substream;
 	apcm->interrupt = ct_atc_pcm_interrupt;
-	runtime->private_data = apcm;
-	runtime->private_free = ct_atc_pcm_free_substream;
 	runtime->hw = ct_pcm_capture_hw;
 	runtime->hw.rate_max = atc->rsr * atc->msr;
 
@@ -298,8 +298,12 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
 	}
 
 	apcm->timer = ct_timer_instance_new(atc->timer, apcm);
-	if (!apcm->timer)
+	if (!apcm->timer) {
+		kfree(apcm);
 		return -ENOMEM;
+	}
+	runtime->private_data = apcm;
+	runtime->private_free = ct_atc_pcm_free_substream;
 
 	return 0;
 }
-- 
cgit v1.2.2


From e2e9566230e0c93d89948cbc799a191d35383d09 Mon Sep 17 00:00:00 2001
From: Peter Rosin <peda@lysator.liu.se>
Date: Wed, 10 Nov 2010 15:55:05 +0100
Subject: ALSA: AT73C213: Rectify misleading comment.

The Atmel SSC can divide by even numbers, not only powers of two.

Signed-off-by: Peter Rosin <peda@axentia.se>
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 1bc56b2b94e2..337a00241a1f 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -155,7 +155,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip)
 	if (max_tries < 1)
 		max_tries = 1;
 
-	/* ssc_div must be a power of 2. */
+	/* ssc_div must be even. */
 	ssc_div = (ssc_div + 1) & ~1UL;
 
 	if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) {
-- 
cgit v1.2.2


From 451a3c24b0135bce54542009b5fde43846c7cf67 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Wed, 17 Nov 2010 16:26:55 +0100
Subject: BKL: remove extraneous #include <smp_lock.h>

The big kernel lock has been removed from all these files at some point,
leaving only the #include.

Remove this too as a cleanup.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
---
 sound/core/info.c       | 1 -
 sound/core/pcm_native.c | 1 -
 sound/core/sound.c      | 1 -
 sound/sound_core.c      | 1 -
 4 files changed, 4 deletions(-)

(limited to 'sound')

diff --git a/sound/core/info.c b/sound/core/info.c
index b70564ed8b37..7077f601da5a 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -23,7 +23,6 @@
 #include <linux/time.h>
 #include <linux/mm.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/string.h>
 #include <sound/core.h>
 #include <sound/minors.h>
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 8bc7cb3db330..e82c1f97d99e 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -22,7 +22,6 @@
 #include <linux/mm.h>
 #include <linux/file.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/pm_qos_params.h>
 #include <linux/uio.h>
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 62a093efb453..66691fe437e6 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -21,7 +21,6 @@
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/device.h>
 #include <linux/moduleparam.h>
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c03bbaefdbc3..5580aced8730 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -104,7 +104,6 @@ module_exit(cleanup_soundcore);
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/types.h>
 #include <linux/kernel.h>
 #include <linux/sound.h>
-- 
cgit v1.2.2


From 0613a59456980161d0cd468bae6c63d772743102 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Mon, 1 Nov 2010 01:14:51 -0400
Subject: ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and
 Headphone controls

BugLink: https://launchpad.net/bugs/669279

The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."

Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.

Reported-and-tested-by: Ben Gamari <bgamari@gmail.com>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/intel8x0.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 400f9ebd243e..629a5494347a 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1864,6 +1864,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.name = "Dell Inspiron 8600",	/* STAC9750/51 */
 		.type = AC97_TUNE_HP_ONLY
 	},
+	{
+		.subvendor = 0x1028,
+		.subdevice = 0x0182,
+		.name = "Dell Latitude D610",	/* STAC9750/51 */
+		.type = AC97_TUNE_HP_ONLY
+	},
 	{
 		.subvendor = 0x1028,
 		.subdevice = 0x0186,
-- 
cgit v1.2.2


From 2fb50f135adba59edf2359effcce83eb17025793 Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Fri, 12 Nov 2010 13:38:04 -0800
Subject: ALSA: sound/ppc: Use printf extension %pR for struct resource

Using %pR standardizes the struct resource output.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/ppc/pmac.c | 12 ++++--------
 1 file changed, 4 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 85081172403f..b47cfd45b3b9 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -1228,10 +1228,8 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
 					       chip->rsrc[i].start + 1,
 					       rnames[i]) == NULL) {
 				printk(KERN_ERR "snd: can't request rsrc "
-				       " %d (%s: 0x%016llx:%016llx)\n",
-				       i, rnames[i],
-				       (unsigned long long)chip->rsrc[i].start,
-				       (unsigned long long)chip->rsrc[i].end);
+				       " %d (%s: %pR)\n",
+				       i, rnames[i], &chip->rsrc[i]);
 				err = -ENODEV;
 				goto __error;
 			}
@@ -1256,10 +1254,8 @@ int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
 					       chip->rsrc[i].start + 1,
 					       rnames[i]) == NULL) {
 				printk(KERN_ERR "snd: can't request rsrc "
-				       " %d (%s: 0x%016llx:%016llx)\n",
-				       i, rnames[i],
-				       (unsigned long long)chip->rsrc[i].start,
-				       (unsigned long long)chip->rsrc[i].end);
+				       " %d (%s: %pR)\n",
+				       i, rnames[i], &chip->rsrc[i]);
 				err = -ENODEV;
 				goto __error;
 			}
-- 
cgit v1.2.2


From c80c1d542744dd7851cc8da748c6ada99680fb4d Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Sun, 14 Nov 2010 19:05:02 -0800
Subject: ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_lib.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a1707cca9c66..b75db8e9cc0f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -223,7 +223,7 @@ static void xrun_log(struct snd_pcm_substream *substream,
 	entry->jiffies = jiffies;
 	entry->pos = pos;
 	entry->period_size = runtime->period_size;
-	entry->buffer_size = runtime->buffer_size;;
+	entry->buffer_size = runtime->buffer_size;
 	entry->old_hw_ptr = runtime->status->hw_ptr;
 	entry->hw_ptr_base = runtime->hw_ptr_base;
 	log->idx = (log->idx + 1) % XRUN_LOG_CNT;
-- 
cgit v1.2.2


From 5dbea6b1f2113f764999b39fd3d79b1354c193d9 Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Mon, 15 Nov 2010 12:14:02 -0800
Subject: ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of
 pci_get_drvdata

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpioctl.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 62895a719fcb..22dbd91811a4 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -435,7 +435,7 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev)
 	struct hpi_message hm;
 	struct hpi_response hr;
 	struct hpi_adapter *pa;
-	pa = (struct hpi_adapter *)pci_get_drvdata(pci_dev);
+	pa = pci_get_drvdata(pci_dev);
 
 	hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM,
 		HPI_SUBSYS_DELETE_ADAPTER);
-- 
cgit v1.2.2


From a0e90acc657990511c83bc69965bfd3c63386d45 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sat, 20 Nov 2010 10:20:35 -0500
Subject: ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup

BugLink: https://launchpad.net/bugs/677830

The original reporter states that the subwoofer does not mute when
inserting headphones.  We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).

Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5f00589cb791..1a7703a49655 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19298,6 +19298,7 @@ static const struct alc_fixup alc662_fixups[] = {
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	{}
-- 
cgit v1.2.2


From 78ac07b0d2b09b1ccb7a41a2e25f71d60b652920 Mon Sep 17 00:00:00 2001
From: Andreas Mohr <andi@lisas.de>
Date: Sun, 21 Nov 2010 12:09:32 +0100
Subject: ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer

. Fix PulseAudio "ALSA driver bug" issue
  (if we have two alternated areas within a 64k DMA buffer, then max
  period size should obviously be 32k only).
  Back references:
   http://pulseaudio.org/wiki/AlsaIssues
   http://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction

When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.

PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.

Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/azt3328.c | 26 ++++++++++++++++++--------
 1 file changed, 18 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 4679ed83a43b..2f3cacbd5528 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1129,10 +1129,11 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip,
 
 		count_areas = size/2;
 		addr_area2 = addr+count_areas;
-		count_areas--; /* max. index */
 		snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n",
 				addr, count_areas, addr_area2, count_areas);
 
+		count_areas--; /* max. index */
+
 		/* build combined I/O buffer length word */
 		lengths = (count_areas << 16) | (count_areas);
 		spin_lock_irqsave(&chip->reg_lock, flags);
@@ -1740,11 +1741,15 @@ static const struct snd_pcm_hardware snd_azf3328_hardware =
 	.rate_max =		AZF_FREQ_66200,
 	.channels_min =		1,
 	.channels_max =		2,
-	.buffer_bytes_max =	65536,
-	.period_bytes_min =	64,
-	.period_bytes_max =	65536,
-	.periods_min =		1,
-	.periods_max =		1024,
+	.buffer_bytes_max =	(64*1024),
+	.period_bytes_min =	1024,
+	.period_bytes_max =	(32*1024),
+	/* We simply have two DMA areas (instead of a list of descriptors
+	   such as other cards); I believe that this is a fixed hardware
+	   attribute and there isn't much driver magic to be done to expand it.
+	   Thus indicate that we have at least and at most 2 periods. */
+	.periods_min =		2,
+	.periods_max =		2,
 	/* FIXME: maybe that card actually has a FIFO?
 	 * Hmm, it seems newer revisions do have one, but we still don't know
 	 * its size... */
@@ -1980,8 +1985,13 @@ snd_azf3328_timer_stop(struct snd_timer *timer)
 	chip = snd_timer_chip(timer);
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	/* disable timer countdown and interrupt */
-	/* FIXME: should we write TIMER_IRQ_ACK here? */
-	snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
+	/* Hmm, should we write TIMER_IRQ_ACK here?
+	   YES indeed, otherwise a rogue timer operation - which prompts
+	   ALSA(?) to call repeated stop() in vain, but NOT start() -
+	   will never end (value 0x03 is kept shown in control byte).
+	   Simply manually poking 0x04 _once_ immediately successfully stops
+	   the hardware/ALSA interrupt activity. */
+	snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x04);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	snd_azf3328_dbgcallleave();
 	return 0;
-- 
cgit v1.2.2


From c0763e687d0283d0db507813ca4462aa4073c5b5 Mon Sep 17 00:00:00 2001
From: Vasiliy Kulikov <segoon@openwall.com>
Date: Sun, 21 Nov 2010 20:40:07 +0300
Subject: ALSA: snd-atmel-abdac: test wrong variable

After clk_get() pclk is checked second time instead of sample_clk check.

Signed-off-by: Vasiliy Kulikov <segoon@openwall.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 f2f41c854221..4e47baada66f 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -420,7 +420,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 		return PTR_ERR(pclk);
 	}
 	sample_clk = clk_get(&pdev->dev, "sample_clk");
-	if (IS_ERR(pclk)) {
+	if (IS_ERR(sample_clk)) {
 		dev_dbg(&pdev->dev, "no sample clock\n");
 		retval = PTR_ERR(pclk);
 		goto out_put_pclk;
-- 
cgit v1.2.2


From 673f7a8984c3a9e2cb1108ce221da1ebbd9e5d09 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 21 Nov 2010 14:01:14 -0500
Subject: ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus
 A52J

BugLink: https://launchpad.net/bugs/677652

The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers.  Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality.  Testing was done
with an alsa-driver build from 2010-11-21.

Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6361f752b5f3..3cfb31e77b16 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3100,6 +3100,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
-- 
cgit v1.2.2


From 1beded5d9ce90256e4a7e7b0e96c317eafe1c513 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 22 Nov 2010 08:58:13 +0100
Subject: ALSA: atmel - Fix the return value in error path

In the commit c0763e687d0283d0db507813ca4462aa4073c5b5
    ALSA: snd-atmel-abdac: test wrong variable
the return value via PTR_ERR() had to be fixed as well.

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 4e47baada66f..6e2409181895 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -422,7 +422,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 	sample_clk = clk_get(&pdev->dev, "sample_clk");
 	if (IS_ERR(sample_clk)) {
 		dev_dbg(&pdev->dev, "no sample clock\n");
-		retval = PTR_ERR(pclk);
+		retval = PTR_ERR(sample_clk);
 		goto out_put_pclk;
 	}
 	clk_enable(pclk);
-- 
cgit v1.2.2


From 01e0f1378c47947b825eac05c98697ab1be1c86f Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Mon, 22 Nov 2010 10:59:36 +0100
Subject: ALSA: hda - Fixed ALC887-VD initial error

ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1a7703a49655..564e6c136ddd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19420,7 +19420,10 @@ static int patch_alc888(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
 		kfree(codec->chip_name);
-		codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+		if (codec->vendor_id == 0x10ec0887)
+			codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+		else
+			codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
 		if (!codec->chip_name) {
 			alc_free(codec);
 			return -ENOMEM;
@@ -19910,7 +19913,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
-- 
cgit v1.2.2


From d090f5976dfcac4935286676825d64e081335e09 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 23 Nov 2010 07:39:58 +0100
Subject: ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using
 the same mux on IDT/STAC"

This reverts commit f41cc2a85d52ac6971299922084ac5ac59dc339d.

The patch broke the digital mic pin handling wrongly.
Reference: bko#23162
	https://bugzilla.kernel.org/show_bug.cgi?id=23162

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index cfd73afad882..5c710807dfe5 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3491,10 +3491,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 				return err;
 		}
 
-		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1)
 			snd_hda_add_imux_item(imux, label, index, NULL);
-			spec->num_analog_muxes++;
-		}
 	}
 
 	return 0;
-- 
cgit v1.2.2


From 6027277e77df2d2893d906c42f5c9f9abcb731e0 Mon Sep 17 00:00:00 2001
From: Manoj Iyer <manoj.iyer@canonical.com>
Date: Tue, 23 Nov 2010 07:43:44 +0100
Subject: ALSA: hda - Enable jack sense for Thinkpad Edge 11

Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models.

Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3cfb31e77b16..846d1ead47fd 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3111,6 +3111,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
  	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
-- 
cgit v1.2.2


From 1657cbd87125a623d28ce8a7ef5ff6959098d425 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Tue, 23 Nov 2010 08:53:32 +0100
Subject: ALSA: hda - Fix wrong ALC269 variant check

The refactoring commit d433a67831ab2c470cc53a3ff9b60f656767be15
    ALSA: hda - Optimize the check of ALC269 codec variants
introduced a wrong check for ALC269-vb type.  This patch corrects it.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 564e6c136ddd..38b63fb79cbd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15104,7 +15104,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
 	if (!spec->adc_nids) { /* wasn't filled automatically? use default */
-		if (spec->codec_variant != ALC269_TYPE_NORMAL) {
+		if (spec->codec_variant == ALC269_TYPE_NORMAL) {
 			spec->adc_nids = alc269_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
 			spec->capsrc_nids = alc269_capsrc_nids;
-- 
cgit v1.2.2


From 48c88e820fb3e35c5925e4743fd13f200891b7b5 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Tue, 23 Nov 2010 08:56:16 +0100
Subject: ALSA: hda - Identify more variants for ALC269

Give more correct chip names for ALC269-variant codecs.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 18 +++++++++++++++---
 1 file changed, 15 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 38b63fb79cbd..0ac6aed0c889 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14623,7 +14623,10 @@ static int alc275_setup_dual_adc(struct hda_codec *codec)
 /* different alc269-variants */
 enum {
 	ALC269_TYPE_NORMAL,
+	ALC269_TYPE_ALC258,
 	ALC269_TYPE_ALC259,
+	ALC269_TYPE_ALC269VB,
+	ALC269_TYPE_ALC270,
 	ALC269_TYPE_ALC271X,
 };
 
@@ -15023,7 +15026,7 @@ static int alc269_fill_coef(struct hda_codec *codec)
 static int patch_alc269(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
+	int board_config, coef;
 	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -15034,14 +15037,23 @@ static int patch_alc269(struct hda_codec *codec)
 
 	alc_auto_parse_customize_define(codec);
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){
+	coef = alc_read_coef_idx(codec, 0);
+	if ((coef & 0x00f0) == 0x0010) {
 		if (codec->bus->pci->subsystem_vendor == 0x1025 &&
 		    spec->cdefine.platform_type == 1) {
 			alc_codec_rename(codec, "ALC271X");
 			spec->codec_variant = ALC269_TYPE_ALC271X;
-		} else {
+		} else if ((coef & 0xf000) == 0x1000) {
+			spec->codec_variant = ALC269_TYPE_ALC270;
+		} else if ((coef & 0xf000) == 0x2000) {
 			alc_codec_rename(codec, "ALC259");
 			spec->codec_variant = ALC269_TYPE_ALC259;
+		} else if ((coef & 0xf000) == 0x3000) {
+			alc_codec_rename(codec, "ALC258");
+			spec->codec_variant = ALC269_TYPE_ALC258;
+		} else {
+			alc_codec_rename(codec, "ALC269VB");
+			spec->codec_variant = ALC269_TYPE_ALC269VB;
 		}
 	} else
 		alc_fix_pll_init(codec, 0x20, 0x04, 15);
-- 
cgit v1.2.2


From d94772070acc5a8f312ab4650cbbf5e78ea9dda2 Mon Sep 17 00:00:00 2001
From: Denis Kuplyakov <dener.kup@gmail.com>
Date: Wed, 24 Nov 2010 06:01:09 +0100
Subject: ALSA: hda - Fix Acer 7730G support

Fixes automatic EAPD configuration on Acer 7730G laptop.

Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++++++++++-------
 1 file changed, 41 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ac6aed0c889..8f7530fc7644 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2013,6 +2013,36 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 	{ }
 };
 
+/*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+/* Bias voltage on for external mic port */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
 /*
  * ALC889 Acer Aspire 8930G model
  */
@@ -2200,6 +2230,16 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x17;
 }
 
+static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->autocfg.speaker_pins[2] = 0x17;
+}
+
 static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -9524,13 +9564,6 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{ } /* end */
-};
-
 static void alc888_6st_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -10328,7 +10361,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.const_channel_count = 6,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.setup = alc888_acer_aspire_6530g_setup,
+		.setup = alc888_acer_aspire_7730g_setup,
 		.init_hook = alc_automute_amp,
 	},
 	[ALC883_MEDION] = {
-- 
cgit v1.2.2


From d4bc99b977e3a1dd10a84a01ebe59ac2ccebf0cd Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 24 Nov 2010 02:44:06 +0000
Subject: ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMI

Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
---
 sound/soc/sh/fsi.c | 19 +++++++++++++++++--
 1 file changed, 17 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 507e709f2807..136414f163e9 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -132,6 +132,8 @@ struct fsi_priv {
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
+	long rate;
+
 	u32 mst_ctrl;
 };
 
@@ -854,10 +856,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	int is_play = fsi_is_play(substream);
+	struct fsi_master *master = fsi_get_master(fsi);
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 
 	fsi_irq_disable(fsi, is_play);
 	fsi_clk_ctrl(fsi, 0);
 
+	set_rate = master->info->set_rate;
+	if (set_rate && fsi->rate)
+		set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
+	fsi->rate = 0;
+
 	pm_runtime_put_sync(dai->dev);
 }
 
@@ -891,9 +900,10 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_master *master = fsi_get_master(fsi);
-	int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 	int fsi_ver = master->core->ver;
 	int is_play = fsi_is_play(substream);
+	long rate = params_rate(params);
 	int ret;
 
 	/* if slave mode, set_rate is not needed */
@@ -901,10 +911,15 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 
 	/* it is error if no set_rate */
+	set_rate = master->info->set_rate;
 	if (!set_rate)
 		return -EIO;
 
-	ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+	ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
+	if (ret < 0) /* error */
+		return ret;
+
+	fsi->rate = rate;
 	if (ret > 0) {
 		u32 data = 0;
 
-- 
cgit v1.2.2


From 22de4e1fe446794acaebdf19dcaff4256d659972 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 19 Nov 2010 07:23:17 +0000
Subject: ARM: mach-shmobile: ap4evb: FSI clock use proper process for ak4642

Current AP4 FSI didn't use set_rate for ak4642,
and used dummy rate when init.
And FSI driver was modified to always call set_rate.

The user which are using FSI set_rate is only AP4 now.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
---
 sound/soc/sh/fsi.c | 8 +-------
 1 file changed, 1 insertion(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 136414f163e9..4c2404b1b862 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -902,18 +902,12 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 	struct fsi_master *master = fsi_get_master(fsi);
 	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 	int fsi_ver = master->core->ver;
-	int is_play = fsi_is_play(substream);
 	long rate = params_rate(params);
 	int ret;
 
-	/* if slave mode, set_rate is not needed */
-	if (!fsi_is_master_mode(fsi, is_play))
-		return 0;
-
-	/* it is error if no set_rate */
 	set_rate = master->info->set_rate;
 	if (!set_rate)
-		return -EIO;
+		return 0;
 
 	ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
 	if (ret < 0) /* error */
-- 
cgit v1.2.2


From cc1c452e509aefc28f7ad2deed75bc69d4f915f7 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 24 Nov 2010 14:17:47 +0100
Subject: ALSA: HDA: Add an extra DAC for Realtek ALC887-VD

The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.

Signed-off-by: David Henningsson <david.henningsson@canonical.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 8f7530fc7644..b0e6b8b47fa9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -18997,6 +18997,8 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 		return 0x02;
 	else if (nid >= 0x0c && nid <= 0x0e)
 		return nid - 0x0c + 0x02;
+	else if (nid == 0x26) /* ALC887-VD has this DAC too */
+		return 0x25;
 	else
 		return 0;
 }
@@ -19005,7 +19007,7 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
 				   hda_nid_t dac)
 {
-	hda_nid_t mix[4];
+	hda_nid_t mix[5];
 	int i, num;
 
 	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
-- 
cgit v1.2.2


From 7167594a3da7dcc33203b85d62e519594baee390 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Thu, 25 Nov 2010 00:08:01 -0200
Subject: ALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixers

The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).

Reference: https://qa.mandriva.com/show_bug.cgi?id=61159

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b0e6b8b47fa9..81a2a49b862c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16943,7 +16943,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+	return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0);
 }
 
 
-- 
cgit v1.2.2


From 5a8cfb4e8ae317d283f84122ed20faa069c5e0c4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 26 Nov 2010 17:11:18 +0100
Subject: ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization

When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far.  This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.

Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.

This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.

Reference:
	https://bugzilla.redhat.com/show_bug.cgi?id=657388

Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 81a2a49b862c..886d7c72936e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1614,6 +1614,7 @@ do_sku:
 		spec->init_amp = ALC_INIT_GPIO3;
 		break;
 	case 5:
+	default:
 		spec->init_amp = ALC_INIT_DEFAULT;
 		break;
 	}
-- 
cgit v1.2.2


From ac70eb1305d5a81efd1e32327d7e79be15a63a5a Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sat, 27 Nov 2010 13:58:04 -0500
Subject: ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for
 MEDION MD2

BugLink: https://launchpad.net/bugs/682199

A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID.  Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.

Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.

Reported-and-tested-by: burningphantom1
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 886d7c72936e..8fddc9d08726 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9865,7 +9865,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
-	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
 	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
-- 
cgit v1.2.2


From 60686aa0086a14f8b15c83a09f3df1eebe3aab3c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 30 Nov 2010 08:14:21 +0100
Subject: ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation

In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal
buffer state in addition to drop of the running streams.  Otherwise the
succeeding access becomes inconsistent.

Tested-by: Amit Nagal <helloin.amit@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/oss/pcm_oss.c | 19 +++++++++++--------
 1 file changed, 11 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 5c8c7dff8ede..b753ec661fcf 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1510,16 +1510,19 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
 static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file)
 {
 	struct snd_pcm_substream *substream;
+	struct snd_pcm_runtime *runtime;
+	int i;
 
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
-	if (substream != NULL) {
-		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
-	}
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
-	if (substream != NULL) {
+	for (i = 0; i < 2; i++) { 
+		substream = pcm_oss_file->streams[i];
+		if (!substream)
+			continue;
+		runtime = substream->runtime;
 		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
+		runtime->oss.prepare = 1;
+		runtime->oss.buffer_used = 0;
+		runtime->oss.prev_hw_ptr_period = 0;
+		runtime->oss.period_ptr = 0;
 	}
 	return 0;
 }
-- 
cgit v1.2.2


From 0defe09ca70daccdc83abd9c3c24cd89ae6a1141 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Wed, 1 Dec 2010 19:16:07 -0500
Subject: ALSA: hda: Use "alienware" model quirk for another SSID

BugLink: https://launchpad.net/bugs/683695

The original reporter states that headphone jacks do not appear to
work.  Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.

Reported-and-tested-by: Cody Thierauf
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 5c710807dfe5..efa4225f5fd6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1627,6 +1627,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
+		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.2


From 5b84ba26a9672e615897234fa5efd3eea2d6b295 Mon Sep 17 00:00:00 2001
From: Tejun Heo <tj@kernel.org>
Date: Sat, 11 Dec 2010 17:51:26 +0100
Subject: sound: don't use flush_scheduled_work()

flush_scheduled_work() is deprecated and scheduled to be removed.

* cancel[_delayed]_work() + flush_scheduled_work() ->
  cancel[_delayed]_work_sync().

* wm8350, wm8753 and soc-core use custom code to cancel a delayed
  work, execute it immediately if it was pending and wait for its
  completion.  This is equivalent to flush_delayed_work_sync().  Use
  it instead.

Signed-off-by: Tejun Heo <tj@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/aoa/core/gpio-feature.c |  7 +++----
 sound/aoa/core/gpio-pmf.c     |  7 +++----
 sound/i2c/other/ak4113.c      |  5 ++---
 sound/i2c/other/ak4114.c      |  5 ++---
 sound/pci/ac97/ac97_codec.c   |  6 ++----
 sound/pci/hda/patch_via.c     |  3 +--
 sound/pci/oxygen/oxygen_lib.c |  6 ++++--
 sound/soc/codecs/wm8350.c     |  9 +--------
 sound/soc/codecs/wm8753.c     | 21 +--------------------
 sound/soc/soc-core.c          | 25 +++----------------------
 10 files changed, 22 insertions(+), 72 deletions(-)

(limited to 'sound')

diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index de8e03afa97b..faa317490545 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -287,10 +287,9 @@ static void ftr_gpio_exit(struct gpio_runtime *rt)
 		free_irq(linein_detect_irq, &rt->line_in_notify);
 	if (rt->line_out_notify.gpio_private)
 		free_irq(lineout_detect_irq, &rt->line_out_notify);
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
 	mutex_destroy(&rt->line_out_notify.mutex);
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 7e267c9379bc..c8d8a1a6f964 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -107,10 +107,9 @@ static void pmf_gpio_exit(struct gpio_runtime *rt)
 
 	/* make sure no work is pending before freeing
 	 * all things */
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index 971a84a4fa77..c424d329f806 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -57,8 +57,7 @@ static void snd_ak4113_free(struct ak4113 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -141,7 +140,7 @@ void snd_ak4113_reinit(struct ak4113 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4113_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 0341451f814c..d9fb537b0b94 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -67,8 +67,7 @@ static void snd_ak4114_free(struct ak4114 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -154,7 +153,7 @@ void snd_ak4114_reinit(struct ak4114 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4114_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a7630e9edf8a..0fc614ce16c1 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1014,8 +1014,7 @@ static int snd_ac97_free(struct snd_ac97 *ac97)
 {
 	if (ac97) {
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-		cancel_delayed_work(&ac97->power_work);
-		flush_scheduled_work();
+		cancel_delayed_work_sync(&ac97->power_work);
 #endif
 		snd_ac97_proc_done(ac97);
 		if (ac97->bus)
@@ -2456,8 +2455,7 @@ void snd_ac97_suspend(struct snd_ac97 *ac97)
 	if (ac97->build_ops->suspend)
 		ac97->build_ops->suspend(ac97);
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-	cancel_delayed_work(&ac97->power_work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&ac97->power_work);
 #endif
 	snd_ac97_powerdown(ac97);
 }
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index d1c3f8defc48..7f4852a478a1 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -263,8 +263,7 @@ static void vt1708_stop_hp_work(struct via_spec *spec)
 		return;
 	snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
 			    !spec->vt1708_jack_detectect);
-	cancel_delayed_work(&spec->vt1708_hp_work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&spec->vt1708_hp_work);
 }
 
 
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index e5ebe56fb0c5..969605fbcb7f 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -557,7 +557,8 @@ static void oxygen_card_free(struct snd_card *card)
 	oxygen_shutdown(chip);
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
-	flush_scheduled_work();
+	flush_work_sync(&chip->spdif_input_bits_work);
+	flush_work_sync(&chip->gpio_work);
 	chip->model.cleanup(chip);
 	kfree(chip->model_data);
 	mutex_destroy(&chip->mutex);
@@ -733,7 +734,8 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
 	spin_unlock_irq(&chip->reg_lock);
 
 	synchronize_irq(chip->irq);
-	flush_scheduled_work();
+	flush_work_sync(&chip->spdif_input_bits_work);
+	flush_work_sync(&chip->gpio_work);
 	chip->interrupt_mask = saved_interrupt_mask;
 
 	pci_disable_device(pci);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 7611add7f8c3..b3e9fac172e5 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1626,7 +1626,6 @@ static int  wm8350_codec_remove(struct snd_soc_codec *codec)
 {
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350 *wm8350 = dev_get_platdata(codec->dev);
-	int ret;
 
 	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1641,15 +1640,9 @@ static int  wm8350_codec_remove(struct snd_soc_codec *codec)
 	priv->hpr.jack = NULL;
 	priv->mic.jack = NULL;
 
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(&codec->delayed_work);
-
 	/* if there was any work waiting then we run it now and
 	 * wait for its completion */
-	if (ret) {
-		schedule_delayed_work(&codec->delayed_work, 0);
-		flush_scheduled_work();
-	}
+	flush_delayed_work_sync(&codec->delayed_work);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8f679a13f2bc..84a23675cba9 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1526,25 +1526,6 @@ static int wm8753_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
-	int ret;
-
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(dwork);
-
-	/* if there was any work waiting then we run it now and
-	 * wait for it's completion */
-	if (ret) {
-		schedule_delayed_work(dwork, 0);
-		flush_scheduled_work();
-	}
-	return ret;
-}
-
 static int wm8753_probe(struct snd_soc_codec *codec)
 {
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
@@ -1604,7 +1585,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 /* power down chip */
 static int wm8753_remove(struct snd_soc_codec *codec)
 {
-	run_delayed_work(&codec->delayed_work);
+	flush_delayed_work_sync(&codec->delayed_work);
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 441285ade024..b54ea9a0a1db 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -67,25 +67,6 @@ static int pmdown_time = 5000;
 module_param(pmdown_time, int, 0);
 MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
 
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
-	int ret;
-
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(dwork);
-
-	/* if there was any work waiting then we run it now and
-	 * wait for it's completion */
-	if (ret) {
-		schedule_delayed_work(dwork, 0);
-		flush_scheduled_work();
-	}
-	return ret;
-}
-
 /* codec register dump */
 static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
@@ -1016,7 +997,7 @@ static int soc_suspend(struct device *dev)
 
 	/* close any waiting streams and save state */
 	for (i = 0; i < card->num_rtd; i++) {
-		run_delayed_work(&card->rtd[i].delayed_work);
+		flush_delayed_work_sync(&card->rtd[i].delayed_work);
 		card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level;
 	}
 
@@ -1687,7 +1668,7 @@ static int soc_remove(struct platform_device *pdev)
 		/* make sure any delayed work runs */
 		for (i = 0; i < card->num_rtd; i++) {
 			struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
-			run_delayed_work(&rtd->delayed_work);
+			flush_delayed_work_sync(&rtd->delayed_work);
 		}
 
 		/* remove and free each DAI */
@@ -1718,7 +1699,7 @@ static int soc_poweroff(struct device *dev)
 	 * now, we're shutting down so no imminent restart. */
 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
-		run_delayed_work(&rtd->delayed_work);
+		flush_delayed_work_sync(&rtd->delayed_work);
 	}
 
 	snd_soc_dapm_shutdown(card);
-- 
cgit v1.2.2


From fdea0571ddca8e3f22448f66d72a034575abea28 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 13 Dec 2010 12:55:39 +0100
Subject: ASoC: Fix merge errors with flush_scheduled_work() removal

delayed_work was moved to dapm in the commit
ce6120cca2589ede530200c7cfe11ac9f144333c
    ASoC: Decouple DAPM from CODECs

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/wm8350.c | 2 +-
 sound/soc/codecs/wm8753.c | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f8a8a6944e65..07ba7e3f6a8c 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1643,7 +1643,7 @@ static int  wm8350_codec_remove(struct snd_soc_codec *codec)
 
 	/* if there was any work waiting then we run it now and
 	 * wait for its completion */
-	flush_delayed_work_sync(&codec->delayed_work);
+	flush_delayed_work_sync(&codec->dapm.delayed_work);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 747457193887..73507e71cb79 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1587,7 +1587,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 /* power down chip */
 static int wm8753_remove(struct snd_soc_codec *codec)
 {
-	flush_delayed_work_sync(&codec->delayed_work);
+	flush_delayed_work_sync(&codec->dapm.delayed_work);
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
-- 
cgit v1.2.2