[PATCH 0/4] ASoC updates

[PATCH 0/4] ASoC updates

[Mark Brown]

This series fixes a couple of bugs (neither of which affect 2.6.27) and
introduces a new API call for marking disconnected pins, mostly as a
placeholder for future development so machine drivers can start using it
now and then benefit when a more substantial implementation follows.

The following changes since commit 3090258fed179996ac4daecb482bad28d84cb054:
  Jean Delvare (1):
        ALSA: ASoC: Convert tlv320aic3x to a new-style i2c driver (v2)

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Mark Brown (4):
      ASoC: Fix inverted input PGA mute bits in WM8903
      ASoC: Fix build of GTA01 audio driver
      ASoC: Allow machine drivers to mark pins as not connected
      ASoC: Use snd_soc_dapm_nc_pin() in GTA01 audio driver

 include/sound/soc-dapm.h           |    1 +
 sound/soc/codecs/wm8903.c          |    4 ++--
 sound/soc/s3c24xx/neo1973_wm8753.c |   17 ++++++++---------
 sound/soc/soc-dapm.c               |   20 ++++++++++++++++++++
 4 files changed, 31 insertions(+), 11 deletions(-)

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Wed, 24 Sep 2008 12:59:07 +0100,
Mark Brown wrote:
> 
> This series fixes a couple of bugs (neither of which affect 2.6.27) and
> introduces a new API call for marking disconnected pins, mostly as a
> placeholder for future development so machine drivers can start using it
> now and then benefit when a more substantial implementation follows.
> 
> The following changes since commit 3090258fed179996ac4daecb482bad28d84cb054:
>   Jean Delvare (1):
>         ALSA: ASoC: Convert tlv320aic3x to a new-style i2c driver (v2)
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Applied.  Thanks.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
  Takashi Iwai (1):
        ALSA: ASoC - Remove unnecessary inclusion of linux/version.h

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

David Anders (1):
      ASoC: Add new parameter to s3c24xx_pcm_enqueue

Marek Vasut (1):
      ASoC: Add Palm/PXA27x unified ASoC audio driver

Mark Brown (1):
      ASoC: Remove core version number

Troy Kisky (1):
      ASoC: TLV320AIC23B Support more sample rates

 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 include/sound/soc.h                       |    2 -
 sound/soc/codecs/tlv320aic23.c            |  227 +++++++++++++++++++-----
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
 sound/soc/soc-core.c                      |    1 -
 8 files changed, 483 insertions(+), 52 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
  Takashi Iwai (1):
        ALSA: ASoC - Remove unnecessary inclusion of linux/version.h

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

This series drops the TLV320AIC23 patch for now and makes the Palm
driver non-modular for the time being due to the platform data dodge
being used.

David Anders (1):
      ASoC: Add new parameter to s3c24xx_pcm_enqueue

Grazvydas Ignotas (1):
      ALSA: ASoC: TWL4030 codec - fix 256*Fs clock

Marek Vasut (1):
      ASoC: Add Palm/PXA27x unified ASoC audio driver

Mark Brown (1):
      ASoC: Remove core version number

 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 include/sound/soc.h                       |    2 -
 sound/soc/codecs/twl4030.c                |    4 +-
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
 sound/soc/soc-core.c                      |    1 -
 8 files changed, 303 insertions(+), 9 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 6 Nov 2008 11:38:32 +0000,
Mark Brown wrote:
> 
> The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
>   Takashi Iwai (1):
>         ALSA: ASoC - Remove unnecessary inclusion of linux/version.h
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled now.


Takashi

> 
> This series drops the TLV320AIC23 patch for now and makes the Palm
> driver non-modular for the time being due to the platform data dodge
> being used.
> 
> David Anders (1):
>       ASoC: Add new parameter to s3c24xx_pcm_enqueue
> 
> Grazvydas Ignotas (1):
>       ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
> 
> Marek Vasut (1):
>       ASoC: Add Palm/PXA27x unified ASoC audio driver
> 
> Mark Brown (1):
>       ASoC: Remove core version number
> 
>  arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
>  include/sound/soc.h                       |    2 -
>  sound/soc/codecs/twl4030.c                |    4 +-
>  sound/soc/pxa/Kconfig                     |    9 +
>  sound/soc/pxa/Makefile                    |    2 +
>  sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
>  sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
>  sound/soc/soc-core.c                      |    1 -
>  8 files changed, 303 insertions(+), 9 deletions(-)
>  create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
>  create mode 100644 sound/soc/pxa/palm27x.c
> 

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d:
  Grazvydas Ignotas (1):
        ALSA: ASoC: TWL4030 codec - fix 256*Fs clock

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Christian Pellegrin (1):
      ASoC: s3c24xx 8 bit sound fix

Hugo Villeneuve (1):
      ASoC: Add Right-Justified mode and Codec clock master to davinci-i2s

Naresh Medisetty (1):
      ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback

Troy Kisky (1):
      ASoC: TLV320AIC23B Support more sample rates

 sound/soc/codecs/tlv320aic23.c  |  222 +++++++++++++++++++++++++++++++--------
 sound/soc/davinci/davinci-i2s.c |   89 ++++++++++++++--
 sound/soc/s3c24xx/s3c24xx-i2s.c |    7 ++
 3 files changed, 263 insertions(+), 55 deletions(-)

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Wed, 12 Nov 2008 11:55:07 +0000,
Mark Brown wrote:
> 
> The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d:
>   Grazvydas Ignotas (1):
>         ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled in, and pushed out.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit fb0ef645f2c546f8297b2fbf9b2b8fff4a7455e8:
  Naresh Medisetty (1):
        ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Jarkko Nikula (2):
      ASoC: Fix supported sample rates of TWL4030 audio codec
      ASoC: OMAP: Add more supported sample rates into McBSP DAI driver

Mark Brown (2):
      ASoC: Revert "ASoC: Add new parameter to s3c24xx_pcm_enqueue"
      ASoC: Add WM8728 codec driver

 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/twl4030.c      |    2 +-
 sound/soc/codecs/wm8728.c       |  574 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8728.h       |   30 ++
 sound/soc/omap/omap-mcbsp.c     |    4 +-
 sound/soc/s3c24xx/s3c24xx-pcm.c |   12 +-
 7 files changed, 616 insertions(+), 12 deletions(-)
 create mode 100644 sound/soc/codecs/wm8728.c
 create mode 100644 sound/soc/codecs/wm8728.h

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Fri, 14 Nov 2008 14:57:06 +0000,
Mark Brown wrote:
> 
> The following changes since commit fb0ef645f2c546f8297b2fbf9b2b8fff4a7455e8:
>   Naresh Medisetty (1):
>         ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, applied now.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2:
  Daniel Mack (1):
        ASoC: tlv320aic3x: headset/button press support

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Mark Brown (4):
      ASoC: Push debugfs files out of the snd_soc_device structure
      ASoC: Remove device from platform suspend and resume operations
      ASoC: Remove platform device from DAI suspend and resume operations
      ASoC: Remove obsolete declaration of struct snd_soc_clock_info

 include/sound/soc-dai.h         |    6 +--
 include/sound/soc.h             |   15 +++----
 sound/soc/atmel/atmel-pcm.c     |    6 +--
 sound/soc/atmel/atmel_ssc_dai.c |    6 +--
 sound/soc/au1x/psc-ac97.c       |    6 +--
 sound/soc/au1x/psc-i2s.c        |    6 +--
 sound/soc/blackfin/bf5xx-ac97.c |    6 +--
 sound/soc/blackfin/bf5xx-i2s.c  |    6 +--
 sound/soc/pxa/pxa-ssp.c         |    6 +--
 sound/soc/pxa/pxa2xx-ac97.c     |    6 +--
 sound/soc/pxa/pxa2xx-i2s.c      |    6 +--
 sound/soc/s3c24xx/s3c2412-i2s.c |   16 +++----
 sound/soc/s3c24xx/s3c24xx-i2s.c |    6 +--
 sound/soc/soc-core.c            |   85 +++++++++++++++++++++-----------------
 14 files changed, 83 insertions(+), 99 deletions(-)

[PATCH 1/4] ASoC: Push debugfs files out of the snd_soc_device structure

[Mark Brown]

This is in preparation for the removal of struct snd_soc_device.

The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |    8 +++--
 sound/soc/soc-core.c |   73 ++++++++++++++++++++++++++++----------------------
 2 files changed, 46 insertions(+), 35 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index ad8141a..3ee608d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -279,6 +279,11 @@ struct snd_soc_codec {
 	/* codec DAI's */
 	struct snd_soc_dai *dai;
 	unsigned int num_dai;
+
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_reg;
+	struct dentry *debugfs_pop_time;
+#endif
 };
 
 /* codec device */
@@ -364,9 +369,6 @@ struct snd_soc_device {
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	void *codec_data;
-#ifdef CONFIG_DEBUG_FS
-	struct dentry	*debugfs_root;
-#endif
 };
 
 /* runtime channel data */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe89260..3411439 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -39,6 +39,10 @@ static DEFINE_MUTEX(pcm_mutex);
 static DEFINE_MUTEX(io_mutex);
 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_root;
+#endif
+
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
  * It can be used to eliminate pops between different playback streams, e.g.
@@ -1002,7 +1006,9 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
 			       size_t count, loff_t *ppos)
 {
 	ssize_t ret;
-	struct snd_soc_device *devdata = file->private_data;
+	struct snd_soc_codec *codec = file->private_data;
+	struct device *card_dev = codec->card->dev;
+	struct snd_soc_device *devdata = card_dev->driver_data;
 	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
@@ -1021,8 +1027,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 	char *start = buf;
 	unsigned long reg, value;
 	int step = 1;
-	struct snd_soc_device *devdata = file->private_data;
-	struct snd_soc_codec *codec = devdata->codec;
+	struct snd_soc_codec *codec = file->private_data;
 
 	buf_size = min(count, (sizeof(buf)-1));
 	if (copy_from_user(buf, user_buf, buf_size))
@@ -1051,44 +1056,36 @@ static const struct file_operations codec_reg_fops = {
 	.write = codec_reg_write_file,
 };
 
-static void soc_init_debugfs(struct snd_soc_device *socdev)
+static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
 {
-	struct dentry *root, *file;
-	struct snd_soc_codec *codec = socdev->codec;
-	root = debugfs_create_dir(dev_name(socdev->dev), NULL);
-	if (IS_ERR(root) || !root)
-		goto exit1;
-
-	file = debugfs_create_file("codec_reg", 0644,
-			root, socdev, &codec_reg_fops);
-	if (!file)
-		goto exit2;
-
-	file = debugfs_create_u32("dapm_pop_time", 0744,
-			root, &codec->pop_time);
-	if (!file)
-		goto exit2;
-	socdev->debugfs_root = root;
-	return;
-exit2:
-	debugfs_remove_recursive(root);
-exit1:
-	dev_err(socdev->dev, "debugfs is not available\n");
+	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
+						 debugfs_root, codec,
+						 &codec_reg_fops);
+	if (!codec->debugfs_reg)
+		printk(KERN_WARNING
+		       "ASoC: Failed to create codec register debugfs file\n");
+
+	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+						     debugfs_root,
+						     &codec->pop_time);
+	if (!codec->debugfs_pop_time)
+		printk(KERN_WARNING
+		       "Failed to create pop time debugfs file\n");
 }
 
-static void soc_cleanup_debugfs(struct snd_soc_device *socdev)
+static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
 {
-	debugfs_remove_recursive(socdev->debugfs_root);
-	socdev->debugfs_root = NULL;
+	debugfs_remove(codec->debugfs_pop_time);
+	debugfs_remove(codec->debugfs_reg);
 }
 
 #else
 
-static inline void soc_init_debugfs(struct snd_soc_device *socdev)
+static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
 {
 }
 
-static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev)
+static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
 {
 }
 #endif
@@ -1305,7 +1302,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
-	soc_init_debugfs(socdev);
+	soc_init_codec_debugfs(socdev->codec);
 	mutex_unlock(&codec->mutex);
 
 out:
@@ -1329,7 +1326,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 #endif
 
 	mutex_lock(&codec->mutex);
-	soc_cleanup_debugfs(socdev);
+	soc_cleanup_codec_debugfs(socdev->codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
@@ -1962,11 +1959,23 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
 static int __devinit snd_soc_init(void)
 {
+#ifdef CONFIG_DEBUG_FS
+	debugfs_root = debugfs_create_dir("asoc", NULL);
+	if (IS_ERR(debugfs_root) || !debugfs_root) {
+		printk(KERN_WARNING
+		       "ASoC: Failed to create debugfs directory\n");
+		debugfs_root = NULL;
+	}
+#endif
+
 	return platform_driver_register(&soc_driver);
 }
 
 static void __exit snd_soc_exit(void)
 {
+#ifdef CONFIG_DEBUG_FS
+	debugfs_remove_recursive(debugfs_root);
+#endif
 	platform_driver_unregister(&soc_driver);
 }
 
-- 
1.5.6.5

[PATCH 2/4] ASoC: Remove device from platform suspend and resume operations

[Mark Brown]

None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h         |    6 ++----
 sound/soc/atmel/atmel-pcm.c |    6 ++----
 sound/soc/soc-core.c        |    4 ++--
 3 files changed, 6 insertions(+), 10 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3ee608d..8ec63c0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -300,10 +300,8 @@ struct snd_soc_platform {
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
 	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 394412f..8507aa1 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -417,8 +417,7 @@ static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm)
 }
 
 #ifdef CONFIG_PM
-static int atmel_pcm_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int atmel_pcm_suspend(struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct atmel_runtime_data *prtd;
@@ -442,8 +441,7 @@ static int atmel_pcm_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int atmel_pcm_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int atmel_pcm_resume(struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct atmel_runtime_data *prtd;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3411439..f83852f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -652,7 +652,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 		if (cpu_dai->suspend && !cpu_dai->ac97_control)
 			cpu_dai->suspend(pdev, cpu_dai);
 		if (platform->suspend)
-			platform->suspend(pdev, cpu_dai);
+			platform->suspend(cpu_dai);
 	}
 
 	/* close any waiting streams and save state */
@@ -741,7 +741,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		if (cpu_dai->resume && !cpu_dai->ac97_control)
 			cpu_dai->resume(pdev, cpu_dai);
 		if (platform->resume)
-			platform->resume(pdev, cpu_dai);
+			platform->resume(cpu_dai);
 	}
 
 	if (card->resume_post)
-- 
1.5.6.5

[PATCH 3/4] ASoC: Remove platform device from DAI suspend and resume operations

[Mark Brown]

None of the DAIs use it except s3c2412-i2s which only uses it for
dev_() printouts.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h         |    6 ++----
 sound/soc/atmel/atmel_ssc_dai.c |    6 ++----
 sound/soc/au1x/psc-ac97.c       |    6 ++----
 sound/soc/au1x/psc-i2s.c        |    6 ++----
 sound/soc/blackfin/bf5xx-ac97.c |    6 ++----
 sound/soc/blackfin/bf5xx-i2s.c  |    6 ++----
 sound/soc/pxa/pxa-ssp.c         |    6 ++----
 sound/soc/pxa/pxa2xx-ac97.c     |    6 ++----
 sound/soc/pxa/pxa2xx-i2s.c      |    6 ++----
 sound/soc/s3c24xx/s3c2412-i2s.c |   16 +++++++---------
 sound/soc/s3c24xx/s3c24xx-i2s.c |    6 ++----
 sound/soc/soc-core.c            |    8 ++++----
 12 files changed, 31 insertions(+), 53 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index a01a24b..e2d5f76 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -191,10 +191,8 @@ struct snd_soc_dai {
 		     struct snd_soc_dai *dai);
 	void (*remove)(struct platform_device *pdev,
 		       struct snd_soc_dai *dai);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* ops */
 	struct snd_soc_dai_ops ops;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 0bb18df..d9b874c 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -628,8 +628,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 
 
 #ifdef CONFIG_PM
-static int atmel_ssc_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *cpu_dai)
+static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
 {
 	struct atmel_ssc_info *ssc_p;
 
@@ -657,8 +656,7 @@ static int atmel_ssc_suspend(struct platform_device *pdev,
 
 
 
-static int atmel_ssc_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *cpu_dai)
+static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 {
 	struct atmel_ssc_info *ssc_p;
 	u32 cr;
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a0bcfea..a1e824d 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -314,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	au1xpsc_ac97_workdata = NULL;
 }
 
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -329,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f4217e7..16f9746 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -339,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	au1xpsc_i2s_workdata = NULL;
 }
 
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
-			       struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -354,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 709bdf0..c602ce1 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -269,8 +269,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 #ifdef CONFIG_PM
-static int bf5xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -285,8 +284,7 @@ static int bf5xx_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int bf5xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	int ret;
 	struct sport_device *sport =
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 6e5036b..9f8ce87 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -222,16 +222,14 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
 }
 
 #ifdef CONFIG_PM
-static int bf5xx_i2s_suspend(struct platform_device *dev,
-			     struct snd_soc_dai *dai)
+static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 402fc5b..73fa10d 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -244,8 +244,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 
 #ifdef CONFIG_PM
 
-static int pxa_ssp_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
 {
 	struct ssp_priv *priv = cpu_dai->private_data;
 
@@ -257,8 +256,7 @@ static int pxa_ssp_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int pxa_ssp_resume(struct platform_device *pdev,
-	struct snd_soc_dai *cpu_dai)
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
 {
 	struct ssp_priv *priv = cpu_dai->private_data;
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index bffbe28..8eed80d 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -87,14 +87,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 };
 
 #ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_suspend();
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_resume();
 }
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index f9a9e2e..314973a 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -293,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 }
 
 #ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -311,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 	return 0;
 }
 
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1c74104..75f87c3 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -649,8 +649,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct platform_device *dev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 	u32 iismod;
@@ -665,25 +664,24 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
 		iismod = readl(i2s->regs + S3C2412_IISMOD);
 
 		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+			pr_warning("%s: RXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+			pr_warning("%s: TXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+			pr_warning("%s: IIS active\n", __func__);
 	}
 
 	return 0;
 }
 
-static int s3c2412_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 
-	dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
-		 dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
 
 	if (dai->active) {
 		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 8d9135f..45fe8f7 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -419,8 +419,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c24xx_i2s_suspend(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -434,8 +433,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int s3c24xx_i2s_resume(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f83852f..2f2a8d9 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -650,7 +650,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && !cpu_dai->ac97_control)
-			cpu_dai->suspend(pdev, cpu_dai);
+			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
 			platform->suspend(cpu_dai);
 	}
@@ -676,7 +676,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->suspend && cpu_dai->ac97_control)
-			cpu_dai->suspend(pdev, cpu_dai);
+			cpu_dai->suspend(cpu_dai);
 	}
 
 	if (card->suspend_post)
@@ -712,7 +712,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && cpu_dai->ac97_control)
-			cpu_dai->resume(pdev, cpu_dai);
+			cpu_dai->resume(cpu_dai);
 	}
 
 	if (codec_dev->resume)
@@ -739,7 +739,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->resume && !cpu_dai->ac97_control)
-			cpu_dai->resume(pdev, cpu_dai);
+			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
 			platform->resume(cpu_dai);
 	}
-- 
1.5.6.5

[PATCH 4/4] ASoC: Remove obsolete declaration of struct snd_soc_clock_info

[Mark Brown]

The struct is never defined.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h |    1 -
 1 files changed, 0 insertions(+), 1 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8ec63c0..79d855d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -152,7 +152,6 @@ struct snd_soc_dai;
 struct snd_soc_codec;
 struct soc_enum;
 struct snd_soc_ac97_ops;
-struct snd_soc_clock_info;
 
 typedef int (*hw_write_t)(void *,const char* ,int);
 typedef int (*hw_read_t)(void *,char* ,int);
-- 
1.5.6.5

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 4 Dec 2008 11:25:43 +0000,
Mark Brown wrote:
> 
> The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2:
>   Daniel Mack (1):
>         ASoC: tlv320aic3x: headset/button press support
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Pulled now.  Thanks.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583:
  Stanley.Miao (1):
        ASoC: TWL4030: hands-free start-up sequence.

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

Alexander Beregalov (1):
      ASoC: switch davinci DPRINTK to pr_debug()

Mark Brown (3):
      ASoC: Ease merge difficulties from new architectures
      ASoC: Complain if we fail to create DAPM controls
      ASoC: Add WM8350 AudioPlus codec driver

 include/linux/mfd/wm8350/audio.h |   38 +-
 sound/soc/Kconfig                |   10 +-
 sound/soc/Makefile               |   12 +-
 sound/soc/codecs/Kconfig         |    4 +
 sound/soc/codecs/Makefile        |    2 +
 sound/soc/codecs/wm8350.c        | 1583 ++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8350.h        |   20 +
 sound/soc/davinci/davinci-pcm.c  |   18 +-
 sound/soc/soc-dapm.c             |    6 +-
 9 files changed, 1669 insertions(+), 24 deletions(-)
 create mode 100644 sound/soc/codecs/wm8350.c
 create mode 100644 sound/soc/codecs/wm8350.h

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 18 Dec 2008 17:30:21 +0000,
Mark Brown wrote:
> 
> The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583:
>   Stanley.Miao (1):
>         ASoC: TWL4030: hands-free start-up sequence.
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

Applied now.  Thanks.


Takashi

> 
> Alexander Beregalov (1):
>       ASoC: switch davinci DPRINTK to pr_debug()
> 
> Mark Brown (3):
>       ASoC: Ease merge difficulties from new architectures
>       ASoC: Complain if we fail to create DAPM controls
>       ASoC: Add WM8350 AudioPlus codec driver
> 
>  include/linux/mfd/wm8350/audio.h |   38 +-
>  sound/soc/Kconfig                |   10 +-
>  sound/soc/Makefile               |   12 +-
>  sound/soc/codecs/Kconfig         |    4 +
>  sound/soc/codecs/Makefile        |    2 +
>  sound/soc/codecs/wm8350.c        | 1583 ++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/wm8350.h        |   20 +
>  sound/soc/davinci/davinci-pcm.c  |   18 +-
>  sound/soc/soc-dapm.c             |    6 +-
>  9 files changed, 1669 insertions(+), 24 deletions(-)
>  create mode 100644 sound/soc/codecs/wm8350.c
>  create mode 100644 sound/soc/codecs/wm8350.h
>