ASoC updates for 2.6.31

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 34e51ce60a210094bd66cf0a75dd8512247618ca:
  Takashi Iwai (1):
        Merge branch 'for-2.6.30' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Alexander Beregalov (1):
      ASoC: n810: replace BUG() with BUG_ON()

Daniel Glöckner (3):
      ASoC: Add driver for s6000 I2S interface
      ASoC: s6105 IP camera machine specific ASoC code
      ASoC: correct s6000 I2S clock polarity

Daniel Ribeiro (1):
      ASoC: pxa-ssp.c fix clock/frame invert

Mark Brown (11):
      ASoC: Display return code when failing to add a DAPM kcontrol
      ASoC: Provide core support for symmetric sample rates
      ASoC: Add WM8988 CODEC driver
      Merge branch 's6000' into for-2.6.31
      ASoC: WM9713 requires symmetric rates on the voice DAI
      ASoC: Factor out application of power for generic widgets
      ASoC: Support DAPM events for DACs and ADCs
      ASoC: Move the WM9713 voice DAC powerdown to a DAPM event
      ASoC: Add WM8960 CODEC driver
      Merge branch 'for-2.6.30' into for-2.6.31
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (1):
      ASoC: tlv320aic23: add DSP_A format support

 include/sound/soc-dai.h        |    1 +
 include/sound/soc-dapm.h       |   10 +
 include/sound/soc.h            |    6 +
 sound/soc/Kconfig              |    1 +
 sound/soc/Makefile             |    1 +
 sound/soc/codecs/Kconfig       |    8 +
 sound/soc/codecs/Makefile      |    4 +
 sound/soc/codecs/tlv320aic23.c |    2 +
 sound/soc/codecs/wm8960.c      |  969 +++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8960.h      |  127 +++++
 sound/soc/codecs/wm8988.c      | 1097 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8988.h      |   60 +++
 sound/soc/codecs/wm9713.c      |   40 +-
 sound/soc/omap/n810.c          |    7 +-
 sound/soc/pxa/pxa-ssp.c        |   11 +-
 sound/soc/s6000/Kconfig        |   19 +
 sound/soc/s6000/Makefile       |   11 +
 sound/soc/s6000/s6000-i2s.c    |  629 +++++++++++++++++++++++
 sound/soc/s6000/s6000-i2s.h    |   25 +
 sound/soc/s6000/s6000-pcm.c    |  497 ++++++++++++++++++
 sound/soc/s6000/s6000-pcm.h    |   35 ++
 sound/soc/s6000/s6105-ipcam.c  |  244 +++++++++
 sound/soc/soc-core.c           |   38 ++
 sound/soc/soc-dapm.c           |  131 +++---
 24 files changed, 3893 insertions(+), 80 deletions(-)
 create mode 100644 sound/soc/codecs/wm8960.c
 create mode 100644 sound/soc/codecs/wm8960.h
 create mode 100644 sound/soc/codecs/wm8988.c
 create mode 100644 sound/soc/codecs/wm8988.h
 create mode 100644 sound/soc/s6000/Kconfig
 create mode 100644 sound/soc/s6000/Makefile
 create mode 100644 sound/soc/s6000/s6000-i2s.c
 create mode 100644 sound/soc/s6000/s6000-i2s.h
 create mode 100644 sound/soc/s6000/s6000-pcm.c
 create mode 100644 sound/soc/s6000/s6000-pcm.h
 create mode 100644 sound/soc/s6000/s6105-ipcam.c

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 6967963d6d5cac40a091d075326f0e3ccb95c58a:
  Mark Brown (1):
        Merge branch 'for-2.6.30' into for-2.6.31

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Ben Dooks (4):
      ASoC: Fix jive_wm8750.c build problems
      ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate
      ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes
      ASoC: Fix include build error in s3c2412-i2s.c

Daniel Mack (1):
      ASoC: pxa-ssp: allow setting of dai format 0

Jarkko Nikula (2):
      ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver
      ASoC: OMAP: Fix FS polarity in OSK5912 machine driver

Mark Brown (4):
      ASoC: Check we have DAI ops when calling via accessor functions
      ASoC: Volume controls are never of boolean type
      ASoC: Request shared rates for WM8903
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (2):
      ASoC: OMAP: Use single-phase for DSP mode
      ASoC: OMAP: Add DSP_A mode support for mcbsp

 sound/soc/codecs/wm8903.c       |    1 +
 sound/soc/omap/omap-mcbsp.c     |   45 ++++++++++++++++++++++++++------------
 sound/soc/omap/osk5912.c        |    4 +-
 sound/soc/pxa/pxa-ssp.c         |    1 +
 sound/soc/s3c24xx/jive_wm8750.c |   12 +++++-----
 sound/soc/s3c24xx/s3c-i2s-v2.c  |   18 ++++++++-------
 sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
 sound/soc/soc-core.c            |   20 ++++++++--------
 8 files changed, 62 insertions(+), 41 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Thu, 16 Apr 2009 14:50:27 +0100,
Mark Brown wrote:
> 
> The following changes since commit 6967963d6d5cac40a091d075326f0e3ccb95c58a:
>   Mark Brown (1):
>         Merge branch 'for-2.6.30' into for-2.6.31
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled to topic/asoc branch now.  Thanks.


Takashi

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 8d98f2246d7c622198ae0b8ca66f1c82b8a25377:
  Mark Brown (1):
        Merge branch 'for-2.6.30' into for-2.6.31

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Jarkko Nikula (1):
      ASoC: OMAP: Update contact addresses

Joonyoung Shim (1):
      ASoC: TWL4030: Add support Voice DAI

Mark Brown (1):
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (1):
      ASoC: TWL4030: Fix for the constraint handling

Philipp Zabel (1):
      ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR

 sound/soc/codecs/twl4030.c    |  258 +++++++++++++++++++++++++++++++++++++----
 sound/soc/codecs/twl4030.h    |   18 +++-
 sound/soc/omap/n810.c         |    4 +-
 sound/soc/omap/omap-mcbsp.c   |    5 +-
 sound/soc/omap/omap-mcbsp.h   |    3 +-
 sound/soc/omap/omap-pcm.c     |    5 +-
 sound/soc/omap/omap-pcm.h     |    3 +-
 sound/soc/omap/omap2evm.c     |    2 +-
 sound/soc/omap/omap3beagle.c  |    2 +-
 sound/soc/omap/omap3pandora.c |    4 +-
 sound/soc/omap/overo.c        |    2 +-
 sound/soc/omap/sdp3430.c      |    2 +-
 sound/soc/pxa/pxa-ssp.c       |   36 +++++--
 13 files changed, 297 insertions(+), 47 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Mon, 20 Apr 2009 12:58:08 +0100,
Mark Brown wrote:
> 
> The following changes since commit 8d98f2246d7c622198ae0b8ca66f1c82b8a25377:
>   Mark Brown (1):
>         Merge branch 'for-2.6.30' into for-2.6.31
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks.

Takashi

> Jarkko Nikula (1):
>       ASoC: OMAP: Update contact addresses
> 
> Joonyoung Shim (1):
>       ASoC: TWL4030: Add support Voice DAI
> 
> Mark Brown (1):
>       Merge branch 'for-2.6.30' into for-2.6.31
> 
> Peter Ujfalusi (1):
>       ASoC: TWL4030: Fix for the constraint handling
> 
> Philipp Zabel (1):
>       ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR
> 
>  sound/soc/codecs/twl4030.c    |  258 +++++++++++++++++++++++++++++++++++++----
>  sound/soc/codecs/twl4030.h    |   18 +++-
>  sound/soc/omap/n810.c         |    4 +-
>  sound/soc/omap/omap-mcbsp.c   |    5 +-
>  sound/soc/omap/omap-mcbsp.h   |    3 +-
>  sound/soc/omap/omap-pcm.c     |    5 +-
>  sound/soc/omap/omap-pcm.h     |    3 +-
>  sound/soc/omap/omap2evm.c     |    2 +-
>  sound/soc/omap/omap3beagle.c  |    2 +-
>  sound/soc/omap/omap3pandora.c |    4 +-
>  sound/soc/omap/overo.c        |    2 +-
>  sound/soc/omap/sdp3430.c      |    2 +-
>  sound/soc/pxa/pxa-ssp.c       |   36 +++++--
>  13 files changed, 297 insertions(+), 47 deletions(-)
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 9b5b0c01598f9782690b09ce6c49f4ba116dde44:
  Mark Brown (1):
        Merge branch 'for-2.6.30' into for-2.6.31

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Eric Miao (2):
      ASoC: simplify the SSP DMA parameters settings by run-time generation
      ASoC: change stereo/mono to 32-bit/16-bit for pxa-ssp

Joonyoung Shim (2):
      ASoC: OMAP: Add checking to detect bufferless pcms
      ASoC: TWL4030: Add VDL path support

Mark Brown (16):
      ASoC: Factor out generic widget power checks
      ASoC: Factor out DAPM power checks for DACs and ADCs
      ASoC: Make the DAPM power check an operation on the widget
      ASoC: Fix offset of freqmode in WM8580 PLL configuration
      ASoC: Fix WM8580 volume update handling for large register changes
      ASoC: Add power supply widget to DAPM
      ASoC: Use DAPM supply widget for WM8903 charge pump
      ASoC: Support CLK_DSP in WM8903
      ASoC: Optimise configuration of WM8903 DC servo
      ASoC: Actively manage the DC servo for WM8903
      ASoC: Remove redundant rate constraint for WM8903
      ASoC: Implement WM8903 digital sidetone support
      Merge commit 'takashi/fix/asoc' into for-2.6.30
      Merge branch 'for-2.6.30' into for-2.6.31
      ASoC: s3c-i2s-v2 needs to declare a license for modular builds
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (2):
      ASoC: OMAP: Add 4 channel support to mcbsp
      ASoC: TWL4030: Add 4 channel TDM support

Russell King - ARM Linux (1):
      ASoC: Fix warning in wm9705

Takashi Iwai (1):
      ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991

 Documentation/sound/alsa/soc/dapm.txt |    1 +
 include/sound/soc-dapm.h              |    9 +-
 sound/soc/codecs/Makefile             |    1 -
 sound/soc/codecs/twl4030.c            |  308 +++++++++++++++++++--------------
 sound/soc/codecs/twl4030.h            |   11 ++
 sound/soc/codecs/wm8580.c             |   16 +-
 sound/soc/codecs/wm8903.c             |  118 ++++++++-----
 sound/soc/codecs/wm9705.c             |    2 +-
 sound/soc/omap/omap-mcbsp.c           |    5 +-
 sound/soc/omap/omap-pcm.c             |    9 +-
 sound/soc/pxa/pxa-ssp.c               |  203 +++++-----------------
 sound/soc/s3c24xx/s3c-i2s-v2.c        |    3 +-
 sound/soc/soc-dapm.c                  |  139 +++++++++++-----
 13 files changed, 438 insertions(+), 387 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Thu, 23 Apr 2009 20:54:20 +0100,
Mark Brown wrote:
> 
> The following changes since commit 9b5b0c01598f9782690b09ce6c49f4ba116dde44:
>   Mark Brown (1):
>         Merge branch 'for-2.6.30' into for-2.6.31
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now to topic/asoc branch.  Thanks.

Takashi

> 
> Eric Miao (2):
>       ASoC: simplify the SSP DMA parameters settings by run-time generation
>       ASoC: change stereo/mono to 32-bit/16-bit for pxa-ssp
> 
> Joonyoung Shim (2):
>       ASoC: OMAP: Add checking to detect bufferless pcms
>       ASoC: TWL4030: Add VDL path support
> 
> Mark Brown (16):
>       ASoC: Factor out generic widget power checks
>       ASoC: Factor out DAPM power checks for DACs and ADCs
>       ASoC: Make the DAPM power check an operation on the widget
>       ASoC: Fix offset of freqmode in WM8580 PLL configuration
>       ASoC: Fix WM8580 volume update handling for large register changes
>       ASoC: Add power supply widget to DAPM
>       ASoC: Use DAPM supply widget for WM8903 charge pump
>       ASoC: Support CLK_DSP in WM8903
>       ASoC: Optimise configuration of WM8903 DC servo
>       ASoC: Actively manage the DC servo for WM8903
>       ASoC: Remove redundant rate constraint for WM8903
>       ASoC: Implement WM8903 digital sidetone support
>       Merge commit 'takashi/fix/asoc' into for-2.6.30
>       Merge branch 'for-2.6.30' into for-2.6.31
>       ASoC: s3c-i2s-v2 needs to declare a license for modular builds
>       Merge branch 'for-2.6.30' into for-2.6.31
> 
> Peter Ujfalusi (2):
>       ASoC: OMAP: Add 4 channel support to mcbsp
>       ASoC: TWL4030: Add 4 channel TDM support
> 
> Russell King - ARM Linux (1):
>       ASoC: Fix warning in wm9705
> 
> Takashi Iwai (1):
>       ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991
> 
>  Documentation/sound/alsa/soc/dapm.txt |    1 +
>  include/sound/soc-dapm.h              |    9 +-
>  sound/soc/codecs/Makefile             |    1 -
>  sound/soc/codecs/twl4030.c            |  308 +++++++++++++++++++--------------
>  sound/soc/codecs/twl4030.h            |   11 ++
>  sound/soc/codecs/wm8580.c             |   16 +-
>  sound/soc/codecs/wm8903.c             |  118 ++++++++-----
>  sound/soc/codecs/wm9705.c             |    2 +-
>  sound/soc/omap/omap-mcbsp.c           |    5 +-
>  sound/soc/omap/omap-pcm.c             |    9 +-
>  sound/soc/pxa/pxa-ssp.c               |  203 +++++-----------------
>  sound/soc/s3c24xx/s3c-i2s-v2.c        |    3 +-
>  sound/soc/soc-dapm.c                  |  139 +++++++++++-----
>  13 files changed, 438 insertions(+), 387 deletions(-)
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9:
  Peter Ujfalusi (1):
        ASoC: TWL4030: Add 4 channel TDM support

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Daniel Mack (3):
      ASoC: add SOC_DOUBLE_EXT macro
      ASoC: cs4270: fix Master Capture Switch polarity
      ASoC: cs4270: add Master Playback Switch

Jon Smirl (3):
      ASoC: Set the MPC5200 i2s driver to BROKEN status.
      ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s
      ASoC: Rename the PSC functions to DMA

Jonathan Cameron (1):
      ASoC WM8940 Driver

Joonyoung Shim (1):
      ASoC: TWL4030: Fix gain control for earpiece amplifier

Mark Brown (10):
      ASoC: Fix S3C64xx IIS device registration and support both ports
      ASoC: S3C2412: Failing to get the I2S clock is an error
      ASoC: Enforce symmetric rates for S3C64xx I2S interface
      ASoC: Include WM8350 register definitions in CODEC header
      ASoC: s3c-i2s-v2 diagnostic improvements
      ASoC: Use our registration function for S3C64xx
      ASoC: Fix logic in WM8350 master clocking check
      Merge branch 'for-2.6.30' into for-2.6.31
      ASoC: Staticise TLV values in WM8940
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (1):
      ASoC: Beagle: Add support for 4 channel

 include/sound/soc.h             |    8 +
 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/cs4270.c       |   44 ++-
 sound/soc/codecs/twl4030.c      |    8 +-
 sound/soc/codecs/wm8350.c       |    2 +-
 sound/soc/codecs/wm8350.h       |    1 +
 sound/soc/codecs/wm8940.c       |  955 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8940.h       |  104 +++++
 sound/soc/fsl/Kconfig           |    6 +-
 sound/soc/fsl/Makefile          |    2 +
 sound/soc/fsl/mpc5200_dma.c     |  457 +++++++++++++++++++
 sound/soc/fsl/mpc5200_dma.h     |   81 ++++
 sound/soc/fsl/mpc5200_psc_i2s.c |  644 ++++-----------------------
 sound/soc/omap/omap3beagle.c    |   26 +-
 sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
 sound/soc/s3c24xx/s3c64xx-i2s.c |  148 ++++--
 sound/soc/s3c24xx/s3c64xx-i2s.h |    2 +-
 18 files changed, 1867 insertions(+), 629 deletions(-)
 create mode 100644 sound/soc/codecs/wm8940.c
 create mode 100644 sound/soc/codecs/wm8940.h
 create mode 100644 sound/soc/fsl/mpc5200_dma.c
 create mode 100644 sound/soc/fsl/mpc5200_dma.h

[PATCH 01/17] ASoC: Beagle: Add support for 4 channel

[Mark Brown]

From: Peter Ujfalusi <peter.ujfalusi@nokia.com>

This patch adds support for the four channel TDM mode
on Beagle board.

Depending on the channel count, the interface needs to be
configured differently (I2S for stereo DSP_A for four channels)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap3beagle.c |   26 ++++++++++++++++++--------
 1 files changed, 18 insertions(+), 8 deletions(-)

diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index 6aa428e..b0cff9f 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -41,23 +41,33 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int fmt;
 	int ret;
 
+	switch (params_channels(params)) {
+	case 2: /* Stereo I2S mode */
+		fmt =	SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	case 4: /* Four channel TDM mode */
+		fmt =	SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_IB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	/* Set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
 	if (ret < 0) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
 		return ret;
 	}
 
 	/* Set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_I2S |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
 	if (ret < 0) {
 		printk(KERN_ERR "can't set cpu DAI configuration\n");
 		return ret;
-- 
1.6.2.4

[PATCH 02/17] ASoC: add SOC_DOUBLE_EXT macro

[Mark Brown]

From: Daniel Mack <daniel@caiaq.de>

Add a macro for double controls with special callback functions.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h |    8 ++++++++
 1 files changed, 8 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index b1f2f88..6ab80bf 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -118,6 +118,14 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+	 xhandler_get, xhandler_put) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.info = snd_soc_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+		 .max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-- 
1.6.2.4

[PATCH 03/17] ASoC: Fix S3C64xx IIS device registration and support both ports

[Mark Brown]

The S3C64xx IIS code had a number of problems with device registration.
The hardware has two IIS ports of which the driver supported only one
at once via a single exported DAI, attempting to identify the DAI to
use based on the dev->id of the ASoC platform device.  As well as
limiting the driver to only supporting one IIS port at once this also
meant that the ID of the soc-audio device (or in future the card device)
had to match the IIS ID.

Fix both problems by converting the driver to register the DAIs based on
probing of platform devices registered by the arch/arm code, using those
platform devices to interact with the clock API.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c64xx-i2s.c |  146 ++++++++++++++++++++++++++-------------
 sound/soc/s3c24xx/s3c64xx-i2s.h |    2 +-
 2 files changed, 98 insertions(+), 50 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 33c5de7..a84c4be 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -120,36 +120,8 @@ EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
 static int s3c64xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	struct device *dev = &pdev->dev;
-	struct s3c_i2sv2_info *i2s;
-	int ret;
-
-	dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
-
-	if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
-		dev_err(dev, "id %d out of range\n", pdev->id);
-		return -EINVAL;
-	}
-
-	i2s = &s3c64xx_i2s[pdev->id];
-
-	ret = s3c_i2sv2_probe(pdev, dai, i2s,
-			      pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
-	if (ret)
-		return ret;
-
-	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
-	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
-
-	i2s->iis_cclk = clk_get(dev, "audio-bus");
-	if (IS_ERR(i2s->iis_cclk)) {
-		dev_err(dev, "failed to get audio-bus");
-		iounmap(i2s->regs);
-		return -ENODEV;
-	}
-
 	/* configure GPIO for i2s port */
-	switch (pdev->id) {
+	switch (dai->id) {
 	case 0:
 		s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
 		s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
@@ -181,35 +153,114 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
 	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
 };
 
-struct snd_soc_dai s3c64xx_i2s_dai = {
-	.name		= "s3c64xx-i2s",
-	.id		= 0,
-	.probe		= s3c64xx_i2s_probe,
-	.playback = {
-		.channels_min	= 2,
-		.channels_max	= 2,
-		.rates		= S3C64XX_I2S_RATES,
-		.formats	= S3C64XX_I2S_FMTS,
+struct snd_soc_dai s3c64xx_i2s_dai[] = {
+	{
+		.name		= "s3c64xx-i2s",
+		.id		= 0,
+		.probe		= s3c64xx_i2s_probe,
+		.playback = {
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= S3C64XX_I2S_RATES,
+			.formats	= S3C64XX_I2S_FMTS,
+		},
+		.capture = {
+			 .channels_min	= 2,
+			 .channels_max	= 2,
+			 .rates		= S3C64XX_I2S_RATES,
+			 .formats	= S3C64XX_I2S_FMTS,
+		 },
+		.ops = &s3c64xx_i2s_dai_ops,
 	},
-	.capture = {
-		.channels_min	= 2,
-		.channels_max	= 2,
-		.rates		= S3C64XX_I2S_RATES,
-		.formats	= S3C64XX_I2S_FMTS,
+	{
+		.name		= "s3c64xx-i2s",
+		.id		= 1,
+		.probe		= s3c64xx_i2s_probe,
+		.playback = {
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= S3C64XX_I2S_RATES,
+			.formats	= S3C64XX_I2S_FMTS,
+		},
+		.capture = {
+			 .channels_min	= 2,
+			 .channels_max	= 2,
+			 .rates		= S3C64XX_I2S_RATES,
+			 .formats	= S3C64XX_I2S_FMTS,
+		 },
+		.ops = &s3c64xx_i2s_dai_ops,
 	},
-	.ops = &s3c64xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
 
+static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
+{
+	struct s3c_i2sv2_info *i2s;
+	struct snd_soc_dai *dai;
+	int ret;
+
+	if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) {
+		dev_err(&pdev->dev, "id %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	i2s = &s3c64xx_i2s[pdev->id];
+	dai = &s3c64xx_i2s_dai[pdev->id];
+	dai->dev = &pdev->dev;
+
+	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+	i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
+	if (IS_ERR(i2s->iis_cclk)) {
+		dev_err(&pdev->dev, "failed to get audio-bus");
+		ret = PTR_ERR(i2s->iis_cclk);
+		goto err;
+	}
+
+	ret = s3c_i2sv2_probe(pdev, dai, i2s,
+			      dai->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
+	if (ret)
+		goto err_clk;
+
+	ret = snd_soc_register_dai(dai);
+	if (ret != 0)
+		goto err_i2sv2;
+
+	return 0;
+
+err_i2sv2:
+	/* Not implemented for I2Sv2 core yet */
+err_clk:
+	clk_put(i2s->iis_cclk);
+err:
+	return ret;
+}
+
+static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev)
+{
+	dev_err(&pdev->dev, "Device removal not yet supported\n");
+	return 0;
+}
+
+static struct platform_driver s3c64xx_iis_driver = {
+	.probe  = s3c64xx_iis_dev_probe,
+	.remove = s3c64xx_iis_dev_remove,
+	.driver = {
+		.name = "s3c64xx-iis",
+		.owner = THIS_MODULE,
+	},
+};
+
 static int __init s3c64xx_i2s_init(void)
 {
-	return  s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+	return platform_driver_register(&s3c64xx_iis_driver);
 }
 module_init(s3c64xx_i2s_init);
 
 static void __exit s3c64xx_i2s_exit(void)
 {
-	snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+	platform_driver_unregister(&s3c64xx_iis_driver);
 }
 module_exit(s3c64xx_i2s_exit);
 
@@ -217,6 +268,3 @@ module_exit(s3c64xx_i2s_exit);
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
 MODULE_LICENSE("GPL");
-
-
-
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index b7ffe3c..597822a 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -24,7 +24,7 @@
 #define S3C64XX_CLKSRC_PCLK	(0)
 #define S3C64XX_CLKSRC_MUX	(1)
 
-extern struct snd_soc_dai s3c64xx_i2s_dai;
+extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
 extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
 
-- 
1.6.2.4

[PATCH 04/17] ASoC: S3C2412: Failing to get the I2S clock is an error

[Mark Brown]

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index b7e0b3f..168a088 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -120,7 +120,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 
 	s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
 	if (s3c2412_i2s.iis_cclk == NULL) {
-		pr_debug("failed to get i2sclk clock\n");
+		pr_err("failed to get i2sclk clock\n");
 		iounmap(s3c2412_i2s.regs);
 		return -ENODEV;
 	}
-- 
1.6.2.4

[PATCH 05/17] ASoC: Enforce symmetric rates for S3C64xx I2S interface

[Mark Brown]

There is only one LRCLK pin on each interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c64xx-i2s.c |    2 ++
 1 files changed, 2 insertions(+), 0 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index a84c4be..c335248 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -171,6 +171,7 @@ struct snd_soc_dai s3c64xx_i2s_dai[] = {
 			 .formats	= S3C64XX_I2S_FMTS,
 		 },
 		.ops = &s3c64xx_i2s_dai_ops,
+		.symmetric_rates = 1,
 	},
 	{
 		.name		= "s3c64xx-i2s",
@@ -189,6 +190,7 @@ struct snd_soc_dai s3c64xx_i2s_dai[] = {
 			 .formats	= S3C64XX_I2S_FMTS,
 		 },
 		.ops = &s3c64xx_i2s_dai_ops,
+		.symmetric_rates = 1,
 	},
 };
 EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
-- 
1.6.2.4

[PATCH 06/17] ASoC: Include WM8350 register definitions in CODEC header

[Mark Brown]

It's expected behaviour for the CODEC header to provide them but the
WM8350 doesn't due to having all the registers together under drivers/mfd.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8350.h |    1 +
 1 files changed, 1 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index d11bd92..d088eb4 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -13,6 +13,7 @@
 #define _WM8350_H
 
 #include <sound/soc.h>
+#include <linux/mfd/wm8350/audio.h>
 
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
-- 
1.6.2.4

[PATCH 07/17] ASoC: s3c-i2s-v2 diagnostic improvements

[Mark Brown]

Say what invalid values we're seeing when we see an invalid value and
ensure that errors are displayed by default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c |   18 ++++++++++++------
 1 files changed, 12 insertions(+), 6 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index ab680aa..aeea49c 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -105,7 +105,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
+			break;
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -132,7 +134,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "TXDIS: Invalid MODE %xin IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
+			break;
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -175,7 +179,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -199,7 +204,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+			dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+				mod & S3C2412_IISMOD_MODE_MASK);
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -281,7 +287,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= IISMOD_MASTER;
 		break;
 	default:
-		pr_debug("unknwon master/slave format\n");
+		pr_err("unknwon master/slave format\n");
 		return -EINVAL;
 	}
 
@@ -298,7 +304,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= S3C2412_IISMOD_SDF_IIS;
 		break;
 	default:
-		pr_debug("Unknown data format\n");
+		pr_err("Unknown data format\n");
 		return -EINVAL;
 	}
 
-- 
1.6.2.4

[PATCH 08/17] ASoC: Use our registration function for S3C64xx

[Mark Brown]

Make sure we get the DAI operations initialised.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c  |   18 ++++++------------
 sound/soc/s3c24xx/s3c64xx-i2s.c |    2 +-
 2 files changed, 7 insertions(+), 13 deletions(-)

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index aeea49c..ab680aa 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -105,9 +105,7 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
-				mod & S3C2412_IISMOD_MODE_MASK);
-			break;
+			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -134,9 +132,7 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "TXDIS: Invalid MODE %xin IISMOD\n",
-				mod & S3C2412_IISMOD_MODE_MASK);
-			break;
+			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -179,8 +175,7 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
-				mod & S3C2412_IISMOD_MODE_MASK);
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
 		}
 
 		writel(mod, regs + S3C2412_IISMOD);
@@ -204,8 +199,7 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 			break;
 
 		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
-				mod & S3C2412_IISMOD_MODE_MASK);
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
 		}
 
 		writel(con, regs + S3C2412_IISCON);
@@ -287,7 +281,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= IISMOD_MASTER;
 		break;
 	default:
-		pr_err("unknwon master/slave format\n");
+		pr_debug("unknwon master/slave format\n");
 		return -EINVAL;
 	}
 
@@ -304,7 +298,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		iismod |= S3C2412_IISMOD_SDF_IIS;
 		break;
 	default:
-		pr_err("Unknown data format\n");
+		pr_debug("Unknown data format\n");
 		return -EINVAL;
 	}
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index c335248..1345fbd 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -225,7 +225,7 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk;
 
-	ret = snd_soc_register_dai(dai);
+	ret = s3c_i2sv2_register_dai(dai);
 	if (ret != 0)
 		goto err_i2sv2;
 
-- 
1.6.2.4

[PATCH 09/17] ASoC WM8940 Driver

[Mark Brown]

From: Jonathan Cameron <jic23@cam.ac.uk>

Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |    4 +
 sound/soc/codecs/Makefile |    2 +
 sound/soc/codecs/wm8940.c |  955 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8940.h |  104 +++++
 4 files changed, 1065 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/wm8940.c
 create mode 100644 sound/soc/codecs/wm8940.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 121d63f..1c19ad5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8940 if I2C
 	select SND_SOC_WM8960 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
@@ -140,6 +141,9 @@ config SND_SOC_WM8900
 config SND_SOC_WM8903
 	tristate
 
+config SND_SOC_WM8940
+        tristate
+
 config SND_SOC_WM8960
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 8116968..3d31b6b 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,7 @@ snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
 snd-soc-wm8960-objs := wm8960.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8988-objs := wm8988.o
@@ -57,6 +58,7 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940)	+= snd-soc-wm8940.o
 obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
 obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
new file mode 100644
index 0000000..26987dc
--- /dev/null
+++ b/sound/soc/codecs/wm8940.c
@@ -0,0 +1,955 @@
+/*
+ * wm8940.c  --  WM8940 ALSA Soc Audio driver
+ *
+ * Author: Jonathan Cameron <jic23@cam.ac.uk>
+ *
+ * Based on wm8510.c
+ *    Copyright  2006 Wolfson Microelectronics PLC.
+ *    Author:  Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Not currently handled:
+ * Notch filter control
+ * AUXMode (inverting vs mixer)
+ * No means to obtain current gain if alc enabled.
+ * No use made of gpio
+ * Fast VMID discharge for power down
+ * Soft Start
+ * DLR and ALR Swaps not enabled
+ * Digital Sidetone not supported
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8940.h"
+
+struct wm8940_priv {
+	unsigned int sysclk;
+	u16 reg_cache[WM8940_CACHEREGNUM];
+	struct snd_soc_codec codec;
+};
+
+static u16 wm8940_reg_defaults[] = {
+	0x8940, /* Soft Reset */
+	0x0000, /* Power 1 */
+	0x0000, /* Power 2 */
+	0x0000, /* Power 3 */
+	0x0010, /* Interface Control */
+	0x0000, /* Companding Control */
+	0x0140, /* Clock Control */
+	0x0000, /* Additional Controls */
+	0x0000, /* GPIO Control */
+	0x0002, /* Auto Increment Control */
+	0x0000, /* DAC Control */
+	0x00FF, /* DAC Volume */
+	0,
+	0,
+	0x0100, /* ADC Control */
+	0x00FF, /* ADC Volume */
+	0x0000, /* Notch Filter 1 Control 1 */
+	0x0000, /* Notch Filter 1 Control 2 */
+	0x0000, /* Notch Filter 2 Control 1 */
+	0x0000, /* Notch Filter 2 Control 2 */
+	0x0000, /* Notch Filter 3 Control 1 */
+	0x0000, /* Notch Filter 3 Control 2 */
+	0x0000, /* Notch Filter 4 Control 1 */
+	0x0000, /* Notch Filter 4 Control 2 */
+	0x0032, /* DAC Limit Control 1 */
+	0x0000, /* DAC Limit Control 2 */
+	0,
+	0,
+	0,
+	0,
+	0,
+	0,
+	0x0038, /* ALC Control 1 */
+	0x000B, /* ALC Control 2 */
+	0x0032, /* ALC Control 3 */
+	0x0000, /* Noise Gate */
+	0x0041, /* PLLN */
+	0x000C, /* PLLK1 */
+	0x0093, /* PLLK2 */
+	0x00E9, /* PLLK3 */
+	0,
+	0,
+	0x0030, /* ALC Control 4 */
+	0,
+	0x0002, /* Input Control */
+	0x0050, /* PGA Gain */
+	0,
+	0x0002, /* ADC Boost Control */
+	0,
+	0x0002, /* Output Control */
+	0x0000, /* Speaker Mixer Control */
+	0,
+	0,
+	0,
+	0x0079, /* Speaker Volume */
+	0,
+	0x0000, /* Mono Mixer Control */
+};
+
+static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
+						 unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+		return -1;
+
+	return cache[reg];
+}
+
+static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
+					  u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+		return -1;
+
+	cache[reg] = value;
+
+	return 0;
+}
+
+static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
+			unsigned int value)
+{
+	int ret;
+	u8 data[3] = { reg,
+		       (value & 0xff00) >> 8,
+		       (value & 0x00ff)
+	};
+
+	wm8940_write_reg_cache(codec, reg, value);
+
+	ret = codec->hw_write(codec->control_data, data, 3);
+
+	if (ret < 0)
+		return ret;
+	else if (ret != 3)
+		return -EIO;
+	return 0;
+}
+
+static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const struct soc_enum wm8940_adc_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
+static const struct soc_enum wm8940_dac_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+
+static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
+static const struct soc_enum wm8940_alc_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+
+static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
+static const struct soc_enum wm8940_mic_bias_level_enum
+= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+
+static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
+static const struct soc_enum wm8940_filter_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+
+DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+
+static const struct snd_kcontrol_new wm8940_snd_controls[] = {
+	SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
+		   6, 1, 0),
+	SOC_ENUM("DAC Companding", wm8940_dac_companding_enum),
+	SOC_ENUM("ADC Companding", wm8940_adc_companding_enum),
+
+	SOC_ENUM("ALC Mode", wm8940_alc_mode_enum),
+	SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0),
+	SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1,
+		       3, 7, 1, wm8940_alc_max_tlv),
+	SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1,
+		       0, 7, 0, wm8940_alc_min_tlv),
+	SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2,
+		       0, 14, 0, wm8940_alc_tar_tlv),
+	SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0),
+	SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0),
+	SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE,
+		   3, 1, 0),
+	SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE,
+		   0, 7, 0),
+
+	SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0),
+	SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0),
+	SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0),
+	SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2,
+		       4, 9, 1, wm8940_lim_thresh_tlv),
+	SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2,
+		       0, 12, 0, wm8940_lim_boost_tlv),
+
+	SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0),
+	SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN,
+		       0, 63, 0, wm8940_pga_vol_tlv),
+	SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL,
+		       0, 255, 0, wm8940_adc_tlv),
+	SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
+		       0, 255, 0, wm8940_adc_tlv),
+	SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
+	SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+		       8, 1, 0, wm8940_capture_boost_vol_tlv),
+	SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
+		       0, 63, 0, wm8940_spk_vol_tlv),
+	SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL,  6, 1, 1),
+
+	SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL,
+		       8, 1, 1, wm8940_att_tlv),
+	SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0),
+
+	SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1),
+	SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX,
+		       7, 1, 1, wm8940_att_tlv),
+
+	SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0),
+	SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum),
+	SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0),
+	SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0),
+	SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0),
+	SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0),
+	SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
+};
+
+DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
+	SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
+	SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
+			    0, 7, 0, wm8940_boost_vol_tlv),
+	SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST,
+			    4, 7, 0, wm8940_boost_vol_tlv),
+};
+
+static const struct snd_kcontrol_new wm8940_micpga_controls[] = {
+	SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0),
+	SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0),
+	SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
+	SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0,
+			   &wm8940_speaker_mixer_controls[0],
+			   ARRAY_SIZE(wm8940_speaker_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0,
+			   &wm8940_mono_mixer_controls[0],
+			   ARRAY_SIZE(wm8940_mono_mixer_controls)),
+	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0),
+
+	SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("MONOOUT"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+	SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
+	SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0),
+	SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0,
+			   &wm8940_micpga_controls[0],
+			   ARRAY_SIZE(wm8940_micpga_controls)),
+	SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0,
+			   &wm8940_input_boost_controls[0],
+			   ARRAY_SIZE(wm8940_input_boost_controls)),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0),
+
+	SND_SOC_DAPM_INPUT("MICN"),
+	SND_SOC_DAPM_INPUT("MICP"),
+	SND_SOC_DAPM_INPUT("AUX"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Mono output mixer */
+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Speaker output mixer */
+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Outputs */
+	{"Mono Out", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono Out"},
+	{"SpkN Out", NULL, "Speaker Mixer"},
+	{"SpkP Out", NULL, "Speaker Mixer"},
+	{"SPKOUTN", NULL, "SpkN Out"},
+	{"SPKOUTP", NULL, "SpkP Out"},
+
+	/*  Microphone PGA */
+	{"Mic PGA", "MICN Switch", "MICN"},
+	{"Mic PGA", "MICP Switch", "MICP"},
+	{"Mic PGA", "AUX Switch", "AUX"},
+
+	/* Boost Mixer */
+	{"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+	{"Boost Mixer", "Mic Volume",  "MICP"},
+	{"Boost Mixer", "Aux Volume", "Aux Input"},
+
+	{"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8940_add_widgets(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+					ARRAY_SIZE(wm8940_dapm_widgets));
+	if (ret)
+		goto error_ret;
+	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	if (ret)
+		goto error_ret;
+	ret = snd_soc_dapm_new_widgets(codec);
+
+error_ret:
+	return ret;
+}
+
+#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+
+static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
+	u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		clk |= 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+	wm8940_write(codec, WM8940_CLOCK, clk);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= (2 << 3);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= (1 << 3);
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= (3 << 3);
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= (3 << 3) | (1 << 7);
+		break;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= (1 << 7);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= (1 << 8);
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= (1 << 8) | (1 << 7);
+		break;
+	}
+
+	wm8940_write(codec, WM8940_IFACE, iface);
+
+	return 0;
+}
+
+static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
+	u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
+	u16 companding =  wm8940_read_reg_cache(codec,
+						WM8940_COMPANDINGCTL) & 0xFFDF;
+	int ret;
+
+	/* LoutR control */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+	    && params_channels(params) == 2)
+		iface |= (1 << 9);
+
+	switch (params_rate(params)) {
+	case SNDRV_PCM_RATE_8000:
+		addcntrl |= (0x5 << 1);
+		break;
+	case SNDRV_PCM_RATE_11025:
+		addcntrl |= (0x4 << 1);
+		break;
+	case SNDRV_PCM_RATE_16000:
+		addcntrl |= (0x3 << 1);
+		break;
+	case SNDRV_PCM_RATE_22050:
+		addcntrl |= (0x2 << 1);
+		break;
+	case SNDRV_PCM_RATE_32000:
+		addcntrl |= (0x1 << 1);
+		break;
+	case SNDRV_PCM_RATE_44100:
+	case SNDRV_PCM_RATE_48000:
+		break;
+	}
+	ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+	if (ret)
+		goto error_ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		companding = companding | (1 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= (1 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= (2 << 5);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= (3 << 5);
+		break;
+	}
+	ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+	if (ret)
+		goto error_ret;
+	ret = wm8940_write(codec, WM8940_IFACE, iface);
+
+error_ret:
+	return ret;
+}
+
+static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+
+	if (mute)
+		mute_reg |= 0x40;
+
+	return wm8940_write(codec, WM8940_DAC, mute_reg);
+}
+
+static int wm8940_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 val;
+	u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		/* Enable thermal shutdown */
+		val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+		ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+		if (ret)
+			break;
+		/* set vmid to 75k */
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* ensure bufioen and biasen */
+		pwr_reg |= (1 << 2) | (1 << 3);
+		/* set vmid to 300k for standby */
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+		break;
+	case SND_SOC_BIAS_OFF:
+		ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+		break;
+	}
+
+	return ret;
+}
+
+struct pll_ {
+	unsigned int pre_scale:2;
+	unsigned int n:4;
+	unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+static void pll_factors(unsigned int target, unsigned int source)
+{
+	unsigned long long Kpart;
+	unsigned int K, Ndiv, Nmod;
+	/* The left shift ist to avoid accuracy loss when right shifting */
+	Ndiv = target / source;
+
+	if (Ndiv > 12) {
+		source <<= 1;
+		/* Multiply by 2 */
+		pll_div.pre_scale = 0;
+		Ndiv = target / source;
+	} else if (Ndiv < 3) {
+		source >>= 2;
+		/* Divide by 4 */
+		pll_div.pre_scale = 3;
+		Ndiv = target / source;
+	} else if (Ndiv < 6) {
+		source >>= 1;
+		/* divide by 2 */
+		pll_div.pre_scale = 2;
+		Ndiv = target / source;
+	} else
+		pll_div.pre_scale = 1;
+
+	if ((Ndiv < 6) || (Ndiv > 12))
+		printk(KERN_WARNING
+			"WM8940 N value %d outwith recommended range!d\n",
+			Ndiv);
+
+	pll_div.n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	K = Kpart & 0xFFFFFFFF;
+
+	/* Check if we need to round */
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	K /= 10;
+
+	pll_div.k = K;
+}
+
+/* Untested at the moment */
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	/* Turn off PLL */
+	reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+	wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+
+	if (freq_in == 0 || freq_out == 0) {
+		/* Clock CODEC directly from MCLK */
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+		wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+		/* Pll power down */
+		wm8940_write(codec, WM8940_PLLN, (1 << 7));
+		return 0;
+	}
+
+	/* Pll is followed by a frequency divide by 4 */
+	pll_factors(freq_out*4, freq_in);
+	if (pll_div.k)
+		wm8940_write(codec, WM8940_PLLN,
+			     (pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
+	else /* No factional component */
+		wm8940_write(codec, WM8940_PLLN,
+			     (pll_div.pre_scale << 4) | pll_div.n);
+	wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+	wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+	wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+	/* Enable the PLL */
+	reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+	wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+
+	/* Run CODEC from PLL instead of MCLK */
+	reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+	wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+
+	return 0;
+}
+
+static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8940_priv *wm8940 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 16934400:
+	case 18432000:
+		wm8940->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+				 int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+	int ret = 0;
+
+	switch (div_id) {
+	case WM8940_BCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
+		ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+		break;
+	case WM8940_MCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
+		ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+		break;
+	case WM8940_OPCLKDIV:
+		reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
+		ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+		break;
+	}
+	return ret;
+}
+
+#define WM8940_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 |				\
+			SNDRV_PCM_FMTBIT_S16_LE |			\
+			SNDRV_PCM_FMTBIT_S20_3LE |			\
+			SNDRV_PCM_FMTBIT_S24_LE |			\
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8940_dai_ops = {
+	.hw_params = wm8940_i2s_hw_params,
+	.set_sysclk = wm8940_set_dai_sysclk,
+	.digital_mute = wm8940_mute,
+	.set_fmt = wm8940_set_dai_fmt,
+	.set_clkdiv = wm8940_set_dai_clkdiv,
+	.set_pll = wm8940_set_dai_pll,
+};
+
+struct snd_soc_dai wm8940_dai = {
+	.name = "WM8940",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8940_RATES,
+		.formats = WM8940_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8940_RATES,
+		.formats = WM8940_FORMATS,
+	},
+	.ops = &wm8940_dai_ops,
+	.symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8940_dai);
+
+static int wm8940_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8940_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int i;
+	int ret;
+	u8 data[3];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware
+	 * Could use auto incremented writes to speed this up
+	 */
+	for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
+		data[0] = i;
+		data[1] = (cache[i] & 0xFF00) >> 8;
+		data[2] = cache[i] & 0x00FF;
+		ret = codec->hw_write(codec->control_data, data, 3);
+		if (ret < 0)
+			goto error_ret;
+		else if (ret != 3) {
+			ret = -EIO;
+			goto error_ret;
+		}
+	}
+	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (ret)
+		goto error_ret;
+	ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
+
+error_ret:
+	return ret;
+}
+
+static struct snd_soc_codec *wm8940_codec;
+
+static int wm8940_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+
+	int ret = 0;
+
+	if (wm8940_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8940_codec;
+	codec = wm8940_codec;
+
+	mutex_init(&codec->mutex);
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	ret = snd_soc_add_controls(codec, wm8940_snd_controls,
+			     ARRAY_SIZE(wm8940_snd_controls));
+	if (ret)
+		goto error_free_pcms;
+	ret = wm8940_add_widgets(codec);
+	if (ret)
+		goto error_free_pcms;
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
+		goto error_free_pcms;
+	}
+
+	return ret;
+
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+static int wm8940_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8940 = {
+	.probe = wm8940_probe,
+	.remove = wm8940_remove,
+	.suspend = wm8940_suspend,
+	.resume = wm8940_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
+
+static int wm8940_register(struct wm8940_priv *wm8940)
+{
+	struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
+	struct snd_soc_codec *codec = &wm8940->codec;
+	int ret;
+	u16 reg;
+	if (wm8940_codec) {
+		dev_err(codec->dev, "Another WM8940 is registered\n");
+		return -EINVAL;
+	}
+
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8940;
+	codec->name = "WM8940";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8940_read_reg_cache;
+	codec->write = wm8940_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8940_set_bias_level;
+	codec->dai = &wm8940_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
+	codec->reg_cache = &wm8940->reg_cache;
+
+	memcpy(codec->reg_cache, wm8940_reg_defaults,
+	       sizeof(wm8940_reg_defaults));
+
+	ret = wm8940_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
+	wm8940_dai.dev = codec->dev;
+
+	wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+	if (ret < 0)
+		return ret;
+
+	if (!pdata)
+		dev_warn(codec->dev, "No platform data supplied\n");
+	else {
+		reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+		ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+		if (ret < 0)
+			return ret;
+	}
+
+
+	wm8940_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8940_dai);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void wm8940_unregister(struct wm8940_priv *wm8940)
+{
+	wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8940_dai);
+	snd_soc_unregister_codec(&wm8940->codec);
+	kfree(wm8940);
+	wm8940_codec = NULL;
+}
+
+static int wm8940_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct wm8940_priv *wm8940;
+	struct snd_soc_codec *codec;
+
+	wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL);
+	if (wm8940 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8940->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	i2c_set_clientdata(i2c, wm8940);
+	codec->control_data = i2c;
+	codec->dev = &i2c->dev;
+
+	return wm8940_register(wm8940);
+}
+
+static int wm8940_i2c_remove(struct i2c_client *client)
+{
+	struct wm8940_priv *wm8940 = i2c_get_clientdata(client);
+
+	wm8940_unregister(wm8940);
+
+	return 0;
+}
+
+static const struct i2c_device_id wm8940_i2c_id[] = {
+	{ "wm8940", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
+
+static struct i2c_driver wm8940_i2c_driver = {
+	.driver = {
+		.name = "WM8940 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8940_i2c_probe,
+	.remove = __devexit_p(wm8940_i2c_remove),
+	.id_table = wm8940_i2c_id,
+};
+
+static int __init wm8940_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&wm8940_i2c_driver);
+	if (ret)
+		printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n",
+		       ret);
+	return ret;
+}
+module_init(wm8940_modinit);
+
+static void __exit wm8940_exit(void)
+{
+	i2c_del_driver(&wm8940_i2c_driver);
+}
+module_exit(wm8940_exit);
+
+MODULE_DESCRIPTION("ASoC WM8940 driver");
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h
new file mode 100644
index 0000000..8410eed
--- /dev/null
+++ b/sound/soc/codecs/wm8940.h
@@ -0,0 +1,104 @@
+/*
+ * wm8940.h -- WM8940 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8940_H
+#define _WM8940_H
+
+struct wm8940_setup_data {
+	/* Vref to analogue output resistance */
+#define WM8940_VROI_1K 0
+#define WM8940_VROI_30K 1
+	unsigned int vroi:1;
+};
+extern struct snd_soc_dai wm8940_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8940;
+
+/* WM8940 register space */
+#define WM8940_SOFTRESET	0x00
+#define WM8940_POWER1		0x01
+#define WM8940_POWER2		0x02
+#define WM8940_POWER3		0x03
+#define WM8940_IFACE		0x04
+#define WM8940_COMPANDINGCTL	0x05
+#define WM8940_CLOCK		0x06
+#define WM8940_ADDCNTRL		0x07
+#define WM8940_GPIO		0x08
+#define WM8940_CTLINT		0x09
+#define WM8940_DAC		0x0A
+#define WM8940_DACVOL		0x0B
+
+#define WM8940_ADC		0x0E
+#define WM8940_ADCVOL		0x0F
+#define WM8940_NOTCH1		0x10
+#define WM8940_NOTCH2		0x11
+#define WM8940_NOTCH3		0x12
+#define WM8940_NOTCH4		0x13
+#define WM8940_NOTCH5		0x14
+#define WM8940_NOTCH6		0x15
+#define WM8940_NOTCH7		0x16
+#define WM8940_NOTCH8		0x17
+#define WM8940_DACLIM1		0x18
+#define WM8940_DACLIM2		0x19
+
+#define WM8940_ALC1		0x20
+#define WM8940_ALC2		0x21
+#define WM8940_ALC3		0x22
+#define WM8940_NOISEGATE	0x23
+#define WM8940_PLLN		0x24
+#define WM8940_PLLK1		0x25
+#define WM8940_PLLK2		0x26
+#define WM8940_PLLK3		0x27
+
+#define WM8940_ALC4		0x2A
+
+#define WM8940_INPUTCTL		0x2C
+#define WM8940_PGAGAIN		0x2D
+
+#define WM8940_ADCBOOST		0x2F
+
+#define WM8940_OUTPUTCTL	0x31
+#define WM8940_SPKMIX		0x32
+
+#define WM8940_SPKVOL		0x36
+
+#define WM8940_MONOMIX		0x38
+
+#define WM8940_CACHEREGNUM  0x57
+
+
+/* Clock divider Id's */
+#define WM8940_BCLKDIV 0
+#define WM8940_MCLKDIV 1
+#define WM8940_OPCLKDIV 2
+
+/* MCLK clock dividers */
+#define WM8940_MCLKDIV_1	0
+#define WM8940_MCLKDIV_1_5	1
+#define WM8940_MCLKDIV_2	2
+#define WM8940_MCLKDIV_3	3
+#define WM8940_MCLKDIV_4	4
+#define WM8940_MCLKDIV_6	5
+#define WM8940_MCLKDIV_8	6
+#define WM8940_MCLKDIV_12	7
+
+/* BCLK clock dividers */
+#define WM8940_BCLKDIV_1 0
+#define WM8940_BCLKDIV_2 1
+#define WM8940_BCLKDIV_4 2
+#define WM8940_BCLKDIV_8 3
+#define WM8940_BCLKDIV_16 4
+#define WM8940_BCLKDIV_32 5
+
+/* PLL Out Dividers */
+#define WM8940_OPCLKDIV_1 0
+#define WM8940_OPCLKDIV_2 1
+#define WM8940_OPCLKDIV_3 2
+#define WM8940_OPCLKDIV_4 3
+
+#endif /* _WM8940_H */
+
-- 
1.6.2.4

[PATCH 10/17] ASoC: cs4270: fix Master Capture Switch polarity

[Mark Brown]

From: Daniel Mack <daniel@caiaq.de>

The control modifies the MUTE register, hence the polarity must be
inverted.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-By: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7fa09a3..3c34fe6 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -486,7 +486,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
 	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
 	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
-	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1)
 };
 
 /*
-- 
1.6.2.4

[PATCH 11/17] ASoC: cs4270: add Master Playback Switch

[Mark Brown]

From: Daniel Mack <daniel@caiaq.de>

This adds a new control named 'Master Playback Switch' for cs4270
codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch
the put function and store the information about manually set mute
controls from userspace. When a manual mute is set, we don't want the
soc core to un-mute the outputs.

Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c |   44 +++++++++++++++++++++++++++++++++++++++-----
 1 files changed, 39 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3c34fe6..ece6ed6 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -109,6 +109,7 @@ struct cs4270_private {
 	unsigned int mclk; /* Input frequency of the MCLK pin */
 	unsigned int mode; /* The mode (I2S or left-justified) */
 	unsigned int slave_mode;
+	unsigned int manual_mute;
 };
 
 /**
@@ -453,7 +454,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 /**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
  * @dai: the SOC DAI
  * @mute: 0 = disable mute, 1 = enable mute
  *
@@ -462,21 +463,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct cs4270_private *cs4270 = codec->private_data;
 	int reg6;
 
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
 		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
-	else
+	else {
 		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 |= cs4270->manual_mute;
+	}
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ * 			 alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs4270_private *cs4270 = codec->private_data;
+	int left = !ucontrol->value.integer.value[0];
+	int right = !ucontrol->value.integer.value[1];
+
+	cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+			      (right ? CS4270_MUTE_DAC_B : 0);
+
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
@@ -486,7 +518,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
 	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
 	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
-	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1)
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+	SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+		snd_soc_get_volsw, cs4270_soc_put_mute),
 };
 
 /*
@@ -506,7 +540,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = {
 	.hw_params	= cs4270_hw_params,
 	.set_sysclk	= cs4270_set_dai_sysclk,
 	.set_fmt	= cs4270_set_dai_fmt,
-	.digital_mute	= cs4270_mute,
+	.digital_mute	= cs4270_dai_mute,
 };
 
 struct snd_soc_dai cs4270_dai = {
-- 
1.6.2.4

[PATCH 12/17] ASoC: Fix logic in WM8350 master clocking check

[Mark Brown]

We need to check only if the WM8350 is master and only when starting
the stream so if either is not true then we can skip the check.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8350.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d099..0275321 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
 	 * required for LRC in master mode. The DACs or ADCs need a
 	 * valid audio path i.e. pin -> ADC or DAC -> pin before
 	 * the LRC will be enabled in master mode. */
-	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
-- 
1.6.2.4

[PATCH 13/17] ASoC: Set the MPC5200 i2s driver to BROKEN status.

[Mark Brown]

From: Jon Smirl <jonsmirl@gmail.com>

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/Kconfig |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9fc9082..e7dd79a 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -21,7 +21,7 @@ config SND_SOC_MPC8610_HPCD
 
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
-	depends on PPC_MPC52xx && PPC_BESTCOMM
+	depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN
 	select SND_SOC_OF_SIMPLE
 	select PPC_BESTCOMM_GEN_BD
 	help
-- 
1.6.2.4

[PATCH 14/17] ASoC: Staticise TLV values in WM8940

[Mark Brown]

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8940.c |   22 +++++++++++-----------
 1 files changed, 11 insertions(+), 11 deletions(-)

diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 26987dc..a66dacc 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -168,16 +168,16 @@ static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
 static const struct soc_enum wm8940_filter_mode_enum
 = SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
 
-DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
-DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
-DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
-DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
-DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
-DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
-DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
-DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
-DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
-DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
 
 static const struct snd_kcontrol_new wm8940_snd_controls[] = {
 	SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
@@ -253,7 +253,7 @@ static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
 	SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
 };
 
-DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
 static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
 	SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
 	SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
-- 
1.6.2.4

[PATCH 15/17] ASoC: TWL4030: Fix gain control for earpiece amplifier

[Mark Brown]

From: Joonyoung Shim <jy0922.shim@samsung.com>

The gain control for earpiece amplifier uses 0dB ~ 12dB according to the
TRM, but the present code is implemented to -6dB ~ 6dB.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |    8 +++++++-
 1 files changed, 7 insertions(+), 1 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205..df7c8c2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		4, 3, 0, output_tvl),
 
 	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
-- 
1.6.2.4

[PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

From: Jon Smirl <jonsmirl@gmail.com>

Basic split of mpc5200 DMA code out from i2s into a standalone file.

[Fixed trailing whitespace -- broonie.]

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/Kconfig           |    4 +
 sound/soc/fsl/Makefile          |    2 +
 sound/soc/fsl/mpc5200_dma.c     |  457 ++++++++++++++++++++++++++++++++++++
 sound/soc/fsl/mpc5200_dma.h     |   81 +++++++
 sound/soc/fsl/mpc5200_psc_i2s.c |  484 +--------------------------------------
 5 files changed, 546 insertions(+), 482 deletions(-)
 create mode 100644 sound/soc/fsl/mpc5200_dma.c
 create mode 100644 sound/soc/fsl/mpc5200_dma.h

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index e7dd79a..d94745b 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,6 +1,9 @@
 config SND_SOC_OF_SIMPLE
 	tristate
 
+config SND_MPC52xx_DMA
+	tristate
+
 # ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
 # for the SSI and the Elo DMA controller.  You will still need to select
 # a platform driver and a codec driver.
@@ -23,6 +26,7 @@ config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
 	depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN
 	select SND_SOC_OF_SIMPLE
+	select SND_MPC52xx_DMA
 	select PPC_BESTCOMM_GEN_BD
 	help
 	  Say Y here to support the MPC5200 PSCs in I2S mode.
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f85134c..7731ef2 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,5 +10,7 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-dma-objs := fsl_dma.o
 obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
 
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 0000000..cccaff4
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,457 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
+{
+	struct psc_i2s *psc_i2s = _psc_i2s;
+	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	u16 isr;
+
+	isr = in_be16(&regs->mpc52xx_psc_isr);
+
+	/* Playback underrun error */
+	if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+		psc_i2s->stats.underrun_count++;
+
+	/* Capture overrun error */
+	if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+		psc_i2s->stats.overrun_count++;
+
+	out_8(&regs->command, 4 << 4);	/* reset the error status */
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue.  Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
+{
+	struct bcom_bd *bd;
+
+	/* Prepare and enqueue the next buffer descriptor */
+	bd = bcom_prepare_next_buffer(s->bcom_task);
+	bd->status = s->period_bytes;
+	bd->data[0] = s->period_next_pt;
+	bcom_submit_next_buffer(s->bcom_task, NULL);
+
+	/* Update for next period */
+	s->period_next_pt += s->period_bytes;
+	if (s->period_next_pt >= s->period_end)
+		s->period_next_pt = s->period_start;
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
+{
+	struct psc_i2s_stream *s = _psc_i2s_stream;
+
+	/* For each finished period, dequeue the completed period buffer
+	 * and enqueue a new one in it's place. */
+	while (bcom_buffer_done(s->bcom_task)) {
+		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+		s->period_current_pt += s->period_bytes;
+		if (s->period_current_pt >= s->period_end)
+			s->period_current_pt = s->period_start;
+		psc_i2s_bcom_enqueue_next_buffer(s);
+		bcom_enable(s->bcom_task);
+	}
+
+	/* If the stream is active, then also inform the PCM middle layer
+	 * of the period finished event. */
+	if (s->active)
+		snd_pcm_period_elapsed(s->stream);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_startup: create a new substream
+ *
+ * This is the first function called when a stream is opened.
+ *
+ * If this is the first stream open, then grab the IRQ and program most of
+ * the PSC registers.
+ */
+int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	int rc;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
+
+	if (!psc_i2s->playback.active &&
+	    !psc_i2s->capture.active) {
+		/* Setup the IRQs */
+		rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
+				 "psc-i2s-status", psc_i2s);
+		rc |= request_irq(psc_i2s->capture.irq,
+				  &psc_i2s_bcom_irq, IRQF_SHARED,
+				  "psc-i2s-capture", &psc_i2s->capture);
+		rc |= request_irq(psc_i2s->playback.irq,
+				  &psc_i2s_bcom_irq, IRQF_SHARED,
+				  "psc-i2s-playback", &psc_i2s->playback);
+		if (rc) {
+			free_irq(psc_i2s->irq, psc_i2s);
+			free_irq(psc_i2s->capture.irq,
+				 &psc_i2s->capture);
+			free_irq(psc_i2s->playback.irq,
+				 &psc_i2s->playback);
+			return -ENODEV;
+		}
+	}
+
+	return 0;
+}
+
+int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+/**
+ * psc_i2s_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct psc_i2s_stream *s;
+	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	u16 imr;
+	u8 psc_cmd;
+	unsigned long flags;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture;
+	else
+		s = &psc_i2s->playback;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
+		" stream_id=%i\n",
+		substream, cmd, substream->pstr->stream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		s->period_bytes = frames_to_bytes(runtime,
+						  runtime->period_size);
+		s->period_start = virt_to_phys(runtime->dma_area);
+		s->period_end = s->period_start +
+				(s->period_bytes * runtime->periods);
+		s->period_next_pt = s->period_start;
+		s->period_current_pt = s->period_start;
+		s->active = 1;
+
+		/* First; reset everything */
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			out_8(&regs->command, MPC52xx_PSC_RST_RX);
+			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+		} else {
+			out_8(&regs->command, MPC52xx_PSC_RST_TX);
+			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+		}
+
+		/* Next, fill up the bestcomm bd queue and enable DMA.
+		 * This will begin filling the PSC's fifo. */
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+			bcom_gen_bd_rx_reset(s->bcom_task);
+		else
+			bcom_gen_bd_tx_reset(s->bcom_task);
+		while (!bcom_queue_full(s->bcom_task))
+			psc_i2s_bcom_enqueue_next_buffer(s);
+		bcom_enable(s->bcom_task);
+
+		/* Due to errata in the i2s mode; need to line up enabling
+		 * the transmitter with a transition on the frame sync
+		 * line */
+
+		spin_lock_irqsave(&psc_i2s->lock, flags);
+		/* first make sure it is low */
+		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+			;
+		/* then wait for the transition to high */
+		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+			;
+		/* Finally, enable the PSC.
+		 * Receiver must always be enabled; even when we only want
+		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
+		psc_cmd = MPC52xx_PSC_RX_ENABLE;
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
+		out_8(&regs->command, psc_cmd);
+		spin_unlock_irqrestore(&psc_i2s->lock, flags);
+
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+		/* Turn off the PSC */
+		s->active = 0;
+		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			if (!psc_i2s->playback.active) {
+				out_8(&regs->command, 2 << 4);	/* reset rx */
+				out_8(&regs->command, 3 << 4);	/* reset tx */
+				out_8(&regs->command, 4 << 4);	/* reset err */
+			}
+		} else {
+			out_8(&regs->command, 3 << 4);	/* reset tx */
+			out_8(&regs->command, 4 << 4);	/* reset err */
+			if (!psc_i2s->capture.active)
+				out_8(&regs->command, 2 << 4);	/* reset rx */
+		}
+
+		bcom_disable(s->bcom_task);
+		while (!bcom_queue_empty(s->bcom_task))
+			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+		break;
+
+	default:
+		dev_dbg(psc_i2s->dev, "invalid command\n");
+		return -EINVAL;
+	}
+
+	/* Update interrupt enable settings */
+	imr = 0;
+	if (psc_i2s->playback.active)
+		imr |= MPC52xx_PSC_IMR_TXEMP;
+	if (psc_i2s->capture.active)
+		imr |= MPC52xx_PSC_IMR_ORERR;
+	out_be16(&regs->isr_imr.imr, imr);
+
+	return 0;
+}
+
+/**
+ * psc_i2s_shutdown: shutdown the data transfer on a stream
+ *
+ * Shutdown the PSC if there are no other substreams open.
+ */
+void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
+
+	/*
+	 * If this is the last active substream, disable the PSC and release
+	 * the IRQ.
+	 */
+	if (!psc_i2s->playback.active &&
+	    !psc_i2s->capture.active) {
+
+		/* Disable all interrupts and reset the PSC */
+		out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
+		out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
+		out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
+		out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
+		out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+
+		/* Release irqs */
+		free_irq(psc_i2s->irq, psc_i2s);
+		free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
+		free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
+	}
+}
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
+	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+	.rate_min = 8000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.period_bytes_max	= 1024 * 1024,
+	.period_bytes_min	= 32,
+	.periods_min		= 2,
+	.periods_max		= 256,
+	.buffer_bytes_max	= 2 * 1024 * 1024,
+	.fifo_size		= 0,
+};
+
+static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture;
+	else
+		s = &psc_i2s->playback;
+
+	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
+
+	s->stream = substream;
+	return 0;
+}
+
+static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+
+	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture;
+	else
+		s = &psc_i2s->playback;
+
+	s->stream = NULL;
+	return 0;
+}
+
+static snd_pcm_uframes_t
+psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_i2s_stream *s;
+	dma_addr_t count;
+
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		s = &psc_i2s->capture;
+	else
+		s = &psc_i2s->playback;
+
+	count = s->period_current_pt - s->period_start;
+
+	return bytes_to_frames(substream->runtime, count);
+}
+
+static struct snd_pcm_ops psc_i2s_pcm_ops = {
+	.open		= psc_i2s_pcm_open,
+	.close		= psc_i2s_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.pointer	= psc_i2s_pcm_pointer,
+};
+
+static u64 psc_i2s_pcm_dmamask = 0xffffffff;
+static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+			   struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
+	int rc = 0;
+
+	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
+		card, dai, pcm);
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (pcm->streams[0].substream) {
+		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+					&pcm->streams[0].substream->dma_buffer);
+		if (rc)
+			goto playback_alloc_err;
+	}
+
+	if (pcm->streams[1].substream) {
+		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+					&pcm->streams[1].substream->dma_buffer);
+		if (rc)
+			goto capture_alloc_err;
+	}
+
+	return 0;
+
+ capture_alloc_err:
+	if (pcm->streams[0].substream)
+		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+ playback_alloc_err:
+	dev_err(card->dev, "Cannot allocate buffer(s)\n");
+	return -ENOMEM;
+}
+
+static void psc_i2s_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+	struct snd_pcm_substream *substream;
+	int stream;
+
+	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (substream) {
+			snd_dma_free_pages(&substream->dma_buffer);
+			substream->dma_buffer.area = NULL;
+			substream->dma_buffer.addr = 0;
+		}
+	}
+}
+
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
+	.name		= "mpc5200-psc-audio",
+	.pcm_ops	= &psc_i2s_pcm_ops,
+	.pcm_new	= &psc_i2s_pcm_new,
+	.pcm_free	= &psc_i2s_pcm_free,
+};
+
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 0000000..9a19e8a
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,81 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+/**
+ * psc_i2s_stream - Data specific to a single stream (playback or capture)
+ * @active:		flag indicating if the stream is active
+ * @psc_i2s:		pointer back to parent psc_i2s data structure
+ * @bcom_task:		bestcomm task structure
+ * @irq:		irq number for bestcomm task
+ * @period_start:	physical address of start of DMA region
+ * @period_end:		physical address of end of DMA region
+ * @period_next_pt:	physical address of next DMA buffer to enqueue
+ * @period_bytes:	size of DMA period in bytes
+ */
+struct psc_i2s_stream {
+	int active;
+	struct psc_i2s *psc_i2s;
+	struct bcom_task *bcom_task;
+	int irq;
+	struct snd_pcm_substream *stream;
+	dma_addr_t period_start;
+	dma_addr_t period_end;
+	dma_addr_t period_next_pt;
+	dma_addr_t period_current_pt;
+	int period_bytes;
+};
+
+/**
+ * psc_i2s - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ *        with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_i2s {
+	char name[32];
+	struct mpc52xx_psc __iomem *psc_regs;
+	struct mpc52xx_psc_fifo __iomem *fifo_regs;
+	unsigned int irq;
+	struct device *dev;
+	struct snd_soc_dai dai;
+	spinlock_t lock;
+	u32 sicr;
+
+	/* per-stream data */
+	struct psc_i2s_stream playback;
+	struct psc_i2s_stream capture;
+
+	/* Statistics */
+	struct {
+		int overrun_count;
+		int underrun_count;
+	} stats;
+};
+
+
+int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai);
+
+int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai);
+
+void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai);
+
+int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai);
+
+extern struct snd_soc_platform psc_i2s_pcm_soc_platform;
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729d..8974b53 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -25,6 +25,8 @@
 #include <sysdev/bestcomm/gen_bd.h>
 #include <asm/mpc52xx_psc.h>
 
+#include "mpc5200_dma.h"
+
 MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
 MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
 MODULE_LICENSE("GPL");
@@ -47,179 +49,6 @@ MODULE_LICENSE("GPL");
 			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
 			 SNDRV_PCM_FMTBIT_S32_BE)
 
-/**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
- * @active:		flag indicating if the stream is active
- * @psc_i2s:		pointer back to parent psc_i2s data structure
- * @bcom_task:		bestcomm task structure
- * @irq:		irq number for bestcomm task
- * @period_start:	physical address of start of DMA region
- * @period_end:		physical address of end of DMA region
- * @period_next_pt:	physical address of next DMA buffer to enqueue
- * @period_bytes:	size of DMA period in bytes
- */
-struct psc_i2s_stream {
-	int active;
-	struct psc_i2s *psc_i2s;
-	struct bcom_task *bcom_task;
-	int irq;
-	struct snd_pcm_substream *stream;
-	dma_addr_t period_start;
-	dma_addr_t period_end;
-	dma_addr_t period_next_pt;
-	dma_addr_t period_current_pt;
-	int period_bytes;
-};
-
-/**
- * psc_i2s - Private driver data
- * @name: short name for this device ("PSC0", "PSC1", etc)
- * @psc_regs: pointer to the PSC's registers
- * @fifo_regs: pointer to the PSC's FIFO registers
- * @irq: IRQ of this PSC
- * @dev: struct device pointer
- * @dai: the CPU DAI for this device
- * @sicr: Base value used in serial interface control register; mode is ORed
- *        with this value.
- * @playback: Playback stream context data
- * @capture: Capture stream context data
- */
-struct psc_i2s {
-	char name[32];
-	struct mpc52xx_psc __iomem *psc_regs;
-	struct mpc52xx_psc_fifo __iomem *fifo_regs;
-	unsigned int irq;
-	struct device *dev;
-	struct snd_soc_dai dai;
-	spinlock_t lock;
-	u32 sicr;
-
-	/* per-stream data */
-	struct psc_i2s_stream playback;
-	struct psc_i2s_stream capture;
-
-	/* Statistics */
-	struct {
-		int overrun_count;
-		int underrun_count;
-	} stats;
-};
-
-/*
- * Interrupt handlers
- */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
-{
-	struct psc_i2s *psc_i2s = _psc_i2s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-	u16 isr;
-
-	isr = in_be16(&regs->mpc52xx_psc_isr);
-
-	/* Playback underrun error */
-	if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
-		psc_i2s->stats.underrun_count++;
-
-	/* Capture overrun error */
-	if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
-		psc_i2s->stats.overrun_count++;
-
-	out_8(&regs->command, 4 << 4);	/* reset the error status */
-
-	return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- * @s: pointer to stream private data structure
- *
- * Enqueues another audio period buffer into the bestcomm queue.
- *
- * Note: The routine must only be called when there is space available in
- * the queue.  Otherwise the enqueue will fail and the audio ring buffer
- * will get out of sync
- */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
-{
-	struct bcom_bd *bd;
-
-	/* Prepare and enqueue the next buffer descriptor */
-	bd = bcom_prepare_next_buffer(s->bcom_task);
-	bd->status = s->period_bytes;
-	bd->data[0] = s->period_next_pt;
-	bcom_submit_next_buffer(s->bcom_task, NULL);
-
-	/* Update for next period */
-	s->period_next_pt += s->period_bytes;
-	if (s->period_next_pt >= s->period_end)
-		s->period_next_pt = s->period_start;
-}
-
-/* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
-{
-	struct psc_i2s_stream *s = _psc_i2s_stream;
-
-	/* For each finished period, dequeue the completed period buffer
-	 * and enqueue a new one in it's place. */
-	while (bcom_buffer_done(s->bcom_task)) {
-		bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-		s->period_current_pt += s->period_bytes;
-		if (s->period_current_pt >= s->period_end)
-			s->period_current_pt = s->period_start;
-		psc_i2s_bcom_enqueue_next_buffer(s);
-		bcom_enable(s->bcom_task);
-	}
-
-	/* If the stream is active, then also inform the PCM middle layer
-	 * of the period finished event. */
-	if (s->active)
-		snd_pcm_period_elapsed(s->stream);
-
-	return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_startup: create a new substream
- *
- * This is the first function called when a stream is opened.
- *
- * If this is the first stream open, then grab the IRQ and program most of
- * the PSC registers.
- */
-static int psc_i2s_startup(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	int rc;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
-
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
-		/* Setup the IRQs */
-		rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
-				 "psc-i2s-status", psc_i2s);
-		rc |= request_irq(psc_i2s->capture.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-capture", &psc_i2s->capture);
-		rc |= request_irq(psc_i2s->playback.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-playback", &psc_i2s->playback);
-		if (rc) {
-			free_irq(psc_i2s->irq, psc_i2s);
-			free_irq(psc_i2s->capture.irq,
-				 &psc_i2s->capture);
-			free_irq(psc_i2s->playback.irq,
-				 &psc_i2s->playback);
-			return -ENODEV;
-		}
-	}
-
-	return 0;
-}
-
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
@@ -258,164 +87,6 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
-{
-	snd_pcm_set_runtime_buffer(substream, NULL);
-	return 0;
-}
-
-/**
- * psc_i2s_trigger: start and stop the DMA transfer.
- *
- * This function is called by ALSA to start, stop, pause, and resume the DMA
- * transfer of data.
- */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct psc_i2s_stream *s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-	u16 imr;
-	u8 psc_cmd;
-	unsigned long flags;
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
-		" stream_id=%i\n",
-		substream, cmd, substream->pstr->stream);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		s->period_bytes = frames_to_bytes(runtime,
-						  runtime->period_size);
-		s->period_start = virt_to_phys(runtime->dma_area);
-		s->period_end = s->period_start +
-				(s->period_bytes * runtime->periods);
-		s->period_next_pt = s->period_start;
-		s->period_current_pt = s->period_start;
-		s->active = 1;
-
-		/* First; reset everything */
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-			out_8(&regs->command, MPC52xx_PSC_RST_RX);
-			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-		} else {
-			out_8(&regs->command, MPC52xx_PSC_RST_TX);
-			out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-		}
-
-		/* Next, fill up the bestcomm bd queue and enable DMA.
-		 * This will begin filling the PSC's fifo. */
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-			bcom_gen_bd_rx_reset(s->bcom_task);
-		else
-			bcom_gen_bd_tx_reset(s->bcom_task);
-		while (!bcom_queue_full(s->bcom_task))
-			psc_i2s_bcom_enqueue_next_buffer(s);
-		bcom_enable(s->bcom_task);
-
-		/* Due to errata in the i2s mode; need to line up enabling
-		 * the transmitter with a transition on the frame sync
-		 * line */
-
-		spin_lock_irqsave(&psc_i2s->lock, flags);
-		/* first make sure it is low */
-		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
-			;
-		/* then wait for the transition to high */
-		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
-			;
-		/* Finally, enable the PSC.
-		 * Receiver must always be enabled; even when we only want
-		 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
-		psc_cmd = MPC52xx_PSC_RX_ENABLE;
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
-		out_8(&regs->command, psc_cmd);
-		spin_unlock_irqrestore(&psc_i2s->lock, flags);
-
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-		/* Turn off the PSC */
-		s->active = 0;
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-			if (!psc_i2s->playback.active) {
-				out_8(&regs->command, 2 << 4);	/* reset rx */
-				out_8(&regs->command, 3 << 4);	/* reset tx */
-				out_8(&regs->command, 4 << 4);	/* reset err */
-			}
-		} else {
-			out_8(&regs->command, 3 << 4);	/* reset tx */
-			out_8(&regs->command, 4 << 4);	/* reset err */
-			if (!psc_i2s->capture.active)
-				out_8(&regs->command, 2 << 4);	/* reset rx */
-		}
-
-		bcom_disable(s->bcom_task);
-		while (!bcom_queue_empty(s->bcom_task))
-			bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-
-		break;
-
-	default:
-		dev_dbg(psc_i2s->dev, "invalid command\n");
-		return -EINVAL;
-	}
-
-	/* Update interrupt enable settings */
-	imr = 0;
-	if (psc_i2s->playback.active)
-		imr |= MPC52xx_PSC_IMR_TXEMP;
-	if (psc_i2s->capture.active)
-		imr |= MPC52xx_PSC_IMR_ORERR;
-	out_be16(&regs->isr_imr.imr, imr);
-
-	return 0;
-}
-
-/**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
- *
- * Shutdown the PSC if there are no other substreams open.
- */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
-
-	/*
-	 * If this is the last active substream, disable the PSC and release
-	 * the IRQ.
-	 */
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
-
-		/* Disable all interrupts and reset the PSC */
-		out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-		out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
-		out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
-		out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-		out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
-
-		/* Release irqs */
-		free_irq(psc_i2s->irq, psc_i2s);
-		free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
-		free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
-	}
-}
-
 /**
  * psc_i2s_set_sysclk: set the clock frequency and direction
  *
@@ -495,157 +166,6 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 };
 
 /* ---------------------------------------------------------------------
- * The PSC I2S 'ASoC platform' driver
- *
- * Can be referenced by an 'ASoC machine' driver
- * This driver only deals with the audio bus; it doesn't have any
- * interaction with the attached codec
- */
-
-static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
-	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
-	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
-		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
-	.rate_min = 8000,
-	.rate_max = 48000,
-	.channels_min = 2,
-	.channels_max = 2,
-	.period_bytes_max	= 1024 * 1024,
-	.period_bytes_min	= 32,
-	.periods_min		= 2,
-	.periods_max		= 256,
-	.buffer_bytes_max	= 2 * 1024 * 1024,
-	.fifo_size		= 0,
-};
-
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
-
-	s->stream = substream;
-	return 0;
-}
-
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	s->stream = NULL;
-	return 0;
-}
-
-static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
-	dma_addr_t count;
-
-	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
-	else
-		s = &psc_i2s->playback;
-
-	count = s->period_current_pt - s->period_start;
-
-	return bytes_to_frames(substream->runtime, count);
-}
-
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
-	.open		= psc_i2s_pcm_open,
-	.close		= psc_i2s_pcm_close,
-	.ioctl		= snd_pcm_lib_ioctl,
-	.pointer	= psc_i2s_pcm_pointer,
-};
-
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-			   struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
-	int rc = 0;
-
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
-		card, dai, pcm);
-
-	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
-	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = 0xffffffff;
-
-	if (pcm->streams[0].substream) {
-		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-					&pcm->streams[0].substream->dma_buffer);
-		if (rc)
-			goto playback_alloc_err;
-	}
-
-	if (pcm->streams[1].substream) {
-		rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-					&pcm->streams[1].substream->dma_buffer);
-		if (rc)
-			goto capture_alloc_err;
-	}
-
-	return 0;
-
- capture_alloc_err:
-	if (pcm->streams[0].substream)
-		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
- playback_alloc_err:
-	dev_err(card->dev, "Cannot allocate buffer(s)\n");
-	return -ENOMEM;
-}
-
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	struct snd_pcm_substream *substream;
-	int stream;
-
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
-
-	for (stream = 0; stream < 2; stream++) {
-		substream = pcm->streams[stream].substream;
-		if (substream) {
-			snd_dma_free_pages(&substream->dma_buffer);
-			substream->dma_buffer.area = NULL;
-			substream->dma_buffer.addr = 0;
-		}
-	}
-}
-
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
-	.name		= "mpc5200-psc-audio",
-	.pcm_ops	= &psc_i2s_pcm_ops,
-	.pcm_new	= &psc_i2s_pcm_new,
-	.pcm_free	= &psc_i2s_pcm_free,
-};
-
-/* ---------------------------------------------------------------------
  * Sysfs attributes for debugging
  */
 
-- 
1.6.2.4

[PATCH 17/17] ASoC: Rename the PSC functions to DMA

[Mark Brown]

From: Jon Smirl <jonsmirl@gmail.com>

Rename the functions in the mpc5200 DMA file from i2s to dma to reflect
the file they are in.

Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/mpc5200_dma.c     |  194 +++++++++++++++++++-------------------
 sound/soc/fsl/mpc5200_dma.h     |   26 +++---
 sound/soc/fsl/mpc5200_psc_i2s.c |  160 ++++++++++++++++----------------
 3 files changed, 190 insertions(+), 190 deletions(-)

diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index cccaff4..c82ef75 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -34,21 +34,21 @@ MODULE_LICENSE("GPL");
 /*
  * Interrupt handlers
  */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
 {
-	struct psc_i2s *psc_i2s = _psc_i2s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	struct psc_dma *psc_dma = _psc_dma;
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
 	u16 isr;
 
 	isr = in_be16(&regs->mpc52xx_psc_isr);
 
 	/* Playback underrun error */
-	if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
-		psc_i2s->stats.underrun_count++;
+	if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+		psc_dma->stats.underrun_count++;
 
 	/* Capture overrun error */
-	if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
-		psc_i2s->stats.overrun_count++;
+	if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+		psc_dma->stats.overrun_count++;
 
 	out_8(&regs->command, 4 << 4);	/* reset the error status */
 
@@ -56,7 +56,7 @@ static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
 }
 
 /**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
  * @s: pointer to stream private data structure
  *
  * Enqueues another audio period buffer into the bestcomm queue.
@@ -65,7 +65,7 @@ static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
  * the queue.  Otherwise the enqueue will fail and the audio ring buffer
  * will get out of sync
  */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
 {
 	struct bcom_bd *bd;
 
@@ -82,9 +82,9 @@ static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
 }
 
 /* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
+static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream)
 {
-	struct psc_i2s_stream *s = _psc_i2s_stream;
+	struct psc_dma_stream *s = _psc_dma_stream;
 
 	/* For each finished period, dequeue the completed period buffer
 	 * and enqueue a new one in it's place. */
@@ -93,7 +93,7 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
 		s->period_current_pt += s->period_bytes;
 		if (s->period_current_pt >= s->period_end)
 			s->period_current_pt = s->period_start;
-		psc_i2s_bcom_enqueue_next_buffer(s);
+		psc_dma_bcom_enqueue_next_buffer(s);
 		bcom_enable(s->bcom_task);
 	}
 
@@ -106,39 +106,39 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
 }
 
 /**
- * psc_i2s_startup: create a new substream
+ * psc_dma_startup: create a new substream
  *
  * This is the first function called when a stream is opened.
  *
  * If this is the first stream open, then grab the IRQ and program most of
  * the PSC registers.
  */
-int psc_i2s_startup(struct snd_pcm_substream *substream,
+int psc_dma_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
 	int rc;
 
-	dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
+	dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream);
 
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
+	if (!psc_dma->playback.active &&
+	    !psc_dma->capture.active) {
 		/* Setup the IRQs */
-		rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
-				 "psc-i2s-status", psc_i2s);
-		rc |= request_irq(psc_i2s->capture.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-capture", &psc_i2s->capture);
-		rc |= request_irq(psc_i2s->playback.irq,
-				  &psc_i2s_bcom_irq, IRQF_SHARED,
-				  "psc-i2s-playback", &psc_i2s->playback);
+		rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+				 "psc-dma-status", psc_dma);
+		rc |= request_irq(psc_dma->capture.irq,
+				  &psc_dma_bcom_irq, IRQF_SHARED,
+				  "psc-dma-capture", &psc_dma->capture);
+		rc |= request_irq(psc_dma->playback.irq,
+				  &psc_dma_bcom_irq, IRQF_SHARED,
+				  "psc-dma-playback", &psc_dma->playback);
 		if (rc) {
-			free_irq(psc_i2s->irq, psc_i2s);
-			free_irq(psc_i2s->capture.irq,
-				 &psc_i2s->capture);
-			free_irq(psc_i2s->playback.irq,
-				 &psc_i2s->playback);
+			free_irq(psc_dma->irq, psc_dma);
+			free_irq(psc_dma->capture.irq,
+				 &psc_dma->capture);
+			free_irq(psc_dma->playback.irq,
+				 &psc_dma->playback);
 			return -ENODEV;
 		}
 	}
@@ -146,7 +146,7 @@ int psc_i2s_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+int psc_dma_hw_free(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
@@ -154,29 +154,29 @@ int psc_i2s_hw_free(struct snd_pcm_substream *substream,
 }
 
 /**
- * psc_i2s_trigger: start and stop the DMA transfer.
+ * psc_dma_trigger: start and stop the DMA transfer.
  *
  * This function is called by ALSA to start, stop, pause, and resume the DMA
  * transfer of data.
  */
-int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct psc_i2s_stream *s;
-	struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+	struct psc_dma_stream *s;
+	struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
 	u16 imr;
 	u8 psc_cmd;
 	unsigned long flags;
 
 	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
+		s = &psc_dma->capture;
 	else
-		s = &psc_i2s->playback;
+		s = &psc_dma->playback;
 
-	dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
+	dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
 		" stream_id=%i\n",
 		substream, cmd, substream->pstr->stream);
 
@@ -207,14 +207,14 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		else
 			bcom_gen_bd_tx_reset(s->bcom_task);
 		while (!bcom_queue_full(s->bcom_task))
-			psc_i2s_bcom_enqueue_next_buffer(s);
+			psc_dma_bcom_enqueue_next_buffer(s);
 		bcom_enable(s->bcom_task);
 
-		/* Due to errata in the i2s mode; need to line up enabling
+		/* Due to errata in the dma mode; need to line up enabling
 		 * the transmitter with a transition on the frame sync
 		 * line */
 
-		spin_lock_irqsave(&psc_i2s->lock, flags);
+		spin_lock_irqsave(&psc_dma->lock, flags);
 		/* first make sure it is low */
 		while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
 			;
@@ -228,7 +228,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			psc_cmd |= MPC52xx_PSC_TX_ENABLE;
 		out_8(&regs->command, psc_cmd);
-		spin_unlock_irqrestore(&psc_i2s->lock, flags);
+		spin_unlock_irqrestore(&psc_dma->lock, flags);
 
 		break;
 
@@ -236,7 +236,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		/* Turn off the PSC */
 		s->active = 0;
 		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-			if (!psc_i2s->playback.active) {
+			if (!psc_dma->playback.active) {
 				out_8(&regs->command, 2 << 4);	/* reset rx */
 				out_8(&regs->command, 3 << 4);	/* reset tx */
 				out_8(&regs->command, 4 << 4);	/* reset err */
@@ -244,7 +244,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		} else {
 			out_8(&regs->command, 3 << 4);	/* reset tx */
 			out_8(&regs->command, 4 << 4);	/* reset err */
-			if (!psc_i2s->capture.active)
+			if (!psc_dma->capture.active)
 				out_8(&regs->command, 2 << 4);	/* reset rx */
 		}
 
@@ -255,15 +255,15 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		break;
 
 	default:
-		dev_dbg(psc_i2s->dev, "invalid command\n");
+		dev_dbg(psc_dma->dev, "invalid command\n");
 		return -EINVAL;
 	}
 
 	/* Update interrupt enable settings */
 	imr = 0;
-	if (psc_i2s->playback.active)
+	if (psc_dma->playback.active)
 		imr |= MPC52xx_PSC_IMR_TXEMP;
-	if (psc_i2s->capture.active)
+	if (psc_dma->capture.active)
 		imr |= MPC52xx_PSC_IMR_ORERR;
 	out_be16(&regs->isr_imr.imr, imr);
 
@@ -271,36 +271,36 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 }
 
 /**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
+ * psc_dma_shutdown: shutdown the data transfer on a stream
  *
  * Shutdown the PSC if there are no other substreams open.
  */
-void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+void psc_dma_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
 
-	dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
+	dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream);
 
 	/*
 	 * If this is the last active substream, disable the PSC and release
 	 * the IRQ.
 	 */
-	if (!psc_i2s->playback.active &&
-	    !psc_i2s->capture.active) {
+	if (!psc_dma->playback.active &&
+	    !psc_dma->capture.active) {
 
 		/* Disable all interrupts and reset the PSC */
-		out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-		out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
-		out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
-		out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-		out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+		out_be16(&psc_dma->psc_regs->isr_imr.imr, 0);
+		out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */
+		out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */
+		out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */
+		out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
 
 		/* Release irqs */
-		free_irq(psc_i2s->irq, psc_i2s);
-		free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
-		free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
+		free_irq(psc_dma->irq, psc_dma);
+		free_irq(psc_dma->capture.irq, &psc_dma->capture);
+		free_irq(psc_dma->playback.irq, &psc_dma->playback);
 	}
 }
 
@@ -312,7 +312,7 @@ void psc_i2s_shutdown(struct snd_pcm_substream *substream,
  * interaction with the attached codec
  */
 
-static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
+static const struct snd_pcm_hardware psc_dma_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
 	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
@@ -329,80 +329,80 @@ static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
 	.fifo_size		= 0,
 };
 
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
+static int psc_dma_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
 
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
+	dev_dbg(psc_dma->dev, "psc_dma_pcm_open(substream=%p)\n", substream);
 
 	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
+		s = &psc_dma->capture;
 	else
-		s = &psc_i2s->playback;
+		s = &psc_dma->playback;
 
-	snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
+	snd_soc_set_runtime_hwparams(substream, &psc_dma_pcm_hardware);
 
 	s->stream = substream;
 	return 0;
 }
 
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
+static int psc_dma_pcm_close(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
 
-	dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
+	dev_dbg(psc_dma->dev, "psc_dma_pcm_close(substream=%p)\n", substream);
 
 	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
+		s = &psc_dma->capture;
 	else
-		s = &psc_i2s->playback;
+		s = &psc_dma->playback;
 
 	s->stream = NULL;
 	return 0;
 }
 
 static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
+psc_dma_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-	struct psc_i2s_stream *s;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+	struct psc_dma_stream *s;
 	dma_addr_t count;
 
 	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-		s = &psc_i2s->capture;
+		s = &psc_dma->capture;
 	else
-		s = &psc_i2s->playback;
+		s = &psc_dma->playback;
 
 	count = s->period_current_pt - s->period_start;
 
 	return bytes_to_frames(substream->runtime, count);
 }
 
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
-	.open		= psc_i2s_pcm_open,
-	.close		= psc_i2s_pcm_close,
+static struct snd_pcm_ops psc_dma_pcm_ops = {
+	.open		= psc_dma_pcm_open,
+	.close		= psc_dma_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-	.pointer	= psc_i2s_pcm_pointer,
+	.pointer	= psc_dma_pcm_pointer,
 };
 
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+static u64 psc_dma_pcm_dmamask = 0xffffffff;
+static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 			   struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
+	size_t size = psc_dma_pcm_hardware.buffer_bytes_max;
 	int rc = 0;
 
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
+	dev_dbg(rtd->socdev->dev, "psc_dma_pcm_new(card=%p, dai=%p, pcm=%p)\n",
 		card, dai, pcm);
 
 	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &psc_i2s_pcm_dmamask;
+		card->dev->dma_mask = &psc_dma_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
@@ -430,13 +430,13 @@ static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	return -ENOMEM;
 }
 
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
+static void psc_dma_pcm_free(struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
 	struct snd_pcm_substream *substream;
 	int stream;
 
-	dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
+	dev_dbg(rtd->socdev->dev, "psc_dma_pcm_free(pcm=%p)\n", pcm);
 
 	for (stream = 0; stream < 2; stream++) {
 		substream = pcm->streams[stream].substream;
@@ -448,10 +448,10 @@ static void psc_i2s_pcm_free(struct snd_pcm *pcm)
 	}
 }
 
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
+struct snd_soc_platform psc_dma_pcm_soc_platform = {
 	.name		= "mpc5200-psc-audio",
-	.pcm_ops	= &psc_i2s_pcm_ops,
-	.pcm_new	= &psc_i2s_pcm_new,
-	.pcm_free	= &psc_i2s_pcm_free,
+	.pcm_ops	= &psc_dma_pcm_ops,
+	.pcm_new	= &psc_dma_pcm_new,
+	.pcm_free	= &psc_dma_pcm_free,
 };
 
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 9a19e8a..a33232c 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -6,9 +6,9 @@
 #define __SOUND_SOC_FSL_MPC5200_DMA_H__
 
 /**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
+ * psc_dma_stream - Data specific to a single stream (playback or capture)
  * @active:		flag indicating if the stream is active
- * @psc_i2s:		pointer back to parent psc_i2s data structure
+ * @psc_dma:		pointer back to parent psc_dma data structure
  * @bcom_task:		bestcomm task structure
  * @irq:		irq number for bestcomm task
  * @period_start:	physical address of start of DMA region
@@ -16,9 +16,9 @@
  * @period_next_pt:	physical address of next DMA buffer to enqueue
  * @period_bytes:	size of DMA period in bytes
  */
-struct psc_i2s_stream {
+struct psc_dma_stream {
 	int active;
-	struct psc_i2s *psc_i2s;
+	struct psc_dma *psc_dma;
 	struct bcom_task *bcom_task;
 	int irq;
 	struct snd_pcm_substream *stream;
@@ -30,7 +30,7 @@ struct psc_i2s_stream {
 };
 
 /**
- * psc_i2s - Private driver data
+ * psc_dma - Private driver data
  * @name: short name for this device ("PSC0", "PSC1", etc)
  * @psc_regs: pointer to the PSC's registers
  * @fifo_regs: pointer to the PSC's FIFO registers
@@ -42,7 +42,7 @@ struct psc_i2s_stream {
  * @playback: Playback stream context data
  * @capture: Capture stream context data
  */
-struct psc_i2s {
+struct psc_dma {
 	char name[32];
 	struct mpc52xx_psc __iomem *psc_regs;
 	struct mpc52xx_psc_fifo __iomem *fifo_regs;
@@ -53,8 +53,8 @@ struct psc_i2s {
 	u32 sicr;
 
 	/* per-stream data */
-	struct psc_i2s_stream playback;
-	struct psc_i2s_stream capture;
+	struct psc_dma_stream playback;
+	struct psc_dma_stream capture;
 
 	/* Statistics */
 	struct {
@@ -64,18 +64,18 @@ struct psc_i2s {
 };
 
 
-int psc_i2s_startup(struct snd_pcm_substream *substream,
+int psc_dma_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai);
 
-int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+int psc_dma_hw_free(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai);
 
-void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+void psc_dma_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai);
 
-int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd,
 			   struct snd_soc_dai *dai);
 
-extern struct snd_soc_platform psc_i2s_pcm_soc_platform;
+extern struct snd_soc_platform psc_dma_pcm_soc_platform;
 
 #endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 8974b53..12a7917 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -54,10 +54,10 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+	struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
 	u32 mode;
 
-	dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+	dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
 		" periods=%i buffer_size=%i  buffer_bytes=%i\n",
 		__func__, substream, params_period_size(params),
 		params_period_bytes(params), params_periods(params),
@@ -77,10 +77,10 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 		mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
 		break;
 	default:
-		dev_dbg(psc_i2s->dev, "invalid format\n");
+		dev_dbg(psc_dma->dev, "invalid format\n");
 		return -EINVAL;
 	}
-	out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
+	out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
@@ -104,8 +104,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 			      int clk_id, unsigned int freq, int dir)
 {
-	struct psc_i2s *psc_i2s = cpu_dai->private_data;
-	dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+	dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
 				cpu_dai, dir);
 	return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
 }
@@ -123,8 +123,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
  */
 static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 {
-	struct psc_i2s *psc_i2s = cpu_dai->private_data;
-	dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+	struct psc_dma *psc_dma = cpu_dai->private_data;
+	dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
 				cpu_dai, format);
 	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
 }
@@ -140,11 +140,11 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai_ops psc_i2s_dai_ops = {
-	.startup	= psc_i2s_startup,
+	.startup	= psc_dma_startup,
 	.hw_params	= psc_i2s_hw_params,
-	.hw_free	= psc_i2s_hw_free,
-	.shutdown	= psc_i2s_shutdown,
-	.trigger	= psc_i2s_trigger,
+	.hw_free	= psc_dma_hw_free,
+	.shutdown	= psc_dma_shutdown,
+	.trigger	= psc_dma_trigger,
 	.set_sysclk	= psc_i2s_set_sysclk,
 	.set_fmt	= psc_i2s_set_fmt,
 };
@@ -172,24 +172,24 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 static ssize_t psc_i2s_status_show(struct device *dev,
 			   struct device_attribute *attr, char *buf)
 {
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+	struct psc_dma *psc_dma = dev_get_drvdata(dev);
 
 	return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
 			"tfnum=%i tfstat=0x%.4x\n",
-			in_be16(&psc_i2s->psc_regs->sr_csr.status),
-			in_be32(&psc_i2s->psc_regs->sicr),
-			in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
-			in_be16(&psc_i2s->fifo_regs->rfstat),
-			in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
-			in_be16(&psc_i2s->fifo_regs->tfstat));
+			in_be16(&psc_dma->psc_regs->sr_csr.status),
+			in_be32(&psc_dma->psc_regs->sicr),
+			in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff,
+			in_be16(&psc_dma->fifo_regs->rfstat),
+			in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff,
+			in_be16(&psc_dma->fifo_regs->tfstat));
 }
 
-static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
+static int *psc_i2s_get_stat_attr(struct psc_dma *psc_dma, const char *name)
 {
 	if (strcmp(name, "playback_underrun") == 0)
-		return &psc_i2s->stats.underrun_count;
+		return &psc_dma->stats.underrun_count;
 	if (strcmp(name, "capture_overrun") == 0)
-		return &psc_i2s->stats.overrun_count;
+		return &psc_dma->stats.overrun_count;
 
 	return NULL;
 }
@@ -197,10 +197,10 @@ static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
 static ssize_t psc_i2s_stat_show(struct device *dev,
 				 struct device_attribute *attr, char *buf)
 {
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+	struct psc_dma *psc_dma = dev_get_drvdata(dev);
 	int *attrib;
 
-	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+	attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name);
 	if (!attrib)
 		return 0;
 
@@ -212,10 +212,10 @@ static ssize_t psc_i2s_stat_store(struct device *dev,
 				  const char *buf,
 				  size_t count)
 {
-	struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+	struct psc_dma *psc_dma = dev_get_drvdata(dev);
 	int *attrib;
 
-	attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+	attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name);
 	if (!attrib)
 		return 0;
 
@@ -238,7 +238,7 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 				      const struct of_device_id *match)
 {
 	phys_addr_t fifo;
-	struct psc_i2s *psc_i2s;
+	struct psc_dma *psc_dma;
 	struct resource res;
 	int size, psc_id, irq, rc;
 	const __be32 *prop;
@@ -265,56 +265,56 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	}
 
 	/* Allocate and initialize the driver private data */
-	psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
-	if (!psc_i2s) {
+	psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+	if (!psc_dma) {
 		iounmap(regs);
 		return -ENOMEM;
 	}
-	spin_lock_init(&psc_i2s->lock);
-	psc_i2s->irq = irq;
-	psc_i2s->psc_regs = regs;
-	psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
-	psc_i2s->dev = &op->dev;
-	psc_i2s->playback.psc_i2s = psc_i2s;
-	psc_i2s->capture.psc_i2s = psc_i2s;
-	snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
+	spin_lock_init(&psc_dma->lock);
+	psc_dma->irq = irq;
+	psc_dma->psc_regs = regs;
+	psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+	psc_dma->dev = &op->dev;
+	psc_dma->playback.psc_dma = psc_dma;
+	psc_dma->capture.psc_dma = psc_dma;
+	snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_id+1);
 
 	/* Fill out the CPU DAI structure */
-	memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
-	psc_i2s->dai.private_data = psc_i2s;
-	psc_i2s->dai.name = psc_i2s->name;
-	psc_i2s->dai.id = psc_id;
+	memcpy(&psc_dma->dai, &psc_i2s_dai_template, sizeof psc_dma->dai);
+	psc_dma->dai.private_data = psc_dma;
+	psc_dma->dai.name = psc_dma->name;
+	psc_dma->dai.id = psc_id;
 
 	/* Find the address of the fifo data registers and setup the
 	 * DMA tasks */
 	fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
-	psc_i2s->capture.bcom_task =
+	psc_dma->capture.bcom_task =
 		bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
-	psc_i2s->playback.bcom_task =
+	psc_dma->playback.bcom_task =
 		bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
-	if (!psc_i2s->capture.bcom_task ||
-	    !psc_i2s->playback.bcom_task) {
+	if (!psc_dma->capture.bcom_task ||
+	    !psc_dma->playback.bcom_task) {
 		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
 		iounmap(regs);
-		kfree(psc_i2s);
+		kfree(psc_dma);
 		return -ENODEV;
 	}
 
 	/* Disable all interrupts and reset the PSC */
-	out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-	out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
-	out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
-	out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-	out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+	out_be16(&psc_dma->psc_regs->isr_imr.imr, 0);
+	out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset transmitter */
+	out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset receiver */
+	out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */
+	out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
 
 	/* Configure the serial interface mode; defaulting to CODEC8 mode */
-	psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+	psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
 			MPC52xx_PSC_SICR_CLKPOL;
 	if (of_get_property(op->node, "fsl,cellslave", NULL))
-		psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
+		psc_dma->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
 				 MPC52xx_PSC_SICR_GENCLK;
-	out_be32(&psc_i2s->psc_regs->sicr,
-		 psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+	out_be32(&psc_dma->psc_regs->sicr,
+		 psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
 
 	/* Check for the codec handle.  If it is not present then we
 	 * are done */
@@ -325,54 +325,54 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	 * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
 	 * Second write: register Normal mode for non loopback
 	 */
-	out_8(&psc_i2s->psc_regs->mode, 0);
-	out_8(&psc_i2s->psc_regs->mode, 0);
+	out_8(&psc_dma->psc_regs->mode, 0);
+	out_8(&psc_dma->psc_regs->mode, 0);
 
 	/* Set the TX and RX fifo alarm thresholds */
-	out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
-	out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
-	out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
-	out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
+	out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+	out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+	out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+	out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
 
 	/* Lookup the IRQ numbers */
-	psc_i2s->playback.irq =
-		bcom_get_task_irq(psc_i2s->playback.bcom_task);
-	psc_i2s->capture.irq =
-		bcom_get_task_irq(psc_i2s->capture.bcom_task);
+	psc_dma->playback.irq =
+		bcom_get_task_irq(psc_dma->playback.bcom_task);
+	psc_dma->capture.irq =
+		bcom_get_task_irq(psc_dma->capture.bcom_task);
 
 	/* Save what we've done so it can be found again later */
-	dev_set_drvdata(&op->dev, psc_i2s);
+	dev_set_drvdata(&op->dev, psc_dma);
 
 	/* Register the SYSFS files */
-	rc = device_create_file(psc_i2s->dev, &dev_attr_status);
-	rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
-	rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
+	rc = device_create_file(psc_dma->dev, &dev_attr_status);
+	rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun);
+	rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun);
 	if (rc)
-		dev_info(psc_i2s->dev, "error creating sysfs files\n");
+		dev_info(psc_dma->dev, "error creating sysfs files\n");
 
-	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+	snd_soc_register_platform(&psc_dma_pcm_soc_platform);
 
 	/* Tell the ASoC OF helpers about it */
-	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
-				     &psc_i2s->dai);
+	of_snd_soc_register_platform(&psc_dma_pcm_soc_platform, op->node,
+				     &psc_dma->dai);
 
 	return 0;
 }
 
 static int __devexit psc_i2s_of_remove(struct of_device *op)
 {
-	struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
+	struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
-	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+	snd_soc_unregister_platform(&psc_dma_pcm_soc_platform);
 
-	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
-	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
+	bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+	bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
 
-	iounmap(psc_i2s->psc_regs);
-	iounmap(psc_i2s->fifo_regs);
-	kfree(psc_i2s);
+	iounmap(psc_dma->psc_regs);
+	iounmap(psc_dma->fifo_regs);
+	kfree(psc_dma);
 	dev_set_drvdata(&op->dev, NULL);
 
 	return 0;
-- 
1.6.2.4

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 16:23:41 +0100,
Mark Brown wrote:
> 
> From: Jon Smirl <jonsmirl@gmail.com>
> 
> Basic split of mpc5200 DMA code out from i2s into a standalone file.
> 
> [Fixed trailing whitespace -- broonie.]
> 
> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
> Acked-by: Grant Likely <grant.likely@secretlab.ca>
> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

This patch conflicts with the last patch I merged to add
SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
(It's in fix/pcm-info-batch branch and not merged to */asoc branch
 yet.  Feel free to pull to your tree by yourself.)
And, the jiffies check shall be anyway fixed in the upstream PCM core
side, thus there is no big merit to hurry to merge the unfinished
changes now.

So, could you back off mpc5200 changes and wait for a while until the
issue regarding PCM jiffies-check is resolved?

Note that the info flag should contain SNDRV_PCM_INFO_BATCH unless the
driver provides the precise position reporting in the pointer
callback.  The flag can be removed once after you implement that
feature, of course.


thanks,

Takashi

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
> At Tue, 28 Apr 2009 16:23:41 +0100,
> Mark Brown wrote:
>>
>> From: Jon Smirl <jonsmirl@gmail.com>
>>
>> Basic split of mpc5200 DMA code out from i2s into a standalone file.
>>
>> [Fixed trailing whitespace -- broonie.]
>>
>> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
>> Acked-by: Grant Likely <grant.likely@secretlab.ca>
>> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
>
> This patch conflicts with the last patch I merged to add
> SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
> (It's in fix/pcm-info-batch branch and not merged to */asoc branch
>  yet.  Feel free to pull to your tree by yourself.)
> And, the jiffies check shall be anyway fixed in the upstream PCM core
> side, thus there is no big merit to hurry to merge the unfinished
> changes now.
>
> So, could you back off mpc5200 changes and wait for a while until the
> issue regarding PCM jiffies-check is resolved?

This is why I was pushing so hard to get the patches that split the
files in. Takashi is going to change the base mpc5200_i2c file and I
am going to have to manually fix up up my 14 patches layered on top.
Git handles file renames fine, but it doesn't handle the case of
splitting a file very well. Once the patches that split the file are
in the tree Git handles things ok.

>
> Note that the info flag should contain SNDRV_PCM_INFO_BATCH unless the
> driver provides the precise position reporting in the pointer
> callback.  The flag can be removed once after you implement that
> feature, of course.
>
>
> thanks,
>
> Takashi
>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 11:59:01 -0400,
Jon Smirl wrote:
> 
> On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
> > At Tue, 28 Apr 2009 16:23:41 +0100,
> > Mark Brown wrote:
> >>
> >> From: Jon Smirl <jonsmirl@gmail.com>
> >>
> >> Basic split of mpc5200 DMA code out from i2s into a standalone file.
> >>
> >> [Fixed trailing whitespace -- broonie.]
> >>
> >> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
> >> Acked-by: Grant Likely <grant.likely@secretlab.ca>
> >> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> >
> > This patch conflicts with the last patch I merged to add
> > SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
> > (It's in fix/pcm-info-batch branch and not merged to */asoc branch
> >  yet.  Feel free to pull to your tree by yourself.)
> > And, the jiffies check shall be anyway fixed in the upstream PCM core
> > side, thus there is no big merit to hurry to merge the unfinished
> > changes now.
> >
> > So, could you back off mpc5200 changes and wait for a while until the
> > issue regarding PCM jiffies-check is resolved?
> 
> This is why I was pushing so hard to get the patches that split the
> files in.

And this is why I am against pulling now :)


Takashi

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 12:51 PM, Takashi Iwai <tiwai@suse.de> wrote:
> At Tue, 28 Apr 2009 11:59:01 -0400,
> Jon Smirl wrote:
>>
>> On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
>> > At Tue, 28 Apr 2009 16:23:41 +0100,
>> > Mark Brown wrote:
>> >>
>> >> From: Jon Smirl <jonsmirl@gmail.com>
>> >>
>> >> Basic split of mpc5200 DMA code out from i2s into a standalone file.
>> >>
>> >> [Fixed trailing whitespace -- broonie.]
>> >>
>> >> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
>> >> Acked-by: Grant Likely <grant.likely@secretlab.ca>
>> >> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
>> >
>> > This patch conflicts with the last patch I merged to add
>> > SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
>> > (It's in fix/pcm-info-batch branch and not merged to */asoc branch
>> >  yet.  Feel free to pull to your tree by yourself.)
>> > And, the jiffies check shall be anyway fixed in the upstream PCM core
>> > side, thus there is no big merit to hurry to merge the unfinished
>> > changes now.
>> >
>> > So, could you back off mpc5200 changes and wait for a while until the
>> > issue regarding PCM jiffies-check is resolved?
>>
>> This is why I was pushing so hard to get the patches that split the
>> files in.
>
> And this is why I am against pulling now :)

I wanted the split files added to 2.6.30 not 2.6.31. If they had gone
into 2.6.30 you would have patched after the split and everything
would have propagated without merge conflicts. In general, changes in
git that cross a file split will cause manual manual merges in
everything pending past the split.

>
>
> Takashi
>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 12:55 PM, Jon Smirl <jonsmirl@gmail.com> wrote:
> On Tue, Apr 28, 2009 at 12:51 PM, Takashi Iwai <tiwai@suse.de> wrote:
>> At Tue, 28 Apr 2009 11:59:01 -0400,
>> Jon Smirl wrote:
>>>
>>> On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
>>> > At Tue, 28 Apr 2009 16:23:41 +0100,
>>> > Mark Brown wrote:
>>> >>
>>> >> From: Jon Smirl <jonsmirl@gmail.com>
>>> >>
>>> >> Basic split of mpc5200 DMA code out from i2s into a standalone file.
>>> >>
>>> >> [Fixed trailing whitespace -- broonie.]
>>> >>
>>> >> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
>>> >> Acked-by: Grant Likely <grant.likely@secretlab.ca>
>>> >> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
>>> >
>>> > This patch conflicts with the last patch I merged to add
>>> > SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
>>> > (It's in fix/pcm-info-batch branch and not merged to */asoc branch
>>> >  yet.  Feel free to pull to your tree by yourself.)
>>> > And, the jiffies check shall be anyway fixed in the upstream PCM core
>>> > side, thus there is no big merit to hurry to merge the unfinished
>>> > changes now.
>>> >
>>> > So, could you back off mpc5200 changes and wait for a while until the
>>> > issue regarding PCM jiffies-check is resolved?
>>>
>>> This is why I was pushing so hard to get the patches that split the
>>> files in.
>>
>> And this is why I am against pulling now :)
>
> I wanted the split files added to 2.6.30 not 2.6.31. If they had gone
> into 2.6.30 you would have patched after the split and everything
> would have propagated without merge conflicts. In general, changes in
> git that cross a file split will cause manual manual merges in
> everything pending past the split.

BTW, pulling the split patches out of 2.6.31 is just going to move the
merge mess onto my local patch stack. To eliminate the merge mess we
need to get the split in and then start patching after the split.

>
>>
>>
>> Takashi
>>
>
>
>
> --
> Jon Smirl
> jonsmirl@gmail.com
>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

On Tue, Apr 28, 2009 at 05:46:11PM +0200, Takashi Iwai wrote:

> And, the jiffies check shall be anyway fixed in the upstream PCM core
> side, thus there is no big merit to hurry to merge the unfinished
> changes now.

The refactoring is a largely orthogonal issue to the jiffies check - the
refactoring is needed to split the DMA code out so that it can be shared
with the AC97 driver John is writing.

> So, could you back off mpc5200 changes and wait for a while until the
> issue regarding PCM jiffies-check is resolved?

Done:

The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9:
  Peter Ujfalusi (1):
        ASoC: TWL4030: Add 4 channel TDM support

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Daniel Mack (3):
      ASoC: add SOC_DOUBLE_EXT macro
      ASoC: cs4270: fix Master Capture Switch polarity
      ASoC: cs4270: add Master Playback Switch

Jon Smirl (1):
      ASoC: Set the MPC5200 i2s driver to BROKEN status.

Jonathan Cameron (1):
      ASoC WM8940 Driver

Joonyoung Shim (1):
      ASoC: TWL4030: Fix gain control for earpiece amplifier

Mark Brown (10):
      ASoC: Fix S3C64xx IIS device registration and support both ports
      ASoC: S3C2412: Failing to get the I2S clock is an error
      ASoC: Enforce symmetric rates for S3C64xx I2S interface
      ASoC: Include WM8350 register definitions in CODEC header
      ASoC: s3c-i2s-v2 diagnostic improvements
      ASoC: Use our registration function for S3C64xx
      ASoC: Fix logic in WM8350 master clocking check
      Merge branch 'for-2.6.30' into for-2.6.31
      ASoC: Staticise TLV values in WM8940
      Merge branch 'for-2.6.30' into for-2.6.31

Peter Ujfalusi (1):
      ASoC: Beagle: Add support for 4 channel

 include/sound/soc.h             |    8 +
 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/cs4270.c       |   44 ++-
 sound/soc/codecs/twl4030.c      |    8 +-
 sound/soc/codecs/wm8350.c       |    2 +-
 sound/soc/codecs/wm8350.h       |    1 +
 sound/soc/codecs/wm8940.c       |  955 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8940.h       |  104 +++++
 sound/soc/fsl/Kconfig           |    2 +-
 sound/soc/omap/omap3beagle.c    |   26 +-
 sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
 sound/soc/s3c24xx/s3c64xx-i2s.c |  148 ++++--
 sound/soc/s3c24xx/s3c64xx-i2s.h |    2 +-
 14 files changed, 1241 insertions(+), 67 deletions(-)
 create mode 100644 sound/soc/codecs/wm8940.c
 create mode 100644 sound/soc/codecs/wm8940.h

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 3:25 PM, Mark Brown <broonie@sirena.org.uk> wrote:
> On Tue, Apr 28, 2009 at 05:46:11PM +0200, Takashi Iwai wrote:
>
>> And, the jiffies check shall be anyway fixed in the upstream PCM core
>> side, thus there is no big merit to hurry to merge the unfinished
>> changes now.
>
> The refactoring is a largely orthogonal issue to the jiffies check - the
> refactoring is needed to split the DMA code out so that it can be shared
> with the AC97 driver John is writing.
>
>> So, could you back off mpc5200 changes and wait for a while until the
>> issue regarding PCM jiffies-check is resolved?
>
> Done:

Now I get to deal with the merge mess in my patch stack. All this did
was move the problem.


>
> The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9:
>  Peter Ujfalusi (1):
>        ASoC: TWL4030: Add 4 channel TDM support
>
> are available in the git repository at:
>
>  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31
>
> Daniel Mack (3):
>      ASoC: add SOC_DOUBLE_EXT macro
>      ASoC: cs4270: fix Master Capture Switch polarity
>      ASoC: cs4270: add Master Playback Switch
>
> Jon Smirl (1):
>      ASoC: Set the MPC5200 i2s driver to BROKEN status.
>
> Jonathan Cameron (1):
>      ASoC WM8940 Driver
>
> Joonyoung Shim (1):
>      ASoC: TWL4030: Fix gain control for earpiece amplifier
>
> Mark Brown (10):
>      ASoC: Fix S3C64xx IIS device registration and support both ports
>      ASoC: S3C2412: Failing to get the I2S clock is an error
>      ASoC: Enforce symmetric rates for S3C64xx I2S interface
>      ASoC: Include WM8350 register definitions in CODEC header
>      ASoC: s3c-i2s-v2 diagnostic improvements
>      ASoC: Use our registration function for S3C64xx
>      ASoC: Fix logic in WM8350 master clocking check
>      Merge branch 'for-2.6.30' into for-2.6.31
>      ASoC: Staticise TLV values in WM8940
>      Merge branch 'for-2.6.30' into for-2.6.31
>
> Peter Ujfalusi (1):
>      ASoC: Beagle: Add support for 4 channel
>
>  include/sound/soc.h             |    8 +
>  sound/soc/codecs/Kconfig        |    4 +
>  sound/soc/codecs/Makefile       |    2 +
>  sound/soc/codecs/cs4270.c       |   44 ++-
>  sound/soc/codecs/twl4030.c      |    8 +-
>  sound/soc/codecs/wm8350.c       |    2 +-
>  sound/soc/codecs/wm8350.h       |    1 +
>  sound/soc/codecs/wm8940.c       |  955 +++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/wm8940.h       |  104 +++++
>  sound/soc/fsl/Kconfig           |    2 +-
>  sound/soc/omap/omap3beagle.c    |   26 +-
>  sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
>  sound/soc/s3c24xx/s3c64xx-i2s.c |  148 ++++--
>  sound/soc/s3c24xx/s3c64xx-i2s.h |    2 +-
>  14 files changed, 1241 insertions(+), 67 deletions(-)
>  create mode 100644 sound/soc/codecs/wm8940.c
>  create mode 100644 sound/soc/codecs/wm8940.h
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 12:55:16 -0400,
Jon Smirl wrote:
> 
> On Tue, Apr 28, 2009 at 12:51 PM, Takashi Iwai <tiwai@suse.de> wrote:
> > At Tue, 28 Apr 2009 11:59:01 -0400,
> > Jon Smirl wrote:
> >>
> >> On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
> >> > At Tue, 28 Apr 2009 16:23:41 +0100,
> >> > Mark Brown wrote:
> >> >>
> >> >> From: Jon Smirl <jonsmirl@gmail.com>
> >> >>
> >> >> Basic split of mpc5200 DMA code out from i2s into a standalone file.
> >> >>
> >> >> [Fixed trailing whitespace -- broonie.]
> >> >>
> >> >> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
> >> >> Acked-by: Grant Likely <grant.likely@secretlab.ca>
> >> >> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> >> >
> >> > This patch conflicts with the last patch I merged to add
> >> > SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
> >> > (It's in fix/pcm-info-batch branch and not merged to */asoc branch
> >> >  yet.  Feel free to pull to your tree by yourself.)
> >> > And, the jiffies check shall be anyway fixed in the upstream PCM core
> >> > side, thus there is no big merit to hurry to merge the unfinished
> >> > changes now.
> >> >
> >> > So, could you back off mpc5200 changes and wait for a while until the
> >> > issue regarding PCM jiffies-check is resolved?
> >>
> >> This is why I was pushing so hard to get the patches that split the
> >> files in.
> >
> > And this is why I am against pulling now :)
> 
> I wanted the split files added to 2.6.30 not 2.6.31. If they had gone
> into 2.6.30 you would have patched after the split and everything
> would have propagated without merge conflicts. In general, changes in
> git that cross a file split will cause manual manual merges in
> everything pending past the split.

Not really so.  The changes done in parallel repositories may
conflict always.  For example, this case would cause the conflict.
The difference is only who resolves the conflict...

In addition, this split is certainly not appropriate for 2.6.30
since it itself has nothing to do with the fix.


Takashi

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

On Tue, Apr 28, 2009 at 12:55:16PM -0400, Jon Smirl wrote:

> I wanted the split files added to 2.6.30 not 2.6.31. If they had gone
> into 2.6.30 you would have patched after the split and everything
> would have propagated without merge conflicts. In general, changes in
> git that cross a file split will cause manual manual merges in
> everything pending past the split.

It wouldn't have helped with the merge issues since bug fixes get merged
up into the 2.6.31 branch very fast.  In any case, we're in the -rc part
of the development cycle - we should be merging targetted bug fixes
only.  Eyebrows would be raised by a diff of that size going in.

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 13:14:09 -0400,
Jon Smirl wrote:
> 
> On Tue, Apr 28, 2009 at 12:55 PM, Jon Smirl <jonsmirl@gmail.com> wrote:
> > On Tue, Apr 28, 2009 at 12:51 PM, Takashi Iwai <tiwai@suse.de> wrote:
> >> At Tue, 28 Apr 2009 11:59:01 -0400,
> >> Jon Smirl wrote:
> >>>
> >>> On Tue, Apr 28, 2009 at 11:46 AM, Takashi Iwai <tiwai@suse.de> wrote:
> >>> > At Tue, 28 Apr 2009 16:23:41 +0100,
> >>> > Mark Brown wrote:
> >>> >>
> >>> >> From: Jon Smirl <jonsmirl@gmail.com>
> >>> >>
> >>> >> Basic split of mpc5200 DMA code out from i2s into a standalone file.
> >>> >>
> >>> >> [Fixed trailing whitespace -- broonie.]
> >>> >>
> >>> >> Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
> >>> >> Acked-by: Grant Likely <grant.likely@secretlab.ca>
> >>> >> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> >>> >
> >>> > This patch conflicts with the last patch I merged to add
> >>> > SNDRV_PCM_INFO_BATCH to some drivers including mpc5200.
> >>> > (It's in fix/pcm-info-batch branch and not merged to */asoc branch
> >>> >  yet.  Feel free to pull to your tree by yourself.)
> >>> > And, the jiffies check shall be anyway fixed in the upstream PCM core
> >>> > side, thus there is no big merit to hurry to merge the unfinished
> >>> > changes now.
> >>> >
> >>> > So, could you back off mpc5200 changes and wait for a while until the
> >>> > issue regarding PCM jiffies-check is resolved?
> >>>
> >>> This is why I was pushing so hard to get the patches that split the
> >>> files in.
> >>
> >> And this is why I am against pulling now :)
> >
> > I wanted the split files added to 2.6.30 not 2.6.31. If they had gone
> > into 2.6.30 you would have patched after the split and everything
> > would have propagated without merge conflicts. In general, changes in
> > git that cross a file split will cause manual manual merges in
> > everything pending past the split.
> 
> BTW, pulling the split patches out of 2.6.31 is just going to move the
> merge mess onto my local patch stack. To eliminate the merge mess we
> need to get the split in and then start patching after the split.

I agree with that.

Also, we need to reconsider which of you change is really needed
since the current problem of the PCM will be solved in the PCM
core side soon later without the change of the driver side.


Takashi

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

On Tue, Apr 28, 2009 at 03:32:18PM -0400, Jon Smirl wrote:

> Now I get to deal with the merge mess in my patch stack. All this did
> was move the problem.

Please send patches against:

git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git mpc5200

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

On Tue, Apr 28, 2009 at 09:37:34PM +0200, Takashi Iwai wrote:

> Also, we need to reconsider which of you change is really needed
> since the current problem of the PCM will be solved in the PCM
> core side soon later without the change of the driver side.

AFAICT all the currently posted patches are needed since they're only
doing the refactoring of the code required to support AC97.  They're
only related to the fixes in that John's board doesn't have I2S but
since the DMA is shared the fixes that John develops while making AC97
work will also fix I2S.

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 3:53 PM, Mark Brown <broonie@sirena.org.uk> wrote:
> On Tue, Apr 28, 2009 at 09:37:34PM +0200, Takashi Iwai wrote:
>
>> Also, we need to reconsider which of you change is really needed
>> since the current problem of the PCM will be solved in the PCM
>> core side soon later without the change of the driver side.
>
> AFAICT all the currently posted patches are needed since they're only
> doing the refactoring of the code required to support AC97.  They're
> only related to the fixes in that John's board doesn't have I2S but
> since the DMA is shared the fixes that John develops while making AC97
> work will also fix I2S.

I purposely sent in refactoring changes that made no functional
changes to the code.

I have been caught in this mess before. This isn't a simple case of
resolving conflicts. What happens is that git isn't smart enough to
track changes across a refactor. That results in big conflicts
covering most of the contents of the files involved. The conflicts in
the refactor then cascade into all of the other patches.

If we put the refactor in front of the fixes git will get everything right.

Why are people going to complain about patches to a driver marked
broken? You can't even compile it without editing the Kconfig.


>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Jon Smirl]

On Tue, Apr 28, 2009 at 3:59 PM, Jon Smirl <jonsmirl@gmail.com> wrote:
> On Tue, Apr 28, 2009 at 3:53 PM, Mark Brown <broonie@sirena.org.uk> wrote:
>> On Tue, Apr 28, 2009 at 09:37:34PM +0200, Takashi Iwai wrote:
>>
>>> Also, we need to reconsider which of you change is really needed
>>> since the current problem of the PCM will be solved in the PCM
>>> core side soon later without the change of the driver side.
>>
>> AFAICT all the currently posted patches are needed since they're only
>> doing the refactoring of the code required to support AC97.  They're
>> only related to the fixes in that John's board doesn't have I2S but
>> since the DMA is shared the fixes that John develops while making AC97
>> work will also fix I2S.
>
> I purposely sent in refactoring changes that made no functional
> changes to the code.
>
> I have been caught in this mess before. This isn't a simple case of
> resolving conflicts. What happens is that git isn't smart enough to
> track changes across a refactor. That results in big conflicts
> covering most of the contents of the files involved. The conflicts in

Thinking about this for a couple minutes, what happens is the three
way merge becomes a four or more way merge depending on how many new
files were created in the the refactor. Git doesn't have an n-way
merge tool so it just kicks the entire files out as conflicts.


> the refactor then cascade into all of the other patches.
>
> If we put the refactor in front of the fixes git will get everything right.
>
> Why are people going to complain about patches to a driver marked
> broken? You can't even compile it without editing the Kconfig.
>
>
>>
>
>
>
> --
> Jon Smirl
> jonsmirl@gmail.com
>



-- 
Jon Smirl
jonsmirl@gmail.com

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Mark Brown]

On Tue, Apr 28, 2009 at 03:59:52PM -0400, Jon Smirl wrote:

> Why are people going to complain about patches to a driver marked
> broken? You can't even compile it without editing the Kconfig.

There is no problem with doing whatever to the driver for 2.6.31 but we
do need to make sure we stay in sync with the other API work that's
going on in 2.6.31.

For 2.6.30 we're outside the merge window so we need to send targetted
fixes for bugs - if we sent your current patches upstream now the
pushback would be that such large refactorings should be merged as part
of the merge window and the bug fixes should be extracted from your
refactoring and merged now.  This is the kernel wide development policy,
not an ALSA or ASoC specific one.

Ideally we wouldn't have to mark the driver broken but would instead be
merging bug fixes to it into 2.6.30 - marking it as broken was an
alternative to that.

For the merges the simplest thing would be to fix the current code
before doing the refactoring, though from the sounds of it the final
fixes required for 2.6.30 will be very small in the driver so I'd expect
this can be resolved fairly simply.  Changes made in the 2.6.30 branch
get merged into the 2.6.31 branch very quickly so any divergence between
the two branches shouldn't exist for long.

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 20:25:45 +0100,
Mark Brown wrote:
> 
> On Tue, Apr 28, 2009 at 05:46:11PM +0200, Takashi Iwai wrote:
> 
> > And, the jiffies check shall be anyway fixed in the upstream PCM core
> > side, thus there is no big merit to hurry to merge the unfinished
> > changes now.
> 
> The refactoring is a largely orthogonal issue to the jiffies check - the
> refactoring is needed to split the DMA code out so that it can be shared
> with the AC97 driver John is writing.
> 
> > So, could you back off mpc5200 changes and wait for a while until the
> > issue regarding PCM jiffies-check is resolved?
> 
> Done:

Thanks, pulled now.


Takashi


> The following changes since commit 8a1f936acdfd53cb0a981f3f80483863dcd84fa9:
>   Peter Ujfalusi (1):
>         ASoC: TWL4030: Add 4 channel TDM support
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31
> 
> Daniel Mack (3):
>       ASoC: add SOC_DOUBLE_EXT macro
>       ASoC: cs4270: fix Master Capture Switch polarity
>       ASoC: cs4270: add Master Playback Switch
> 
> Jon Smirl (1):
>       ASoC: Set the MPC5200 i2s driver to BROKEN status.
> 
> Jonathan Cameron (1):
>       ASoC WM8940 Driver
> 
> Joonyoung Shim (1):
>       ASoC: TWL4030: Fix gain control for earpiece amplifier
> 
> Mark Brown (10):
>       ASoC: Fix S3C64xx IIS device registration and support both ports
>       ASoC: S3C2412: Failing to get the I2S clock is an error
>       ASoC: Enforce symmetric rates for S3C64xx I2S interface
>       ASoC: Include WM8350 register definitions in CODEC header
>       ASoC: s3c-i2s-v2 diagnostic improvements
>       ASoC: Use our registration function for S3C64xx
>       ASoC: Fix logic in WM8350 master clocking check
>       Merge branch 'for-2.6.30' into for-2.6.31
>       ASoC: Staticise TLV values in WM8940
>       Merge branch 'for-2.6.30' into for-2.6.31
> 
> Peter Ujfalusi (1):
>       ASoC: Beagle: Add support for 4 channel
> 
>  include/sound/soc.h             |    8 +
>  sound/soc/codecs/Kconfig        |    4 +
>  sound/soc/codecs/Makefile       |    2 +
>  sound/soc/codecs/cs4270.c       |   44 ++-
>  sound/soc/codecs/twl4030.c      |    8 +-
>  sound/soc/codecs/wm8350.c       |    2 +-
>  sound/soc/codecs/wm8350.h       |    1 +
>  sound/soc/codecs/wm8940.c       |  955 +++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/wm8940.h       |  104 +++++
>  sound/soc/fsl/Kconfig           |    2 +-
>  sound/soc/omap/omap3beagle.c    |   26 +-
>  sound/soc/s3c24xx/s3c2412-i2s.c |    2 +-
>  sound/soc/s3c24xx/s3c64xx-i2s.c |  148 ++++--
>  sound/soc/s3c24xx/s3c64xx-i2s.h |    2 +-
>  14 files changed, 1241 insertions(+), 67 deletions(-)
>  create mode 100644 sound/soc/codecs/wm8940.c
>  create mode 100644 sound/soc/codecs/wm8940.h
> 

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 20:53:53 +0100,
Mark Brown wrote:
> 
> On Tue, Apr 28, 2009 at 09:37:34PM +0200, Takashi Iwai wrote:
> 
> > Also, we need to reconsider which of you change is really needed
> > since the current problem of the PCM will be solved in the PCM
> > core side soon later without the change of the driver side.
> 
> AFAICT all the currently posted patches are needed since they're only
> doing the refactoring of the code required to support AC97.  They're
> only related to the fixes in that John's board doesn't have I2S but
> since the DMA is shared the fixes that John develops while making AC97
> work will also fix I2S.

OK.  I thought that he rewrote the pointer callback to return
jiffies-based values and it could be superfluous.  Other than that,
it should be all fine.

Basically I don't mind to merge Jon's patch series now for 2.6.31 once
after the current merge conflicts are resolved.
There won't be many changes on 2.6.30 for that driver any more.
At most, a patch will be just to remove BROKEN from kconfig again
(hopefully in time).


Takashi

Re: [PATCH 16/17] ASoC: Basic split of mpc5200 DMA code out from mpc5200_psc_i2s

[Takashi Iwai]

At Tue, 28 Apr 2009 15:59:52 -0400,
Jon Smirl wrote:
> 
> On Tue, Apr 28, 2009 at 3:53 PM, Mark Brown <broonie@sirena.org.uk> wrote:
> > On Tue, Apr 28, 2009 at 09:37:34PM +0200, Takashi Iwai wrote:
> >
> >> Also, we need to reconsider which of you change is really needed
> >> since the current problem of the PCM will be solved in the PCM
> >> core side soon later without the change of the driver side.
> >
> > AFAICT all the currently posted patches are needed since they're only
> > doing the refactoring of the code required to support AC97.  They're
> > only related to the fixes in that John's board doesn't have I2S but
> > since the DMA is shared the fixes that John develops while making AC97
> > work will also fix I2S.
> 
> I purposely sent in refactoring changes that made no functional
> changes to the code.
> 
> I have been caught in this mess before. This isn't a simple case of
> resolving conflicts.

Why do you believe that this case will be like that, too?

> What happens is that git isn't smart enough to
> track changes across a refactor. That results in big conflicts
> covering most of the contents of the files involved. The conflicts in
> the refactor then cascade into all of the other patches.
> 
> If we put the refactor in front of the fixes git will get everything right.

Well, you can do it by yourself on your local tree, too.  topgit or
other tools can serve well for such use cases.
Then either let Mark or me pull your tree, or straighten the patches
to submit.

> Why are people going to complain about patches to a driver marked
> broken? You can't even compile it without editing the Kconfig.

Because we broke it.


Takashi

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 8560b9321f9050968f393ce1ec67e47c1a0bd5cf:
  Takashi Iwai (1):
        Merge branch 'fix/asoc' into topic/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Jinyoung Park (1):
      ASoC: Fix errors in WM8990

Jonathan Cameron (1):
      ASoC: IMote2 ASoC Support

Lopez Cruz, Misael (2):
      ASoC: TWL4030: Add VDL analog bypass
      ASoC: TWL4030: Add voice digital loopback: sidetone

Mark Brown (15):
      ASoC: Fix error message formatting in s3c64xx-i2s driver
      ASoC: Check for supported CPUs when building s3c-i2s-v2
      ASoC: Make S3C64xx clock export function to return struct clk
      ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bit
      ASoC: Fix boot warnings from S3C IISv2
      ASoC: Allow use of resource from the platform device for S3C IISv2
      ASoC: Display the clock rate used as the basis for rate calculation
      ASoC: Display S3C IISv2 mode and MS errors by default
      ASoC: Staticise txctrl and rxctrl for S3C IISv2
      ASoC: Use platform device resource for S3C64xx IISv2
      ASoC: Use a shared define for AC97 CODEC data formats
      ASoC: Remove unused DAI format defines
      Merge commit 'takashi/topic/asoc' into for-2.6.31
      Merge branch 'for-2.6.30' into for-2.6.31
      ASoC: Remove redundant codec pointer from DAIs

Peter Ujfalusi (1):
      ASoC: TWL4030: Add VIBRA output

 include/sound/soc-dai.h         |   28 ++-------
 sound/soc/codecs/ac97.c         |    4 +-
 sound/soc/codecs/ad1980.c       |    4 +-
 sound/soc/codecs/twl4030.c      |  121 +++++++++++++++++++++++++++++++++++++--
 sound/soc/codecs/wm8990.c       |   40 +++++++-------
 sound/soc/codecs/wm9705.c       |    4 +-
 sound/soc/codecs/wm9712.c       |    6 +-
 sound/soc/codecs/wm9713.c       |    6 +-
 sound/soc/pxa/Kconfig           |    9 +++
 sound/soc/pxa/Makefile          |    2 +
 sound/soc/pxa/imote2.c          |  114 ++++++++++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c-i2s-v2.c  |   87 ++++++++++++++++++++++++----
 sound/soc/s3c24xx/s3c64xx-i2s.c |   15 ++---
 sound/soc/s3c24xx/s3c64xx-i2s.h |    4 +-
 14 files changed, 362 insertions(+), 82 deletions(-)
 create mode 100644 sound/soc/pxa/imote2.c

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Tue, 5 May 2009 11:01:23 +0100,
Mark Brown wrote:
> 
> The following changes since commit 8560b9321f9050968f393ce1ec67e47c1a0bd5cf:
>   Takashi Iwai (1):
>         Merge branch 'fix/asoc' into topic/asoc
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled this one, too.  Thanks.


Takashi

> Jinyoung Park (1):
>       ASoC: Fix errors in WM8990
> 
> Jonathan Cameron (1):
>       ASoC: IMote2 ASoC Support
> 
> Lopez Cruz, Misael (2):
>       ASoC: TWL4030: Add VDL analog bypass
>       ASoC: TWL4030: Add voice digital loopback: sidetone
> 
> Mark Brown (15):
>       ASoC: Fix error message formatting in s3c64xx-i2s driver
>       ASoC: Check for supported CPUs when building s3c-i2s-v2
>       ASoC: Make S3C64xx clock export function to return struct clk
>       ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bit
>       ASoC: Fix boot warnings from S3C IISv2
>       ASoC: Allow use of resource from the platform device for S3C IISv2
>       ASoC: Display the clock rate used as the basis for rate calculation
>       ASoC: Display S3C IISv2 mode and MS errors by default
>       ASoC: Staticise txctrl and rxctrl for S3C IISv2
>       ASoC: Use platform device resource for S3C64xx IISv2
>       ASoC: Use a shared define for AC97 CODEC data formats
>       ASoC: Remove unused DAI format defines
>       Merge commit 'takashi/topic/asoc' into for-2.6.31
>       Merge branch 'for-2.6.30' into for-2.6.31
>       ASoC: Remove redundant codec pointer from DAIs
> 
> Peter Ujfalusi (1):
>       ASoC: TWL4030: Add VIBRA output
> 
>  include/sound/soc-dai.h         |   28 ++-------
>  sound/soc/codecs/ac97.c         |    4 +-
>  sound/soc/codecs/ad1980.c       |    4 +-
>  sound/soc/codecs/twl4030.c      |  121 +++++++++++++++++++++++++++++++++++++--
>  sound/soc/codecs/wm8990.c       |   40 +++++++-------
>  sound/soc/codecs/wm9705.c       |    4 +-
>  sound/soc/codecs/wm9712.c       |    6 +-
>  sound/soc/codecs/wm9713.c       |    6 +-
>  sound/soc/pxa/Kconfig           |    9 +++
>  sound/soc/pxa/Makefile          |    2 +
>  sound/soc/pxa/imote2.c          |  114 ++++++++++++++++++++++++++++++++++++
>  sound/soc/s3c24xx/s3c-i2s-v2.c  |   87 ++++++++++++++++++++++++----
>  sound/soc/s3c24xx/s3c64xx-i2s.c |   15 ++---
>  sound/soc/s3c24xx/s3c64xx-i2s.h |    4 +-
>  14 files changed, 362 insertions(+), 82 deletions(-)
>  create mode 100644 sound/soc/pxa/imote2.c
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit bbd993077d788589a86a718ba7a7895ba5e71a17:
  Mark Brown (1):
        ASoC: Remove redundant codec pointer from DAIs

are available in the git repository at:

  ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

(git:// doesn't seem to be updating so...).

Daniel Mack (2):
      ASoC: cs4270: introduce CS4270_I2C_INCR
      ASoC: cs4270: add power management support

Mark Brown (2):
      ASoC: Add headers to match patterns in MAINTAINERS
      ASoC: Fix file patterns for PXA sound drivers

Peter Ujfalusi (1):
      ASoC: TWL4030: Fix typo in twl4030_codec_mute function

 MAINTAINERS                |    4 ++-
 sound/soc/codecs/cs4270.c  |   61 ++++++++++++++++++++++++++++++++++++++++++-
 sound/soc/codecs/twl4030.c |    2 +-
 3 files changed, 63 insertions(+), 4 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Fri, 8 May 2009 13:57:11 +0100,
Mark Brown wrote:
> 
> The following changes since commit bbd993077d788589a86a718ba7a7895ba5e71a17:
>   Mark Brown (1):
>         ASoC: Remove redundant codec pointer from DAIs
> 
> are available in the git repository at:
> 
>   ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31
> 
> (git:// doesn't seem to be updating so...).

It appears now.  Thanks, pulled now.


Takashi

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit bec4c99e8637b5b8bd4b0513eacb51da25885e3b:
  Mark Brown (1):
        ASoC: Fix file patterns for PXA sound drivers

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Joonyoung Shim (1):
      ASoC: TWL4030: change DAPM for analog microphone selection

Mark Brown (3):
      ASoC: Fix up CODEC DAI formats for big endian CPUs
      ASoC: Enforce symmetric rates for PXA2xx I2S
      Merge branch 'for-2.6.30' into for-2.6.31

Mike Rapoport (3):
      ASoC: soc-core: fix crash when removing not instantiated card
      pxa2xx-ac97: fix reset gpio mode setting
      ASoC: em-x270: make the driver support also eXeda and CM-X300 machines

Roel Kluin (1):
      ASoC: remove driver_data direct access of struct device

Sergey Lapin (1):
      ASoC: AFEB9260 driver

 sound/arm/pxa2xx-ac97-lib.c        |    2 +-
 sound/soc/atmel/Kconfig            |    8 ++
 sound/soc/atmel/Makefile           |    1 +
 sound/soc/atmel/snd-soc-afeb9260.c |  203 ++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/twl4030.c         |   52 ++++------
 sound/soc/codecs/wm8400.c          |    4 +-
 sound/soc/codecs/wm8731.c          |    4 +-
 sound/soc/codecs/wm8753.c          |    4 +-
 sound/soc/pxa/Kconfig              |    4 +-
 sound/soc/pxa/em-x270.c            |    9 +-
 sound/soc/pxa/pxa2xx-i2s.c         |    1 +
 sound/soc/soc-core.c               |   43 ++++++++
 12 files changed, 289 insertions(+), 46 deletions(-)
 create mode 100644 sound/soc/atmel/snd-soc-afeb9260.c

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit dd4a416442a2fb4bd761163277d37ab6ce75ee4e:
  Takashi Iwai (1):
        Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Anuj Aggarwal (1):
      ASoC: Added OMAP3 EVM support in ASoC.

David Brownell (3):
      ASoC: davinci-pcm buildfixes
      ASoC: DaVinci I2S updates
      ASoC: DaVinci EVM board support buildfixes

Jon Smirl (1):
      ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format

Karl Beldan (5):
      ASoC: pxa2xx-i2s: Proper initialization
      ASoC: pxa2xx-i2s: Proper hw initialization
      ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
      ASoC: pxa2xx-i2s: Fix inappropriate release of i2s clock
      ASoC: pxa2xx-i2s: Fix suspend/resume

Marek Vasut (1):
      ASoC: Support AC97 link off by default on WM9712

Mark Brown (3):
      ASoC: Point at kernel.org git
      Merge branch 'pxa2xx-i2s' into for-2.6.31
      Merge branch 'for-2.6.30' into for-2.6.31

 MAINTAINERS                              |    2 +-
 arch/arm/mach-davinci/include/mach/asp.h |   25 +++++
 include/sound/soc-dai.h                  |    3 +-
 sound/soc/codecs/wm9712.c                |    2 +
 sound/soc/davinci/Kconfig                |    7 +-
 sound/soc/davinci/davinci-evm.c          |   63 +++++++++++--
 sound/soc/davinci/davinci-i2s.c          |   26 +++++-
 sound/soc/davinci/davinci-pcm.c          |   71 +++++++++------
 sound/soc/omap/Kconfig                   |    8 ++
 sound/soc/omap/Makefile                  |    2 +
 sound/soc/omap/omap3evm.c                |  147 ++++++++++++++++++++++++++++++
 sound/soc/pxa/pxa2xx-i2s.c               |   38 +++++---
 12 files changed, 330 insertions(+), 64 deletions(-)
 create mode 100644 arch/arm/mach-davinci/include/mach/asp.h
 create mode 100644 sound/soc/omap/omap3evm.c

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Fri, 15 May 2009 09:26:31 +0100,
Mark Brown wrote:
> 
> The following changes since commit dd4a416442a2fb4bd761163277d37ab6ce75ee4e:
>   Takashi Iwai (1):
>         Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks.


Takashi


> Anuj Aggarwal (1):
>       ASoC: Added OMAP3 EVM support in ASoC.
> 
> David Brownell (3):
>       ASoC: davinci-pcm buildfixes
>       ASoC: DaVinci I2S updates
>       ASoC: DaVinci EVM board support buildfixes
> 
> Jon Smirl (1):
>       ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format
> 
> Karl Beldan (5):
>       ASoC: pxa2xx-i2s: Proper initialization
>       ASoC: pxa2xx-i2s: Proper hw initialization
>       ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately
>       ASoC: pxa2xx-i2s: Fix inappropriate release of i2s clock
>       ASoC: pxa2xx-i2s: Fix suspend/resume
> 
> Marek Vasut (1):
>       ASoC: Support AC97 link off by default on WM9712
> 
> Mark Brown (3):
>       ASoC: Point at kernel.org git
>       Merge branch 'pxa2xx-i2s' into for-2.6.31
>       Merge branch 'for-2.6.30' into for-2.6.31
> 
>  MAINTAINERS                              |    2 +-
>  arch/arm/mach-davinci/include/mach/asp.h |   25 +++++
>  include/sound/soc-dai.h                  |    3 +-
>  sound/soc/codecs/wm9712.c                |    2 +
>  sound/soc/davinci/Kconfig                |    7 +-
>  sound/soc/davinci/davinci-evm.c          |   63 +++++++++++--
>  sound/soc/davinci/davinci-i2s.c          |   26 +++++-
>  sound/soc/davinci/davinci-pcm.c          |   71 +++++++++------
>  sound/soc/omap/Kconfig                   |    8 ++
>  sound/soc/omap/Makefile                  |    2 +
>  sound/soc/omap/omap3evm.c                |  147 ++++++++++++++++++++++++++++++
>  sound/soc/pxa/pxa2xx-i2s.c               |   38 +++++---
>  12 files changed, 330 insertions(+), 64 deletions(-)
>  create mode 100644 arch/arm/mach-davinci/include/mach/asp.h
>  create mode 100644 sound/soc/omap/omap3evm.c
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 5b740ea975c4ce3da12ac21b56f9e43354ca4327:
  Julia Lawall (1):
        sound: use dev_set_drvdata

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Mark Brown (4):
      ASoC: Split DAPM power checks from sequencing of power changes
      ASoC: Make DAPM sysfs entries non-optional
      ASoC: Integrate bias management with DAPM power management
      ASoC: Add debug trace for bias level transitions

Misael Lopez Cruz (1):
      ASoC: TWL4030: Enable/disable voice digital filters

 include/sound/soc-dapm.h   |    5 +-
 include/sound/soc.h        |    3 +
 sound/soc/codecs/twl4030.c |   37 +++++++++
 sound/soc/codecs/twl4030.h |    2 +-
 sound/soc/soc-core.c       |   61 +++------------
 sound/soc/soc-dapm.c       |  187 ++++++++++++++++++++++++++++++--------------
 6 files changed, 183 insertions(+), 112 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Mon, 18 May 2009 16:04:53 +0100,
Mark Brown wrote:
> 
> The following changes since commit 5b740ea975c4ce3da12ac21b56f9e43354ca4327:
>   Julia Lawall (1):
>         sound: use dev_set_drvdata
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks, pulled now.


Takashi

> Mark Brown (4):
>       ASoC: Split DAPM power checks from sequencing of power changes
>       ASoC: Make DAPM sysfs entries non-optional
>       ASoC: Integrate bias management with DAPM power management
>       ASoC: Add debug trace for bias level transitions
> 
> Misael Lopez Cruz (1):
>       ASoC: TWL4030: Enable/disable voice digital filters
> 
>  include/sound/soc-dapm.h   |    5 +-
>  include/sound/soc.h        |    3 +
>  sound/soc/codecs/twl4030.c |   37 +++++++++
>  sound/soc/codecs/twl4030.h |    2 +-
>  sound/soc/soc-core.c       |   61 +++------------
>  sound/soc/soc-dapm.c       |  187 ++++++++++++++++++++++++++++++--------------
>  6 files changed, 183 insertions(+), 112 deletions(-)
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit f83fba8baab9e95fff0fe2be0e1e32a1650bdd7f:
  Mark Brown (1):
        ASoC: Add debug trace for bias level transitions

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Atsushi Nemoto (1):
      ASoC: Add TXx9 AC link controller driver (v3)

Lopez Cruz, Misael (2):
      ASoC: TWL4030: Add control for selecting codec operation mode
      ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3

Peter Ujfalusi (1):
      ASoC: TWL4030: Fix Analog capture path for AUXR

 sound/soc/Kconfig                 |    1 +
 sound/soc/Makefile                |    1 +
 sound/soc/codecs/twl4030.c        |   49 ++++-
 sound/soc/omap/sdp3430.c          |   87 +++++++-
 sound/soc/txx9/Kconfig            |   29 +++
 sound/soc/txx9/Makefile           |   11 +
 sound/soc/txx9/txx9aclc-ac97.c    |  255 ++++++++++++++++++++++
 sound/soc/txx9/txx9aclc-generic.c |   98 +++++++++
 sound/soc/txx9/txx9aclc.c         |  430 +++++++++++++++++++++++++++++++++++++
 sound/soc/txx9/txx9aclc.h         |   83 +++++++
 10 files changed, 1033 insertions(+), 11 deletions(-)
 create mode 100644 sound/soc/txx9/Kconfig
 create mode 100644 sound/soc/txx9/Makefile
 create mode 100644 sound/soc/txx9/txx9aclc-ac97.c
 create mode 100644 sound/soc/txx9/txx9aclc-generic.c
 create mode 100644 sound/soc/txx9/txx9aclc.c
 create mode 100644 sound/soc/txx9/txx9aclc.h

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Tue, 19 May 2009 20:27:32 +0100,
Mark Brown wrote:
> 
> The following changes since commit f83fba8baab9e95fff0fe2be0e1e32a1650bdd7f:
>   Mark Brown (1):
>         ASoC: Add debug trace for bias level transitions
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks!


Takashi


> Atsushi Nemoto (1):
>       ASoC: Add TXx9 AC link controller driver (v3)
> 
> Lopez Cruz, Misael (2):
>       ASoC: TWL4030: Add control for selecting codec operation mode
>       ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3
> 
> Peter Ujfalusi (1):
>       ASoC: TWL4030: Fix Analog capture path for AUXR
> 
>  sound/soc/Kconfig                 |    1 +
>  sound/soc/Makefile                |    1 +
>  sound/soc/codecs/twl4030.c        |   49 ++++-
>  sound/soc/omap/sdp3430.c          |   87 +++++++-
>  sound/soc/txx9/Kconfig            |   29 +++
>  sound/soc/txx9/Makefile           |   11 +
>  sound/soc/txx9/txx9aclc-ac97.c    |  255 ++++++++++++++++++++++
>  sound/soc/txx9/txx9aclc-generic.c |   98 +++++++++
>  sound/soc/txx9/txx9aclc.c         |  430 +++++++++++++++++++++++++++++++++++++
>  sound/soc/txx9/txx9aclc.h         |   83 +++++++
>  10 files changed, 1033 insertions(+), 11 deletions(-)
>  create mode 100644 sound/soc/txx9/Kconfig
>  create mode 100644 sound/soc/txx9/Makefile
>  create mode 100644 sound/soc/txx9/txx9aclc-ac97.c
>  create mode 100644 sound/soc/txx9/txx9aclc-generic.c
>  create mode 100644 sound/soc/txx9/txx9aclc.c
>  create mode 100644 sound/soc/txx9/txx9aclc.h
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit e24805dd85283ac0912b9c400768a4d171b400ff:
  Atsushi Nemoto (1):
        ASoC: Add TXx9 AC link controller driver (v3)

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Mark Brown (2):
      ASoC: Make snd_soc_read() and snd_soc_write() functions
      ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver

Peter Ujfalusi (5):
      ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs
      ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event
      ASoC: TWL4030: Add support for platform dependent configuration
      ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal
      ASoC: TWL4030: Differentiate the playback streams

 include/sound/soc.h        |   17 +-
 include/sound/wm9081.h     |   25 +
 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/twl4030.c |  285 ++++++---
 sound/soc/codecs/twl4030.h |    5 +
 sound/soc/codecs/wm9081.c  | 1532 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm9081.h  |  787 +++++++++++++++++++++++
 sound/soc/omap/sdp3430.c   |    7 +
 9 files changed, 2559 insertions(+), 105 deletions(-)
 create mode 100644 include/sound/wm9081.h
 create mode 100644 sound/soc/codecs/wm9081.c
 create mode 100644 sound/soc/codecs/wm9081.h

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Fri, 22 May 2009 21:21:24 +0100,
Mark Brown wrote:
> 
> The following changes since commit e24805dd85283ac0912b9c400768a4d171b400ff:
>   Atsushi Nemoto (1):
>         ASoC: Add TXx9 AC link controller driver (v3)
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks, pulled now.


Takashi

> Mark Brown (2):
>       ASoC: Make snd_soc_read() and snd_soc_write() functions
>       ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver
> 
> Peter Ujfalusi (5):
>       ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs
>       ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event
>       ASoC: TWL4030: Add support for platform dependent configuration
>       ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal
>       ASoC: TWL4030: Differentiate the playback streams
> 
>  include/sound/soc.h        |   17 +-
>  include/sound/wm9081.h     |   25 +
>  sound/soc/codecs/Kconfig   |    4 +
>  sound/soc/codecs/Makefile  |    2 +
>  sound/soc/codecs/twl4030.c |  285 ++++++---
>  sound/soc/codecs/twl4030.h |    5 +
>  sound/soc/codecs/wm9081.c  | 1532 ++++++++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/wm9081.h  |  787 +++++++++++++++++++++++
>  sound/soc/omap/sdp3430.c   |    7 +
>  9 files changed, 2559 insertions(+), 105 deletions(-)
>  create mode 100644 include/sound/wm9081.h
>  create mode 100644 sound/soc/codecs/wm9081.c
>  create mode 100644 sound/soc/codecs/wm9081.h
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 86ed3669f068b514ab85ffd548456a342b3fb8d3:
  Mark Brown (1):
        ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Jon Smirl (4):
      ASoC: Codec for STAC9766 used on the Efika
      ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2s
      ASoC: Rename the PSC functions to DMA
      ASoC: Add a few more mpc5200 PSC defines

Mark Brown (2):
      ASoC: Fix WM9081 PowerPC compiler issues
      ASoC: Fix minor issues in STAC9766 driver

Peter Ujfalusi (3):
      ASoC: TWL4030: Handsfree pop removal redesign
      ASoC: TWL4030: Add shadow register
      ASoC: TWL4030: HandsfreeL/R mute DAPM switch

 arch/powerpc/include/asm/mpc52xx_psc.h |   11 +
 sound/soc/codecs/Kconfig               |    4 +
 sound/soc/codecs/Makefile              |    2 +
 sound/soc/codecs/stac9766.c            |  463 +++++++++++++++++++++++
 sound/soc/codecs/stac9766.h            |   21 +
 sound/soc/codecs/twl4030.c             |   99 ++++-
 sound/soc/codecs/twl4030.h             |    7 +-
 sound/soc/codecs/wm9081.c              |    6 +-
 sound/soc/fsl/Kconfig                  |    4 +
 sound/soc/fsl/Makefile                 |    2 +
 sound/soc/fsl/mpc5200_dma.c            |  458 ++++++++++++++++++++++
 sound/soc/fsl/mpc5200_dma.h            |   81 ++++
 sound/soc/fsl/mpc5200_psc_i2s.c        |  645 ++++----------------------------
 13 files changed, 1217 insertions(+), 586 deletions(-)
 create mode 100644 sound/soc/codecs/stac9766.c
 create mode 100644 sound/soc/codecs/stac9766.h
 create mode 100644 sound/soc/fsl/mpc5200_dma.c
 create mode 100644 sound/soc/fsl/mpc5200_dma.h

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Mon, 25 May 2009 13:07:57 +0100,
Mark Brown wrote:
> 
> The following changes since commit 86ed3669f068b514ab85ffd548456a342b3fb8d3:
>   Mark Brown (1):
>         ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks.


Takashi

> Jon Smirl (4):
>       ASoC: Codec for STAC9766 used on the Efika
>       ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2s
>       ASoC: Rename the PSC functions to DMA
>       ASoC: Add a few more mpc5200 PSC defines
> 
> Mark Brown (2):
>       ASoC: Fix WM9081 PowerPC compiler issues
>       ASoC: Fix minor issues in STAC9766 driver
> 
> Peter Ujfalusi (3):
>       ASoC: TWL4030: Handsfree pop removal redesign
>       ASoC: TWL4030: Add shadow register
>       ASoC: TWL4030: HandsfreeL/R mute DAPM switch
> 
>  arch/powerpc/include/asm/mpc52xx_psc.h |   11 +
>  sound/soc/codecs/Kconfig               |    4 +
>  sound/soc/codecs/Makefile              |    2 +
>  sound/soc/codecs/stac9766.c            |  463 +++++++++++++++++++++++
>  sound/soc/codecs/stac9766.h            |   21 +
>  sound/soc/codecs/twl4030.c             |   99 ++++-
>  sound/soc/codecs/twl4030.h             |    7 +-
>  sound/soc/codecs/wm9081.c              |    6 +-
>  sound/soc/fsl/Kconfig                  |    4 +
>  sound/soc/fsl/Makefile                 |    2 +
>  sound/soc/fsl/mpc5200_dma.c            |  458 ++++++++++++++++++++++
>  sound/soc/fsl/mpc5200_dma.h            |   81 ++++
>  sound/soc/fsl/mpc5200_psc_i2s.c        |  645 ++++----------------------------
>  13 files changed, 1217 insertions(+), 586 deletions(-)
>  create mode 100644 sound/soc/codecs/stac9766.c
>  create mode 100644 sound/soc/codecs/stac9766.h
>  create mode 100644 sound/soc/fsl/mpc5200_dma.c
>  create mode 100644 sound/soc/fsl/mpc5200_dma.h
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 0f89bdcac61536c5cb2a095a514657019573afb4:
  Peter Ujfalusi (1):
        ASoC: TWL4030: HandsfreeL/R mute DAPM switch

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Jon Smirl (4):
      ASoC: Main rewite of the mpc5200 audio DMA code
      ASoC: AC97 driver for mpc5200
      ASoC: Support for AC97 on Phytec pmc030 base board.
      ASoC: Fabric bindings for STAC9766 on the Efika

Mark Brown (2):
      ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
      ASoC: Switch FSL SSI DAI over to symmetric_rates

 sound/soc/fsl/Kconfig               |   28 +++-
 sound/soc/fsl/Makefile              |    5 +
 sound/soc/fsl/efika-audio-fabric.c  |   90 +++++++
 sound/soc/fsl/fsl_ssi.c             |    9 +-
 sound/soc/fsl/mpc5200_dma.c         |  442 ++++++++++++++++++++++-------------
 sound/soc/fsl/mpc5200_dma.h         |   33 ++--
 sound/soc/fsl/mpc5200_psc_ac97.c    |  331 ++++++++++++++++++++++++++
 sound/soc/fsl/mpc5200_psc_ac97.h    |   15 ++
 sound/soc/fsl/mpc5200_psc_i2s.c     |  247 ++++----------------
 sound/soc/fsl/mpc5200_psc_i2s.h     |   12 +
 sound/soc/fsl/pcm030-audio-fabric.c |   90 +++++++
 11 files changed, 905 insertions(+), 397 deletions(-)
 create mode 100644 sound/soc/fsl/efika-audio-fabric.c
 create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c
 create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h
 create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h
 create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Wed, 27 May 2009 11:48:49 +0100,
Mark Brown wrote:
> 
> The following changes since commit 0f89bdcac61536c5cb2a095a514657019573afb4:
>   Peter Ujfalusi (1):
>         ASoC: TWL4030: HandsfreeL/R mute DAPM switch
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks, pulled now.


Takashi

> 
> Jon Smirl (4):
>       ASoC: Main rewite of the mpc5200 audio DMA code
>       ASoC: AC97 driver for mpc5200
>       ASoC: Support for AC97 on Phytec pmc030 base board.
>       ASoC: Fabric bindings for STAC9766 on the Efika
> 
> Mark Brown (2):
>       ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
>       ASoC: Switch FSL SSI DAI over to symmetric_rates
> 
>  sound/soc/fsl/Kconfig               |   28 +++-
>  sound/soc/fsl/Makefile              |    5 +
>  sound/soc/fsl/efika-audio-fabric.c  |   90 +++++++
>  sound/soc/fsl/fsl_ssi.c             |    9 +-
>  sound/soc/fsl/mpc5200_dma.c         |  442 ++++++++++++++++++++++-------------
>  sound/soc/fsl/mpc5200_dma.h         |   33 ++--
>  sound/soc/fsl/mpc5200_psc_ac97.c    |  331 ++++++++++++++++++++++++++
>  sound/soc/fsl/mpc5200_psc_ac97.h    |   15 ++
>  sound/soc/fsl/mpc5200_psc_i2s.c     |  247 ++++----------------
>  sound/soc/fsl/mpc5200_psc_i2s.h     |   12 +
>  sound/soc/fsl/pcm030-audio-fabric.c |   90 +++++++
>  11 files changed, 905 insertions(+), 397 deletions(-)
>  create mode 100644 sound/soc/fsl/efika-audio-fabric.c
>  create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c
>  create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h
>  create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h
>  create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 08d15f034e94251606479d7ca9070994c2e2fcf0:
  Mark Brown (1):
        ASoC: Switch FSL SSI DAI over to symmetric_rates

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Chaithrika U S (1):
      ASoC: Add dummy S/PDIF codec support

Jon Smirl (1):
      ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()

Mark Brown (1):
      ASoC: Initialise dev for the dummy S/PDIF DAI

Roel Kluin (1):
      ASoC: correct print specifiers for unsigneds

 sound/soc/atmel/playpaq_wm8510.c     |    2 +-
 sound/soc/codecs/Kconfig             |    4 ++
 sound/soc/codecs/Makefile            |    2 +
 sound/soc/codecs/spdif_transciever.c |   71 ++++++++++++++++++++++++++++++++++
 sound/soc/codecs/spdif_transciever.h |   17 ++++++++
 sound/soc/codecs/tlv320aic23.c       |    4 +-
 sound/soc/codecs/uda134x.c           |    4 +-
 sound/soc/codecs/wm8350.c            |    2 +-
 sound/soc/codecs/wm8400.c            |    4 +-
 sound/soc/codecs/wm8510.c            |    2 +-
 sound/soc/codecs/wm8580.c            |    4 +-
 sound/soc/codecs/wm8753.c            |    2 +-
 sound/soc/codecs/wm8900.c            |    6 +-
 sound/soc/codecs/wm8990.c            |    2 +-
 sound/soc/codecs/wm9713.c            |    2 +-
 sound/soc/fsl/mpc5200_psc_ac97.c     |   30 +++++++--------
 sound/soc/pxa/pxa-ssp.c              |    4 +-
 sound/soc/s3c24xx/s3c-i2s-v2.c       |    4 +-
 sound/soc/sh/ssi.c                   |    2 +-
 19 files changed, 130 insertions(+), 38 deletions(-)
 create mode 100644 sound/soc/codecs/spdif_transciever.c
 create mode 100644 sound/soc/codecs/spdif_transciever.h

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Thu, 28 May 2009 19:04:30 +0100,
Mark Brown wrote:
> 
> The following changes since commit 08d15f034e94251606479d7ca9070994c2e2fcf0:
>   Mark Brown (1):
>         ASoC: Switch FSL SSI DAI over to symmetric_rates
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks, pulled now.


Takashi

> Chaithrika U S (1):
>       ASoC: Add dummy S/PDIF codec support
> 
> Jon Smirl (1):
>       ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
> 
> Mark Brown (1):
>       ASoC: Initialise dev for the dummy S/PDIF DAI
> 
> Roel Kluin (1):
>       ASoC: correct print specifiers for unsigneds
> 
>  sound/soc/atmel/playpaq_wm8510.c     |    2 +-
>  sound/soc/codecs/Kconfig             |    4 ++
>  sound/soc/codecs/Makefile            |    2 +
>  sound/soc/codecs/spdif_transciever.c |   71 ++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/spdif_transciever.h |   17 ++++++++
>  sound/soc/codecs/tlv320aic23.c       |    4 +-
>  sound/soc/codecs/uda134x.c           |    4 +-
>  sound/soc/codecs/wm8350.c            |    2 +-
>  sound/soc/codecs/wm8400.c            |    4 +-
>  sound/soc/codecs/wm8510.c            |    2 +-
>  sound/soc/codecs/wm8580.c            |    4 +-
>  sound/soc/codecs/wm8753.c            |    2 +-
>  sound/soc/codecs/wm8900.c            |    6 +-
>  sound/soc/codecs/wm8990.c            |    2 +-
>  sound/soc/codecs/wm9713.c            |    2 +-
>  sound/soc/fsl/mpc5200_psc_ac97.c     |   30 +++++++--------
>  sound/soc/pxa/pxa-ssp.c              |    4 +-
>  sound/soc/s3c24xx/s3c-i2s-v2.c       |    4 +-
>  sound/soc/sh/ssi.c                   |    2 +-
>  19 files changed, 130 insertions(+), 38 deletions(-)
>  create mode 100644 sound/soc/codecs/spdif_transciever.c
>  create mode 100644 sound/soc/codecs/spdif_transciever.h
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6:
  Mark Brown (1):
        ASoC: Initialise dev for the dummy S/PDIF DAI

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Cliff Cai (3):
      ASoC: SSM2602: remove unsupported sample rates
      ASoC: Blackfin: set the transfer size according the ac97_frame size
      ASoC: SSM2602: assign last substream to the master when shutting down

Peter Ujfalusi (2):
      ASoC: TWL4030: Use reg_cache in twl4030_init_chip
      ASoC: TWL4030: Check the interface format for 4 channel mode

Sonic Zhang (1):
      ASoC: Blackfin: document how anomaly 05000250 is handled

 sound/soc/blackfin/bf5xx-ac97.c  |    9 +++++++++
 sound/soc/blackfin/bf5xx-sport.c |    4 ++--
 sound/soc/codecs/ssm2602.c       |   33 +++++++++++++++++++--------------
 sound/soc/codecs/twl4030.c       |   16 +++++++++++-----
 4 files changed, 41 insertions(+), 21 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Wed, 3 Jun 2009 11:51:50 +0100,
Mark Brown wrote:
> 
> The following changes since commit 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6:
>   Mark Brown (1):
>         ASoC: Initialise dev for the dummy S/PDIF DAI
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks, pulled now.


Takashi

> Cliff Cai (3):
>       ASoC: SSM2602: remove unsupported sample rates
>       ASoC: Blackfin: set the transfer size according the ac97_frame size
>       ASoC: SSM2602: assign last substream to the master when shutting down
> 
> Peter Ujfalusi (2):
>       ASoC: TWL4030: Use reg_cache in twl4030_init_chip
>       ASoC: TWL4030: Check the interface format for 4 channel mode
> 
> Sonic Zhang (1):
>       ASoC: Blackfin: document how anomaly 05000250 is handled
> 
>  sound/soc/blackfin/bf5xx-ac97.c  |    9 +++++++++
>  sound/soc/blackfin/bf5xx-sport.c |    4 ++--
>  sound/soc/codecs/ssm2602.c       |   33 +++++++++++++++++++--------------
>  sound/soc/codecs/twl4030.c       |   16 +++++++++++-----
>  4 files changed, 41 insertions(+), 21 deletions(-)
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27:
  Takashi Iwai (1):
        ASoC: Kill BUS_ID_SIZE

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Barry Song (1):
      ASoC: Blackfin: update the bf5xx_i2s_resume parameters

Cliff Cai (1):
      ASoC: Blackfin: keep better track of SPORT configuration state

 sound/soc/blackfin/bf5xx-i2s.c |   10 +++++++---
 1 files changed, 7 insertions(+), 3 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Sun, 21 Jun 2009 14:15:31 +0100,
Mark Brown wrote:
> 
> The following changes since commit ef39412622b6e8f09c383de9565b07e93553fc27:
>   Takashi Iwai (1):
>         ASoC: Kill BUS_ID_SIZE
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks.


Takashi

> Barry Song (1):
>       ASoC: Blackfin: update the bf5xx_i2s_resume parameters
> 
> Cliff Cai (1):
>       ASoC: Blackfin: keep better track of SPORT configuration state
> 
>  sound/soc/blackfin/bf5xx-i2s.c |   10 +++++++---
>  1 files changed, 7 insertions(+), 3 deletions(-)
> 

Re: ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3:
  Linus Torvalds (1):
        Linux 2.6.31-rc1

are available in the git repository at:

  ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Peter Ujfalusi (1):
      MAINTAINERS: Add entry for twl4030 series soc codec driver

 MAINTAINERS |    7 +++++++
 1 files changed, 7 insertions(+), 0 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Thu, 25 Jun 2009 14:09:39 +0100,
Mark Brown wrote:
> 
> The following changes since commit 28d0325ce6e0a52f53d8af687e6427fee59004d3:
>   Linus Torvalds (1):
>         Linux 2.6.31-rc1
> 
> are available in the git repository at:
> 
>   ssh://master.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Thanks.  This gave me a merge error (because other entries have been
added to MAINTAINERS) so I corrected manually.

So, I based my 2.6.32 asoc branch on this to avoid the doubly merge
corrections, and I pulled your 2.6.32 branch there, too.

Now everything should be settled doen, you can forward gracefully by
just pulling from my branches :)


Takashi

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 10121a12e2380fb34b6e646a8e367add06d036c3:
  Takashi Iwai (1):
        Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Grant Likely (2):
      ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig
      ASoC: remove BROKEN from Efika and pcm030 fabric drivers

Janusz Krzysztofik (1):
      ASoC: OMAP: fix OMAP1510 broken PCM pointer callback

Mark Brown (1):
      ASoC: Only disable pxa2xx-i2s clocks if we enabled them

 sound/soc/fsl/Kconfig      |    6 +++---
 sound/soc/omap/omap-pcm.c  |   11 +++++++----
 sound/soc/pxa/pxa2xx-i2s.c |    7 ++++++-
 3 files changed, 16 insertions(+), 8 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Wed, 1 Jul 2009 21:45:05 +0100,
Mark Brown wrote:
> 
> The following changes since commit 10121a12e2380fb34b6e646a8e367add06d036c3:
>   Takashi Iwai (1):
>         Merge branch 'for-2.6.31' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled now.  Thanks.


Takashi

> Grant Likely (2):
>       ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig
>       ASoC: remove BROKEN from Efika and pcm030 fabric drivers
> 
> Janusz Krzysztofik (1):
>       ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
> 
> Mark Brown (1):
>       ASoC: Only disable pxa2xx-i2s clocks if we enabled them
> 
>  sound/soc/fsl/Kconfig      |    6 +++---
>  sound/soc/omap/omap-pcm.c  |   11 +++++++----
>  sound/soc/pxa/pxa2xx-i2s.c |    7 ++++++-
>  3 files changed, 16 insertions(+), 8 deletions(-)
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit da9ff1f796e81976935407251815838bef9868d4:
  Mark Brown (1):
        ASoC: Only disable pxa2xx-i2s clocks if we enabled them

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Grant Likely (2):
      ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
      ASoC: add locking to mpc5200-psc-ac97 driver

Lars-Peter Clausen (1):
      ASoC: Fix wm8753 register cache size and initialization

Mark Brown (1):
      ASoC: Fix register cache initialisation for WM8753

 sound/soc/codecs/wm8753.c        |    6 +++---
 sound/soc/fsl/mpc5200_dma.c      |    1 +
 sound/soc/fsl/mpc5200_dma.h      |    1 +
 sound/soc/fsl/mpc5200_psc_ac97.c |   17 ++++++++++++++++-
 4 files changed, 21 insertions(+), 4 deletions(-)

Re: ASoC updates for 2.6.31

[Takashi Iwai]

At Mon, 6 Jul 2009 21:49:01 +0100,
Mark Brown wrote:
> 
> The following changes since commit da9ff1f796e81976935407251815838bef9868d4:
>   Mark Brown (1):
>         ASoC: Only disable pxa2xx-i2s clocks if we enabled them
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Pulled both branches now.  Thanks.


Takashi

> Grant Likely (2):
>       ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
>       ASoC: add locking to mpc5200-psc-ac97 driver
> 
> Lars-Peter Clausen (1):
>       ASoC: Fix wm8753 register cache size and initialization
> 
> Mark Brown (1):
>       ASoC: Fix register cache initialisation for WM8753
> 
>  sound/soc/codecs/wm8753.c        |    6 +++---
>  sound/soc/fsl/mpc5200_dma.c      |    1 +
>  sound/soc/fsl/mpc5200_dma.h      |    1 +
>  sound/soc/fsl/mpc5200_psc_ac97.c |   17 ++++++++++++++++-
>  4 files changed, 21 insertions(+), 4 deletions(-)
> 

ASoC updates for 2.6.31

[Mark Brown]

The following changes since commit 637a935aaba2f05e2178c9d1b714d7a2c36c8b44:
  Lars-Peter Clausen (1):
        ASoC: Fix wm8753 register cache size and initialization

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31

Daniel Mack (1):
      ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free

 sound/arm/pxa2xx-pcm-lib.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)