[PATCH 00/18] ASoC updates

[PATCH 00/18] ASoC updates

[Mark Brown]

The following changes since commit 28a1d869560a49d960ba2a3b450ec965712e5560:
  Daniel Mack (1):
        ASoC: tlv320aic3x: control additions and cleanups

are available in the git repository at:

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

Mark Brown (9):
      ASoC: Add card registration API
      ASoC: Add DAI registration API
      ASoC: Register platform DAIs
      ASoC: Add platform registration API
      ASoC: Register platform drivers
      ASoC: Register non-AC97 codec DAIs
      ASoC: Initial framework for dynamic card instantiation
      ASoC: Wait for non-AC97 codec DAIs before instantiating
      ASoC: Convert WM8900 to allow registration by machine code

Peter Ujfalusi (9):
      ASoC: TWL4030: Correct DAPM_DAC with power control
      ASoC: TWL4030: Add Analog PGA control switch to DAPM
      ASoC: TWL4030: Add DAPM event handler for output MUX selection
      ASoC: TWL4030: DAPM mapping of the Earpiece output
      ASoC: TWL4030: DAPM mapping of the PreDriv outputs
      ASoC: TWL4030: DAPM mapping of the Headset outputs
      ASoC: TWL4030: DAPM mapping of the Carkit outputs
      ASoC: TWL4030: DAPM mapping of the Handsfree outputs
      ASoC: TWL4030: Do not alter the Headset output volume on power-up/down

 include/sound/soc-dai.h             |    8 +
 include/sound/soc.h                 |   10 ++
 sound/soc/atmel/atmel-pcm.c         |   12 ++
 sound/soc/atmel/atmel_ssc_dai.c     |   11 ++
 sound/soc/au1x/dbdma2.c             |    3 +-
 sound/soc/au1x/psc-ac97.c           |    3 +-
 sound/soc/au1x/psc-i2s.c            |    3 +-
 sound/soc/blackfin/bf5xx-ac97-pcm.c |   12 ++
 sound/soc/blackfin/bf5xx-ac97.c     |   12 ++
 sound/soc/blackfin/bf5xx-i2s-pcm.c  |   12 ++
 sound/soc/blackfin/bf5xx-i2s.c      |   12 ++
 sound/soc/codecs/ad73311.c          |   12 ++
 sound/soc/codecs/ak4535.c           |   12 ++
 sound/soc/codecs/cs4270.c           |   12 ++
 sound/soc/codecs/pcm3008.c          |   12 ++
 sound/soc/codecs/ssm2602.c          |   12 ++
 sound/soc/codecs/tlv320aic23.c      |   12 ++
 sound/soc/codecs/tlv320aic26.c      |   15 ++-
 sound/soc/codecs/tlv320aic3x.c      |   12 ++
 sound/soc/codecs/twl4030.c          |  277 +++++++++++++++++++++++++++++++--
 sound/soc/codecs/uda134x.c          |   12 ++
 sound/soc/codecs/uda1380.c          |   12 ++
 sound/soc/codecs/wm8510.c           |   12 ++
 sound/soc/codecs/wm8580.c           |   12 ++
 sound/soc/codecs/wm8728.c           |   12 ++
 sound/soc/codecs/wm8731.c           |   12 ++
 sound/soc/codecs/wm8750.c           |   12 ++
 sound/soc/codecs/wm8753.c           |   12 ++
 sound/soc/codecs/wm8900.c           |  105 ++++---------
 sound/soc/codecs/wm8903.c           |   12 ++
 sound/soc/codecs/wm8971.c           |   12 ++
 sound/soc/codecs/wm8990.c           |   12 ++
 sound/soc/davinci/davinci-i2s.c     |   12 ++
 sound/soc/davinci/davinci-pcm.c     |   12 ++
 sound/soc/fsl/fsl_dma.c             |   12 ++
 sound/soc/fsl/fsl_ssi.c             |   10 ++
 sound/soc/fsl/mpc5200_psc_i2s.c     |    4 +
 sound/soc/omap/omap-mcbsp.c         |   13 ++
 sound/soc/omap/omap-pcm.c           |   12 ++
 sound/soc/pxa/pxa-ssp.c             |   12 ++
 sound/soc/pxa/pxa2xx-ac97.c         |   12 ++
 sound/soc/pxa/pxa2xx-i2s.c          |   13 ++-
 sound/soc/pxa/pxa2xx-pcm.c          |   12 ++
 sound/soc/s3c24xx/s3c2412-i2s.c     |   13 ++
 sound/soc/s3c24xx/s3c2443-ac97.c    |   13 ++
 sound/soc/s3c24xx/s3c24xx-i2s.c     |   12 ++
 sound/soc/s3c24xx/s3c24xx-pcm.c     |   12 ++
 sound/soc/sh/dma-sh7760.c           |   12 ++
 sound/soc/sh/hac.c                  |   12 ++
 sound/soc/sh/ssi.c                  |   12 ++
 sound/soc/soc-core.c                |  291 +++++++++++++++++++++++++++++++++--
 51 files changed, 1102 insertions(+), 110 deletions(-)

[PATCH 01/18] ASoC: TWL4030: Correct DAPM_DAC with power control

[Mark Brown]

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

Add all four DACs to dapm_widgets with power switch.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   15 +++++++++++----
 1 files changed, 11 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 3c9fdf2..3543bf6 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -616,8 +616,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("OUTL"),
 	SND_SOC_DAPM_OUTPUT("OUTR"),
 
-	SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0),
+	/* DACs */
+	SND_SOC_DAPM_DAC("DACR1", "Right Front Playback",
+			TWL4030_REG_AVDAC_CTL, 0, 0),
+	SND_SOC_DAPM_DAC("DACL1", "Left Front Playback",
+			TWL4030_REG_AVDAC_CTL, 1, 0),
+	SND_SOC_DAPM_DAC("DACR2", "Right Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 2, 0),
+	SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 3, 0),
 
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
@@ -625,8 +632,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 
 static const struct snd_soc_dapm_route intercon[] = {
 	/* outputs */
-	{"OUTL", NULL, "DACL"},
-	{"OUTR", NULL, "DACR"},
+	{"OUTL", NULL, "DACL2"},
+	{"OUTR", NULL, "DACR2"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 02/18] ASoC: TWL4030: Add Analog PGA control switch to DAPM

[Mark Brown]

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

Add all four APGA switch to DAPM routing and widgets.
Add user control for DA enable for all APGA as normal
control.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   25 +++++++++++++++++++++++--
 1 files changed, 23 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 3543bf6..4293ec7 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -562,6 +562,12 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
 		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
 		3, 0x12, 1, analog_tlv),
+	SOC_DOUBLE_R("DAC1 Analog Playback Switch",
+		TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+		1, 1, 0),
+	SOC_DOUBLE_R("DAC2 Analog Playback Switch",
+		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+		1, 1, 0),
 
 	/* Separate output gain controls */
 	SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
@@ -626,14 +632,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback",
 			TWL4030_REG_AVDAC_CTL, 3, 0),
 
+	/* Analog PGAs */
+	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
+			0, 0, NULL, 0),
+
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
+	{"ARXL1_APGA", NULL, "DACL1"},
+	{"ARXR1_APGA", NULL, "DACR1"},
+	{"ARXL2_APGA", NULL, "DACL2"},
+	{"ARXR2_APGA", NULL, "DACR2"},
+
 	/* outputs */
-	{"OUTL", NULL, "DACL2"},
-	{"OUTR", NULL, "DACR2"},
+	{"OUTL", NULL, "ARXL2_APGA"},
+	{"OUTR", NULL, "ARXR2_APGA"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 03/18] ASoC: TWL4030: Add DAPM event handler for output MUX selection

[Mark Brown]

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

DAPM event handler is set to filter out invalid MUX settings
for certain outputs.
Earpiece:
 - 0 = Off
 - 1 = DACL1
 - 2 = DACL2
 - 3 = *** Invalid ***
 - 4 = DACR1

PreDriveL/R:
 - 0 = Off/Off
 - 1 = DACL1/DACR1
 - 2 = DACL2/DACR2
 - 3 = *** Invalid/Invalid ***
 - 4 = DACR2/DACL2

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   24 ++++++++++++++++++++++++
 1 files changed, 24 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 4293ec7..9d10783 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -190,6 +190,30 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	int ret = 0;
+	int val;
+
+	switch (e->reg) {
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_EAR_CTL:
+		val = w->value >> e->shift_l;
+		if (val == 3) {
+			printk(KERN_WARNING
+			"Invalid MUX setting for register 0x%02x (%d)\n",
+			      e->reg, val);
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
-- 
1.5.6.5

[PATCH 04/18] ASoC: TWL4030: DAPM mapping of the Earpiece output

[Mark Brown]

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

Adds DAPM muxing, routing for the Earpiece output.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   27 +++++++++++++++++++++++++++
 1 files changed, 27 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 9d10783..1da4617 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -190,6 +190,19 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+/* Earpiece */
+static const char *twl4030_earpiece_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid",
+		"DACR1"};
+
+static const struct soc_enum twl4030_earpiece_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1,
+			ARRAY_SIZE(twl4030_earpiece_texts),
+			twl4030_earpiece_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
+SOC_DAPM_ENUM("Route", twl4030_earpiece_enum);
+
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -645,6 +658,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 
 	SND_SOC_DAPM_OUTPUT("OUTL"),
 	SND_SOC_DAPM_OUTPUT("OUTR"),
+	SND_SOC_DAPM_OUTPUT("EARPIECE"),
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DACR1", "Right Front Playback",
@@ -666,6 +680,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
 			0, 0, NULL, 0),
 
+	/* Output MUX controls */
+	/* Earpiece */
+	SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_earpiece_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
 };
@@ -676,9 +696,16 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ARXL2_APGA", NULL, "DACL2"},
 	{"ARXR2_APGA", NULL, "DACR2"},
 
+	/* Internal playback routings */
+	/* Earpiece */
+	{"Earpiece Mux", "DACL1", "ARXL1_APGA"},
+	{"Earpiece Mux", "DACL2", "ARXL2_APGA"},
+	{"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+
 	/* outputs */
 	{"OUTL", NULL, "ARXL2_APGA"},
 	{"OUTR", NULL, "ARXR2_APGA"},
+	{"EARPIECE", NULL, "Earpiece Mux"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 05/18] ASoC: TWL4030: DAPM mapping of the PreDriv outputs

[Mark Brown]

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

Adds DAPM muxing, routing for the PreDrive outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   45 ++++++++++++++++++++++++++++++++++++++++++++
 1 files changed, 45 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 1da4617..c508344 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -203,6 +203,32 @@ static const struct soc_enum twl4030_earpiece_enum =
 static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
 SOC_DAPM_ENUM("Route", twl4030_earpiece_enum);
 
+/* PreDrive Left */
+static const char *twl4030_predrivel_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid",
+		"DACR2"};
+
+static const struct soc_enum twl4030_predrivel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1,
+			ARRAY_SIZE(twl4030_predrivel_texts),
+			twl4030_predrivel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
+SOC_DAPM_ENUM("Route", twl4030_predrivel_enum);
+
+/* PreDrive Right */
+static const char *twl4030_predriver_texts[] =
+		{"Off", "DACR1", "DACR2", "Invalid",
+		"DACL2"};
+
+static const struct soc_enum twl4030_predriver_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1,
+			ARRAY_SIZE(twl4030_predriver_texts),
+			twl4030_predriver_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
+SOC_DAPM_ENUM("Route", twl4030_predriver_enum);
+
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -659,6 +685,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("OUTL"),
 	SND_SOC_DAPM_OUTPUT("OUTR"),
 	SND_SOC_DAPM_OUTPUT("EARPIECE"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DACR1", "Right Front Playback",
@@ -685,6 +713,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_earpiece_control, outmixer_event,
 		SND_SOC_DAPM_PRE_REG),
+	/* PreDrivL/R */
+	SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predrivel_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predriver_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
 
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
@@ -701,11 +736,21 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Earpiece Mux", "DACL1", "ARXL1_APGA"},
 	{"Earpiece Mux", "DACL2", "ARXL2_APGA"},
 	{"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+	/* PreDrivL */
+	{"PredriveL Mux", "DACL1", "ARXL1_APGA"},
+	{"PredriveL Mux", "DACL2", "ARXL2_APGA"},
+	{"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+	/* PreDrivR */
+	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
+	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
+	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
 
 	/* outputs */
 	{"OUTL", NULL, "ARXL2_APGA"},
 	{"OUTR", NULL, "ARXR2_APGA"},
 	{"EARPIECE", NULL, "Earpiece Mux"},
+	{"PREDRIVEL", NULL, "PredriveL Mux"},
+	{"PREDRIVER", NULL, "PredriveR Mux"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 06/18] ASoC: TWL4030: DAPM mapping of the Headset outputs

[Mark Brown]

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

Adds DAPM muxing, routing for the Headset outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   39 +++++++++++++++++++++++++++++++++++++++
 1 files changed, 39 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index c508344..86ff5a9 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -229,6 +229,30 @@ static const struct soc_enum twl4030_predriver_enum =
 static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
 SOC_DAPM_ENUM("Route", twl4030_predriver_enum);
 
+/* Headset Left */
+static const char *twl4030_hsol_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_hsol_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
+			ARRAY_SIZE(twl4030_hsol_texts),
+			twl4030_hsol_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
+SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+
+/* Headset Right */
+static const char *twl4030_hsor_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_hsor_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
+			ARRAY_SIZE(twl4030_hsor_texts),
+			twl4030_hsor_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
+SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -687,6 +711,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("EARPIECE"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DACR1", "Right Front Playback",
@@ -720,6 +746,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_predriver_control, outmixer_event,
 		SND_SOC_DAPM_PRE_REG),
+	/* HeadsetL/R */
+	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsor_control),
 
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
@@ -744,6 +775,12 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
 	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
 	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetL */
+	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
+	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetR */
+	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
+	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
 
 	/* outputs */
 	{"OUTL", NULL, "ARXL2_APGA"},
@@ -751,6 +788,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"EARPIECE", NULL, "Earpiece Mux"},
 	{"PREDRIVEL", NULL, "PredriveL Mux"},
 	{"PREDRIVER", NULL, "PredriveR Mux"},
+	{"HSOL", NULL, "HeadsetL Mux"},
+	{"HSOR", NULL, "HeadsetR Mux"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 07/18] ASoC: TWL4030: DAPM mapping of the Carkit outputs

[Mark Brown]

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

Adds DAPM muxing, routing for the Carkit outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   37 +++++++++++++++++++++++++++++++++++++
 1 files changed, 37 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 86ff5a9..08c33e9 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -253,6 +253,30 @@ static const struct soc_enum twl4030_hsor_enum =
 static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
 SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
 
+/* Carkit Left */
+static const char *twl4030_carkitl_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_carkitl_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitl_texts),
+			twl4030_carkitl_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+
+/* Carkit Right */
+static const char *twl4030_carkitr_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_carkitr_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitr_texts),
+			twl4030_carkitr_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -751,6 +775,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		&twl4030_dapm_hsol_control),
 	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_hsor_control),
+	/* CarkitL/R */
+	SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitl_control),
+	SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitr_control),
 
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
@@ -781,6 +810,12 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* HeadsetR */
 	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
 	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+	/* CarkitL */
+	{"CarkitL Mux", "DACL1", "ARXL1_APGA"},
+	{"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+	/* CarkitR */
+	{"CarkitR Mux", "DACR1", "ARXR1_APGA"},
+	{"CarkitR Mux", "DACR2", "ARXR2_APGA"},
 
 	/* outputs */
 	{"OUTL", NULL, "ARXL2_APGA"},
@@ -790,6 +825,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"PREDRIVER", NULL, "PredriveR Mux"},
 	{"HSOL", NULL, "HeadsetL Mux"},
 	{"HSOR", NULL, "HeadsetR Mux"},
+	{"CARKITL", NULL, "CarkitL Mux"},
+	{"CARKITR", NULL, "CarkitR Mux"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 08/18] ASoC: TWL4030: DAPM mapping of the Handsfree outputs

[Mark Brown]

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

Adds DAPM muxing, routing for the Handsfree outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   41 +++++++++++++++++++++++++++++++++++++++++
 1 files changed, 41 insertions(+), 0 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 08c33e9..d0612a4 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -277,6 +277,30 @@ static const struct soc_enum twl4030_carkitr_enum =
 static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
 SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
 
+/* Handsfree Left */
+static const char *twl4030_handsfreel_texts[] =
+		{"Voice", "DACL1", "DACL2", "DACR2"};
+
+static const struct soc_enum twl4030_handsfreel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreel_texts),
+			twl4030_handsfreel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+
+/* Handsfree Right */
+static const char *twl4030_handsfreer_texts[] =
+		{"Voice", "DACR1", "DACR2", "DACL2"};
+
+static const struct soc_enum twl4030_handsfreer_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreer_texts),
+			twl4030_handsfreer_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+
 static int outmixer_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -737,6 +761,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
 	SND_SOC_DAPM_OUTPUT("HSOL"),
 	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DACR1", "Right Front Playback",
@@ -780,6 +806,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		&twl4030_dapm_carkitl_control),
 	SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_carkitr_control),
+	/* HandsfreeL/R */
+	SND_SOC_DAPM_MUX("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
+		&twl4030_dapm_handsfreel_control),
+	SND_SOC_DAPM_MUX("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
+		&twl4030_dapm_handsfreer_control),
 
 	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
@@ -816,6 +847,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* CarkitR */
 	{"CarkitR Mux", "DACR1", "ARXR1_APGA"},
 	{"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeL */
+	{"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
+	{"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
+	{"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeR */
+	{"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
+	{"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
+	{"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
 
 	/* outputs */
 	{"OUTL", NULL, "ARXL2_APGA"},
@@ -827,6 +866,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HSOR", NULL, "HeadsetR Mux"},
 	{"CARKITL", NULL, "CarkitL Mux"},
 	{"CARKITR", NULL, "CarkitR Mux"},
+	{"HFL", NULL, "HandsfreeL Mux"},
+	{"HFR", NULL, "HandsfreeR Mux"},
 
 	/* inputs */
 	{"ADCL", NULL, "INL"},
-- 
1.5.6.5

[PATCH 09/18] ASoC: TWL4030: Do not alter the Headset output volume on power-up/down

[Mark Brown]

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

There is a separate gain control for the Headset output already.
Do not reset the gain to 0 dB at power up.
In power-down, there is no need to set the Headset output gain
to power-down mode, since if the CODECPDZ is in powered off this
setting has no effect.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c |   12 ++----------
 1 files changed, 2 insertions(+), 10 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index d0612a4..358aa2b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -887,7 +887,7 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
 {
-	u8 anamicl, regmisc1, byte, popn, hsgain;
+	u8 anamicl, regmisc1, byte, popn;
 	int i = 0;
 
 	/* set CODECPDZ to turn on codec */
@@ -925,10 +925,6 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	popn |=	TWL4030_VMID_EN;
 	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
 
-	/* enable output stage and gain setting */
-	hsgain = TWL4030_HSR_GAIN_0DB | TWL4030_HSL_GAIN_0DB;
-	twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain);
-
 	/* enable anti-pop ramp */
 	popn |= TWL4030_RAMP_EN;
 	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
@@ -936,17 +932,13 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 
 static void twl4030_power_down(struct snd_soc_codec *codec)
 {
-	u8 popn, hsgain;
+	u8 popn;
 
 	/* disable anti-pop ramp */
 	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
 	popn &= ~TWL4030_RAMP_EN;
 	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
 
-	/* disable output stage and gain setting */
-	hsgain = TWL4030_HSR_GAIN_PWR_DOWN | TWL4030_HSL_GAIN_PWR_DOWN;
-	twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain);
-
 	/* disable bias out */
 	popn &= ~TWL4030_VMID_EN;
 	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-- 
1.5.6.5

[PATCH 10/18] ASoC: Add card registration API

[Mark Brown]

ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting.  Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |    5 ++++
 sound/soc/soc-core.c |   62 ++++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 67 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 79d855d..4a578b5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -333,6 +333,11 @@ struct snd_soc_dai_link  {
 /* SoC card */
 struct snd_soc_card {
 	char *name;
+	struct device *dev;
+
+	struct list_head list;
+
+	int instantiated;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2f2a8d9..44fbd71 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -43,6 +43,12 @@ static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 static struct dentry *debugfs_root;
 #endif
 
+static DEFINE_MUTEX(client_mutex);
+static LIST_HEAD(card_list);
+
+static int snd_soc_register_card(struct snd_soc_card *card);
+static int snd_soc_unregister_card(struct snd_soc_card *card);
+
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
  * It can be used to eliminate pops between different playback streams, e.g.
@@ -784,6 +790,14 @@ static int soc_probe(struct platform_device *pdev)
 	/* Bodge while we push things out of socdev */
 	card->socdev = socdev;
 
+	/* Bodge while we unpick instantiation */
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register card\n");
+		return ret;
+	}
+
 	if (card->probe) {
 		ret = card->probe(pdev);
 		if (ret < 0)
@@ -863,6 +877,8 @@ static int soc_remove(struct platform_device *pdev)
 	if (card->remove)
 		card->remove(pdev);
 
+	snd_soc_unregister_card(card);
+
 	return 0;
 }
 
@@ -1957,6 +1973,52 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
+/**
+ * snd_soc_register_card - Register a card with the ASoC core
+ *
+ * @param card Card to register
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_register_card(struct snd_soc_card *card)
+{
+	if (!card->name || !card->dev)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&card->list);
+	card->instantiated = 0;
+
+	mutex_lock(&client_mutex);
+	list_add(&card->list, &card_list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Registered card '%s'\n", card->name);
+
+	return 0;
+}
+
+/**
+ * snd_soc_unregister_card - Unregister a card with the ASoC core
+ *
+ * @param card Card to unregister
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_unregister_card(struct snd_soc_card *card)
+{
+	mutex_lock(&client_mutex);
+	list_del(&card->list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
+
+	return 0;
+}
+
 static int __devinit snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-- 
1.5.6.5

[PATCH 11/18] ASoC: Add DAI registration API

[Mark Brown]

Add API calls to register and unregister DAIs with the core.  Currently
these APIs are ineffective.  Since multiple DAIs for a given device are
a common case bulk variants are provided.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h |    8 ++++
 sound/soc/soc-core.c    |   83 +++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 91 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index e2d5f76..24247f7 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -100,6 +100,12 @@ struct snd_soc_dai_ops;
 struct snd_soc_dai;
 struct snd_ac97_bus_ops;
 
+/* Digital Audio Interface registration */
+int snd_soc_register_dai(struct snd_soc_dai *dai);
+void snd_soc_unregister_dai(struct snd_soc_dai *dai);
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
+
 /* Digital Audio Interface clocking API.*/
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir);
@@ -186,6 +192,8 @@ struct snd_soc_dai {
 	unsigned int id;
 	int ac97_control;
 
+	struct device *dev;
+
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,
 		     struct snd_soc_dai *dai);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 44fbd71..03460b0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -45,6 +45,7 @@ static struct dentry *debugfs_root;
 
 static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(card_list);
+static LIST_HEAD(dai_list);
 
 static int snd_soc_register_card(struct snd_soc_card *card);
 static int snd_soc_unregister_card(struct snd_soc_card *card);
@@ -2019,6 +2020,88 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
+/**
+ * snd_soc_register_dai - Register a DAI with the ASoC core
+ *
+ * @param dai DAI to register
+ */
+int snd_soc_register_dai(struct snd_soc_dai *dai)
+{
+	if (!dai->name)
+		return -EINVAL;
+
+	/* The device should become mandatory over time */
+	if (!dai->dev)
+		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
+
+	INIT_LIST_HEAD(&dai->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&dai->list, &dai_list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered DAI '%s'\n", dai->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dai);
+
+/**
+ * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
+ *
+ * @param dai DAI to unregister
+ */
+void snd_soc_unregister_dai(struct snd_soc_dai *dai)
+{
+	mutex_lock(&client_mutex);
+	list_del(&dai->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered DAI '%s'\n", dai->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
+
+/**
+ * snd_soc_register_dais - Register multiple DAIs with the ASoC core
+ *
+ * @param dai Array of DAIs to register
+ * @param count Number of DAIs
+ */
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i, ret;
+
+	for (i = 0; i < count; i++) {
+		ret = snd_soc_register_dai(&dai[i]);
+		if (ret != 0)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	for (i--; i >= 0; i--)
+		snd_soc_unregister_dai(&dai[i]);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dais);
+
+/**
+ * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
+ *
+ * @param dai Array of DAIs to unregister
+ * @param count Number of DAIs
+ */
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i;
+
+	for (i = 0; i < count; i++)
+		snd_soc_unregister_dai(&dai[i]);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
+
 static int __devinit snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-- 
1.5.6.5

[PATCH 12/18] ASoC: Register platform DAIs

[Mark Brown]

Register all platform DAIs with the core.  In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel_ssc_dai.c  |   11 +++++++++++
 sound/soc/au1x/psc-ac97.c        |    3 ++-
 sound/soc/au1x/psc-i2s.c         |    3 ++-
 sound/soc/blackfin/bf5xx-ac97.c  |   12 ++++++++++++
 sound/soc/blackfin/bf5xx-i2s.c   |   12 ++++++++++++
 sound/soc/davinci/davinci-i2s.c  |   12 ++++++++++++
 sound/soc/fsl/fsl_ssi.c          |   10 ++++++++++
 sound/soc/omap/omap-mcbsp.c      |   13 +++++++++++++
 sound/soc/pxa/pxa-ssp.c          |   12 ++++++++++++
 sound/soc/pxa/pxa2xx-ac97.c      |   12 ++++++++++++
 sound/soc/pxa/pxa2xx-i2s.c       |   13 ++++++++++++-
 sound/soc/s3c24xx/s3c2412-i2s.c  |   13 +++++++++++++
 sound/soc/s3c24xx/s3c2443-ac97.c |   13 +++++++++++++
 sound/soc/s3c24xx/s3c24xx-i2s.c  |   12 ++++++++++++
 sound/soc/sh/hac.c               |   12 ++++++++++++
 sound/soc/sh/ssi.c               |   12 ++++++++++++
 16 files changed, 172 insertions(+), 3 deletions(-)

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index d9b874c..c9f02ed 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -772,6 +772,17 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 };
 EXPORT_SYMBOL_GPL(atmel_ssc_dai);
 
+static int __devinit atmel_ssc_modinit(void)
+{
+	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_init(snd_soc_init);
+
+static void __exit snd_soc_exit(void)
+{
+	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+
 /* Module information */
 MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
 MODULE_DESCRIPTION("ATMEL SSC ASoC Interface");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a1e824d..f0e30ae 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -371,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 static int __init au1xpsc_ac97_init(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_ac97_dai);
 }
 
 static void __exit au1xpsc_ac97_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 }
 
 module_init(au1xpsc_ac97_init);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 16f9746..f916de4 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -396,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai);
 static int __init au1xpsc_i2s_init(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_i2s_dai);
 }
 
 static void __exit au1xpsc_i2s_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 }
 
 module_init(au1xpsc_i2s_init);
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index c602ce1..ad3efee 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -431,6 +431,18 @@ struct snd_soc_dai bfin_ac97_dai = {
 };
 EXPORT_SYMBOL_GPL(bfin_ac97_dai);
 
+static int __devinit bfin_ac97_init(void)
+{
+	return snd_soc_register_dai(&bfin_ac97_dai);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_ac97_dai);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 9f8ce87..0d58d2b 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -313,6 +313,18 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
+static int __devinit bfin_i2s_init(void)
+{
+	return snd_soc_register_dai(&bfin_i2s_dai);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_i2s_dai);
+}
+module_exit(bfin_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 8b99efb..d89fc2f 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -481,6 +481,18 @@ struct snd_soc_dai davinci_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
+static int __devinit davinci_i2s_init(void)
+{
+	return snd_soc_register_dai(&davinci_i2s_dai);
+}
+module_init(davinci_i2s_init);
+
+static void __exit davinci_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&davinci_i2s_dai);
+}
+module_exit(davinci_i2s_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 52c290b..c6d6eb7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -673,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->private_data = ssi_private;
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
+	fsl_ssi_dai->dev = ssi_info->dev;
+
+	ret = snd_soc_register_dai(fsl_ssi_dai);
+	if (ret != 0) {
+		dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+		kfree(fsl_ssi_dai);
+		return NULL;
+	}
 
 	return fsl_ssi_dai;
 }
@@ -690,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 
 	device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
 
+	snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
 	kfree(ssi_private);
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index e8f1314..41cab20 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -499,6 +499,19 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
+static int __devinit omap_mcbsp_init(void)
+{
+	return snd_soc_register_dais(omap_mcbsp_dai,
+				     ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(omap_mcbsp_init);
+
+static void __exit omap_mcbsp_exit(void)
+{
+	snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(omap_mcbsp_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 73fa10d..3587f2f 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -913,6 +913,18 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
 
+static int __devinit pxa_ssp_init(void)
+{
+	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
 /* Module information */
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 8eed80d..f6249d5 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -228,6 +228,18 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __devinit pxa_ac97_init(void)
+{
+	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 314973a..517991f 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
 static int pxa2xx_i2s_probe(struct platform_device *dev)
 {
+	int ret;
+
 	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	pxa_i2s_dai.dev = &dev->dev;
+	ret = snd_soc_register_dai(&pxa_i2s_dai);
+	if (ret != 0)
+		clk_put(clk_i2s);
+
+	return ret;
 }
 
 static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
 {
+	snd_soc_unregister_dai(&pxa_i2s_dai);
 	clk_put(clk_i2s);
 	clk_i2s = ERR_PTR(-ENOENT);
 	return 0;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 75f87c3..2cf0507 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -736,6 +736,19 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
+static int __devinit s3c2412_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c2412_i2s_dai);
+}
+module_init(s3c2412_i2s_init);
+
+static void __exit s3c2412_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2412_i2s_dai);
+}
+module_exit(s3c2412_i2s_exit);
+
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index f0bc9b7..7360c6d 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -396,6 +396,19 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __devinit s3c2443_ac97_init(void)
+{
+	return snd_soc_register_dai(&s3c2443_ac97_dai);
+}
+module_init(s3c2443_ac97_init);
+
+static void __exit s3c2443_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2443_ac97_dai);
+}
+module_exit(s3c2443_ac97_exit);
+
+
 MODULE_AUTHOR("Graeme Gregory");
 MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 45fe8f7..897b1ac 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -482,6 +482,18 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
+static int __devinit s3c24xx_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c24xx_i2s_dai);
+}
+module_init(s3c24xx_i2s_init);
+
+static void __exit s3c24xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c24xx_i2s_dai);
+}
+module_exit(s3c24xx_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index c435913..9169bad 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -314,6 +314,18 @@ struct snd_soc_dai sh4_hac_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_hac_dai);
 
+static int __devinit sh4_hac_init(void)
+{
+	return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_init(sh4_hac_init);
+
+static void __exit sh4_hac_exit(void)
+{
+	snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_exit(sh4_hac_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index fed544a..9093588 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -392,6 +392,18 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_ssi_dai);
 
+static int __devinit sh4_ssi_init(void)
+{
+	return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_init(sh4_ssi_init);
+
+static void __exit sh4_ssi_exit(void)
+{
+	snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_exit(sh4_ssi_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
-- 
1.5.6.5

[PATCH 13/18] ASoC: Add platform registration API

[Mark Brown]

ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |    5 +++++
 sound/soc/soc-core.c |   38 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 43 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4a578b5..ce3661d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -149,6 +149,7 @@ struct snd_soc_ops;
 struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
+struct snd_soc_platform;
 struct snd_soc_codec;
 struct soc_enum;
 struct snd_soc_ac97_ops;
@@ -158,6 +159,9 @@ typedef int (*hw_read_t)(void *,char* ,int);
 
 extern struct snd_ac97_bus_ops soc_ac97_ops;
 
+int snd_soc_register_platform(struct snd_soc_platform *platform);
+void snd_soc_unregister_platform(struct snd_soc_platform *platform);
+
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
@@ -296,6 +300,7 @@ struct snd_soc_codec_device {
 /* SoC platform interface */
 struct snd_soc_platform {
 	char *name;
+	struct list_head list;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 03460b0..ffae370 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -46,6 +46,7 @@ static struct dentry *debugfs_root;
 static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(card_list);
 static LIST_HEAD(dai_list);
+static LIST_HEAD(platform_list);
 
 static int snd_soc_register_card(struct snd_soc_card *card);
 static int snd_soc_unregister_card(struct snd_soc_card *card);
@@ -2102,6 +2103,43 @@ void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
 
+/**
+ * snd_soc_register_platform - Register a platform with the ASoC core
+ *
+ * @param platform platform to register
+ */
+int snd_soc_register_platform(struct snd_soc_platform *platform)
+{
+	if (!platform->name)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&platform->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&platform->list, &platform_list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered platform '%s'\n", platform->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+
+/**
+ * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ *
+ * @param platform platform to unregister
+ */
+void snd_soc_unregister_platform(struct snd_soc_platform *platform)
+{
+	mutex_lock(&client_mutex);
+	list_del(&platform->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered platform '%s'\n", platform->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
+
 static int __devinit snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-- 
1.5.6.5

[PATCH 14/18] ASoC: Register platform drivers

[Mark Brown]

This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c         |   12 ++++++++++++
 sound/soc/au1x/dbdma2.c             |    3 ++-
 sound/soc/blackfin/bf5xx-ac97-pcm.c |   12 ++++++++++++
 sound/soc/blackfin/bf5xx-i2s-pcm.c  |   12 ++++++++++++
 sound/soc/davinci/davinci-pcm.c     |   12 ++++++++++++
 sound/soc/fsl/fsl_dma.c             |   12 ++++++++++++
 sound/soc/fsl/mpc5200_psc_i2s.c     |    4 ++++
 sound/soc/omap/omap-pcm.c           |   12 ++++++++++++
 sound/soc/pxa/pxa2xx-pcm.c          |   12 ++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c     |   12 ++++++++++++
 sound/soc/sh/dma-sh7760.c           |   12 ++++++++++++
 11 files changed, 114 insertions(+), 1 deletions(-)

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 8507aa1..d6bcb4e 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -477,6 +477,18 @@ struct snd_soc_platform atmel_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(atmel_soc_platform);
 
+static int __devinit atmel_pcm_modinit(void)
+{
+	return snd_soc_register_platform(&atmel_soc_platform);
+}
+module_init(atmel_pcm_modinit);
+
+static void __exit atmel_pcm_exit(void)
+{
+	snd_soc_unregister_platform(&atmel_soc_platform);
+}
+module_exit(atmel_pcm_modexit);
+
 MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
 MODULE_DESCRIPTION("Atmel PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 1466d93..74c823d 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return 0;
+	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }
 
 static void __exit au1xpsc_audio_dbdma_exit(void)
 {
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }
 
 module_init(au1xpsc_audio_dbdma_init);
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index d3d51bc..5b27e0d 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -451,6 +451,18 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
 
+static int __devinit bfin_ac97_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_ac97_soc_platform);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 61fccf9..c58b12a 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
+static int __devinit bfin_i2s_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 76feaa6..f1b6e02 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -384,6 +384,18 @@ struct snd_soc_platform davinci_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(davinci_soc_platform);
 
+static int __devinit davinci_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&davinci_soc_platform);
+}
+module_init(davinci_soc_platform_init);
+
+static void __exit davinci_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&davinci_soc_platform);
+}
+module_exit(davinci_soc_platform_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index bf92331..646c807 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
+static int __devinit fsl_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9ad8f9a..9eb1ce1 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -828,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
+	snd_soc_register_platform(&psc_i2s_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);
@@ -841,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op)
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
+	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
 	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
 	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e9084fd..9940de2 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -354,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(omap_soc_platform);
 
+static int __devinit omap_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 0f6b7bb..4fa1578 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
+static int __devinit pxa2xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index e13e614..ea5a9ca 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,6 +465,18 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
+static int __devinit s3c24xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&s3c24xx_soc_platform);
+}
+module_init(s3c24xx_soc_platform_init);
+
+static void __exit s3c24xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&s3c24xx_soc_platform);
+}
+module_exit(s3c24xx_soc_platform_exit);
+
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 9faa126..39ffca0 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,6 +348,18 @@ struct snd_soc_platform sh7760_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(sh7760_soc_platform);
 
+static int __devinit sh7760_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&sh7760_soc_platform);
+}
+module_init(sh7760_soc_platform_init);
+
+static void __exit sh7760_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&sh7760_soc_platform);
+}
+module_exit(sh7760_soc_platform_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
-- 
1.5.6.5

[PATCH 15/18] ASoC: Register non-AC97 codec DAIs

[Mark Brown]

Currently this is done at module probe time since ASoC ties in codec
device probe to the instantiation of the entire ASoC device. Subsequent
patches will refactor the codec drivers to handle probing separately.
Note that the core does not yet use this information.

AC97 is special since the codec is controlled over the AC97 link but
we want to give the machine driver a chance to set up the system before
trying to instantiate since it may need to do configuration before the
AC97 link will operate

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ad73311.c     |   12 ++++++++++++
 sound/soc/codecs/ak4535.c      |   12 ++++++++++++
 sound/soc/codecs/cs4270.c      |   12 ++++++++++++
 sound/soc/codecs/pcm3008.c     |   12 ++++++++++++
 sound/soc/codecs/ssm2602.c     |   12 ++++++++++++
 sound/soc/codecs/tlv320aic23.c |   12 ++++++++++++
 sound/soc/codecs/tlv320aic26.c |   15 ++++++++++++---
 sound/soc/codecs/tlv320aic3x.c |   12 ++++++++++++
 sound/soc/codecs/twl4030.c     |   14 +++++++++++++-
 sound/soc/codecs/uda134x.c     |   12 ++++++++++++
 sound/soc/codecs/uda1380.c     |   12 ++++++++++++
 sound/soc/codecs/wm8510.c      |   12 ++++++++++++
 sound/soc/codecs/wm8580.c      |   12 ++++++++++++
 sound/soc/codecs/wm8728.c      |   12 ++++++++++++
 sound/soc/codecs/wm8731.c      |   12 ++++++++++++
 sound/soc/codecs/wm8750.c      |   12 ++++++++++++
 sound/soc/codecs/wm8753.c      |   12 ++++++++++++
 sound/soc/codecs/wm8900.c      |   12 ++++++++++++
 sound/soc/codecs/wm8903.c      |   12 ++++++++++++
 sound/soc/codecs/wm8971.c      |   12 ++++++++++++
 sound/soc/codecs/wm8990.c      |   12 ++++++++++++
 21 files changed, 253 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 500f9f3..e32f550 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -98,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
 
+static int __devinit ad73311_init(void)
+{
+	return snd_soc_register_dai(&ad73311_dai);
+}
+module_init(ad73311_init);
+
+static void __exit ad73311_exit(void)
+{
+	snd_soc_unregister_dai(&ad73311_dai);
+}
+module_exit(ad73311_exit);
+
 MODULE_DESCRIPTION("ASoC ad73311 driver");
 MODULE_AUTHOR("Cliff Cai ");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 23062c9..94148fb 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -688,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
 
+static int __devinit ak4535_modinit(void)
+{
+	return snd_soc_register_dai(&ak4535_dai);
+}
+module_init(ak4535_modinit);
+
+static void __exit ak4535_exit(void)
+{
+	snd_soc_unregister_dai(&ak4535_dai);
+}
+module_exit(ak4535_exit);
+
 MODULE_DESCRIPTION("Soc AK4535 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 4667a07..73aaf24 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -774,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
+static int __devinit cs4270_init(void)
+{
+	return snd_soc_register_dai(&cs4270_dai);
+}
+module_init(cs4270_init);
+
+static void __exit cs4270_exit(void)
+{
+	snd_soc_unregister_dai(&cs4270_dai);
+}
+module_exit(cs4270_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index a586255..f333e88 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -195,6 +195,18 @@ struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
 
+static int __devinit pcm3008_init(void)
+{
+	return snd_soc_register_dai(&pcm3008_dai);
+}
+module_init(pcm3008_init);
+
+static void __exit pcm3008_exit(void)
+{
+	snd_soc_unregister_dai(&pcm3008_dai);
+}
+module_exit(pcm3008_exit);
+
 MODULE_DESCRIPTION("Soc PCM3008 driver");
 MODULE_AUTHOR("Hugo Villeneuve");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 9738449..77fdcb4 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -793,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
 
+static int __devinit ssm2602_modinit(void)
+{
+	return snd_soc_register_dai(&ssm2602_dai);
+}
+module_init(ssm2602_modinit);
+
+static void __exit ssm2602_exit(void)
+{
+	snd_soc_unregister_dai(&ssm2602_dai);
+}
+module_exit(ssm2602_exit);
+
 MODULE_DESCRIPTION("ASoC ssm2602 driver");
 MODULE_AUTHOR("Cliff Cai");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index d209bec..eac449b 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -847,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
 
+static int __devinit tlv320aic23_modinit(void)
+{
+	return snd_soc_register_dai(&tlv320aic23_dai);
+}
+module_init(tlv320aic23_modinit);
+
+static void __exit tlv320aic23_exit(void)
+{
+	snd_soc_unregister_dai(&tlv320aic23_dai);
+}
+module_exit(tlv320aic23_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
 MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e33fb7e..29f2f1a 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -426,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
 static int aic26_spi_probe(struct spi_device *spi)
 {
 	struct aic26 *aic26;
-	int rc, i, reg;
+	int ret, i, reg;
 
 	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
 
@@ -456,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi)
 	aic26->codec.reg_cache_size = AIC26_NUM_REGS;
 	aic26->codec.reg_cache = aic26->reg_cache;
 
+	aic26_dai.dev = &spi->dev;
+	ret = snd_soc_register_dai(&aic26_dai);
+	if (ret != 0) {
+		dev_err(&spi->dev, "Failed to register DAI: %d\n", ret);
+		kfree(aic26);
+		return ret;
+	}
+
 	/* Reset the codec to power on defaults */
 	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
 
@@ -474,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi)
 
 	/* Register the sysfs files for debugging */
 	/* Create SysFS files */
-	rc = device_create_file(&spi->dev, &dev_attr_keyclick);
-	if (rc)
+	ret = device_create_file(&spi->dev, &dev_attr_keyclick);
+	if (ret)
 		dev_info(&spi->dev, "error creating sysfs files\n");
 
 #if defined(CONFIG_SND_SOC_OF_SIMPLE)
@@ -492,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi)
 {
 	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
 
+	snd_soc_unregister_dai(&aic26_dai);
 	kfree(aic26);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6a05829..ccd5759 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1411,6 +1411,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
 
+static int __devinit aic3x_modinit(void)
+{
+	return snd_soc_register_dai(&aic3x_dai);
+}
+module_init(aic3x_modinit);
+
+static void __exit aic3x_exit(void)
+{
+	snd_soc_unregister_dai(&aic3x_dai);
+}
+module_exit(aic3x_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 358aa2b..373daa4 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -358,7 +358,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 	.put = snd_soc_put_volsw_r2_twl4030, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		.max = xmax, .invert = xinvert} }
+		 .rshift = xshift, .max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
 	SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
 			       xinvert, tlv_array)
@@ -1275,6 +1275,18 @@ struct snd_soc_codec_device soc_codec_dev_twl4030 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
 
+static int __devinit twl4030_init(void)
+{
+	return snd_soc_register_dai(&twl4030_dai);
+}
+module_init(twl4030_init);
+
+static void __exit twl4030_exit(void)
+{
+	snd_soc_unregister_dai(&twl4030_dai);
+}
+module_exit(twl4030_exit);
+
 MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
 MODULE_AUTHOR("Steve Sakoman");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 58de749..8e035b5 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -651,6 +651,18 @@ struct snd_soc_codec_device soc_codec_dev_uda134x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
 
+static int __devinit uda134x_init(void)
+{
+	return snd_soc_register_dai(&uda134x_dai);
+}
+module_init(uda134x_init);
+
+static void __exit uda134x_exit(void)
+{
+	snd_soc_unregister_dai(&uda134x_dai);
+}
+module_exit(uda134x_exit);
+
 MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
 MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 4249165..55a99b6 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -841,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
 
+static int __devinit uda1380_modinit(void)
+{
+	return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_init(uda1380_modinit);
+
+static void __exit uda1380_exit(void)
+{
+	snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_exit(uda1380_exit);
+
 MODULE_AUTHOR("Giorgio Padrin");
 MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 126c70f..a2af04b 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -889,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
 
+static int __devinit wm8510_modinit(void)
+{
+	return snd_soc_register_dai(&wm8510_dai);
+}
+module_init(wm8510_modinit);
+
+static void __exit wm8510_exit(void)
+{
+	snd_soc_unregister_dai(&wm8510_dai);
+}
+module_exit(wm8510_exit);
+
 MODULE_DESCRIPTION("ASoC WM8510 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 572a31d..391ec29 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1042,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
 
+static int __devinit wm8580_modinit(void)
+{
+	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_init(wm8580_modinit);
+
+static void __exit wm8580_exit(void)
+{
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_exit(wm8580_exit);
+
 MODULE_DESCRIPTION("ASoC WM8580 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 28f12c6..d905e25 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -568,6 +568,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8728 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
 
+static int __devinit wm8728_modinit(void)
+{
+	return snd_soc_register_dai(&wm8728_dai);
+}
+module_init(wm8728_modinit);
+
+static void __exit wm8728_exit(void)
+{
+	snd_soc_unregister_dai(&wm8728_dai);
+}
+module_exit(wm8728_exit);
+
 MODULE_DESCRIPTION("ASoC WM8728 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 403dea1..7b455a6 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -793,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
+static int __devinit wm8731_modinit(void)
+{
+	return snd_soc_register_dai(&wm8731_dai);
+}
+module_init(wm8731_modinit);
+
+static void __exit wm8731_exit(void)
+{
+	snd_soc_unregister_dai(&wm8731_dai);
+}
+module_exit(wm8731_exit);
+
 MODULE_DESCRIPTION("ASoC WM8731 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 979446f..84a6307 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -1085,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
 
+static int __devinit wm8750_modinit(void)
+{
+	return snd_soc_register_dai(&wm8750_dai);
+}
+module_init(wm8750_modinit);
+
+static void __exit wm8750_exit(void)
+{
+	snd_soc_unregister_dai(&wm8750_dai);
+}
+module_exit(wm8750_exit);
+
 MODULE_DESCRIPTION("ASoC WM8750 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 96c0453..1caca30 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1874,6 +1874,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
+static int __devinit wm8753_modinit(void)
+{
+	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_init(wm8753_modinit);
+
+static void __exit wm8753_exit(void)
+{
+	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_exit(wm8753_exit);
+
 MODULE_DESCRIPTION("ASoC WM8753 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 29cd839..02bb1c9 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1530,6 +1530,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
 
+static int __devinit wm8900_modinit(void)
+{
+	return snd_soc_register_dai(&wm8900_dai);
+}
+module_init(wm8900_modinit);
+
+static void __exit wm8900_exit(void)
+{
+	snd_soc_unregister_dai(&wm8900_dai);
+}
+module_exit(wm8900_exit);
+
 MODULE_DESCRIPTION("ASoC WM8900 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 3c83b79..5d8fe7e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1809,6 +1809,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
 
+static int __devinit wm8903_modinit(void)
+{
+	return snd_soc_register_dai(&wm8903_dai);
+}
+module_init(wm8903_modinit);
+
+static void __exit wm8903_exit(void)
+{
+	snd_soc_unregister_dai(&wm8903_dai);
+}
+module_exit(wm8903_exit);
+
 MODULE_DESCRIPTION("ASoC WM8903 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 53e6937..2979fc4 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -935,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = {
 
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
 
+static int __devinit wm8971_modinit(void)
+{
+	return snd_soc_register_dai(&wm8971_dai);
+}
+module_init(wm8971_modinit);
+
+static void __exit wm8971_exit(void)
+{
+	snd_soc_unregister_dai(&wm8971_dai);
+}
+module_exit(wm8971_exit);
+
 MODULE_DESCRIPTION("ASoC WM8971 driver");
 MODULE_AUTHOR("Lab126");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5c5128b..53e71aa 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1643,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
 
+static int __devinit wm8990_modinit(void)
+{
+	return snd_soc_register_dai(&wm8990_dai);
+}
+module_init(wm8990_modinit);
+
+static void __exit wm8990_exit(void)
+{
+	snd_soc_unregister_dai(&wm8990_dai);
+}
+module_exit(wm8990_exit);
+
 MODULE_DESCRIPTION("ASoC WM8990 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
-- 
1.5.6.5

[PATCH 16/18] ASoC: Initial framework for dynamic card instantiation

[Mark Brown]

Use the lists of platforms, platform DAIs and cards to check to see that
everything has registered. Since relationships are still specified by
direct references to the structures in the drivers and the drivers all
register everything at modprobe there should be no practical effect yet.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c |   97 ++++++++++++++++++++++++++++++++++++++++----------
 1 files changed, 78 insertions(+), 19 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ffae370..717db0e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -780,30 +780,54 @@ static int soc_resume(struct platform_device *pdev)
 #define soc_resume	NULL
 #endif
 
-/* probes a new socdev */
-static int soc_probe(struct platform_device *pdev)
-{
-	int ret = 0, i;
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_platform *platform = card->platform;
-	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-
-	/* Bodge while we push things out of socdev */
-	card->socdev = socdev;
+static void snd_soc_instantiate_card(struct snd_soc_card *card)
+{
+	struct platform_device *pdev = container_of(card->dev,
+						    struct platform_device,
+						    dev);
+	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
+	struct snd_soc_platform *platform;
+	struct snd_soc_dai *dai;
+	int i, found, ret;
+
+	if (card->instantiated)
+		return;
+
+	found = 0;
+	list_for_each_entry(platform, &platform_list, list)
+		if (card->platform == platform) {
+			found = 1;
+			break;
+		}
+	if (!found) {
+		dev_dbg(card->dev, "Platform %s not registered\n",
+			card->platform->name);
+		return;
+	}
 
-	/* Bodge while we unpick instantiation */
-	card->dev = &pdev->dev;
-	ret = snd_soc_register_card(card);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "Failed to register card\n");
-		return ret;
+	for (i = 0; i < card->num_links; i++) {
+		found = 0;
+		list_for_each_entry(dai, &dai_list, list)
+			if (card->dai_link[i].cpu_dai == dai) {
+				found = 1;
+				break;
+			}
+		if (!found) {
+			dev_dbg(card->dev, "DAI %s not registered\n",
+				card->dai_link[i].cpu_dai->name);
+			return;
+		}
 	}
 
+	/* Note that we do not current check for codec components */
+
+	dev_dbg(card->dev, "All components present, instantiating\n");
+
+	/* Found everything, bring it up */
 	if (card->probe) {
 		ret = card->probe(pdev);
 		if (ret < 0)
-			return ret;
+			return;
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -834,7 +858,9 @@ static int soc_probe(struct platform_device *pdev)
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
-	return 0;
+	card->instantiated = 1;
+
+	return;
 
 platform_err:
 	if (codec_dev->remove)
@@ -849,8 +875,38 @@ cpu_dai_err:
 
 	if (card->remove)
 		card->remove(pdev);
+}
 
-	return ret;
+/*
+ * Attempt to initialise any uninitalised cards.  Must be called with
+ * client_mutex.
+ */
+static void snd_soc_instantiate_cards(void)
+{
+	struct snd_soc_card *card;
+	list_for_each_entry(card, &card_list, list)
+		snd_soc_instantiate_card(card);
+}
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
+
+	/* Bodge while we push things out of socdev */
+	card->socdev = socdev;
+
+	/* Bodge while we unpick instantiation */
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register card\n");
+		return ret;
+	}
+
+	return 0;
 }
 
 /* removes a socdev */
@@ -1994,6 +2050,7 @@ static int snd_soc_register_card(struct snd_soc_card *card)
 
 	mutex_lock(&client_mutex);
 	list_add(&card->list, &card_list);
+	snd_soc_instantiate_cards();
 	mutex_unlock(&client_mutex);
 
 	dev_dbg(card->dev, "Registered card '%s'\n", card->name);
@@ -2039,6 +2096,7 @@ int snd_soc_register_dai(struct snd_soc_dai *dai)
 
 	mutex_lock(&client_mutex);
 	list_add(&dai->list, &dai_list);
+	snd_soc_instantiate_cards();
 	mutex_unlock(&client_mutex);
 
 	pr_debug("Registered DAI '%s'\n", dai->name);
@@ -2117,6 +2175,7 @@ int snd_soc_register_platform(struct snd_soc_platform *platform)
 
 	mutex_lock(&client_mutex);
 	list_add(&platform->list, &platform_list);
+	snd_soc_instantiate_cards();
 	mutex_unlock(&client_mutex);
 
 	pr_debug("Registered platform '%s'\n", platform->name);
-- 
1.5.6.5

[PATCH 17/18] ASoC: Wait for non-AC97 codec DAIs before instantiating

[Mark Brown]

This will allow codec drivers to be refactored to allow them to be
registered out of line with the ASoC device registration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c |   27 ++++++++++++++++++++++++++-
 1 files changed, 26 insertions(+), 1 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 717db0e..76a89eb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -788,7 +788,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
 	struct snd_soc_platform *platform;
 	struct snd_soc_dai *dai;
-	int i, found, ret;
+	int i, found, ret, ac97;
 
 	if (card->instantiated)
 		return;
@@ -805,6 +805,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		return;
 	}
 
+	ac97 = 0;
 	for (i = 0; i < card->num_links; i++) {
 		found = 0;
 		list_for_each_entry(dai, &dai_list, list)
@@ -817,8 +818,32 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 				card->dai_link[i].cpu_dai->name);
 			return;
 		}
+
+		if (card->dai_link[i].cpu_dai->ac97_control)
+			ac97 = 1;
 	}
 
+	/* If we have AC97 in the system then don't wait for the
+	 * codec.  This will need revisiting if we have to handle
+	 * systems with mixed AC97 and non-AC97 parts.  Only check for
+	 * DAIs currently; we can't do this per link since some AC97
+	 * codecs have non-AC97 DAIs.
+	 */
+	if (!ac97)
+		for (i = 0; i < card->num_links; i++) {
+			found = 0;
+			list_for_each_entry(dai, &dai_list, list)
+				if (card->dai_link[i].codec_dai == dai) {
+					found = 1;
+					break;
+				}
+			if (!found) {
+				dev_dbg(card->dev, "DAI %s not registered\n",
+					card->dai_link[i].codec_dai->name);
+				return;
+			}
+		}
+
 	/* Note that we do not current check for codec components */
 
 	dev_dbg(card->dev, "All components present, instantiating\n");
-- 
1.5.6.5

[PATCH 18/18] ASoC: Convert WM8900 to allow registration by machine code

[Mark Brown]

This makes use of the support for delayed DAI registration to allow the
WM8900 I2C device to be registered by general platform/architecture code
rather than as part of the ASoC device probe.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8900.c |   97 +++++++++-----------------------------------
 1 files changed, 20 insertions(+), 77 deletions(-)

diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 02bb1c9..34e58af 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1382,32 +1382,21 @@ priv_err:
 	return ret;
 }
 
-static struct snd_soc_device *wm8900_socdev;
+static struct i2c_client *wm8900_client;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
 static int wm8900_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8900_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8900_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "failed to initialise WM8900\n");
-	return ret;
+	wm8900_client = i2c;
+	wm8900_dai.dev = &i2c->dev;
+	return snd_soc_register_dai(&wm8900_dai);
 }
 
 static int wm8900_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	snd_soc_unregister_dai(&wm8900_dai);
+	wm8900_dai.dev = NULL;
+	wm8900_client = NULL;
 	return 0;
 }
 
@@ -1427,57 +1416,17 @@ static struct i2c_driver wm8900_i2c_driver = {
 	.id_table = wm8900_i2c_id,
 };
 
-static int wm8900_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8900_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8900_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8900", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8900_i2c_driver);
-	return -ENODEV;
-}
-#endif
-
 static int wm8900_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8900_setup_data *setup;
 	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	dev_info(&pdev->dev, "WM8900 Audio Codec\n");
+	if (!wm8900_client) {
+		dev_err(&pdev->dev, "I2C client not yet instantiated\n");
+		return -ENODEV;
+	}
 
-	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
@@ -1490,15 +1439,13 @@ static int wm8900_probe(struct platform_device *pdev)
 
 	codec->set_bias_level = wm8900_set_bias_level;
 
-	wm8900_socdev = socdev;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8900_add_i2c_device(pdev, setup);
-	}
-#else
-#error Non-I2C interfaces not yet supported
-#endif
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	codec->control_data = wm8900_client;
+
+	ret = wm8900_init(socdev);
+	if (ret != 0)
+		kfree(codec);
+
 	return ret;
 }
 
@@ -1513,10 +1460,6 @@ static int wm8900_remove(struct platform_device *pdev)
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8900_i2c_driver);
-#endif
 	kfree(codec);
 
 	return 0;
@@ -1532,13 +1475,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
 
 static int __devinit wm8900_modinit(void)
 {
-	return snd_soc_register_dai(&wm8900_dai);
+	return i2c_add_driver(&wm8900_i2c_driver);
 }
 module_init(wm8900_modinit);
 
 static void __exit wm8900_exit(void)
 {
-	snd_soc_unregister_dai(&wm8900_dai);
+	i2c_del_driver(&wm8900_i2c_driver);
 }
 module_exit(wm8900_exit);
 
-- 
1.5.6.5

Re: [PATCH 14/18] ASoC: Register platform drivers

[Takashi Iwai]

At Tue,  9 Dec 2008 14:01:38 +0000,
Mark Brown wrote:
> diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
> index 8507aa1..d6bcb4e 100644
> --- a/sound/soc/atmel/atmel-pcm.c
> +++ b/sound/soc/atmel/atmel-pcm.c
> @@ -477,6 +477,18 @@ struct snd_soc_platform atmel_soc_platform = {
>  };
>  EXPORT_SYMBOL_GPL(atmel_soc_platform);
>  
> +static int __devinit atmel_pcm_modinit(void)
> +{
> +	return snd_soc_register_platform(&atmel_soc_platform);
> +}
> +module_init(atmel_pcm_modinit);
> +
> +static void __exit atmel_pcm_exit(void)

Typo either here...

> +{
> +	snd_soc_unregister_platform(&atmel_soc_platform);
> +}
> +module_exit(atmel_pcm_modexit);

... or here.


Takashi

Re: [PATCH 12/18] ASoC: Register platform DAIs

[Takashi Iwai]

At Tue,  9 Dec 2008 14:01:36 +0000,
Mark Brown wrote:
> 
> diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
> index d9b874c..c9f02ed 100644
> --- a/sound/soc/atmel/atmel_ssc_dai.c
> +++ b/sound/soc/atmel/atmel_ssc_dai.c
> @@ -772,6 +772,17 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
>  };
>  EXPORT_SYMBOL_GPL(atmel_ssc_dai);
>  
> +static int __devinit atmel_ssc_modinit(void)
> +{
> +	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
> +}
> +module_init(snd_soc_init);

Typo.

> +
> +static void __exit snd_soc_exit(void)
> +{
> +	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
> +}
> +

No module_exit() is defined.


Takashi

Re: [PATCH 10/18] ASoC: Add card registration API

[Takashi Iwai]

At Tue,  9 Dec 2008 14:01:34 +0000,
Mark Brown wrote:
> 
> ASoC v2 allows cards, codecs and platforms to instantiate separately,
> with the overall ASoC device only being instantiated once all the
> required components have registered. As part of backporting Liam's work
> introduce an initial version of the card registration functions. At
> present these do nothing active and are internal only, they will be
> exposed to machine drivers after further backporting.  Adding this now
> allows the datastructures used for dynamic card instantiation to be
> built up gradually.
> 
> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(snip)
> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
> index 2f2a8d9..44fbd71 100644
> --- a/sound/soc/soc-core.c
> +++ b/sound/soc/soc-core.c
(snip)
> @@ -784,6 +790,14 @@ static int soc_probe(struct platform_device *pdev)
>  	/* Bodge while we push things out of socdev */
>  	card->socdev = socdev;
>  
> +	/* Bodge while we unpick instantiation */
> +	card->dev = &pdev->dev;
> +	ret = snd_soc_register_card(card);
> +	if (ret != 0) {
> +		dev_err(&pdev->dev, "Failed to register card\n");
> +		return ret;
> +	}
> +

I guess unregister call is missing in the error path of soc_probe()?


Takashi

Re: [PATCH 10/18] ASoC: Add card registration API

[Takashi Iwai]

At Tue, 09 Dec 2008 15:17:16 +0100,
I wrote:
> 
> At Tue,  9 Dec 2008 14:01:34 +0000,
> Mark Brown wrote:
> > 
> > ASoC v2 allows cards, codecs and platforms to instantiate separately,
> > with the overall ASoC device only being instantiated once all the
> > required components have registered. As part of backporting Liam's work
> > introduce an initial version of the card registration functions. At
> > present these do nothing active and are internal only, they will be
> > exposed to machine drivers after further backporting.  Adding this now
> > allows the datastructures used for dynamic card instantiation to be
> > built up gradually.
> > 
> > Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> (snip)
> > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
> > index 2f2a8d9..44fbd71 100644
> > --- a/sound/soc/soc-core.c
> > +++ b/sound/soc/soc-core.c
> (snip)
> > @@ -784,6 +790,14 @@ static int soc_probe(struct platform_device *pdev)
> >  	/* Bodge while we push things out of socdev */
> >  	card->socdev = socdev;
> >  
> > +	/* Bodge while we unpick instantiation */
> > +	card->dev = &pdev->dev;
> > +	ret = snd_soc_register_card(card);
> > +	if (ret != 0) {
> > +		dev_err(&pdev->dev, "Failed to register card\n");
> > +		return ret;
> > +	}
> > +
> 
> I guess unregister call is missing in the error path of soc_probe()?

Looking further, the patch 16/18 changes the code again, thus
the error check there will be irrelevant.  Then no need to fix here...


Takashi

Re: [PATCH 06/18] ASoC: TWL4030: DAPM mapping of the Headset outputs

[stanley.miao]

On Tue, 2008-12-09 at 14:01 +0000, Mark Brown wrote:
> From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
> 
> Adds DAPM muxing, routing for the Headset outputs.
> 
> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> ---
>  sound/soc/codecs/twl4030.c |   39 +++++++++++++++++++++++++++++++++++++++
>  1 files changed, 39 insertions(+), 0 deletions(-)
> 
<snip>
>  
>  	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
>  	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
> @@ -744,6 +775,12 @@ static const struct snd_soc_dapm_route intercon[] = {
>  	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
>  	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
>  	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
> +	/* HeadsetL */
> +	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
> +	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},

The control names "DACR1","DACR2","DACL1","DACL2" have the same name
with the widgets in the first patch. I suggest you change a different
name to avoid confusion.

Stanley.

> +	/* HeadsetR */
> +	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
> +	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
>  
>  	/* outputs */
>  	{"OUTL", NULL, "ARXL2_APGA"},
> @@ -751,6 +788,8 @@ static const struct snd_soc_dapm_route intercon[] = {
>  	{"EARPIECE", NULL, "Earpiece Mux"},
>  	{"PREDRIVEL", NULL, "PredriveL Mux"},
>  	{"PREDRIVER", NULL, "PredriveR Mux"},
> +	{"HSOL", NULL, "HeadsetL Mux"},
> +	{"HSOR", NULL, "HeadsetR Mux"},
>  
>  	/* inputs */
>  	{"ADCL", NULL, "INL"},

Re: [PATCH 06/18] ASoC: TWL4030: DAPM mapping of the Headset outputs

[Peter Ujfalusi]

On Wednesday 10 December 2008 10:27:44 ext stanley.miao wrote:
> On Tue, 2008-12-09 at 14:01 +0000, Mark Brown wrote:
> > From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
> >
> > Adds DAPM muxing, routing for the Headset outputs.
> >
> > Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
> > Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> > ---
> >  sound/soc/codecs/twl4030.c |   39
> > +++++++++++++++++++++++++++++++++++++++ 1 files changed, 39
> > insertions(+), 0 deletions(-)
>
> <snip>
>
> >  	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
> >  	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
> > @@ -744,6 +775,12 @@ static const struct snd_soc_dapm_route intercon[] =
> > { {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
> >  	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
> >  	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
> > +	/* HeadsetL */
> > +	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
> > +	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
>
> The control names "DACR1","DACR2","DACL1","DACL2" have the same name
> with the widgets in the first patch. I suggest you change a different
> name to avoid confusion.

Yes, this can be confusing... I'm thinking of changing either the control 
names like this:
"DACR1" -> "DAC Right1" or "Audio Out R1" or "Out R1"
"DACR2" -> "DAC Right2" or "Audio Out R2" or "Out R1"
...
I think the "DAC Right1" type reflects better the functionality of the mux...

or make the same name change, but with the SND_SOC_DAPM_DAC

>
> Stanley.

-- 
Péter

>
> > +	/* HeadsetR */
> > +	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
> > +	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
> >
> >  	/* outputs */
> >  	{"OUTL", NULL, "ARXL2_APGA"},
> > @@ -751,6 +788,8 @@ static const struct snd_soc_dapm_route intercon[] = {
> >  	{"EARPIECE", NULL, "Earpiece Mux"},
> >  	{"PREDRIVEL", NULL, "PredriveL Mux"},
> >  	{"PREDRIVER", NULL, "PredriveR Mux"},
> > +	{"HSOL", NULL, "HeadsetL Mux"},
> > +	{"HSOR", NULL, "HeadsetR Mux"},
> >
> >  	/* inputs */
> >  	{"ADCL", NULL, "INL"},

Re: [PATCH 00/18] ASoC updates

[Takashi Iwai]

At Tue, 9 Dec 2008 14:01:01 +0000,
Mark Brown wrote:
> 
> The following changes since commit 28a1d869560a49d960ba2a3b450ec965712e5560:
>   Daniel Mack (1):
>         ASoC: tlv320aic3x: control additions and cleanups
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, now I pulled this.  But it seems that there are few build
problems.


Takashi

Re: [PATCH 00/18] ASoC updates

[Mark Brown]

On Tue, Dec 09, 2008 at 03:39:08PM +0100, Takashi Iwai wrote:

> Thanks, now I pulled this.  But it seems that there are few build
> problems.

Any beyond the Atmel ones you pointed out already for which:

The following changes since commit f0752331b89ce79063f765545dd7dd5f49d9a713:
  Mark Brown (1):
        ASoC: Convert WM8900 to allow registration by machine code

are available in the git repository at:

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

Mark Brown (1):
      ASoC: Fix typos in Atmel module registration

 sound/soc/atmel/atmel-pcm.c     |    2 +-
 sound/soc/atmel/atmel_ssc_dai.c |    5 +++--
 2 files changed, 4 insertions(+), 3 deletions(-)

[PATCH] ASoC: Fix typos in Atmel module registration

[Mark Brown]

I wish I had boards which work with unmodified kernels :/

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c     |    2 +-
 sound/soc/atmel/atmel_ssc_dai.c |    5 +++--
 2 files changed, 4 insertions(+), 3 deletions(-)

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index d6bcb4e..027eb13 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -483,7 +483,7 @@ static int __devinit atmel_pcm_modinit(void)
 }
 module_init(atmel_pcm_modinit);
 
-static void __exit atmel_pcm_exit(void)
+static void __exit atmel_pcm_modexit(void)
 {
 	snd_soc_unregister_platform(&atmel_soc_platform);
 }
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index c9f02ed..87904b6 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -776,12 +776,13 @@ static int __devinit atmel_ssc_modinit(void)
 {
 	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
 }
-module_init(snd_soc_init);
+module_init(atmel_ssc_modinit);
 
-static void __exit snd_soc_exit(void)
+static void __exit atmel_ssc_modexit(void)
 {
 	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
 }
+module_exit(atmel_ssc_modexit);
 
 /* Module information */
 MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
-- 
1.5.6.5

Re: [PATCH 00/18] ASoC updates

[Takashi Iwai]

At Tue, 9 Dec 2008 15:01:11 +0000,
Mark Brown wrote:
> 
> On Tue, Dec 09, 2008 at 03:39:08PM +0100, Takashi Iwai wrote:
> 
> > Thanks, now I pulled this.  But it seems that there are few build
> > problems.
> 
> Any beyond the Atmel ones you pointed out already for which:
> 
> The following changes since commit f0752331b89ce79063f765545dd7dd5f49d9a713:
>   Mark Brown (1):
>         ASoC: Convert WM8900 to allow registration by machine code
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled this one now.


Takashi

> 
> Mark Brown (1):
>       ASoC: Fix typos in Atmel module registration
> 
>  sound/soc/atmel/atmel-pcm.c     |    2 +-
>  sound/soc/atmel/atmel_ssc_dai.c |    5 +++--
>  2 files changed, 4 insertions(+), 3 deletions(-)
>