From f9a3fba2ce8622977c5373d2449eb71705613721 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Wed, 31 Dec 2008 10:08:37 +0200
Subject: ASoC: TWL4030: Make the enum filter generic for twl4030

Modify the enum filter to more generic that it will filter
out the enums with text "Invalid".
The enum filter also required for the capture path.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 30 ++++++++++++++----------------
 1 file changed, 14 insertions(+), 16 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 51848880504a..2c279cd8deb5 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -298,25 +298,23 @@ static const struct soc_enum twl4030_handsfreer_enum =
 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,
+/*
+ * This function filters out the non valid mux settings, named as "Invalid"
+ * in the enum texts.
+ * Just refuse to set an invalid mux mode.
+ */
+static int twl4030_enum_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;
+	val = w->value >> e->shift_l;
+	if (!strcmp("Invalid", e->texts[val])) {
+		printk(KERN_WARNING "Invalid MUX setting on 0x%02x (%d)\n",
+			e->reg, val);
+		ret = -1;
 	}
 
 	return ret;
@@ -810,14 +808,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	/* Output MUX controls */
 	/* Earpiece */
 	SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_earpiece_control, outmixer_event,
+		&twl4030_dapm_earpiece_control, twl4030_enum_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,
+		&twl4030_dapm_predrivel_control, twl4030_enum_event,
 		SND_SOC_DAPM_PRE_REG),
 	SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_predriver_control, outmixer_event,
+		&twl4030_dapm_predriver_control, twl4030_enum_event,
 		SND_SOC_DAPM_PRE_REG),
 	/* HeadsetL/R */
 	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-- 
cgit v1.2.3


From 276c62225a7c98737510483dcaec6af7e7965389 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi
Date: Wed, 31 Dec 2008 10:08:38 +0200
Subject: ASoC: TWL4030: DAPM based capture implementation

This patch adds DAPM implementaion for the capture path
on twlx030.

TWL has two physical ADC and two digital microphone (stereo) connections.
The CPU interface has four microphone channels.
For simplicity the microphone channel paths are named as:
TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid
TX2 (Left/Right)

Input routing (simplified version):
There is two levels of mux settings for TWL in input path:
Analog input mux:
 ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic}
 ADCR <- {Off, Sub mic, AUXR}

Analog/Digital mux:
TX1 Analog mode:
 TX1L <- ADCL
 TX1R <- ADCR
TX1 Digital mode:
 TX1L <- Digimic0 (Left)
 TX1R <- Digimic0 (Right)

TX2 Analog mode:
 TX2L <- ADCL
 TX2R <- ADCR
TX2 Digital mode:
 TX2L <- Digimic1 (Left)
 TX2R <- Digimic1 (Right)

The patch provides the following user controls for the capture path:
Mux settings:
"TX1 Capture Route": {Analog, Digimic0}
"TX2 Capture Route": {Analog, Digimic1}
"Analog Left Capture Route":  {Off, Main Mic, Headset Mic, AUXL, Carkit Mic}
"Analog Right Capture Route": {Off, Sub Mic, AUXR}

Volume/Gain controls:
"TX1 Digital Capture Volume": Stereo gain control for TX1 path
"TX2 Digital Capture Volume": Stereo gain control for TX2 path
"Analog Capture Volume":      Stereo gain control for the analog path only

Important things for the board files:
Microphone bias:
"Mic Bias 1":       Bias for Main mic or for digimic0 (analog or digital path)
"Mic Bias 2":       Bias for Sub mic or for digimic1 (analog or digital path)
"Headset Mic Bias": Bias for Headset mic

When the routing configured correctly only the needed components will be
powered/enabled.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 347 +++++++++++++++++++++++----------------------
 sound/soc/codecs/twl4030.h |   7 +
 2 files changed, 182 insertions(+), 172 deletions(-)

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 2c279cd8deb5..31e44e346dc8 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -298,6 +298,55 @@ static const struct soc_enum twl4030_handsfreer_enum =
 static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
 SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
 
+/* Left analog microphone selection */
+static const char *twl4030_analoglmic_texts[] =
+		{"Off", "Main mic", "Headset mic", "Invalid", "AUXL",
+		 "Invalid", "Invalid", "Invalid", "Carkit mic"};
+
+static const struct soc_enum twl4030_analoglmic_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0,
+			ARRAY_SIZE(twl4030_analoglmic_texts),
+			twl4030_analoglmic_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
+SOC_DAPM_ENUM("Route", twl4030_analoglmic_enum);
+
+/* Right analog microphone selection */
+static const char *twl4030_analogrmic_texts[] =
+		{"Off", "Sub mic", "Invalid", "Invalid", "AUXR"};
+
+static const struct soc_enum twl4030_analogrmic_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0,
+			ARRAY_SIZE(twl4030_analogrmic_texts),
+			twl4030_analogrmic_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
+SOC_DAPM_ENUM("Route", twl4030_analogrmic_enum);
+
+/* TX1 L/R Analog/Digital microphone selection */
+static const char *twl4030_micpathtx1_texts[] =
+		{"Analog", "Digimic0"};
+
+static const struct soc_enum twl4030_micpathtx1_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0,
+			ARRAY_SIZE(twl4030_micpathtx1_texts),
+			twl4030_micpathtx1_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control =
+SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
+
+/* TX2 L/R Analog/Digital microphone selection */
+static const char *twl4030_micpathtx2_texts[] =
+		{"Analog", "Digimic1"};
+
+static const struct soc_enum twl4030_micpathtx2_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2,
+			ARRAY_SIZE(twl4030_micpathtx2_texts),
+			twl4030_micpathtx2_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
+SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
+
 /*
  * This function filters out the non valid mux settings, named as "Invalid"
  * in the enum texts.
@@ -320,6 +369,36 @@ static int twl4030_enum_event(struct snd_soc_dapm_widget *w,
 	return ret;
 }
 
+static int micpath_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+	unsigned char adcmicsel, micbias_ctl;
+
+	adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL);
+	micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL);
+	/* Prepare the bits for the given TX path:
+	 * shift_l == 0: TX1 microphone path
+	 * shift_l == 2: TX2 microphone path */
+	if (e->shift_l) {
+		/* TX2 microphone path */
+		if (adcmicsel & TWL4030_TX2IN_SEL)
+			micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */
+		else
+			micbias_ctl &= ~TWL4030_MICBIAS2_CTL;
+	} else {
+		/* TX1 microphone path */
+		if (adcmicsel & TWL4030_TX1IN_SEL)
+			micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */
+		else
+			micbias_ctl &= ~TWL4030_MICBIAS1_CTL;
+	}
+
+	twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl);
+
+	return 0;
+}
+
 static int handsfree_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
@@ -501,162 +580,6 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
 	return err;
 }
 
-static int twl4030_get_left_input(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = kcontrol->private_data;
-	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
-	int result = 0;
-
-	/* one bit must be set a time */
-	reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
-			| TWL4030_MAINMIC_EN;
-	if (reg != 0) {
-		result++;
-		while ((reg & 1) == 0) {
-			result++;
-			reg >>= 1;
-		}
-	}
-
-	ucontrol->value.integer.value[0] = result;
-	return 0;
-}
-
-static int twl4030_put_left_input(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = kcontrol->private_data;
-	int value = ucontrol->value.integer.value[0];
-	u8 anamicl, micbias, avadc_ctl;
-
-	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
-	anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
-			| TWL4030_MAINMIC_EN);
-	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
-	micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN);
-	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
-
-	switch (value) {
-	case 1:
-		anamicl |= TWL4030_MAINMIC_EN;
-		micbias |= TWL4030_MICBIAS1_EN;
-		break;
-	case 2:
-		anamicl |= TWL4030_HSMIC_EN;
-		micbias |= TWL4030_HSMICBIAS_EN;
-		break;
-	case 3:
-		anamicl |= TWL4030_AUXL_EN;
-		break;
-	case 4:
-		anamicl |= TWL4030_CKMIC_EN;
-		break;
-	default:
-		break;
-	}
-
-	/* If some input is selected, enable amp and ADC */
-	if (value != 0) {
-		anamicl |= TWL4030_MICAMPL_EN;
-		avadc_ctl |= TWL4030_ADCL_EN;
-	} else {
-		anamicl &= ~TWL4030_MICAMPL_EN;
-		avadc_ctl &= ~TWL4030_ADCL_EN;
-	}
-
-	twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl);
-	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
-	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
-
-	return 1;
-}
-
-static int twl4030_get_right_input(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = kcontrol->private_data;
-	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
-	int value = 0;
-
-	reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN;
-	switch (reg) {
-	case TWL4030_SUBMIC_EN:
-		value = 1;
-		break;
-	case TWL4030_AUXR_EN:
-		value = 2;
-		break;
-	default:
-		break;
-	}
-
-	ucontrol->value.integer.value[0] = value;
-	return 0;
-}
-
-static int twl4030_put_right_input(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = kcontrol->private_data;
-	int value = ucontrol->value.integer.value[0];
-	u8 anamicr, micbias, avadc_ctl;
-
-	anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
-	anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN);
-	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
-	micbias &= ~TWL4030_MICBIAS2_EN;
-	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
-
-	switch (value) {
-	case 1:
-		anamicr |= TWL4030_SUBMIC_EN;
-		micbias |= TWL4030_MICBIAS2_EN;
-		break;
-	case 2:
-		anamicr |= TWL4030_AUXR_EN;
-		break;
-	default:
-		break;
-	}
-
-	if (value != 0) {
-		anamicr |= TWL4030_MICAMPR_EN;
-		avadc_ctl |= TWL4030_ADCR_EN;
-	} else {
-		anamicr &= ~TWL4030_MICAMPR_EN;
-		avadc_ctl &= ~TWL4030_ADCR_EN;
-	}
-
-	twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr);
-	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
-	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
-
-	return 1;
-}
-
-static const char *twl4030_left_in_sel[] = {
-	"None",
-	"Main Mic",
-	"Headset Mic",
-	"Line In",
-	"Carkit Mic",
-};
-
-static const char *twl4030_right_in_sel[] = {
-	"None",
-	"Sub Mic",
-	"Line In",
-};
-
-static const struct soc_enum twl4030_left_input_mux =
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel),
-		twl4030_left_in_sel);
-
-static const struct soc_enum twl4030_right_input_mux =
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel),
-		twl4030_right_in_sel);
-
 /*
  * FGAIN volume control:
  * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
@@ -739,18 +662,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
 
 	/* Common capture gain controls */
-	SOC_DOUBLE_R_TLV("Capture Volume",
+	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
 		TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
 		0, 0x1f, 0, digital_capture_tlv),
+	SOC_DOUBLE_R_TLV("TX2 Digital Capture Volume",
+		TWL4030_REG_AVTXL2PGA, TWL4030_REG_AVTXR2PGA,
+		0, 0x1f, 0, digital_capture_tlv),
 
-	SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN,
+	SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
 		0, 3, 5, 0, input_gain_tlv),
-
-	/* Input source controls */
-	SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux,
-		twl4030_get_left_input, twl4030_put_left_input),
-	SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux,
-		twl4030_get_right_input, twl4030_put_right_input),
 };
 
 /* add non dapm controls */
@@ -770,9 +690,19 @@ static int twl4030_add_controls(struct snd_soc_codec *codec)
 }
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
-	SND_SOC_DAPM_INPUT("INL"),
-	SND_SOC_DAPM_INPUT("INR"),
-
+	/* Left channel inputs */
+	SND_SOC_DAPM_INPUT("MAINMIC"),
+	SND_SOC_DAPM_INPUT("HSMIC"),
+	SND_SOC_DAPM_INPUT("AUXL"),
+	SND_SOC_DAPM_INPUT("CARKITMIC"),
+	/* Right channel inputs */
+	SND_SOC_DAPM_INPUT("SUBMIC"),
+	SND_SOC_DAPM_INPUT("AUXR"),
+	/* Digital microphones (Stereo) */
+	SND_SOC_DAPM_INPUT("DIGIMIC0"),
+	SND_SOC_DAPM_INPUT("DIGIMIC1"),
+
+	/* Outputs */
 	SND_SOC_DAPM_OUTPUT("OUTL"),
 	SND_SOC_DAPM_OUTPUT("OUTR"),
 	SND_SOC_DAPM_OUTPUT("EARPIECE"),
@@ -835,8 +765,50 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		&twl4030_dapm_handsfreer_control, handsfree_event,
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 
-	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
+	/* Introducing four virtual ADC, since TWL4030 have four channel for
+	   capture */
+	SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture",
+		SND_SOC_NOPM, 0, 0),
+
+	/* Analog/Digital mic path selection.
+	   TX1 Left/Right: either analog Left/Right or Digimic0
+	   TX2 Left/Right: either analog Left/Right or Digimic1 */
+	SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_micpathtx1_control, micpath_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
+		SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_micpathtx2_control, micpath_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
+		SND_SOC_DAPM_POST_REG),
+
+	/* Analog input muxes with power switch for the physical ADCL/R */
+	SND_SOC_DAPM_MUX_E("Analog Left Capture Route",
+		TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control,
+		twl4030_enum_event, SND_SOC_DAPM_PRE_REG),
+	SND_SOC_DAPM_MUX_E("Analog Right Capture Route",
+		TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control,
+		twl4030_enum_event, SND_SOC_DAPM_PRE_REG),
+
+	SND_SOC_DAPM_PGA("Analog Left Amplifier",
+		TWL4030_REG_ANAMICL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Analog Right Amplifier",
+		TWL4030_REG_ANAMICR, 4, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Digimic0 Enable",
+		TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Digimic1 Enable",
+		TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
+	SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
@@ -892,9 +864,39 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HFL", NULL, "HandsfreeL Mux"},
 	{"HFR", NULL, "HandsfreeR Mux"},
 
-	/* inputs */
-	{"ADCL", NULL, "INL"},
-	{"ADCR", NULL, "INR"},
+	/* Capture path */
+	{"Analog Left Capture Route", "Main mic", "MAINMIC"},
+	{"Analog Left Capture Route", "Headset mic", "HSMIC"},
+	{"Analog Left Capture Route", "AUXL", "AUXL"},
+	{"Analog Left Capture Route", "Carkit mic", "CARKITMIC"},
+
+	{"Analog Right Capture Route", "Sub mic", "SUBMIC"},
+	{"Analog Right Capture Route", "AUXR", "AUXR"},
+
+	{"Analog Left Amplifier", NULL, "Analog Left Capture Route"},
+	{"Analog Right Amplifier", NULL, "Analog Right Capture Route"},
+
+	{"Digimic0 Enable", NULL, "DIGIMIC0"},
+	{"Digimic1 Enable", NULL, "DIGIMIC1"},
+
+	/* TX1 Left capture path */
+	{"TX1 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
+	/* TX1 Right capture path */
+	{"TX1 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
+	/* TX2 Left capture path */
+	{"TX2 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
+	/* TX2 Right capture path */
+	{"TX2 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
+
+	{"ADC Virtual Left1", NULL, "TX1 Capture Route"},
+	{"ADC Virtual Right1", NULL, "TX1 Capture Route"},
+	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
+	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
+
 };
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
@@ -921,6 +923,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	twl4030_write(codec, TWL4030_REG_ANAMICL,
 		anamicl | TWL4030_CNCL_OFFSET_START);
 
+
 	/* wait for offset cancellation to complete */
 	do {
 		/* this takes a little while, so don't slam i2c */
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 54615c76802b..442e5a828617 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -147,6 +147,13 @@
 #define TWL4030_AVADC_CLK_PRIORITY	0x04
 #define TWL4030_ADCR_EN			0x02
 
+/* TWL4030_REG_ADCMICSEL (0x08) Fields */
+
+#define TWL4030_DIGMIC1_EN		0x08
+#define TWL4030_TX2IN_SEL		0x04
+#define TWL4030_DIGMIC0_EN		0x02
+#define TWL4030_TX1IN_SEL		0x01
+
 /* AUDIO_IF (0x0E) Fields */
 
 #define TWL4030_AIF_SLAVE_EN		0x80
-- 
cgit v1.2.3


From 42a6e66f1e40a930d093c33ba0bb9d8d8e4555ed Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Mon, 29 Dec 2008 11:23:02 +0100
Subject: ALSA: sound/usb: use USB API functions rather than constants

This set of patches introduces calls to the following set of functions:

usb_endpoint_dir_in(epd)
usb_endpoint_dir_out(epd)
usb_endpoint_is_bulk_in(epd)
usb_endpoint_is_bulk_out(epd)
usb_endpoint_is_int_in(epd)
usb_endpoint_is_int_out(epd)
usb_endpoint_num(epd)
usb_endpoint_type(epd)
usb_endpoint_xfer_bulk(epd)
usb_endpoint_xfer_control(epd)
usb_endpoint_xfer_int(epd)
usb_endpoint_xfer_isoc(epd)

In some cases, introducing one of these functions is not possible, and it
just replaces an explicit integer value by one of the following constants:

USB_ENDPOINT_XFER_BULK
USB_ENDPOINT_XFER_CONTROL
USB_ENDPOINT_XFER_INT
USB_ENDPOINT_XFER_ISOC

An extract of the semantic patch that makes these changes is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@r1@ struct usb_endpoint_descriptor *epd; @@

- ((epd->bmAttributes & \(USB_ENDPOINT_XFERTYPE_MASK\|3\)) ==
- \(USB_ENDPOINT_XFER_CONTROL\|0\))
+ usb_endpoint_xfer_control(epd)

@r5@ struct usb_endpoint_descriptor *epd; @@

- ((epd->bEndpointAddress & \(USB_ENDPOINT_DIR_MASK\|0x80\)) ==
-  \(USB_DIR_IN\|0x80\))
+ usb_endpoint_dir_in(epd)

@inc@
@@

#include <linux/usb.h>

@depends on !inc && (r1||r5)@
@@

+ #include <linux/usb.h>
  #include <linux/usb/...>
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbmidi.c  | 42 +++++++++++++++++++++---------------------
 sound/usb/usbmixer.c |  6 +++---
 2 files changed, 24 insertions(+), 24 deletions(-)

diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 6d9f9b135c62..3a9a9fecd292 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1392,8 +1392,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
 	for (i = 0; i < intfd->bNumEndpoints; ++i) {
 		hostep = &hostif->endpoint[i];
 		ep = get_ep_desc(hostep);
-		if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK &&
-		    (ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+		if (usb_endpoint_type(ep) != USB_ENDPOINT_XFER_BULK &&
+		    usb_endpoint_type(ep) != USB_ENDPOINT_XFER_INT)
 			continue;
 		ms_ep = (struct usb_ms_endpoint_descriptor*)hostep->extra;
 		if (hostep->extralen < 4 ||
@@ -1401,15 +1401,15 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
 		    ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
 		    ms_ep->bDescriptorSubtype != MS_GENERAL)
 			continue;
-		if ((ep->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_OUT) {
+		if (usb_endpoint_dir_out(ep)) {
 			if (endpoints[epidx].out_ep) {
 				if (++epidx >= MIDI_MAX_ENDPOINTS) {
 					snd_printk(KERN_WARNING "too many endpoints\n");
 					break;
 				}
 			}
-			endpoints[epidx].out_ep = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-			if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+			endpoints[epidx].out_ep = usb_endpoint_num(ep);
+			if (usb_endpoint_xfer_int(ep))
 				endpoints[epidx].out_interval = ep->bInterval;
 			else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW)
 				/*
@@ -1428,8 +1428,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
 					break;
 				}
 			}
-			endpoints[epidx].in_ep = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-			if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+			endpoints[epidx].in_ep = usb_endpoint_num(ep);
+			if (usb_endpoint_xfer_int(ep))
 				endpoints[epidx].in_interval = ep->bInterval;
 			else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW)
 				endpoints[epidx].in_interval = 1;
@@ -1495,20 +1495,20 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi,
 
 	for (i = 0; i < intfd->bNumEndpoints; ++i) {
 		epd = get_endpoint(hostif, i);
-		if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK &&
-		    (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+		if (usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK &&
+		    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_INT)
 			continue;
 		if (out_eps < max_endpoints &&
-		    (epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_OUT) {
-			endpoint[out_eps].out_ep = epd->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-			if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+		    usb_endpoint_dir_out(epd)) {
+			endpoint[out_eps].out_ep = usb_endpoint_num(epd);
+			if (usb_endpoint_xfer_int(epd))
 				endpoint[out_eps].out_interval = epd->bInterval;
 			++out_eps;
 		}
 		if (in_eps < max_endpoints &&
-		    (epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_IN) {
-			endpoint[in_eps].in_ep = epd->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-			if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+		    usb_endpoint_dir_in(epd)) {
+			endpoint[in_eps].in_ep = usb_endpoint_num(epd);
+			if (usb_endpoint_xfer_int(epd))
 				endpoint[in_eps].in_interval = epd->bInterval;
 			++in_eps;
 		}
@@ -1607,21 +1607,21 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
 	}
 
 	epd = get_endpoint(hostif, 0);
-	if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_IN ||
-	    (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) {
+	if (usb_endpoint_dir_out(epd) ||
+	    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_INT) {
 		snd_printdd(KERN_ERR "endpoint[0] isn't interrupt\n");
 		return -ENXIO;
 	}
 	epd = get_endpoint(hostif, 2);
-	if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_OUT ||
-	    (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK) {
+	if (usb_endpoint_dir_in(epd) ||
+	    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK) {
 		snd_printdd(KERN_ERR "endpoint[2] isn't bulk output\n");
 		return -ENXIO;
 	}
 	if (endpoint->out_cables > 0x0001) {
 		epd = get_endpoint(hostif, 4);
-		if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_OUT ||
-		    (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK) {
+		if (usb_endpoint_dir_in(epd) ||
+		    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK) {
 			snd_printdd(KERN_ERR "endpoint[4] isn't bulk output\n");
 			return -ENXIO;
 		}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index a49246113e75..9ce626f0e36b 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1755,11 +1755,11 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 	if (get_iface_desc(hostif)->bNumEndpoints < 1)
 		return 0;
 	ep = get_endpoint(hostif, 0);
-	if ((ep->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_IN ||
-	    (ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+	if (usb_endpoint_dir_out(ep) ||
+	    usb_endpoint_type(ep) != USB_ENDPOINT_XFER_INT)
 		return 0;
 
-	epnum = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+	epnum = usb_endpoint_num(ep);
 	buffer_length = le16_to_cpu(ep->wMaxPacketSize);
 	transfer_buffer = kmalloc(buffer_length, GFP_KERNEL);
 	if (!transfer_buffer)
-- 
cgit v1.2.3


From 3fea2cb0451b9009af32d1418ea77cc674fe7e02 Mon Sep 17 00:00:00 2001
From: Wu Fengguang
Date: Fri, 26 Dec 2008 12:20:43 +0800
Subject: ALSA: hda - fix name for ALC1200

Move the more specific preset for ALC1200 above the general one for
ALC888, so that it will have the chance to get matched and selected.

Reported-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0bd4e6bf354d..69a251bb6b06 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -16638,9 +16638,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
 	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 },
-	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
 	  .patch = patch_alc883 },
+	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
 	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
 	{} /* terminator */
 };
-- 
cgit v1.2.3


From 06bf3e15f64aacfb068fed5002b6544f870cc638 Mon Sep 17 00:00:00 2001
From: Chris Bagwell
Date: Thu, 1 Jan 2009 10:32:08 +0100
Subject: LSA: hda - Add HP Acacia detection

Add automatic mapping of HP Acacia motherboards to 3stack-hp.  Allows
for greater then 2 channel audio by enabling Channel Mode option in mixer.

Motherboard specs:
http://h10025.www1.hp.com/ewfrf/wc/document?docname=c01321559&lc=en&dlc=en&cc=us&product=3829353&os=2093&lang=en#

Signed-off-by: Chris Bagwell <chris at cnpbagwell dot com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 69a251bb6b06..9065ebf9c065 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8467,6 +8467,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
-- 
cgit v1.2.3


From 9bef6489d72abd8f598aede92be3854a69324c50 Mon Sep 17 00:00:00 2001
From: Stephen Ware
Date: Wed, 31 Dec 2008 14:39:23 -0800
Subject: ASoC: Fix pxa2xx-pcm checks for invalid DMA channels

Set the invalid dma channel to -1 (and check properly for it) in
pxa2xx_pcm_hw_free().  Was assuming 0 is an invalid channel number but 0
is a valid pxa dma channel num.

Signed-off-by: stephen <stephen.ware@eqware.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/pxa2xx-pcm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index c670d08e7c9e..53b9fb127a6d 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -61,9 +61,9 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	__pxa2xx_pcm_hw_free(substream);
 
-	if (prtd->dma_ch) {
+	if (prtd->dma_ch >= 0) {
 		pxa_free_dma(prtd->dma_ch);
-		prtd->dma_ch = 0;
+		prtd->dma_ch = -1;
 	}
 
 	return 0;
-- 
cgit v1.2.3


From ac11a2b35cc25c77d28218aaf60e7f7c6c7ee5d3 Mon Sep 17 00:00:00 2001
From: Mark Brown
Date: Thu, 1 Jan 2009 12:18:17 +0000
Subject: ASoC: Clean up kerneldoc warnings

Almost all parameters that have been misnamed in the comments.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 46 ++++++++++++++++++++++++----------------------
 sound/soc/soc-dapm.c | 10 +++++-----
 2 files changed, 29 insertions(+), 27 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b098c0b4c584..f73c1341437c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1300,6 +1300,8 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
 /**
  * snd_soc_new_pcms - create new sound card and pcms
  * @socdev: the SoC audio device
+ * @idx: ALSA card index
+ * @xid: card identification
  *
  * Create a new sound card based upon the codec and interface pcms.
  *
@@ -1472,7 +1474,7 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
  * snd_soc_cnew - create new control
  * @_template: control template
  * @data: control private data
- * @lnng_name: control long name
+ * @long_name: control long name
  *
  * Create a new mixer control from a template control.
  *
@@ -1522,7 +1524,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
 /**
  * snd_soc_get_enum_double - enumerated double mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a double enumerated mixer.
  *
@@ -1551,7 +1553,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
 /**
  * snd_soc_put_enum_double - enumerated double mixer put callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a double enumerated mixer.
  *
@@ -1668,7 +1670,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
 /**
  * snd_soc_get_volsw - single mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a single mixer control.
  *
@@ -1707,7 +1709,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
 /**
  * snd_soc_put_volsw - single mixer put callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a single mixer control.
  *
@@ -1775,7 +1777,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
 /**
  * snd_soc_get_volsw_2r - double mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a double mixer control that spans 2 registers.
  *
@@ -1812,7 +1814,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
 /**
  * snd_soc_put_volsw_2r - double mixer set callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a double mixer control that spans 2 registers.
  *
@@ -1882,7 +1884,7 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
 /**
  * snd_soc_get_volsw_s8 - signed mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a signed mixer control.
  *
@@ -1909,7 +1911,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
 /**
  * snd_soc_put_volsw_sgn - signed mixer put callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a signed mixer control.
  *
@@ -1954,7 +1956,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 /**
  * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
  * @dai: DAI
- * @clk_id: DAI specific clock divider ID
+ * @div_id: DAI specific clock divider ID
  * @div: new clock divisor.
  *
  * Configures the clock dividers. This is used to derive the best DAI bit and
@@ -2060,7 +2062,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 /**
  * snd_soc_register_card - Register a card with the ASoC core
  *
- * @param card Card to register
+ * @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
@@ -2087,7 +2089,7 @@ static int snd_soc_register_card(struct snd_soc_card *card)
 /**
  * snd_soc_unregister_card - Unregister a card with the ASoC core
  *
- * @param card Card to unregister
+ * @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
@@ -2107,7 +2109,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
- * @param dai DAI to register
+ * @dai: DAI to register
  */
 int snd_soc_register_dai(struct snd_soc_dai *dai)
 {
@@ -2134,7 +2136,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dai);
 /**
  * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
  *
- * @param dai DAI to unregister
+ * @dai: DAI to unregister
  */
 void snd_soc_unregister_dai(struct snd_soc_dai *dai)
 {
@@ -2149,8 +2151,8 @@ 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
+ * @dai: Array of DAIs to register
+ * @count: Number of DAIs
  */
 int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
 {
@@ -2175,8 +2177,8 @@ 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
+ * @dai: Array of DAIs to unregister
+ * @count: Number of DAIs
  */
 void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
 {
@@ -2190,7 +2192,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
 /**
  * snd_soc_register_platform - Register a platform with the ASoC core
  *
- * @param platform platform to register
+ * @platform: platform to register
  */
 int snd_soc_register_platform(struct snd_soc_platform *platform)
 {
@@ -2213,7 +2215,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform);
 /**
  * snd_soc_unregister_platform - Unregister a platform from the ASoC core
  *
- * @param platform platform to unregister
+ * @platform: platform to unregister
  */
 void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 {
@@ -2228,7 +2230,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
 /**
  * snd_soc_register_codec - Register a codec with the ASoC core
  *
- * @param codec codec to register
+ * @codec: codec to register
  */
 int snd_soc_register_codec(struct snd_soc_codec *codec)
 {
@@ -2255,7 +2257,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec);
 /**
  * snd_soc_unregister_codec - Unregister a codec from the ASoC core
  *
- * @param codec codec to unregister
+ * @codec: codec to unregister
  */
 void snd_soc_unregister_codec(struct snd_soc_codec *codec)
 {
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8863eddbac02..6c79ca6df0bf 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1077,7 +1077,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
 /**
  * snd_soc_dapm_get_volsw - dapm mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a dapm mixer control.
  *
@@ -1122,7 +1122,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
 /**
  * snd_soc_dapm_put_volsw - dapm mixer set callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a dapm mixer control.
  *
@@ -1193,7 +1193,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
 /**
  * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to get the value of a dapm enumerated double mixer control.
  *
@@ -1221,7 +1221,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
 /**
  * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
  * @kcontrol: mixer control
- * @uinfo: control element information
+ * @ucontrol: control element information
  *
  * Callback to set the value of a dapm enumerated double mixer control.
  *
@@ -1419,7 +1419,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 
 /**
  * snd_soc_dapm_enable_pin - enable pin.
- * @snd_soc_codec: SoC codec
+ * @codec: SoC codec
  * @pin: pin name
  *
  * Enables input/output pin and it's parents or children widgets iff there is
-- 
cgit v1.2.3


From f4e9749f451747f7cdd334eae951357f839c57f2 Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Thu, 1 Jan 2009 18:14:35 +0100
Subject: ALSA: Use usb_set/get_intfdata

Use the USB functions usb_get_intfdata and usb_set_intfdata instead of
dev_get_drvdata and dev_set_drvdata, respectively.

The semantic patch that makes this change for the usb_get_intfdata case is
as follows: (http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@header@
@@

#include <linux/usb.h>

@same depends on header@
position p;
@@

usb_get_intfdata@p(...) { ... }

@depends on header@
position _p!=same.p;
identifier _f;
struct usb_interface*intf;
@@

_f@_p(...) { <+...
- dev_get_drvdata(&intf->dev)
+ usb_get_intfdata(intf)
...+> }
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/caiaq/caiaq-device.c | 4 ++--
 sound/usb/usbaudio.c           | 8 ++++----
 sound/usb/usx2y/us122l.c       | 4 ++--
 sound/usb/usx2y/usbusx2y.c     | 4 ++--
 4 files changed, 10 insertions(+), 10 deletions(-)

diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index b143ef7152f7..a62500e387a6 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -446,7 +446,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
 	if (!card)
 		return -ENOMEM;
 			
-	dev_set_drvdata(&intf->dev, card);
+	usb_set_intfdata(intf, card);
 	ret = init_card(caiaqdev(card));
 	if (ret < 0) {
 		log("unable to init card! (ret=%d)\n", ret);
@@ -460,7 +460,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
 static void snd_disconnect(struct usb_interface *intf)
 {
 	struct snd_usb_caiaqdev *dev;
-	struct snd_card *card = dev_get_drvdata(&intf->dev);
+	struct snd_card *card = usb_get_intfdata(intf);
 
 	debug("%s(%p)\n", __func__, intf);
 
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index bbd70d5814a0..c709b9563226 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -3709,7 +3709,7 @@ static int usb_audio_probe(struct usb_interface *intf,
 	void *chip;
 	chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id);
 	if (chip) {
-		dev_set_drvdata(&intf->dev, chip);
+		usb_set_intfdata(intf, chip);
 		return 0;
 	} else
 		return -EIO;
@@ -3718,13 +3718,13 @@ static int usb_audio_probe(struct usb_interface *intf,
 static void usb_audio_disconnect(struct usb_interface *intf)
 {
 	snd_usb_audio_disconnect(interface_to_usbdev(intf),
-				 dev_get_drvdata(&intf->dev));
+				 usb_get_intfdata(intf));
 }
 
 #ifdef CONFIG_PM
 static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 {
-	struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev);
+	struct snd_usb_audio *chip = usb_get_intfdata(intf);
 	struct list_head *p;
 	struct snd_usb_stream *as;
 
@@ -3744,7 +3744,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 
 static int usb_audio_resume(struct usb_interface *intf)
 {
-	struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev);
+	struct snd_usb_audio *chip = usb_get_intfdata(intf);
 
 	if (chip == (void *)-1L)
 		return 0;
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index c2515b680f9f..73e59f4403a4 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -589,7 +589,7 @@ static int snd_us122l_suspend(struct usb_interface *intf, pm_message_t message)
 	struct us122l *us122l;
 	struct list_head *p;
 
-	card = dev_get_drvdata(&intf->dev);
+	card = usb_get_intfdata(intf);
 	if (!card)
 		return 0;
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -615,7 +615,7 @@ static int snd_us122l_resume(struct usb_interface *intf)
 	struct list_head *p;
 	int err;
 
-	card = dev_get_drvdata(&intf->dev);
+	card = usb_get_intfdata(intf);
 	if (!card)
 		return 0;
 
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index e5981a630314..ca26c532e77e 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -392,7 +392,7 @@ static int snd_usX2Y_probe(struct usb_interface *intf, const struct usb_device_i
 	void *chip;
 	chip = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id);
 	if (chip) {
-		dev_set_drvdata(&intf->dev, chip);
+		usb_set_intfdata(intf, chip);
 		return 0;
 	} else
 		return -EIO;
@@ -401,7 +401,7 @@ static int snd_usX2Y_probe(struct usb_interface *intf, const struct usb_device_i
 static void snd_usX2Y_disconnect(struct usb_interface *intf)
 {
 	usX2Y_usb_disconnect(interface_to_usbdev(intf),
-				 dev_get_drvdata(&intf->dev));
+				 usb_get_intfdata(intf));
 }
 
 MODULE_DEVICE_TABLE(usb, snd_usX2Y_usb_id_table);
-- 
cgit v1.2.3


From bc7a166dd1530965aa80966f267235f067c5fddf Mon Sep 17 00:00:00 2001
From: Ulrich Dangel
Date: Fri, 2 Jan 2009 19:30:13 +0100
Subject: ALSA: hda - add basic jack reporting functions to patch_conexant.c

Added functions to report jack sense.
As CXT5051_PORTB_EVENT has the same value as CONEXANT_MIC_EVENT two input
devices for the microphone will be created if using CXT5051.

Signed-off-by: Ulrich Dangel <uli@spamt.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 111 +++++++++++++++++++++++++++++++++++++++--
 1 file changed, 108 insertions(+), 3 deletions(-)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index b20e1cede00b..e0eebfbec351 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -25,6 +25,8 @@
 #include <linux/slab.h>
 #include <linux/pci.h>
 #include <sound/core.h>
+#include <sound/jack.h>
+
 #include "hda_codec.h"
 #include "hda_local.h"
 
@@ -37,8 +39,21 @@
 #define CONEXANT_HP_EVENT	0x37
 #define CONEXANT_MIC_EVENT	0x38
 
+/* Conexant 5051 specific */
+
+#define CXT5051_SPDIF_OUT	0x1C
+#define CXT5051_PORTB_EVENT	0x38
+#define CXT5051_PORTC_EVENT	0x39
 
 
+struct conexant_jack {
+
+	hda_nid_t nid;
+	int type;
+	struct snd_jack *jack;
+
+};
+
 struct conexant_spec {
 
 	struct snd_kcontrol_new *mixers[5];
@@ -83,6 +98,9 @@ struct conexant_spec {
 
 	unsigned int spdif_route;
 
+	/* jack detection */
+	struct snd_array jacks;
+
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
 	struct hda_input_mux private_imux;
@@ -329,6 +347,86 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol,
 				     &spec->cur_mux[adc_idx]);
 }
 
+static int conexant_add_jack(struct hda_codec *codec,
+		hda_nid_t nid, int type)
+{
+	struct conexant_spec *spec;
+	struct conexant_jack *jack;
+	const char *name;
+
+	spec = codec->spec;
+	snd_array_init(&spec->jacks, sizeof(*jack), 32);
+	jack = snd_array_new(&spec->jacks);
+	name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
+
+	if (!jack)
+		return -ENOMEM;
+
+	jack->nid = nid;
+	jack->type = type;
+
+	return snd_jack_new(codec->bus->card, name, type, &jack->jack);
+}
+
+static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct conexant_spec *spec = codec->spec;
+	struct conexant_jack *jacks = spec->jacks.list;
+
+	if (jacks) {
+		int i;
+		for (i = 0; i < spec->jacks.used; i++) {
+			if (jacks->nid == nid) {
+				unsigned int present;
+				present = snd_hda_codec_read(codec, nid, 0,
+						AC_VERB_GET_PIN_SENSE, 0) &
+					AC_PINSENSE_PRESENCE;
+
+				present = (present) ? jacks->type : 0 ;
+
+				snd_jack_report(jacks->jack,
+						present);
+			}
+			jacks++;
+		}
+	}
+}
+
+static int conexant_init_jacks(struct hda_codec *codec)
+{
+#ifdef CONFIG_SND_JACK
+	struct conexant_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_init_verbs; i++) {
+		const struct hda_verb *hv;
+
+		hv = spec->init_verbs[i];
+		while (hv->nid) {
+			int err = 0;
+			switch (hv->param ^ AC_USRSP_EN) {
+			case CONEXANT_HP_EVENT:
+				err = conexant_add_jack(codec, hv->nid,
+						SND_JACK_HEADPHONE);
+				conexant_report_jack(codec, hv->nid);
+				break;
+			case CXT5051_PORTC_EVENT:
+			case CONEXANT_MIC_EVENT:
+				err = conexant_add_jack(codec, hv->nid,
+						SND_JACK_MICROPHONE);
+				conexant_report_jack(codec, hv->nid);
+				break;
+			}
+			if (err < 0)
+				return err;
+			++hv;
+		}
+	}
+#endif
+	return 0;
+
+}
+
 static int conexant_init(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -341,6 +439,16 @@ static int conexant_init(struct hda_codec *codec)
 
 static void conexant_free(struct hda_codec *codec)
 {
+#ifdef CONFIG_SND_JACK
+	struct conexant_spec *spec = codec->spec;
+	if (spec->jacks.list) {
+		struct conexant_jack *jacks = spec->jacks.list;
+		int i;
+		for (i = 0; i < spec->jacks.used; i++)
+			snd_device_free(codec->bus->card, &jacks[i].jack);
+		snd_array_free(&spec->jacks);
+	}
+#endif
 	kfree(codec->spec);
 }
 
@@ -1526,9 +1634,6 @@ static int patch_cxt5047(struct hda_codec *codec)
 /* Conexant 5051 specific */
 static hda_nid_t cxt5051_dac_nids[1] = { 0x10 };
 static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 };
-#define CXT5051_SPDIF_OUT	0x1C
-#define CXT5051_PORTB_EVENT	0x38
-#define CXT5051_PORTC_EVENT	0x39
 
 static struct hda_channel_mode cxt5051_modes[1] = {
 	{ 2, NULL },
-- 
cgit v1.2.3


From acf26c0cad5ba00dcafa633805e4660e90c1eac0 Mon Sep 17 00:00:00 2001
From: Ulrich Dangel
Date: Fri, 2 Jan 2009 19:30:14 +0100
Subject: ALSA: hda - cxt5051 report jack state

Signed-off-by: Ulrich Dangel <uli@spamt.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e0eebfbec351..75de40aaab0a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1713,6 +1713,7 @@ static void cxt5051_hp_automute(struct hda_codec *codec)
 static void cxt5051_hp_unsol_event(struct hda_codec *codec,
 				   unsigned int res)
 {
+	int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20;
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5051_hp_automute(codec);
@@ -1724,6 +1725,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec,
 		cxt5051_portc_automic(codec);
 		break;
 	}
+	conexant_report_jack(codec, nid);
 }
 
 static struct snd_kcontrol_new cxt5051_mixers[] = {
@@ -1798,6 +1800,7 @@ static struct hda_verb cxt5051_init_verbs[] = {
 static int cxt5051_init(struct hda_codec *codec)
 {
 	conexant_init(codec);
+	conexant_init_jacks(codec);
 	if (codec->patch_ops.unsol_event) {
 		cxt5051_hp_automute(codec);
 		cxt5051_portb_automic(codec);
-- 
cgit v1.2.3


From 913ae5a24efd27deef4fc154953871b62d0d99cd Mon Sep 17 00:00:00 2001
From: Julia Lawall
Date: Sat, 3 Jan 2009 17:54:53 +0100
Subject: ALSA: sound/usb: Use negated usb_endpoint_xfer_control, etc

This patch extends 42a6e66f1e40a930d093c33ba0bb9d8d8e4555ed by using
usb_endpoint_xfer_control, usb_endpoint_xfer_isoc, usb_endpoint_xfer_bulk,
and usb_endpoint_xfer_int in the negated case as well.

This patch also rewrites some calls to usb_endpoint_dir_in as negated calls
to !usb_endpoint_dir_out, and vice versa, to better correspond to the
intent of the original code.

The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@ struct usb_endpoint_descriptor *epd; @@

- (usb_endpoint_type(epd) != \(USB_ENDPOINT_XFER_CONTROL\|0\))
+ !usb_endpoint_xfer_control(epd)

@@ struct usb_endpoint_descriptor *epd; @@

- (usb_endpoint_type(epd) != \(USB_ENDPOINT_XFER_ISOC\|1\))
+ !usb_endpoint_xfer_isoc(epd)

@@ struct usb_endpoint_descriptor *epd; @@

- (usb_endpoint_type(epd) != \(USB_ENDPOINT_XFER_BULK\|2\))
+ !usb_endpoint_xfer_bulk(epd)

@@ struct usb_endpoint_descriptor *epd; @@

- (usb_endpoint_type(epd) != \(USB_ENDPOINT_XFER_INT\|3\))
+ !usb_endpoint_xfer_int(epd)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbmidi.c  | 17 +++++++----------
 sound/usb/usbmixer.c |  3 +--
 2 files changed, 8 insertions(+), 12 deletions(-)

diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 3a9a9fecd292..320641ab5be7 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1392,8 +1392,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
 	for (i = 0; i < intfd->bNumEndpoints; ++i) {
 		hostep = &hostif->endpoint[i];
 		ep = get_ep_desc(hostep);
-		if (usb_endpoint_type(ep) != USB_ENDPOINT_XFER_BULK &&
-		    usb_endpoint_type(ep) != USB_ENDPOINT_XFER_INT)
+		if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
 			continue;
 		ms_ep = (struct usb_ms_endpoint_descriptor*)hostep->extra;
 		if (hostep->extralen < 4 ||
@@ -1495,8 +1494,8 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi,
 
 	for (i = 0; i < intfd->bNumEndpoints; ++i) {
 		epd = get_endpoint(hostif, i);
-		if (usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK &&
-		    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_INT)
+		if (!usb_endpoint_xfer_bulk(epd) &&
+		    !usb_endpoint_xfer_int(epd))
 			continue;
 		if (out_eps < max_endpoints &&
 		    usb_endpoint_dir_out(epd)) {
@@ -1607,21 +1606,19 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
 	}
 
 	epd = get_endpoint(hostif, 0);
-	if (usb_endpoint_dir_out(epd) ||
-	    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_INT) {
+	if (!usb_endpoint_dir_in(epd) || !usb_endpoint_xfer_int(epd)) {
 		snd_printdd(KERN_ERR "endpoint[0] isn't interrupt\n");
 		return -ENXIO;
 	}
 	epd = get_endpoint(hostif, 2);
-	if (usb_endpoint_dir_in(epd) ||
-	    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK) {
+	if (!usb_endpoint_dir_out(epd) || !usb_endpoint_xfer_bulk(epd)) {
 		snd_printdd(KERN_ERR "endpoint[2] isn't bulk output\n");
 		return -ENXIO;
 	}
 	if (endpoint->out_cables > 0x0001) {
 		epd = get_endpoint(hostif, 4);
-		if (usb_endpoint_dir_in(epd) ||
-		    usb_endpoint_type(epd) != USB_ENDPOINT_XFER_BULK) {
+		if (!usb_endpoint_dir_out(epd) ||
+		    !usb_endpoint_xfer_bulk(epd)) {
 			snd_printdd(KERN_ERR "endpoint[4] isn't bulk output\n");
 			return -ENXIO;
 		}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 9ce626f0e36b..00397c8a765b 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1755,8 +1755,7 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
 	if (get_iface_desc(hostif)->bNumEndpoints < 1)
 		return 0;
 	ep = get_endpoint(hostif, 0);
-	if (usb_endpoint_dir_out(ep) ||
-	    usb_endpoint_type(ep) != USB_ENDPOINT_XFER_INT)
+	if (!usb_endpoint_dir_in(ep) || !usb_endpoint_xfer_int(ep))
 		return 0;
 
 	epnum = usb_endpoint_num(ep);
-- 
cgit v1.2.3


From 5cf1c00b0ef3ba964b2ad268a55c278cf43f798f Mon Sep 17 00:00:00 2001
From: David Brownell
Date: Mon, 5 Jan 2009 02:08:30 -0800
Subject: ASoC: fix davinci-sffsdr buglet

Minor bugfix:  now that DaVinci kernels can support multiple
boards, board-specific ASoC components need to verify they're
running on the right board before initializing.

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/davinci-sffsdr.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index f67579d52765..4935d1bcbd8d 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -24,6 +24,7 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
+#include <asm/mach-types.h>
 #include <asm/plat-sffsdr/sffsdr-fpga.h>
 
 #include <mach/mcbsp.h>
@@ -115,6 +116,9 @@ static int __init sffsdr_init(void)
 {
 	int ret;
 
+	if (!machine_is_sffsdr())
+		return -EINVAL;
+
 	sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
 	if (!sffsdr_snd_device) {
 		printk(KERN_ERR "platform device allocation failed\n");
-- 
cgit v1.2.3


From 8eca75382e012b74b98526a1679ada2a1849024b Mon Sep 17 00:00:00 2001
From: Alan Horstmann
Date: Mon, 5 Jan 2009 18:30:04 +0100
Subject: ALSA: ice1724 - Fix a typo in IEC958 PCM name

Fix trivial name string typo as reported in bug 2552.

Signed-off-by: Alan Horstmann <gineera@aspect135.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/ice1724.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 0dfa0540ce2c..bb8d8c766b9d 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -1239,7 +1239,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
 	if (ice->force_pdma4 || ice->force_rdma1)
 		name = "ICE1724 Secondary";
 	else
-		name = "IEC1724 IEC958";
+		name = "ICE1724 IEC958";
 	err = snd_pcm_new(ice->card, name, device, play, capt, &pcm);
 	if (err < 0)
 		return err;
-- 
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