Merge branch 'for-2.6.32' of...

Merge branch 'for-2.6.32' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

include/sound/soc-dapm.h

@@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
+void snd_soc_dapm_shutdown(struct snd_soc_device *socdev);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);

include/sound/soc.h

@@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform);
 int snd_soc_register_codec(struct snd_soc_codec *codec);
 void snd_soc_unregister_codec(struct snd_soc_codec *codec);
 
+#ifdef CONFIG_PM
+int snd_soc_suspend_device(struct device *dev);
+int snd_soc_resume_device(struct device *dev);
+#endif
+
 /* 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);
@@ -216,9 +221,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 
 /* codec register bit access */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value);
+				unsigned int mask, unsigned int value);
 int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value);
+				unsigned int mask, unsigned int value);
 
 int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num);
@@ -369,8 +374,6 @@ struct snd_soc_codec {
 	enum snd_soc_bias_level bias_level;
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
-	struct list_head up_list;
-	struct list_head down_list;
 
 	/* codec DAI's */
 	struct snd_soc_dai *dai;

include/sound/uda1380.h

+/*
+ * UDA1380 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __UDA1380_H
+#define __UDA1380_H
+
+struct uda1380_platform_data {
+	int gpio_power;
+	int gpio_reset;
+	int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL  1
+};
+
+#endif /* __UDA1380_H */

sound/soc/blackfin/bf5xx-ad73311.c

@@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
 	.codec_dev = &soc_codec_dev_ad73311,
 };
 
-static struct platform_device *bf52x_ad73311_snd_device;
+static struct platform_device *bf5xx_ad73311_snd_device;
 
 static int __init bf5xx_ad73311_init(void)
 {
 	int ret;
 
 	pr_debug("%s enter\n", __func__);
-	bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!bf52x_ad73311_snd_device)
+	bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!bf5xx_ad73311_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
-	bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
-	ret = platform_device_add(bf52x_ad73311_snd_device);
+	platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
+	bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev;
+	ret = platform_device_add(bf5xx_ad73311_snd_device);
 
 	if (ret)
-		platform_device_put(bf52x_ad73311_snd_device);
+		platform_device_put(bf5xx_ad73311_snd_device);
 
 	return ret;
 }
@@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void)
 static void __exit bf5xx_ad73311_exit(void)
 {
 	pr_debug("%s enter\n", __func__);
-	platform_device_unregister(bf52x_ad73311_snd_device);
+	platform_device_unregister(bf5xx_ad73311_snd_device);
 }
 
 module_init(bf5xx_ad73311_init);

sound/soc/blackfin/bf5xx-ssm2602.c

@@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
 	.codec_data = &bf5xx_ssm2602_setup,
 };
 
-static struct platform_device *bf52x_ssm2602_snd_device;
+static struct platform_device *bf5xx_ssm2602_snd_device;
 
 static int __init bf5xx_ssm2602_init(void)
 {
 	int ret;
 
 	pr_debug("%s enter\n", __func__);
-	bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!bf52x_ssm2602_snd_device)
+	bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!bf5xx_ssm2602_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(bf52x_ssm2602_snd_device,
+	platform_set_drvdata(bf5xx_ssm2602_snd_device,
 				&bf5xx_ssm2602_snd_devdata);
-	bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev;
-	ret = platform_device_add(bf52x_ssm2602_snd_device);
+	bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev;
+	ret = platform_device_add(bf5xx_ssm2602_snd_device);
 
 	if (ret)
-		platform_device_put(bf52x_ssm2602_snd_device);
+		platform_device_put(bf5xx_ssm2602_snd_device);
 
 	return ret;
 }
@@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void)
 static void __exit bf5xx_ssm2602_exit(void)
 {
 	pr_debug("%s enter\n", __func__);
-	platform_device_unregister(bf52x_ssm2602_snd_device);
+	platform_device_unregister(bf5xx_ssm2602_snd_device);
 }
 
 module_init(bf5xx_ssm2602_init);

sound/soc/codecs/Kconfig

@@ -39,6 +39,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8940 if I2C
 	select SND_SOC_WM8960 if I2C
+	select SND_SOC_WM8961 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
@@ -156,6 +157,9 @@ config SND_SOC_WM8940
 config SND_SOC_WM8960
 	tristate
 
+config SND_SOC_WM8961
+	tristate
+
 config SND_SOC_WM8971
 	tristate
 

sound/soc/codecs/Makefile

@@ -27,6 +27,7 @@ snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8940-objs := wm8940.o
 snd-soc-wm8960-objs := wm8960.o
+snd-soc-wm8961-objs := wm8961.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
@@ -65,6 +66,7 @@ obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8940)	+= snd-soc-wm8940.o
 obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8961)	+= snd-soc-wm8961.o
 obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
 obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o

sound/soc/codecs/twl4030.c

@@ -712,7 +712,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
 
 	reg = twl4030_read_reg_cache(w->codec, m->reg);
 
-	if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
+	/*
+	 * bypass_state[0:3] - analog HiFi bypass
+	 * bypass_state[4]   - analog voice bypass
+	 * bypass_state[5]   - digital voice bypass
+	 * bypass_state[6:7] - digital HiFi bypass
+	 */
+	if (m->reg == TWL4030_REG_VSTPGA) {
+		/* Voice digital bypass */
+		if (reg)
+			twl4030->bypass_state |= (1 << 5);
+		else
+			twl4030->bypass_state &= ~(1 << 5);
+	} else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
 		/* Analog bypass */
 		if (reg & (1 << m->shift))
 			twl4030->bypass_state |=
@@ -726,12 +738,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
 			twl4030->bypass_state |= (1 << 4);
 		else
 			twl4030->bypass_state &= ~(1 << 4);
-	} else if (m->reg == TWL4030_REG_VSTPGA) {
-		/* Voice digital bypass */
-		if (reg)
-			twl4030->bypass_state |= (1 << 5);
-		else
-			twl4030->bypass_state &= ~(1 << 5);
 	} else {
 		/* Digital bypass */
 		if (reg & (0x7 << m->shift))
@@ -924,7 +930,7 @@ static const struct soc_enum twl4030_op_modes_enum =
 			ARRAY_SIZE(twl4030_op_modes_texts),
 			twl4030_op_modes_texts);
 
-int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -1005,6 +1011,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
  */
 static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
 
+/* AVADC clock priority */
+static const char *twl4030_avadc_clk_priority_texts[] = {
+	"Voice high priority", "HiFi high priority"
+};
+
+static const struct soc_enum twl4030_avadc_clk_priority_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2,
+			ARRAY_SIZE(twl4030_avadc_clk_priority_texts),
+			twl4030_avadc_clk_priority_texts);
+
 static const char *twl4030_rampdelay_texts[] = {
 	"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
 	"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
@@ -1106,6 +1122,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
 		0, 3, 5, 0, input_gain_tlv),
 
+	SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum),
+
 	SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
 
 	SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
@@ -1609,8 +1627,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 
 	 /* If the substream has 4 channel, do the necessary setup */
 	if (params_channels(params) == 4) {
-		u8 format, mode;
-
 		format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
 		mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
 
@@ -1948,7 +1964,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFM:
 		format &= ~(TWL4030_VIF_SLAVE_EN);
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:

sound/soc/codecs/uda1380.h

@@ -72,14 +72,6 @@
 #define R22_SKIP_DCFIL	0x0002
 #define R23_AGC_EN	0x0001
 
-struct uda1380_setup_data {
-	int            i2c_bus;
-	unsigned short i2c_address;
-	int            dac_clk;
-#define UDA1380_DAC_CLK_SYSCLK 0
-#define UDA1380_DAC_CLK_WSPLL  1
-};
-
 #define UDA1380_DAI_DUPLEX	0 /* playback and capture on single DAI */
 #define UDA1380_DAI_PLAYBACK	1 /* playback DAI */
 #define UDA1380_DAI_CAPTURE	2 /* capture DAI */

sound/soc/codecs/wm8350.c

@@ -406,7 +406,6 @@ static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
 static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
 static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
 static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
-static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
 static const char *wm8350_adcfilter[] = { "None", "High Pass" };
 static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
 static const char *wm8350_lr[] = { "Left", "Right" };
@@ -416,7 +415,6 @@ static const struct soc_enum wm8350_enum[] = {
 	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
 	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
 	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
-	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
 	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
 	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
 	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
@@ -444,10 +442,9 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = {
 				0, 255, 0, dac_pcm_tlv),
 	SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
 	SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
-	SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
-	SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
-	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
-	SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+	SOC_ENUM("Capture PCM Filter", wm8350_enum[4]),
+	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]),
+	SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]),
 	SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
 				WM8350_ADC_DIGITAL_VOLUME_L,
 				WM8350_ADC_DIGITAL_VOLUME_R,
@@ -993,6 +990,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *codec_dai)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
 	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
 	    ~WM8350_AIF_WL_MASK;
 
@@ -1012,6 +1010,19 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+
+	/* The sloping stopband filter is recommended for use with
+	 * lower sample rates to improve performance.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (params_rate(params) < 24000)
+			wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
+					WM8350_DAC_SB_FILT);
+		else
+			wm8350_clear_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
+					  WM8350_DAC_SB_FILT);
+	}
+
 	return 0;
 }
 
@@ -1660,6 +1671,21 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m)
+{
+	return snd_soc_suspend_device(&pdev->dev);
+}
+
+static int wm8350_codec_resume(struct platform_device *pdev)
+{
+	return snd_soc_resume_device(&pdev->dev);
+}
+#else
+#define wm8350_codec_suspend NULL
+#define wm8350_codec_resume NULL
+#endif
+
 static struct platform_driver wm8350_codec_driver = {
 	.driver = {
 		   .name = "wm8350-codec",
@@ -1667,6 +1693,8 @@ static struct platform_driver wm8350_codec_driver = {
 		   },
 	.probe = wm8350_codec_probe,
 	.remove = __devexit_p(wm8350_codec_remove),
+	.suspend = wm8350_codec_suspend,
+	.resume = wm8350_codec_resume,
 };
 
 static __init int wm8350_init(void)

sound/soc/codecs/wm8400.c

@@ -1553,6 +1553,21 @@ static int __exit wm8400_codec_remove(struct platform_device *dev)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&pdev->dev);
+}
+
+static int wm8400_pdev_resume(struct platform_device *pdev)
+{
+	return snd_soc_resume_device(&pdev->dev);
+}
+#else
+#define wm8400_pdev_suspend NULL
+#define wm8400_pdev_resume NULL
+#endif
+
 static struct platform_driver wm8400_codec_driver = {
 	.driver = {
 		.name = "wm8400-codec",
@@ -1560,6 +1575,8 @@ static struct platform_driver wm8400_codec_driver = {
 	},
 	.probe = wm8400_codec_probe,
 	.remove	= __exit_p(wm8400_codec_remove),
+	.suspend = wm8400_pdev_suspend,
+	.resume = wm8400_pdev_resume,
 };
 
 static int __init wm8400_codec_init(void)

sound/soc/codecs/wm8580.c

@@ -24,6 +24,8 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -187,15 +189,22 @@ struct pll_state {
 	unsigned int out;
 };
 
+#define WM8580_NUM_SUPPLIES 3
+static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = {
+	"AVDD",
+	"DVDD",
+	"PVDD",
+};
+
 /* codec private data */
 struct wm8580_priv {
 	struct snd_soc_codec codec;
+	struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES];
 	u16 reg_cache[WM8580_MAX_REGISTER + 1];
 	struct pll_state a;
 	struct pll_state b;
 };
 
-
 /*
  * read wm8580 register cache
  */
@@ -922,11 +931,28 @@ static int wm8580_register(struct wm8580_priv *wm8580)
 
 	memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg));
 
+	for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++)
+		wm8580->supplies[i].supply = wm8580_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8580->supplies),
+				 wm8580->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies),
+				    wm8580->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_regulator_get;
+	}
+
 	/* Get the codec into a known state */
 	ret = wm8580_write(codec, WM8580_RESET, 0);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to reset codec: %d\n", ret);
-		goto err;
+		goto err_regulator_enable;
 	}
 
 	for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++)
@@ -939,7 +965,7 @@ static int wm8580_register(struct wm8580_priv *wm8580)
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		goto err;
+		goto err_regulator_enable;
 	}
 
 	ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
@@ -952,6 +978,10 @@ static int wm8580_register(struct wm8580_priv *wm8580)
 
 err_codec:
 	snd_soc_unregister_codec(codec);
+err_regulator_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
+err_regulator_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
 err:
 	kfree(wm8580);
 	return ret;
@@ -962,6 +992,8 @@ static void wm8580_unregister(struct wm8580_priv *wm8580)
 	wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
 	snd_soc_unregister_codec(&wm8580->codec);
+	regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
+	regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies);
 	kfree(wm8580);
 	wm8580_codec = NULL;
 }
@@ -995,6 +1027,21 @@ static int wm8580_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8580_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8580_i2c_suspend NULL
+#define wm8580_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8580_i2c_id[] = {
 	{ "wm8580", 0 },
 	{ }
@@ -1008,6 +1055,8 @@ static struct i2c_driver wm8580_i2c_driver = {
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
+	.suspend =  wm8580_i2c_suspend,
+	.resume =   wm8580_i2c_resume,
 	.id_table = wm8580_i2c_id,
 };
 #endif

sound/soc/codecs/wm8731.c

@@ -460,6 +460,7 @@ struct snd_soc_dai wm8731_dai = {
 };
 EXPORT_SYMBOL_GPL(wm8731_dai);
 
+#ifdef CONFIG_PM
 static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -488,6 +489,10 @@ static int wm8731_resume(struct platform_device *pdev)
 	wm8731_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
+#else
+#define wm8731_suspend NULL
+#define wm8731_resume NULL
+#endif
 
 static int wm8731_probe(struct platform_device *pdev)
 {
@@ -680,6 +685,21 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8731_spi_resume(struct spi_device *spi)
+{
+	return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8731_spi_suspend NULL
+#define wm8731_spi_resume NULL
+#endif
+
 static struct spi_driver wm8731_spi_driver = {
 	.driver = {
 		.name	= "wm8731",
@@ -687,6 +707,8 @@ static struct spi_driver wm8731_spi_driver = {
 		.owner	= THIS_MODULE,
 	},
 	.probe		= wm8731_spi_probe,
+	.suspend	= wm8731_spi_suspend,
+	.resume		= wm8731_spi_resume,
 	.remove		= __devexit_p(wm8731_spi_remove),
 };
 #endif /* CONFIG_SPI_MASTER */
@@ -720,6 +742,21 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&i2c->dev);
+}
+
+static int wm8731_i2c_resume(struct i2c_client *i2c)
+{
+	return snd_soc_resume_device(&i2c->dev);
+}
+#else
+#define wm8731_i2c_suspend NULL
+#define wm8731_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8731_i2c_id[] = {
 	{ "wm8731", 0 },
 	{ }
@@ -733,6 +770,8 @@ static struct i2c_driver wm8731_i2c_driver = {
 	},
 	.probe =    wm8731_i2c_probe,
 	.remove =   __devexit_p(wm8731_i2c_remove),
+	.suspend =  wm8731_i2c_suspend,
+	.resume =   wm8731_i2c_resume,
 	.id_table = wm8731_i2c_id,
 };
 #endif

sound/soc/codecs/wm8753.c

@@ -1766,6 +1766,21 @@ static int wm8753_i2c_remove(struct i2c_client *client)
         return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8753_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8753_i2c_suspend NULL
+#define wm8753_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8753_i2c_id[] = {
 	{ "wm8753", 0 },
 	{ }
@@ -1779,6 +1794,8 @@ static struct i2c_driver wm8753_i2c_driver = {
 	},
 	.probe =    wm8753_i2c_probe,
 	.remove =   wm8753_i2c_remove,
+	.suspend =  wm8753_i2c_suspend,
+	.resume =   wm8753_i2c_resume,
 	.id_table = wm8753_i2c_id,
 };
 #endif
@@ -1834,6 +1851,22 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8753_spi_resume(struct spi_device *spi)
+{
+	return snd_soc_resume_device(&spi->dev);
+}
+
+#else
+#define wm8753_spi_suspend NULL
+#define wm8753_spi_resume NULL
+#endif
+
 static struct spi_driver wm8753_spi_driver = {
 	.driver = {
 		.name	= "wm8753",
@@ -1842,6 +1875,8 @@ static struct spi_driver wm8753_spi_driver = {
 	},
 	.probe		= wm8753_spi_probe,
 	.remove		= __devexit_p(wm8753_spi_remove),
+	.suspend	= wm8753_spi_suspend,
+	.resume		= wm8753_spi_resume,
 };
 #endif
 

sound/soc/codecs/wm8900.c

@@ -116,6 +116,7 @@
 #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c
 
 #define WM8900_REG_DACCTRL_MUTE          0x004
+#define WM8900_REG_DACCTRL_DAC_SB_FILT   0x100
 #define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400
 
 #define WM8900_REG_AUDIO3_ADCLRC_DIR    0x0800
@@ -439,7 +440,6 @@ SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1),
 SOC_ENUM("DAC Mute Rate", dac_mute_rate),
 SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0),
 SOC_ENUM("DAC Deemphasis", dac_deemphasis),
-SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0),
 SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL,
 	   12, 1, 0),
 
@@ -743,6 +743,17 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
 
 	wm8900_write(codec, WM8900_REG_AUDIO1, reg);
 
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		reg = wm8900_read(codec, WM8900_REG_DACCTRL);
+
+		if (params_rate(params) <= 24000)
+			reg |= WM8900_REG_DACCTRL_DAC_SB_FILT;
+		else
+			reg &= ~WM8900_REG_DACCTRL_DAC_SB_FILT;
+
+		wm8900_write(codec, WM8900_REG_DACCTRL, reg);
+	}
+
 	return 0;
 }
 
@@ -1388,6 +1399,21 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8900_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8900_i2c_suspend NULL
+#define wm8900_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8900_i2c_id[] = {
 	{ "wm8900", 0 },
 	{ }
@@ -1401,6 +1427,8 @@ static struct i2c_driver wm8900_i2c_driver = {
 	},
 	.probe = wm8900_i2c_probe,
 	.remove = __devexit_p(wm8900_i2c_remove),
+	.suspend = wm8900_i2c_suspend,
+	.resume = wm8900_i2c_resume,
 	.id_table = wm8900_i2c_id,
 };
 

sound/soc/codecs/wm8903.c

@@ -715,8 +715,6 @@ SOC_ENUM("DAC Soft Mute Rate", soft_mute),
 SOC_ENUM("DAC Mute Mode", mute_mode),
 SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0),
 SOC_ENUM("DAC De-emphasis", dac_deemphasis),
-SOC_SINGLE("DAC Sloping Stopband Filter Switch",
-	   WM8903_DAC_DIGITAL_1, 11, 1, 0),
 SOC_ENUM("DAC Companding Mode", dac_companding),
 SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0),
 
@@ -1373,12 +1371,19 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 	u16 aif3 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_3);
 	u16 clock0 = wm8903_read(codec, WM8903_CLOCK_RATES_0);
 	u16 clock1 = wm8903_read(codec, WM8903_CLOCK_RATES_1);
+	u16 dac_digital1 = wm8903_read(codec, WM8903_DAC_DIGITAL_1);
 
 	if (substream == wm8903->slave_substream) {
 		dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
 		return 0;
 	}
 
+	/* Enable sloping stopband filter for low sample rates */
+	if (fs <= 24000)
+		dac_digital1 |= WM8903_DAC_SB_FILT;
+	else
+		dac_digital1 &= ~WM8903_DAC_SB_FILT;
+
 	/* Configure sample rate logic for DSP - choose nearest rate */
 	dsp_config = 0;
 	best_val = abs(sample_rates[dsp_config].rate - fs);
@@ -1503,6 +1508,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 	wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
 	wm8903_write(codec, WM8903_AUDIO_INTERFACE_2, aif2);
 	wm8903_write(codec, WM8903_AUDIO_INTERFACE_3, aif3);
+	wm8903_write(codec, WM8903_DAC_DIGITAL_1, dac_digital1);
 
 	return 0;
 }
@@ -1721,6 +1727,21 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8903_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8903_i2c_suspend NULL
+#define wm8903_i2c_resume NULL
+#endif
+
 /* i2c codec control layer */
 static const struct i2c_device_id wm8903_i2c_id[] = {
        { "wm8903", 0 },
@@ -1735,6 +1756,8 @@ static struct i2c_driver wm8903_i2c_driver = {
 	},
 	.probe    = wm8903_i2c_probe,
 	.remove   = __devexit_p(wm8903_i2c_remove),
+	.suspend  = wm8903_i2c_suspend,
+	.resume   = wm8903_i2c_resume,
 	.id_table = wm8903_i2c_id,
 };
 

sound/soc/codecs/wm8940.c

@@ -916,6 +916,21 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8940_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8940_i2c_suspend NULL
+#define wm8940_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8940_i2c_id[] = {
 	{ "wm8940", 0 },
 	{ }
@@ -929,6 +944,8 @@ static struct i2c_driver wm8940_i2c_driver = {
 	},
 	.probe = wm8940_i2c_probe,
 	.remove = __devexit_p(wm8940_i2c_remove),
+	.suspend = wm8940_i2c_suspend,
+	.resume = wm8940_i2c_resume,
 	.id_table = wm8940_i2c_id,
 };
 

sound/soc/codecs/wm8960.c

@@ -927,6 +927,21 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8960_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8960_i2c_suspend NULL
+#define wm8960_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8960_i2c_id[] = {
 	{ "wm8960", 0 },
 	{ }
@@ -940,6 +955,8 @@ static struct i2c_driver wm8960_i2c_driver = {
 	},
 	.probe =    wm8960_i2c_probe,
 	.remove =   __devexit_p(wm8960_i2c_remove),
+	.suspend =  wm8960_i2c_suspend,
+	.resume =   wm8960_i2c_resume,
 	.id_table = wm8960_i2c_id,
 };
 

sound/soc/codecs/wm8988.c

@@ -981,6 +981,21 @@ static int wm8988_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm8988_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm8988_i2c_suspend NULL
+#define wm8988_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm8988_i2c_id[] = {
 	{ "wm8988", 0 },
 	{ }
@@ -994,6 +1009,8 @@ static struct i2c_driver wm8988_i2c_driver = {
 	},
 	.probe = wm8988_i2c_probe,
 	.remove = wm8988_i2c_remove,
+	.suspend = wm8988_i2c_suspend,
+	.resume = wm8988_i2c_resume,
 	.id_table = wm8988_i2c_id,
 };
 #endif
@@ -1051,6 +1068,21 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&spi->dev);
+}
+
+static int wm8988_spi_resume(struct spi_device *spi)
+{
+	return snd_soc_resume_device(&spi->dev);
+}
+#else
+#define wm8988_spi_suspend NULL
+#define wm8988_spi_resume NULL
+#endif
+
 static struct spi_driver wm8988_spi_driver = {
 	.driver = {
 		.name	= "wm8988",
@@ -1059,6 +1091,8 @@ static struct spi_driver wm8988_spi_driver = {
 	},
 	.probe		= wm8988_spi_probe,
 	.remove		= __devexit_p(wm8988_spi_remove),
+	.suspend	= wm8988_spi_suspend,
+	.resume		= wm8988_spi_resume,
 };
 #endif
 

sound/soc/codecs/wm9081.c

@@ -1492,6 +1492,21 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg)
+{
+	return snd_soc_suspend_device(&client->dev);
+}
+
+static int wm9081_i2c_resume(struct i2c_client *client)
+{
+	return snd_soc_resume_device(&client->dev);
+}
+#else
+#define wm9081_i2c_suspend NULL
+#define wm9081_i2c_resume NULL
+#endif
+
 static const struct i2c_device_id wm9081_i2c_id[] = {
 	{ "wm9081", 0 },
 	{ }
@@ -1505,6 +1520,8 @@ static struct i2c_driver wm9081_i2c_driver = {
 	},
 	.probe =    wm9081_i2c_probe,
 	.remove =   __devexit_p(wm9081_i2c_remove),
+	.suspend =  wm9081_i2c_suspend,
+	.resume =   wm9081_i2c_resume,
 	.id_table = wm9081_i2c_id,
 };
 

sound/soc/omap/Kconfig

@@ -72,4 +72,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE
 	help
 	  Say Y if you want to add support for SoC audio on the Beagleboard.
 
+config SND_OMAP_SOC_ZOOM2
+	tristate "SoC Audio support for Zoom2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on Zoom2 board.
 

sound/soc/omap/Makefile

@@ -14,6 +14,7 @@ snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
+snd-soc-zoom2-objs := zoom2.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
+obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o

sound/soc/omap/sdp3430.c

@@ -96,7 +96,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
 	ret = snd_soc_dai_set_fmt(codec_dai,
 				SND_SOC_DAIFMT_DSP_A |
 				SND_SOC_DAIFMT_IB_NF |
-				SND_SOC_DAIFMT_CBS_CFM);
+				SND_SOC_DAIFMT_CBM_CFM);
 	if (ret) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
 		return ret;

sound/soc/omap/zoom2.c

+/*
+ * zoom2.c  --  SoC audio for Zoom2
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define ZOOM2_HEADSET_MUX_GPIO	(OMAP_MAX_GPIO_LINES + 15)
+
+static int zoom2_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops zoom2_ops = {
+	.hw_params = zoom2_hw_params,
+};
+
+static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				SND_SOC_DAIFMT_DSP_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops zoom2_voice_ops = {
+	.hw_params = zoom2_hw_voice_params,
+};
+
+/* Zoom2 machine DAPM */
+static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Stereophone:  HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Aux In: AUXL, AUXR */
+	{"Aux In", NULL, "AUXL"},
+	{"Aux In", NULL, "AUXR"},
+};
+
+static int zoom2_twl4030_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	/* Add Zoom2 specific widgets */
+	ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+				ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up Zoom2 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Zoom2 connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_enable_pin(codec, "Headset Mic");
+	snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(codec, "Aux In");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+	ret = snd_soc_dapm_sync(codec);
+
+	return ret;
+}
+
+static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+	unsigned short reg;
+
+	/* Enable voice interface */
+	reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+	reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+	codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link zoom2_dai[] = {
+	{
+		.name = "TWL4030 I2S",
+		.stream_name = "TWL4030 Audio",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+		.init = zoom2_twl4030_init,
+		.ops = &zoom2_ops,
+	},
+	{
+		.name = "TWL4030 PCM",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai = &omap_mcbsp_dai[1],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+		.init = zoom2_twl4030_voice_init,
+		.ops = &zoom2_voice_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_zoom2 = {
+	.name = "Zoom2",
+	.platform = &omap_soc_platform,
+	.dai_link = zoom2_dai,
+	.num_links = ARRAY_SIZE(zoom2_dai),
+};
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+	.ramp_delay_value = 2,	/* 81 ms */
+	.sysclk = 26000,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device zoom2_snd_devdata = {
+	.card = &snd_soc_zoom2,
+	.codec_dev = &soc_codec_dev_twl4030,
+	.codec_data = &twl4030_setup,
+};
+
+static struct platform_device *zoom2_snd_device;
+
+static int __init zoom2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_zoom2()) {
+		pr_debug("Not Zoom2!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "Zoom2 SoC init\n");
+
+	zoom2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!zoom2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata);
+	zoom2_snd_devdata.dev = &zoom2_snd_device->dev;
+	*(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+	*(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+
+	ret = platform_device_add(zoom2_snd_device);
+	if (ret)
+		goto err1;
+
+	BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
+	gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(zoom2_snd_device);
+
+	return ret;
+}
+module_init(zoom2_soc_init);
+
+static void __exit zoom2_soc_exit(void)
+{
+	gpio_free(ZOOM2_HEADSET_MUX_GPIO);
+
+	platform_device_unregister(zoom2_snd_device);
+}
+module_exit(zoom2_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC Zoom2");
+MODULE_LICENSE("GPL");
+

sound/soc/pxa/magician.c

@@ -20,12 +20,14 @@
 #include <linux/platform_device.h>
 #include <linux/delay.h>
 #include <linux/gpio.h>
+#include <linux/i2c.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include <sound/uda1380.h>
 
 #include <mach/magician.h>
 #include <asm/mach-types.h>
@@ -447,34 +449,47 @@ static struct snd_soc_card snd_soc_card_magician = {
 	.platform = &pxa2xx_soc_platform,
 };
 
-/* magician audio private data */
-static struct uda1380_setup_data magician_uda1380_setup = {
-	.i2c_address = 0x18,
-	.dac_clk = UDA1380_DAC_CLK_WSPLL,
-};
-
 /* magician audio subsystem */
 static struct snd_soc_device magician_snd_devdata = {
 	.card = &snd_soc_card_magician,
 	.codec_dev = &soc_codec_dev_uda1380,
-	.codec_data = &magician_uda1380_setup,
 };
 
 static struct platform_device *magician_snd_device;
 
+/*
+ * FIXME: move into magician board file once merged into the pxa tree
+ */
+static struct uda1380_platform_data uda1380_info = {
+	.gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
+	.gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
+	.dac_clk    = UDA1380_DAC_CLK_WSPLL,
+};
+
+static struct i2c_board_info i2c_board_info[] = {
+	{
+		I2C_BOARD_INFO("uda1380", 0x18),
+		.platform_data = &uda1380_info,
+	},
+};
+
 static int __init magician_init(void)
 {
 	int ret;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
 
 	if (!machine_is_magician())
 		return -ENODEV;
 
-	ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
-	if (ret)
-		goto err_request_power;
-	ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
-	if (ret)
-		goto err_request_reset;
+	adapter = i2c_get_adapter(0);
+	if (!adapter)
+		return -ENODEV;
+	client = i2c_new_device(adapter, i2c_board_info);
+	i2c_put_adapter(adapter);
+	if (!client)
+		return -ENODEV;
+
 	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
 	if (ret)
 		goto err_request_spk;
@@ -491,14 +506,8 @@ static int __init magician_init(void)
 	if (ret)
 		goto err_request_in_sel1;
 
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
 	gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
 
-	/* we may need to have the clock running here - pH5 */
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
-	udelay(5);
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
-
 	magician_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!magician_snd_device) {
 		ret = -ENOMEM;
@@ -526,10 +535,6 @@ static int __init magician_init(void)
 err_request_ep:
 	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
 err_request_spk:
-	gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
-err_request_reset:
-	gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
-err_request_power:
 	return ret;
 }
 
@@ -540,15 +545,12 @@ static void __exit magician_exit(void)
 	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
 	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
 	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
-	gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
 
 	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
 	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
 	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
 	gpio_free(EGPIO_MAGICIAN_EP_POWER);
 	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
-	gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
-	gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
 }
 
 module_init(magician_init);

sound/soc/pxa/pxa-ssp.c

@@ -457,31 +457,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	ssp_write_reg(ssp, SSCR0, sscr0);
-	ssp_write_reg(ssp, SSCR1, sscr1);
-	ssp_write_reg(ssp, SSPSP, sspsp);
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		sspsp |= SSPSP_SFRMP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		sspsp |= SSPSP_SCMODE(2);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
+		break;
+	default:
+		return -EINVAL;
+	}
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		sscr0 |= SSCR0_PSP;
 		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
-
 		/* See hw_params() */
-		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-		case SND_SOC_DAIFMT_NB_NF:
-			sspsp |= SSPSP_SFRMP;
-			break;
-		case SND_SOC_DAIFMT_NB_IF:
-			break;
-		case SND_SOC_DAIFMT_IB_IF:
-			sspsp |= SSPSP_SCMODE(2);
-			break;
-		case SND_SOC_DAIFMT_IB_NF:
-			sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
-			break;
-		default:
-			return -EINVAL;
-		}
 		break;
 
 	case SND_SOC_DAIFMT_DSP_A:
@@ -489,22 +485,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_DSP_B:
 		sscr0 |= SSCR0_MOD | SSCR0_PSP;
 		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
-
-		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-		case SND_SOC_DAIFMT_NB_NF:
-			sspsp |= SSPSP_SFRMP;
-			break;
-		case SND_SOC_DAIFMT_NB_IF:
-			break;
-		case SND_SOC_DAIFMT_IB_IF:
-			sspsp |= SSPSP_SCMODE(2);
-			break;
-		case SND_SOC_DAIFMT_IB_NF:
-			sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
-			break;
-		default:
-			return -EINVAL;
-		}
 		break;
 
 	default:

sound/soc/soc-core.c

@@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev)
 	return 0;
 }
 
+/**
+ * snd_soc_suspend_device: Notify core of device suspend
+ *
+ * @dev: Device being suspended.
+ *
+ * In order to ensure that the entire audio subsystem is suspended in a
+ * coordinated fashion ASoC devices should suspend themselves when
+ * called by ASoC.  When the standard kernel suspend process asks the
+ * device to suspend it should call this function to initiate a suspend
+ * of the entire ASoC card.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_suspend_device(struct device *dev)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_suspend_device);
+
+/**
+ * snd_soc_resume_device: Notify core of device resume
+ *
+ * @dev: Device being resumed.
+ *
+ * In order to ensure that the entire audio subsystem is resumed in a
+ * coordinated fashion ASoC devices should resume themselves when called
+ * by ASoC.  When the standard kernel resume process asks the device
+ * to resume it should call this function.  Once all the components of
+ * the card have notified that they are ready to be resumed the card
+ * will be resumed.
+ *
+ * \note Currently this function is stubbed out.
+ */
+int snd_soc_resume_device(struct device *dev)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_resume_device);
+
 #else
 #define soc_suspend	NULL
 #define soc_resume	NULL
@@ -981,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static void soc_shutdown(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
+
+	if (!card->instantiated)
+		return;
+
+	/* Flush out pmdown_time work - we actually do want to run it
+	 * now, we're shutting down so no imminent restart. */
+	run_delayed_work(&card->delayed_work);
+
+	snd_soc_dapm_shutdown(socdev);
+}
+
 /* ASoC platform driver */
 static struct platform_driver soc_driver = {
 	.driver		= {
@@ -991,6 +1045,7 @@ static struct platform_driver soc_driver = {
 	.remove		= soc_remove,
 	.suspend	= soc_suspend,
 	.resume		= soc_resume,
+	.shutdown	= soc_shutdown,
 };
 
 /* create a new pcm */
@@ -1264,10 +1319,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
  * Returns 1 for change else 0.
  */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value)
+				unsigned int mask, unsigned int value)
 {
 	int change;
-	unsigned short old, new;
+	unsigned int old, new;
 
 	mutex_lock(&io_mutex);
 	old = snd_soc_read(codec, reg);
@@ -1294,10 +1349,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits);
  * Returns 1 for change else 0.
  */
 int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value)
+				unsigned int mask, unsigned int value)
 {
 	int change;
-	unsigned short old, new;
+	unsigned int old, new;
 
 	mutex_lock(&io_mutex);
 	old = snd_soc_read(codec, reg);
@@ -1586,7 +1641,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned short val, bitmask;
+	unsigned int val, bitmask;
 
 	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
 		;
@@ -1615,8 +1670,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned short val;
-	unsigned short mask, bitmask;
+	unsigned int val;
+	unsigned int mask, bitmask;
 
 	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
 		;
@@ -1652,7 +1707,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned short reg_val, val, mux;
+	unsigned int reg_val, val, mux;
 
 	reg_val = snd_soc_read(codec, e->reg);
 	val = (reg_val >> e->shift_l) & e->mask;
@@ -1691,8 +1746,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned short val;
-	unsigned short mask;
+	unsigned int val;
+	unsigned int mask;
 
 	if (ucontrol->value.enumerated.item[0] > e->max - 1)
 		return -EINVAL;
@@ -1852,7 +1907,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
-	unsigned short val, val2, val_mask;
+	unsigned int val, val2, val_mask;
 
 	val = (ucontrol->value.integer.value[0] & mask);
 	if (invert)
@@ -1918,7 +1973,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
 	unsigned int reg2 = mc->rreg;
 	unsigned int shift = mc->shift;
 	int max = mc->max;
-	unsigned int mask = (1<<fls(max))-1;
+	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 
 	ucontrol->value.integer.value[0] =
@@ -1958,7 +2013,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 	int err;
-	unsigned short val, val2, val_mask;
+	unsigned int val, val2, val_mask;
 
 	val_mask = mask << shift;
 	val = (ucontrol->value.integer.value[0] & mask);
@@ -2050,7 +2105,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	unsigned int reg = mc->reg;
 	int min = mc->min;
-	unsigned short val;
+	unsigned int val;
 
 	val = (ucontrol->value.integer.value[0]+min) & 0xff;
 	val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;