Merge remote branch 'broonie-asoc/for-2.6.37' into for-2.6.37

sound/soc/atmel/atmel_ssc_dai.c

@@ -789,13 +789,14 @@ static struct snd_soc_dai_driver atmel_ssc_dai[NUM_SSC_DEVICES] = {
 
 static __devinit int asoc_ssc_probe(struct platform_device *pdev)
 {
-	return snd_soc_register_dais(&pdev->dev, atmel_ssc_dai,
-			ARRAY_SIZE(atmel_ssc_dai));
+	BUG_ON(pdev->id < 0);
+	BUG_ON(pdev->id >= ARRAY_SIZE(atmel_ssc_dai));
+	return snd_soc_register_dai(&pdev->dev, &atmel_ssc_dai[pdev->id]);
 }
 
 static int __devexit asoc_ssc_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(atmel_ssc_dai));
+	snd_soc_unregister_dai(&pdev->dev);
 	return 0;
 }
 
@@ -809,6 +810,56 @@ static struct platform_driver asoc_ssc_driver = {
 	.remove = __devexit_p(asoc_ssc_remove),
 };
 
+/**
+ * atmel_ssc_set_audio - Allocate the specified SSC for audio use.
+ */
+int atmel_ssc_set_audio(int ssc_id)
+{
+	struct ssc_device *ssc;
+	static struct platform_device *dma_pdev;
+	struct platform_device *ssc_pdev;
+	int ret;
+
+	if (ssc_id < 0 || ssc_id >= ARRAY_SIZE(atmel_ssc_dai))
+		return -EINVAL;
+
+	/* Allocate a dummy device for DMA if we don't have one already */
+	if (!dma_pdev) {
+		dma_pdev = platform_device_alloc("atmel-pcm-audio", -1);
+		if (!dma_pdev)
+			return -ENOMEM;
+
+		ret = platform_device_add(dma_pdev);
+		if (ret < 0) {
+			platform_device_put(dma_pdev);
+			dma_pdev = NULL;
+			return ret;
+		}
+	}
+
+	ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
+	if (!ssc_pdev) {
+		ssc_free(ssc);
+		return -ENOMEM;
+	}
+
+	/* If we can grab the SSC briefly to parent the DAI device off it */
+	ssc = ssc_request(ssc_id);
+	if (IS_ERR(ssc))
+		pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
+			PTR_ERR(ssc));
+	else
+		ssc_pdev->dev.parent = &(ssc->pdev->dev);
+	ssc_free(ssc);
+
+	ret = platform_device_add(ssc_pdev);
+	if (ret < 0)
+		platform_device_put(ssc_pdev);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(atmel_ssc_set_audio);
+
 static int __init snd_atmel_ssc_init(void)
 {
 	return platform_driver_register(&asoc_ssc_driver);

sound/soc/atmel/atmel_ssc_dai.h

@@ -117,4 +117,6 @@ struct atmel_ssc_info {
 	struct atmel_ssc_state ssc_state;
 };
 
+int atmel_ssc_set_audio(int ssc);
+
 #endif /* _AT91_SSC_DAI_H */

sound/soc/atmel/sam9g20_wm8731.c

@@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 			"at91sam9g20ek_wm8731 "
 			": at91sam9g20ek_wm8731_init() called\n");
 
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
 		MCLK_RATE, SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret);
@@ -183,8 +183,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 	.cpu_dai_name = "atmel-ssc-dai.0",
 	.codec_dai_name = "wm8731-hifi",
 	.init = at91sam9g20ek_wm8731_init,
-	.platform_name = "atmel_pcm-audio",
-	.codec_name = "wm8731-codec.0-001a",
+	.platform_name = "atmel-pcm-audio",
+	.codec_name = "wm8731-codec.0-001b",
 	.ops = &at91sam9g20ek_ops,
 };
 
@@ -205,6 +205,12 @@ static int __init at91sam9g20ek_init(void)
 	if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc()))
 		return -ENODEV;
 
+	ret = atmel_ssc_set_audio(0);
+	if (ret != 0) {
+		pr_err("Failed to set SSC 0 for audio: %d\n", ret);
+		return ret;
+	}
+
 	/*
 	 * Codec MCLK is supplied by PCK0 - set it up.
 	 */

sound/soc/au1x/db1200.c

@@ -49,7 +49,7 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream)
 	int ret;
 
 	/* WM8731 has its own 12MHz crystal */
-	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
 				12000000, SND_SOC_CLOCK_IN);
 
 	/* codec is bitclock and lrclk master */

sound/soc/codecs/88pm860x-codec.h

+/*
+ * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ *	Haojian Zhuang <haojian.zhuang@marvell.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.
+ */
+
+#ifndef __88PM860X_H
+#define __88PM860X_H
+
+/* The offset of these registers are 0xb0 */
+#define PM860X_PCM_IFACE_1		0x00
+#define PM860X_PCM_IFACE_2		0x01
+#define PM860X_PCM_IFACE_3		0x02
+#define PM860X_PCM_RATE			0x03
+#define PM860X_EC_PATH			0x04
+#define PM860X_SIDETONE_L_GAIN		0x05
+#define PM860X_SIDETONE_R_GAIN		0x06
+#define PM860X_SIDETONE_SHIFT		0x07
+#define PM860X_ADC_OFFSET_1		0x08
+#define PM860X_ADC_OFFSET_2		0x09
+#define PM860X_DMIC_DELAY		0x0a
+
+#define PM860X_I2S_IFACE_1		0x0b
+#define PM860X_I2S_IFACE_2		0x0c
+#define PM860X_I2S_IFACE_3		0x0d
+#define PM860X_I2S_IFACE_4		0x0e
+#define PM860X_EQUALIZER_N0_1		0x0f
+#define PM860X_EQUALIZER_N0_2		0x10
+#define PM860X_EQUALIZER_N1_1		0x11
+#define PM860X_EQUALIZER_N1_2		0x12
+#define PM860X_EQUALIZER_D1_1		0x13
+#define PM860X_EQUALIZER_D1_2		0x14
+#define PM860X_LOFI_GAIN_LEFT		0x15
+#define PM860X_LOFI_GAIN_RIGHT		0x16
+#define PM860X_HIFIL_GAIN_LEFT		0x17
+#define PM860X_HIFIL_GAIN_RIGHT		0x18
+#define PM860X_HIFIR_GAIN_LEFT		0x19
+#define PM860X_HIFIR_GAIN_RIGHT		0x1a
+#define PM860X_DAC_OFFSET		0x1b
+#define PM860X_OFFSET_LEFT_1		0x1c
+#define PM860X_OFFSET_LEFT_2		0x1d
+#define PM860X_OFFSET_RIGHT_1		0x1e
+#define PM860X_OFFSET_RIGHT_2		0x1f
+#define PM860X_ADC_ANA_1		0x20
+#define PM860X_ADC_ANA_2		0x21
+#define PM860X_ADC_ANA_3		0x22
+#define PM860X_ADC_ANA_4		0x23
+#define PM860X_ANA_TO_ANA		0x24
+#define PM860X_HS1_CTRL			0x25
+#define PM860X_HS2_CTRL			0x26
+#define PM860X_LO1_CTRL			0x27
+#define PM860X_LO2_CTRL			0x28
+#define PM860X_EAR_CTRL_1		0x29
+#define PM860X_EAR_CTRL_2		0x2a
+#define PM860X_AUDIO_SUPPLIES_1		0x2b
+#define PM860X_AUDIO_SUPPLIES_2		0x2c
+#define PM860X_ADC_EN_1			0x2d
+#define PM860X_ADC_EN_2			0x2e
+#define PM860X_DAC_EN_1			0x2f
+#define PM860X_DAC_EN_2			0x31
+#define PM860X_AUDIO_CAL_1		0x32
+#define PM860X_AUDIO_CAL_2		0x33
+#define PM860X_AUDIO_CAL_3		0x34
+#define PM860X_AUDIO_CAL_4		0x35
+#define PM860X_AUDIO_CAL_5		0x36
+#define PM860X_ANA_INPUT_SEL_1		0x37
+#define PM860X_ANA_INPUT_SEL_2		0x38
+
+#define PM860X_PCM_IFACE_4		0x39
+#define PM860X_I2S_IFACE_5		0x3a
+
+#define PM860X_SHORTS			0x3b
+#define PM860X_PLL_ADJ_1		0x3c
+#define PM860X_PLL_ADJ_2		0x3d
+
+/* bits definition */
+#define PM860X_CLK_DIR_IN		0
+#define PM860X_CLK_DIR_OUT		1
+
+#define PM860X_DET_HEADSET		(1 << 0)
+#define PM860X_DET_MIC			(1 << 1)
+#define PM860X_DET_HOOK			(1 << 2)
+#define PM860X_SHORT_HEADSET		(1 << 3)
+#define PM860X_SHORT_LINEOUT		(1 << 4)
+#define PM860X_DET_MASK			0x1F
+
+extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+				 int, int, int, int);
+extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+				  int);
+
+#endif	/* __88PM860X_H */

sound/soc/codecs/Kconfig

@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
@@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS
 
           If unsure select "N".
 
+config SND_SOC_88PM860X
+	tristate
+
 config SND_SOC_WM_HUBS
 	tristate
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y

sound/soc/codecs/Makefile

+snd-soc-88pm860x-objs := 88pm860x-codec.o
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
@@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm9090-objs := wm9090.o
 
+obj-$(CONFIG_SND_SOC_88PM860X)	+= snd-soc-88pm860x.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o

sound/soc/codecs/cx20442.c

@@ -318,7 +318,7 @@ EXPORT_SYMBOL_GPL(v253_ops);
  */
 
 static struct snd_soc_dai_driver cx20442_dai = {
-	.name = "cx20442-hifi",
+	.name = "cx20442-voice",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,

sound/soc/codecs/tlv320aic3x.c

@@ -12,11 +12,11 @@
  *
  * Notes:
  *  The AIC3X is a driver for a low power stereo audio
- *  codecs aic31, aic32, aic33.
+ *  codecs aic31, aic32, aic33, aic3007.
  *
  *  It supports full aic33 codec functionality.
- *  The compatibility with aic32, aic31 is as follows:
- *        aic32        |        aic31
+ *  The compatibility with aic32, aic31 and aic3007 is as follows:
+ *    aic32/aic3007    |        aic31
  *  ---------------------------------------
  *   MONO_LOUT -> N/A  |  MONO_LOUT -> N/A
  *                     |  IN1L -> LINE1L
@@ -70,6 +70,10 @@ struct aic3x_priv {
 	unsigned int sysclk;
 	int master;
 	int gpio_reset;
+#define AIC3X_MODEL_3X 0
+#define AIC3X_MODEL_33 1
+#define AIC3X_MODEL_3007 2
+	u16 model;
 };
 
 /*
@@ -361,6 +365,14 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
+/*
+ * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
+	SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
+
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
 SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -589,6 +601,15 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("LINE2R"),
 };
 
+static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = {
+	/* Class-D outputs */
+	SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("SPOP"),
+	SND_SOC_DAPM_OUTPUT("SPOM"),
+};
+
 static const struct snd_soc_dapm_route intercon[] = {
 	/* Left Output */
 	{"Left DAC Mux", "DAC_L1", "Left DAC"},
@@ -759,14 +780,30 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
 };
 
+static const struct snd_soc_dapm_route intercon_3007[] = {
+	/* Class-D outputs */
+	{"Left Class-D Out", NULL, "Left Line Out"},
+	{"Right Class-D Out", NULL, "Left Line Out"},
+	{"SPOP", NULL, "Left Class-D Out"},
+	{"SPOM", NULL, "Right Class-D Out"},
+};
+
 static int aic3x_add_widgets(struct snd_soc_codec *codec)
 {
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+
 	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* set up audio path interconnects */
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
+	if (aic3x->model == AIC3X_MODEL_3007) {
+		snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets,
+			ARRAY_SIZE(aic3007_dapm_widgets));
+		snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007));
+	}
+
 	return 0;
 }
 
@@ -1117,6 +1154,7 @@ static struct snd_soc_dai_driver aic3x_dai = {
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
 	.ops = &aic3x_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state)
@@ -1150,6 +1188,7 @@ static int aic3x_resume(struct snd_soc_codec *codec)
  */
 static int aic3x_init(struct snd_soc_codec *codec)
 {
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int reg;
 
 	aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
@@ -1218,6 +1257,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
 	aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
+	if (aic3x->model == AIC3X_MODEL_3007) {
+		/* Class-D speaker driver init; datasheet p. 46 */
+		aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D);
+		aic3x_write(codec, 0xD, 0x0D);
+		aic3x_write(codec, 0x8, 0x5C);
+		aic3x_write(codec, 0x8, 0x5D);
+		aic3x_write(codec, 0x8, 0x5C);
+		aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00);
+		aic3x_write(codec, CLASSD_CTRL, 0);
+	}
+
 	/* off, with power on */
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1243,6 +1293,8 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 
 	snd_soc_add_controls(codec, aic3x_snd_controls,
 			     ARRAY_SIZE(aic3x_snd_controls));
+	if (aic3x->model == AIC3X_MODEL_3007)
+		snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
 
 	aic3x_add_widgets(codec);
 
@@ -1274,6 +1326,14 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
  * 0x18, 0x19, 0x1A, 0x1B
  */
 
+static const struct i2c_device_id aic3x_i2c_id[] = {
+	[AIC3X_MODEL_3X] = { "tlv320aic3x", 0 },
+	[AIC3X_MODEL_33] = { "tlv320aic33", 0 },
+	[AIC3X_MODEL_3007] = { "tlv320aic3007", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
+
 /*
  * If the i2c layer weren't so broken, we could pass this kind of data
  * around
@@ -1285,6 +1345,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 	struct aic3x_setup_data *setup = pdata->setup;
 	struct aic3x_priv *aic3x;
 	int ret, i;
+	const struct i2c_device_id *tbl;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
 	if (aic3x == NULL) {
@@ -1305,6 +1366,12 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		gpio_direction_output(aic3x->gpio_reset, 0);
 	}
 
+	for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) {
+		if (!strcmp(tbl->name, id->name))
+			break;
+	}
+	aic3x->model = tbl - aic3x_i2c_id;
+
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		aic3x->supplies[i].supply = aic3x_supply_names[i];
 
@@ -1359,13 +1426,6 @@ static int aic3x_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-static const struct i2c_device_id aic3x_i2c_id[] = {
-	{ "tlv320aic3x", 0 },
-	{ "tlv320aic33", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
-
 /* machine i2c codec control layer */
 static struct i2c_driver aic3x_i2c_driver = {
 	.driver = {

sound/soc/codecs/tlv320aic3x.h

@@ -111,6 +111,8 @@
 #define DACL1_2_MONOLOPM_VOL		75
 #define DACR1_2_MONOLOPM_VOL		78
 #define MONOLOPM_CTRL			79
+/* Class-D speaker driver on tlv320aic3007 */
+#define CLASSD_CTRL			73
 /* Line Output Plus/Minus control registers */
 #define LINE2L_2_LLOPM_VOL		80
 #define LINE2L_2_RLOPM_VOL		87

sound/soc/codecs/wm8731.c

@@ -46,6 +46,7 @@ struct wm8731_priv {
 	struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES];
 	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
+	int sysclk_type;
 };
 
 
@@ -110,6 +111,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
 	&wm8731_output_mixer_controls[0],
 	ARRAY_SIZE(wm8731_output_mixer_controls)),
@@ -127,7 +129,18 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
 SND_SOC_DAPM_INPUT("LLINEIN"),
 };
 
+static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
+			    struct snd_soc_dapm_widget *sink)
+{
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
+
+	return wm8731->sysclk_type == WM8731_SYSCLK_MCLK;
+}
+
 static const struct snd_soc_dapm_route intercon[] = {
+	{"DAC", NULL, "OSC", wm8731_check_osc},
+	{"ADC", NULL, "OSC", wm8731_check_osc},
+
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
 	{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -285,6 +298,15 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
+	switch (clk_id) {
+	case WM8731_SYSCLK_XTAL:
+	case WM8731_SYSCLK_MCLK:
+		wm8731->sysclk_type = clk_id;
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	switch (freq) {
 	case 11289600:
 	case 12000000:
@@ -292,9 +314,14 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	case 16934400:
 	case 18432000:
 		wm8731->sysclk = freq;
-		return 0;
-	}
+		break;
+	default:
 		return -EINVAL;
+	}
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
 }
 
 

sound/soc/codecs/wm8731.h

@@ -31,7 +31,9 @@
 
 #define WM8731_CACHEREGNUM 	10
 
-#define WM8731_SYSCLK	0
+#define WM8731_SYSCLK_XTAL 1
+#define WM8731_SYSCLK_MCLK 2
+
 #define WM8731_DAI		0
 
 #endif

sound/soc/fsl/Kconfig

 config SND_MPC52xx_DMA
 	tristate
 
-# ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
-# for the SSI and the Elo DMA controller.  You will still need to select
-# a platform driver and a codec driver.
-config SND_SOC_MPC8610
+# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and
+# an Elo DMA controller, such as the MPC8610 and P1022.  You will still need to
+# select a platform driver and a codec driver.
+config SND_SOC_POWERPC_SSI
 	tristate
-	depends on MPC8610
+	depends on FSL_SOC
 
 config SND_SOC_MPC8610_HPCD
 	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
 	# I2C is necessary for the CS4270 driver
 	depends on MPC8610_HPCD && I2C
-	select SND_SOC_MPC8610
+	select SND_SOC_POWERPC_SSI
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
 	default y if MPC8610_HPCD
 	help
 	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
 
+config SND_SOC_P1022_DS
+	tristate "ALSA SoC support for the Freescale P1022 DS board"
+	# I2C is necessary for the WM8776 driver
+	depends on P1022_DS && I2C
+	select SND_SOC_POWERPC_SSI
+	select SND_SOC_WM8776
+	default y if P1022_DS
+	help
+	  Say Y if you want to enable audio on the Freescale P1022 DS board.
+	  This will also include the Wolfson Microelectronics WM8776 codec
+	  driver.
+
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
 	depends on PPC_MPC52xx && PPC_BESTCOMM

sound/soc/fsl/Makefile

@@ -2,10 +2,14 @@
 snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
 obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
 
-# MPC8610 Platform Support
+# P1022 DS Machine Support
+snd-soc-p1022-ds-objs := p1022_ds.o
+obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
+
+# Freescale PowerPC SSI/DMA Platform Support
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-dma-objs := fsl_dma.o
-obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o

sound/soc/fsl/fsl_dma.c

@@ -23,6 +23,7 @@
 #include <linux/gfp.h>
 #include <linux/of_platform.h>
 #include <linux/list.h>
+#include <linux/slab.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -305,22 +306,30 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = fsl_dma_dmamask;
 
+	/* Some codecs have separate DAIs for playback and capture, so we
+	 * should allocate a DMA buffer only for the streams that are valid.
+	 */
+
+	if (dai->driver->playback.channels_min) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[0].substream->dma_buffer);
 		if (ret) {
-		dev_err(card->dev, "can't allocate playback dma buffer\n");
+			dev_err(card->dev, "can't alloc playback dma buffer\n");
 			return ret;
 		}
+	}
 
+	if (dai->driver->capture.channels_min) {
 		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
 			fsl_dma_hardware.buffer_bytes_max,
 			&pcm->streams[1].substream->dma_buffer);
 		if (ret) {
 			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
-		dev_err(card->dev, "can't allocate capture dma buffer\n");
+			dev_err(card->dev, "can't alloc capture dma buffer\n");
 			return ret;
 		}
+	}
 
 	return 0;
 }
@@ -887,11 +896,11 @@ static struct snd_pcm_ops fsl_dma_ops = {
 	.pointer	= fsl_dma_pointer,
 };
 
-static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
+static int __devinit fsl_soc_dma_probe(struct platform_device *pdev,
 				       const struct of_device_id *match)
  {
 	struct dma_object *dma;
-	struct device_node *np = of_dev->dev.of_node;
+	struct device_node *np = pdev->dev.of_node;
 	struct device_node *ssi_np;
 	struct resource res;
 	const uint32_t *iprop;
@@ -900,13 +909,13 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	/* Find the SSI node that points to us. */
 	ssi_np = find_ssi_node(np);
 	if (!ssi_np) {
-		dev_err(&of_dev->dev, "cannot find parent SSI node\n");
+		dev_err(&pdev->dev, "cannot find parent SSI node\n");
 		return -ENODEV;
 	}
 
 	ret = of_address_to_resource(ssi_np, 0, &res);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not determine resources for %s\n",
+		dev_err(&pdev->dev, "could not determine resources for %s\n",
 			ssi_np->full_name);
 		of_node_put(ssi_np);
 		return ret;
@@ -914,7 +923,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 
 	dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
 	if (!dma) {
-		dev_err(&of_dev->dev, "could not allocate dma object\n");
+		dev_err(&pdev->dev, "could not allocate dma object\n");
 		of_node_put(ssi_np);
 		return -ENOMEM;
 	}
@@ -937,9 +946,9 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 
 	of_node_put(ssi_np);
 
-	ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
+	ret = snd_soc_register_platform(&pdev->dev, &dma->dai);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not register platform\n");
+		dev_err(&pdev->dev, "could not register platform\n");
 		kfree(dma);
 		return ret;
 	}
@@ -947,16 +956,16 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	dma->channel = of_iomap(np, 0);
 	dma->irq = irq_of_parse_and_map(np, 0);
 
-	dev_set_drvdata(&of_dev->dev, dma);
+	dev_set_drvdata(&pdev->dev, dma);
 
 	return 0;
 }
 
-static int __devexit fsl_soc_dma_remove(struct of_device *of_dev)
+static int __devexit fsl_soc_dma_remove(struct platform_device *pdev)
 {
-	struct dma_object *dma = dev_get_drvdata(&of_dev->dev);
+	struct dma_object *dma = dev_get_drvdata(&pdev->dev);
 
-	snd_soc_unregister_platform(&of_dev->dev);
+	snd_soc_unregister_platform(&pdev->dev);
 	iounmap(dma->channel);
 	irq_dispose_mapping(dma->irq);
 	kfree(dma);

sound/soc/fsl/fsl_ssi.c

@@ -624,13 +624,13 @@ static void make_lowercase(char *s)
 	}
 }
 
-static int __devinit fsl_ssi_probe(struct of_device *of_dev,
+static int __devinit fsl_ssi_probe(struct platform_device *pdev,
 				   const struct of_device_id *match)
 {
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
 	struct device_attribute *dev_attr = NULL;
-	struct device_node *np = of_dev->dev.of_node;
+	struct device_node *np = pdev->dev.of_node;
 	const char *p, *sprop;
 	const uint32_t *iprop;
 	struct resource res;
@@ -645,14 +645,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 
 	/* Check for a codec-handle property. */
 	if (!of_get_property(np, "codec-handle", NULL)) {
-		dev_err(&of_dev->dev, "missing codec-handle property\n");
+		dev_err(&pdev->dev, "missing codec-handle property\n");
 		return -ENODEV;
 	}
 
 	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (!sprop || strcmp(sprop, "i2s-slave")) {
-		dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop);
+		dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
 		return -ENODEV;
 	}
 
@@ -661,7 +661,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p),
 			      GFP_KERNEL);
 	if (!ssi_private) {
-		dev_err(&of_dev->dev, "could not allocate DAI object\n");
+		dev_err(&pdev->dev, "could not allocate DAI object\n");
 		return -ENOMEM;
 	}
 
@@ -675,7 +675,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	/* Get the addresses and IRQ */
 	ret = of_address_to_resource(np, 0, &res);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not determine device resources\n");
+		dev_err(&pdev->dev, "could not determine device resources\n");
 		kfree(ssi_private);
 		return ret;
 	}
@@ -703,19 +703,19 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	dev_attr->attr.mode = S_IRUGO;
 	dev_attr->show = fsl_sysfs_ssi_show;
 
-	ret = device_create_file(&of_dev->dev, dev_attr);
+	ret = device_create_file(&pdev->dev, dev_attr);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not create sysfs %s file\n",
+		dev_err(&pdev->dev, "could not create sysfs %s file\n",
 			ssi_private->dev_attr.attr.name);
 		goto error;
 	}
 
 	/* Register with ASoC */
-	dev_set_drvdata(&of_dev->dev, ssi_private);
+	dev_set_drvdata(&pdev->dev, ssi_private);
 
-	ret = snd_soc_register_dai(&of_dev->dev, &ssi_private->cpu_dai_drv);
+	ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
 	if (ret) {
-		dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret);
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		goto error;
 	}
 
@@ -733,20 +733,20 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	make_lowercase(name);
 
 	ssi_private->pdev =
-		platform_device_register_data(&of_dev->dev, name, 0, NULL, 0);
+		platform_device_register_data(&pdev->dev, name, 0, NULL, 0);
 	if (IS_ERR(ssi_private->pdev)) {
 		ret = PTR_ERR(ssi_private->pdev);
-		dev_err(&of_dev->dev, "failed to register platform: %d\n", ret);
+		dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
 		goto error;
 	}
 
 	return 0;
 
 error:
-	snd_soc_unregister_dai(&of_dev->dev);
-	dev_set_drvdata(&of_dev->dev, NULL);
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
 	if (dev_attr)
-		device_remove_file(&of_dev->dev, dev_attr);
+		device_remove_file(&pdev->dev, dev_attr);
 	irq_dispose_mapping(ssi_private->irq);
 	iounmap(ssi_private->ssi);
 	kfree(ssi_private);
@@ -754,16 +754,16 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	return ret;
 }
 
-static int fsl_ssi_remove(struct of_device *of_dev)
+static int fsl_ssi_remove(struct platform_device *pdev)
 {
-	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev);
+	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
 
 	platform_device_unregister(ssi_private->pdev);
-	snd_soc_unregister_dai(&of_dev->dev);
-	device_remove_file(&of_dev->dev, &ssi_private->dev_attr);
+	snd_soc_unregister_dai(&pdev->dev);
+	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 
 	kfree(ssi_private);
-	dev_set_drvdata(&of_dev->dev, NULL);
+	dev_set_drvdata(&pdev->dev, NULL);
 
 	return 0;
 }

sound/soc/fsl/mpc8610_hpcd.c

@@ -13,6 +13,7 @@
 #include <linux/module.h>
 #include <linux/interrupt.h>
 #include <linux/of_device.h>
+#include <linux/slab.h>
 #include <sound/soc.h>
 #include <asm/fsl_guts.h>
 
@@ -323,9 +324,10 @@ static int get_dma_channel(struct device_node *ssi_np,
 static int mpc8610_hpcd_probe(struct platform_device *pdev)
 {
 	struct device *dev = pdev->dev.parent;
-	/* of_dev is the OF device for the SSI node that probed us */
-	struct of_device *of_dev = container_of(dev, struct of_device, dev);
-	struct device_node *np = of_dev->dev.of_node;
+	/* ssi_pdev is the platform device for the SSI node that probed us */
+	struct platform_device *ssi_pdev =
+		container_of(dev, struct platform_device, dev);
+	struct device_node *np = ssi_pdev->dev.of_node;
 	struct device_node *codec_np = NULL;
 	struct platform_device *sound_device = NULL;
 	struct mpc8610_hpcd_data *machine_data;
@@ -348,7 +350,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	if (!machine_data)
 		return -ENOMEM;
 
-	machine_data->dai[0].cpu_dai_name = dev_name(&of_dev->dev);
+	machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	machine_data->dai[0].ops = &mpc8610_hpcd_ops;
 
 	/* Determine the codec name, it will be used as the codec DAI name */

sound/soc/omap/ams-delta.c

@@ -584,7 +584,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
 	.name = "CX20442",
 	.stream_name = "CX20442",
 	.cpu_dai_name ="omap-mcbsp-dai.0",
-	.codec_dai_name = "cx20442-hifi",
+	.codec_dai_name = "cx20442-voice",
 	.init = ams_delta_cx20442_init,
 	.platform_name = "omap-pcm-audio",
 	.codec_name = "cx20442-codec",

sound/soc/pxa/Kconfig

@@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X
 	  Say Y if you want to add support for SoC audio on
 	  Palm T|X, T5, E2 or LifeDrive handheld computer.
 
+config SND_SOC_SAARB
+	tristate "SoC Audio support for Marvell Saarb"
+	depends on SND_PXA2XX_SOC && MACH_SAARB
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
+config SND_SOC_TAVOREVB3
+	tristate "SoC Audio support for Marvell Tavor EVB3"
+	depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE

sound/soc/pxa/Makefile

@@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
+snd-soc-saarb-objs := saarb.o
+snd-soc-tavorevb3-objs := tavorevb3.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
@@ -38,6 +40,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
+obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
+obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o

sound/soc/pxa/corgi.c

@@ -149,7 +149,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;

sound/soc/pxa/e740_wm9705.c

@@ -198,6 +198,9 @@ static int __init e740_init(void)
 static void __exit e740_exit(void)
 {
 	platform_device_unregister(e740_snd_device);
+	gpio_free(GPIO_E740_WM9705_nAVDD2);
+	gpio_free(GPIO_E740_AMP_ON);
+	gpio_free(GPIO_E740_MIC_ON);
 }
 
 module_init(e740_init);

sound/soc/pxa/poodle.c

@@ -128,7 +128,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;

sound/soc/pxa/pxa2xx-ac97.c

@@ -24,7 +24,6 @@
 #include <mach/dma.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)

sound/soc/pxa/saarb.c

+/*
+ * saarb.c -- SoC audio for saarb
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.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/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *saarb_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* saarb machine dapm widgets */
+static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* saarb machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int saarb_i2s_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->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	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)
+		return ret;
+	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)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+
+	return ret;
+}
+
+static struct snd_soc_ops saarb_i2s_ops = {
+	.hw_params	= saarb_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link saarb_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= saarb_pm860x_init,
+		.ops		= &saarb_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_saarb = {
+	.name = "Saarb",
+	.dai_link = saarb_dai,
+	.num_links = ARRAY_SIZE(saarb_dai),
+};
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
+				  ARRAY_SIZE(saarb_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init saarb_init(void)
+{
+	int ret;
+
+	if (!machine_is_saarb())
+		return -ENODEV;
+	saarb_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!saarb_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
+
+	ret = platform_device_add(saarb_snd_device);
+	if (ret)
+		platform_device_put(saarb_snd_device);
+
+	return ret;
+}
+
+static void __exit saarb_exit(void)
+{
+	platform_device_unregister(saarb_snd_device);
+}
+
+module_init(saarb_init);
+module_exit(saarb_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
+MODULE_LICENSE("GPL");

sound/soc/pxa/tavorevb3.c

+/*
+ * tavorevb3.c -- SoC audio for Tavor EVB3
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.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/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *evb3_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* tavorevb3 machine dapm widgets */
+static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* tavorevb3 machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int evb3_i2s_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->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	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)
+		return ret;
+
+	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)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+	return ret;
+}
+
+static struct snd_soc_ops evb3_i2s_ops = {
+	.hw_params	= evb3_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link evb3_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= evb3_pm860x_init,
+		.ops		= &evb3_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_evb3 = {
+	.name = "Tavor EVB3",
+	.dai_link = evb3_dai,
+	.num_links = ARRAY_SIZE(evb3_dai),
+};
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
+				  ARRAY_SIZE(evb3_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init tavorevb3_init(void)
+{
+	int ret;
+
+	if (!machine_is_tavorevb3())
+		return -ENODEV;
+	evb3_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!evb3_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
+
+	ret = platform_device_add(evb3_snd_device);
+	if (ret)
+		platform_device_put(evb3_snd_device);
+
+	return ret;
+}
+
+static void __exit tavorevb3_exit(void)
+{
+	platform_device_unregister(evb3_snd_device);
+}
+
+module_init(tavorevb3_init);
+module_exit(tavorevb3_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
+MODULE_LICENSE("GPL");

sound/soc/soc-core.c

@@ -3122,10 +3122,12 @@ int snd_soc_register_codec(struct device *dev,
 		fixup_codec_formats(&dai_drv[i].capture);
 	}
 
-	/* register DAIs */
+	/* register any DAIs */
+	if (num_dai) {
 		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
 		if (ret < 0)
 			goto error;
+	}
 
 	mutex_lock(&client_mutex);
 	list_add(&codec->list, &codec_list);
@@ -3164,6 +3166,7 @@ void snd_soc_unregister_codec(struct device *dev)
 	return;
 
 found:
+	if (codec->num_dai)
 		for (i = 0; i < codec->num_dai; i++)
 			snd_soc_unregister_dai(dev);