Merge remote-tracking branches 'asoc/fix/ux500' and 'asoc/fix/wm8994' into for-3.7

13 files changed, 96 insertions, 63 deletions

diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index bcbf1d00aa8..99f32f7c069 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,8 +1,9 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
 
-snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
-obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
+ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
+snd-soc-core-objs += soc-dmaengine-pcm.o
+endif
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 185d8dd3639..f379b085c39 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -178,6 +178,12 @@
 #define DA9055_AIF_WORD_S24_LE		(2 << 2)
 #define DA9055_AIF_WORD_S32_LE		(3 << 2)
 
+/* MIC_L_CTRL bit fields */
+#define DA9055_MIC_L_MUTE_EN		(1 << 6)
+
+/* MIC_R_CTRL bit fields */
+#define DA9055_MIC_R_MUTE_EN		(1 << 6)
+
 /* MIXIN_L_CTRL bit fields */
 #define DA9055_MIXIN_L_MIX_EN		(1 << 3)
 
@@ -476,7 +482,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	u8 reg_val, adc_left, adc_right;
+	u8 reg_val, adc_left, adc_right, mic_left, mic_right;
 	int avg_left_data, avg_right_data, offset_l, offset_r;
 
 	if (ucontrol->value.integer.value[0]) {
@@ -485,6 +491,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
 		 * offsets must be done first
 		 */
 
+		/* Save current values from Mic control registers */
+		mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL);
+		mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL);
+
+		/* Mute Mic PGA Left and Right */
+		snd_soc_update_bits(codec, DA9055_MIC_L_CTRL,
+				    DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN);
+		snd_soc_update_bits(codec, DA9055_MIC_R_CTRL,
+				    DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN);
+
 		/* Save current values from ADC control registers */
 		adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL);
 		adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL);
@@ -520,6 +536,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
 		/* Restore original values of ADC control registers */
 		snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left);
 		snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right);
+
+		/* Restore original values of Mic control registers */
+		snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left);
+		snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right);
 	}
 
 	return snd_soc_put_volsw(kcontrol, ucontrol);
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index e8f97af7592..00b85cc1b9a 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"VIBRA DAC", NULL, "Vibra Playback"},
 
 	/* ADC -> Stream mapping */
-	{"ADC Left", NULL, "Legacy Capture"},
-	{"ADC Left", NULL, "Capture"},
-	{"ADC Right", NULL, "Legacy Capture"},
-	{"ADC Right", NULL, "Capture"},
+	{"Legacy Capture" , NULL, "ADC Left"},
+	{"Capture", NULL, "ADC Left"},
+	{"Legacy Capture", NULL, "ADC Right"},
+	{"Capture" , NULL, "ADC Right"},
 
 	/* Capture path */
 	{"Analog Left Capture Route", "Headset Mic", "HSMIC"},
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index efa93dbb019..eab64a19398 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
 		 WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0,
 		 digital_tlv),
 SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
-	   WM2200_SPK1R_MUTE_SHIFT, 1, 0),
+	   WM2200_SPK1R_MUTE_SHIFT, 1, 1),
 };
 
 WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c,
 
 	switch (wm2200->rev) {
 	case 0:
+	case 1:
 		ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch,
 					    ARRAY_SIZE(wm2200_reva_patch));
 		if (ret != 0) {
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 2b2dadc54da..3fddc7ad112 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1045,6 +1045,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 		      struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994 *control = codec->control_data;
 	int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
 	int i;
@@ -1063,6 +1064,10 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
+		/* Don't enable timeslot 2 if not in use */
+		if (wm8994->channels[0] <= 2)
+			mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA);
+
 		val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1);
 		if ((val & WM8994_AIF1ADCL_SRC) &&
 		    (val & WM8994_AIF1ADCR_SRC))
@@ -2687,7 +2692,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	bclk_rate = params_rate(params) * 4;
+	bclk_rate = params_rate(params);
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bclk_rate *= 16;
@@ -2708,6 +2713,17 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	wm8994->channels[id] = params_channels(params);
+	switch (params_channels(params)) {
+	case 1:
+	case 2:
+		bclk_rate *= 2;
+		break;
+	default:
+		bclk_rate *= 4;
+		break;
+	}
+
 	/* Try to find an appropriate sample rate; look for an exact match. */
 	for (i = 0; i < ARRAY_SIZE(srs); i++)
 		if (srs[i].rate == params_rate(params))
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index f142ec198db..ccbce5791e9 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -77,6 +77,7 @@ struct wm8994_priv {
 	int sysclk_rate[2];
 	int mclk[2];
 	int aifclk[2];
+	int channels[2];
 	struct wm8994_fll_config fll[2], fll_suspend[2];
 	struct completion fll_locked[2];
 	bool fll_locked_irq;
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index dc0ee762662..d8e96b2cd03 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -575,56 +575,53 @@ static struct snd_soc_card ams_delta_audio_card = {
 };
 
 /* Module init/exit */
-static struct platform_device *ams_delta_audio_platform_device;
-static struct platform_device *cx20442_platform_device;
-
-static int __init ams_delta_module_init(void)
+static __devinit int ams_delta_probe(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = &ams_delta_audio_card;
 	int ret;
 
-	if (!(machine_is_ams_delta()))
-		return -ENODEV;
-
-	ams_delta_audio_platform_device =
-			platform_device_alloc("soc-audio", -1);
-	if (!ams_delta_audio_platform_device)
-		return -ENOMEM;
+	card->dev = &pdev->dev;
 
-	platform_set_drvdata(ams_delta_audio_platform_device,
-				&ams_delta_audio_card);
-
-	ret = platform_device_add(ams_delta_audio_platform_device);
-	if (ret)
-		goto err;
-
-	/*
-	 * Codec platform device could be registered from elsewhere (board?),
-	 * but I do it here as it makes sense only if used with the card.
-	 */
-	cx20442_platform_device =
-		platform_device_register_simple("cx20442-codec", -1, NULL, 0);
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		card->dev = NULL;
+		return ret;
+	}
 	return 0;
-err:
-	platform_device_put(ams_delta_audio_platform_device);
-	return ret;
 }
-late_initcall(ams_delta_module_init);
 
-static void __exit ams_delta_module_exit(void)
+static int __devexit ams_delta_remove(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
 	if (tty_unregister_ldisc(N_V253) != 0)
-		dev_warn(&ams_delta_audio_platform_device->dev,
+		dev_warn(&pdev->dev,
 			"failed to unregister V253 line discipline\n");
 
 	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
 			ARRAY_SIZE(ams_delta_hook_switch_gpios),
 			ams_delta_hook_switch_gpios);
 
-	platform_device_unregister(cx20442_platform_device);
-	platform_device_unregister(ams_delta_audio_platform_device);
+	snd_soc_unregister_card(card);
+	card->dev = NULL;
+	return 0;
 }
-module_exit(ams_delta_module_exit);
+
+#define DRV_NAME "ams-delta-audio"
+
+static struct platform_driver ams_delta_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = ams_delta_probe,
+	.remove = __devexit_p(ams_delta_remove),
+};
+
+module_platform_driver(ams_delta_driver);
 
 MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
 MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 4a73ef3ae12..a57a4e68dcc 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -216,7 +216,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
 	twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
 	twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
 	twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
-	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
 	twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
 	twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
 	twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index c02b001ee4b..56965bb3275 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -40,7 +40,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <plat/omap_hwmod.h>
 #include "omap-mcpdm.h"
 #include "omap-pcm.h"
 
@@ -260,13 +259,9 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
 	mutex_lock(&mcpdm->mutex);
 
 	if (!dai->active) {
-		/* Enable watch dog for ES above ES 1.0 to avoid saturation */
-		if (omap_rev() != OMAP4430_REV_ES1_0) {
-			u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+		u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
 
-			omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL,
-					 ctrl | MCPDM_WD_EN);
-		}
+		omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN);
 		omap_mcpdm_open_streams(mcpdm);
 	}
 	mutex_unlock(&mcpdm->mutex);
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 73ac5463c9e..e834faf859f 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -15,13 +15,13 @@
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
+#include <linux/platform_data/dma-mmp_tdma.h>
 #include <linux/platform_data/mmp_audio.h>
 #include <sound/pxa2xx-lib.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <mach/sram.h>
 #include <sound/dmaengine_pcm.h>
 
 struct mmp_dma_data {
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
index 5dc10dfc0d4..b0d46d63d55 100644
--- a/sound/soc/samsung/bells.c
+++ b/sound/soc/samsung/bells.c
@@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = {
 	{
 		.name = "Sub",
 		.stream_name = "Sub",
-		.cpu_dai_name = "wm5102-aif3",
+		.cpu_dai_name = "wm5110-aif3",
 		.codec_dai_name = "wm9081-hifi",
 		.codec_name = "wm9081.1-006c",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = {
 	{
 		.name = "Sub",
 		.stream_name = "Sub",
-		.cpu_dai_name = "wm5102-aif3",
+		.cpu_dai_name = "wm5110-aif3",
 		.codec_dai_name = "wm9081-hifi",
 		.codec_name = "wm9081.1-006c",
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 5328ae5539f..9d7f30774a4 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -20,6 +20,7 @@
 #include <linux/sh_dma.h>
 #include <linux/slab.h>
 #include <linux/module.h>
+#include <linux/workqueue.h>
 #include <sound/soc.h>
 #include <sound/sh_fsi.h>
 
@@ -223,7 +224,7 @@ struct fsi_stream {
 	 */
 	struct dma_chan		*chan;
 	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
-	struct tasklet_struct	tasklet;
+	struct work_struct	work;
 	dma_addr_t		dma;
 };
 
@@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data)
 	snd_pcm_period_elapsed(io->substream);
 }
 
-static void fsi_dma_do_tasklet(unsigned long data)
+static void fsi_dma_do_work(struct work_struct *work)
 {
-	struct fsi_stream *io = (struct fsi_stream *)data;
+	struct fsi_stream *io = container_of(work, struct fsi_stream, work);
 	struct fsi_priv *fsi = fsi_stream_to_priv(io);
 	struct snd_soc_dai *dai;
 	struct dma_async_tx_descriptor *desc;
@@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data)
 	 * FIXME
 	 *
 	 * In DMAEngine case, codec and FSI cannot be started simultaneously
-	 * since FSI is using tasklet.
+	 * since FSI is using the scheduler work queue.
 	 * Therefore, in capture case, probably FSI FIFO will have got
 	 * overflow error in this point.
 	 * in that case, DMA cannot start transfer until error was cleared.
@@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param)
 
 static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 {
-	tasklet_schedule(&io->tasklet);
+	schedule_work(&io->work);
 
 	return 0;
 }
@@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev
 		return fsi_stream_probe(fsi, dev);
 	}
 
-	tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
+	INIT_WORK(&io->work, fsi_dma_do_work);
 
 	return 0;
 }
 
 static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
 {
-	tasklet_kill(&io->tasklet);
+	cancel_work_sync(&io->work);
 
 	fsi_stream_stop(fsi, io);
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fa0fd8ddae9..1ab5fe04bfc 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -22,7 +22,7 @@
 
 /**
  * snd_soc_jack_new - Create a new jack
- * @card:  ASoC card
+ * @codec: ASoC codec
  * @id:    an identifying string for this jack
  * @type:  a bitmask of enum snd_jack_type values that can be detected by
  *         this jack
@@ -133,12 +133,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
 
 /**
  * snd_soc_jack_get_type - Based on the mic bias value, this function returns
- * the type of jack from the zones delcared in the jack type
+ * the type of jack from the zones declared in the jack type
  *
+ * @jack:  ASoC jack
  * @micbias_voltage:  mic bias voltage at adc channel when jack is plugged in
  *
  * Based on the mic bias value passed, this function helps identify
- * the type of jack from the already delcared jack zones
+ * the type of jack from the already declared jack zones
  */
 int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
 {