Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixes

24 files changed, 125 insertions, 80 deletions

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 6e240918189..bfee60c4d4c 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index b310702c646..ac35222ad0d 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Driver for Atmel AC97 controller");
-MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 2ca6f4f85b4..e3569bdd3b6 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -27,7 +27,6 @@
 #include "hpioctl.h"
 
 #include <linux/pci.h>
-#include <linux/version.h>
 #include <linux/init.h>
 #include <linux/jiffies.h>
 #include <linux/slab.h>
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index f16bc8aad6e..e083122ca55 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 			&((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
 		desc->addr = cpu_to_le32(addr);
 		desc->size = cpu_to_le32(period_bytes);
-		desc->ctlreserved = cpu_to_le32(PRD_EOP);
+		desc->ctlreserved = cpu_to_le16(PRD_EOP);
 		desc_addr += sizeof(struct cs5535audio_dma_desc);
 		addr += period_bytes;
 	}
@@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
 	lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods];
 	lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
 	lastdesc->size = 0;
-	lastdesc->ctlreserved = cpu_to_le32(PRD_JMP);
+	lastdesc->ctlreserved = cpu_to_le16(PRD_JMP);
 	jmpprd_addr = cpu_to_le32(lastdesc->addr +
 				  (sizeof(struct cs5535audio_dma_desc)*periods));
 
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b05f7be9dc1..e3e853153d1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e,
 		snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl);
 		goto out_fail;
 	} else
-		strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl);
+		strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1);
 
 	for (i = 0; i < e->sad_count; i++) {
 		if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 694b9daf691..7bbc5f237a5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3074,6 +3074,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 };
 
 static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
@@ -4389,6 +4390,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15069, .name = "CX20585",
 	  .patch = patch_cxt5066 },
+	{ .id = 0x14f1506c, .name = "CX20588",
+	  .patch = patch_cxt5066 },
 	{ .id = 0x14f1506e, .name = "CX20590",
 	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15097, .name = "CX20631",
@@ -4417,6 +4420,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
 MODULE_ALIAS("snd-hda-codec-id:14f15067");
 MODULE_ALIAS("snd-hda-codec-id:14f15068");
 MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506c");
 MODULE_ALIAS("snd-hda-codec-id:14f1506e");
 MODULE_ALIAS("snd-hda-codec-id:14f15097");
 MODULE_ALIAS("snd-hda-codec-id:14f15098");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 61a774b3d3c..b48fb43b544 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
 
 static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol,
-				 getput_call_t func)
+				 getput_call_t func, bool check_adc_switch)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	int err;
+	int i, err = 0;
 
 	mutex_lock(&codec->control_mutex);
-	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
-						      3, 0, HDA_INPUT);
-	err = func(kcontrol, ucontrol);
+	if (check_adc_switch && spec->dual_adc_switch) {
+		for (i = 0; i < spec->num_adc_nids; i++) {
+			kcontrol->private_value =
+				HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+						    3, 0, HDA_INPUT);
+			err = func(kcontrol, ucontrol);
+			if (err < 0)
+				goto error;
+		}
+	} else {
+		i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+		kcontrol->private_value =
+			HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+					    3, 0, HDA_INPUT);
+		err = func(kcontrol, ucontrol);
+	}
+ error:
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
@@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_get);
+				     snd_hda_mixer_amp_volume_get, false);
 }
 
 static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_put);
+				     snd_hda_mixer_amp_volume_put, true);
 }
 
 /* capture mixer elements */
@@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_get);
+				     snd_hda_mixer_amp_switch_get, false);
 }
 
 static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_put);
+				     snd_hda_mixer_amp_switch_put, true);
 }
 
 #define _DEFINE_CAPMIX(num) \
@@ -4883,7 +4896,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
 	SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
 	SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
-	SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
 	SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
 	SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -12600,6 +12612,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
  */
 enum {
 	PINFIX_FSC_H270,
+	PINFIX_HP_Z200,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -12612,9 +12625,17 @@ static const struct alc_fixup alc262_fixups[] = {
 			{ }
 		}
 	},
+	[PINFIX_HP_Z200] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x16, 0x99130120 }, /* internal speaker */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
 	SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
 	{}
 };
@@ -12731,6 +12752,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
 			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+			   ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13872,7 +13895,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c952582fb21..f43bb0eaed8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
 	struct via_spec *spec = codec->spec;
 	hda_nid_t nid = kcontrol->private_value;
 	unsigned int pinsel = ucontrol->value.enumerated.item[0];
+	unsigned int parm0, parm1;
 	/* Get Independent Mode index of headphone pin widget */
 	spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
 		? 1 : 0;
-	if (spec->codec_type == VT1718S)
+	if (spec->codec_type == VT1718S) {
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+		/* Set correct mute switch for MW3 */
+		parm0 = spec->hp_independent_mode ?
+			       AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+		parm1 = spec->hp_independent_mode ?
+			       AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+	}
 	else
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -4283,9 +4294,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
-	/* Setup default input of Front HP to MW9 */
-	{0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
 	/* PW9 PW10 Output enable */
 	{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
 	{0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4294,10 +4302,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* Enable Boost Volume backdoor */
 	{0x1, 0xf88, 0x8},
 	/* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4307,8 +4315,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
 	{0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
 	{0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Unmute MW4's index 0 */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ }
 };
 
@@ -4456,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
 			if (err < 0)
 				return err;
 		} else if (i == AUTO_SEQ_FRONT) {
+			/* add control to mixer index 0 */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Master Front Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Master Front Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
 			/* Front */
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 3f08afc0f0d..c8e402fc378 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -896,11 +896,11 @@ struct hdspm {
 	unsigned char max_channels_in;
 	unsigned char max_channels_out;
 
-	char *channel_map_in;
-	char *channel_map_out;
+	signed char *channel_map_in;
+	signed char *channel_map_out;
 
-	char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
-	char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
+	signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs;
+	signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs;
 
 	char **port_names_in;
 	char **port_names_out;
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c8..f1fd95bb641 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	pr_debug("%s enter\n", __func__);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		diff = sport_curr_offset_tx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	} else {
 		diff = sport_curr_offset_rx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	}
+
+	/*
+	 * TX at least can report one frame beyond the end of the
+	 * buffer if we hit the wraparound case - clamp to within the
+	 * buffer as the ALSA APIs require.
+	 */
+	if (diff == snd_pcm_lib_buffer_bytes(substream))
+		diff = 0;
+
+	frames = bytes_to_frames(substream->runtime, diff);
+
 	return frames;
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1..65f46047b1c 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	default:
 		return -EINVAL;
 	}
-	snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+	snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
 	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
 
 	/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d394..7859bdcc93d 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 		dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
 	}
 
-	/* Configure PLL */
+	/**
+	 * Configure PLL
+	 * fsref = (mclk * PLLM) / 2048
+	 * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+	 */
 	pval = 1;
-	jval = (fsref == 44100) ? 7 : 8;
-	dval = (fsref == 44100) ? 5264 : 1920;
+	/* compute J portion of multiplier */
+	jval = fsref / (aic26->mclk / 2048);
+	/* compute fractional DDDD component of multiplier */
+	dval = fsref - (jval * (aic26->mclk / 2048));
+	dval = (10000 * dval) / (aic26->mclk / 2048);
+	dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
 	qval = 0;
 	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
 	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c26..789453d44ec 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 
 		/* Sync reg_cache with the hardware */
 		codec->cache_only = 0;
-		for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+		for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
 			snd_soc_write(codec, i, cache[i]);
 		if (aic3x->model == AIC3X_MODEL_3007)
 			aic3x_init_3007(codec);
 		codec->cache_sync = 0;
 	} else {
+		/*
+		 * Do soft reset to this codec instance in order to clear
+		 * possible VDD leakage currents in case the supply regulators
+		 * remain on
+		 */
+		snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+		codec->cache_sync = 1;
 		aic3x->power = 0;
 		/* HW writes are needless when bias is off */
 		codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4..76b4361e9b8 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
 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],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
 static const struct snd_soc_dapm_route wm8731_intercon[] = {
 	{"DAC", NULL, "OSC", wm8731_check_osc},
 	{"ADC", NULL, "OSC", wm8731_check_osc},
+	{"DAC", NULL, "ACTIVE"},
+	{"ADC", NULL, "ACTIVE"},
 
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
-			      struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* set active */
-	snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
-	return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
-			    struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* deactivate */
-	if (!codec->active) {
-		udelay(50);
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
-	}
-}
-
 static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 		snd_soc_write(codec, WM8731_PWR, 0xffff);
 		regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
 				       wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops wm8731_dai_ops = {
-	.prepare	= wm8731_pcm_prepare,
 	.hw_params	= wm8731_hw_params,
-	.shutdown	= wm8731_shutdown,
 	.digital_mute	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
 	.set_fmt	= wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73c..6af23d06870 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360..c2fc0356c2a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
 				    WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
 				    reg);
+
+		msleep(5);
 	}
 
 	wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd..bb699bb55a5 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
 
 if SND_IMX_SOC
 
-config SND_MXC_SOC_SSI
-	tristate
-
 config SND_MXC_SOC_FIQ
 	tristate
 
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
 	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
 	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
 	select SND_SOC_WM8350
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
 	tristate "SoC audio support for Visstrim M10 boards"
 	depends on MACH_IMX27_VISSTRIM_M10
 	select SND_SOC_TVL320AIC32X4
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_MX2
 	help
 	  Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
 	select SND_SOC_WM9712
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
 		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index b2ed764fd89..43fdc24f7e8 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca753..61fceb09cdb 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
 MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
 MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d..fab20a54e86 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c005ceb70c9..039b9532b27 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
 	switch (control) {
-	case SND_SOC_CUSTOM:
-		break;
-
 	case SND_SOC_I2C:
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 		codec->hw_write = (hw_write_t)i2c_master_send;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc..b194be09e74 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		 "%s", card->name);
 	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->long_name ? card->long_name : card->name);
-	snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
-		 "%s", card->driver_name ? card->driver_name : card->name);
+	if (card->driver_name)
+		strlcpy(card->snd_card->driver, card->driver_name,
+			sizeof(card->snd_card->driver));
 
 	if (card->late_probe) {
 		ret = card->late_probe(card);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548..95f03c10b4f 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (i2sclock % (2 * srate))
 		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
 
+	if (!i2s->clk_refs)
+		clk_enable(i2s->clk_i2s);
+
 	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
 
 	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
 		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
 		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
+	if (!i2s->clk_refs)
+		clk_disable(i2s->clk_i2s);
+
 	return 0;
 }
 
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 337a00241a1..4dd051bdf4f 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void)
 }
 module_exit(at73c213_exit);
 
-MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt@atmel.com>");
+MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt@samfundet.no>");
 MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC");
 MODULE_LICENSE("GPL");