Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (32 commits)
  ALSA: hda: Conexant: Allow different output types to share DAC
  ASoC: Correct element count for WM8996 sidetone HPF
  ASoC: Tegra: wm8903 machine driver: Drop Ventana support
  ASoC: Add samsung maintainer
  ASoC: Add Springbank I/O card to Speyside Kconfig
  ALSA: hda/conexant - Enable ADC-switching for auto-mic mode, too
  ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser
  ALSA: hda - Update jack-sense info even when no automute is set
  ALSA: hda - Fix output-path initialization for Realtek auto-parser
  sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put
  sound/soc/fsl/p1022_ds.c: add missing of_node_put
  sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree
  sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
  ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
  sound/soc/fsl/fsl_dma.c: add missing of_node_put
  ASoC: Clear completions from late WM8996 FLL lock IRQs
  ASoC: Clear any outstanding WM8962 FLL lock completions before waiting
  ASoC: Ensure we only run Speyside WM8962 bias level callbacks once
  ASoC: Fix configuration of WM8996 input enables
  ASoC: WM8996 record paths need AIFCLK
  ...

24 files changed, 148 insertions, 77 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index d94292065359..1a8cc600067d 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5532,6 +5532,7 @@ F:	include/media/*7146*
 
 SAMSUNG AUDIO (ASoC) DRIVERS
 M:	Jassi Brar <jassisinghbrar@gmail.com>
+M:	Sangbeom Kim <sbkim73@samsung.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Supported
 F:	sound/soc/samsung
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc9499453..7696d05b9356 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #define MAX_AUTO_DACS	5
 
+#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
+
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int type)
+			      struct pin_dac_pair *filled, int nums, 
+			      int type)
 {
-	int i, nums;
+	int i, start = nums;
 
-	nums = 0;
-	for (i = 0; i < num_pins; i++) {
+	for (i = 0; i < num_pins; i++, nums++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		nums++;
+		if (filled[nums].dac) 
+			continue;
+		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
 	}
 	return nums;
 }
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-				  dacs, &rest, spec->dac_info,
-				  AUTO_PIN_LINE_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_HP_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_SPEAKER_OUT);
+			  dacs, &rest, spec->dac_info, 0,
+			  AUTO_PIN_LINE_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_HP_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac)
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic) {
+	if (imux->num_items > 1) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
+		else if (nid & DAC_SLAVE_FLAG)
+			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-	if (caps & AC_AMPCAP_NUM_STEPS)
-		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	if (dac && !(dac & DAC_SLAVE_FLAG)) {
+		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+		if (caps & AC_AMPCAP_NUM_STEPS)
+			return cx_auto_add_pb_volume(codec, dac, name, idx);
+	}
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		if (!spec->dac_info[i].dac)
-			continue;
+		hda_nid_t dac = spec->dac_info[i].dac;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+		err = try_add_pb_volume(codec, dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fcb11af9ad24..7cabd7317163 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute)
-		return;
 	spec->jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
+	if (!spec->automute)
+		return;
 	update_speakers(codec);
 }
 
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute || !spec->detect_line)
-		return;
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
+	if (!spec->automute || !spec->detect_line)
+		return;
 	update_speakers(codec);
 }
 
@@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
+	hda_nid_t pin, dac;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.hp_nid);
+	if (pin) {
+		dac = spec->multiout.hp_nid;
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+	}
 	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
-					spec->multiout.extra_out_nid[0]);
+	if (pin) {
+		dac = spec->multiout.extra_out_nid[0];
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+	}
 }
 
 /*
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c033cb7..a118a0fb9d81 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
 
 	switch (params_rate(params)) {
 	case 48000:
-		clk = 12288000;
+		clk = 24576000;
 		break;
 	}
 
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5ffe68..eedb6f5e5823 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
 	int sysclk;
 };
 
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
-	0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
 /*
  * AD193X volume/mute/de-emphasis etc. controls
  */
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
 
 	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
-	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+		| (word_len << AD193X_DAC_WORD_LEN_SHFT);
 	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
 
 	reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
 	.probe = 	ad193x_probe,
-	.reg_cache_default = ad193x_reg,
-	.reg_cache_size = AD193X_NUM_REGS,
-	.reg_word_size = sizeof(u16),
 };
 
 #if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b5497877..cccc2e8e5fbd 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
 #define AD193X_DAC_LEFT_HIGH    (1 << 3)
 #define AD193X_DAC_BCLK_INV     (1 << 7)
 #define AD193X_DAC_CTRL2        0x804
-#define AD193X_DAC_WORD_LEN_MASK	0xC
+#define AD193X_DAC_WORD_LEN_SHFT        3
+#define AD193X_DAC_WORD_LEN_MASK        0x18
 #define AD193X_DAC_MASTER_MUTE  1
 #define AD193X_DAC_CHNL_MUTE    0x805
 #define AD193X_DACL1_MUTE       0
@@ -63,7 +64,7 @@
 #define AD193X_ADC_CTRL1        0x80f
 #define AD193X_ADC_SERFMT_MASK		0x60
 #define AD193X_ADC_SERFMT_STEREO	(0 << 5)
-#define AD193X_ADC_SERFMT_TDM		(1 << 2)
+#define AD193X_ADC_SERFMT_TDM		(1 << 5)
 #define AD193X_ADC_SERFMT_AUX		(2 << 5)
 #define AD193X_ADC_WORD_LEN_MASK	0x3
 #define AD193X_ADC_CTRL2        0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d1f34c..fbd7eb9e61ce 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+		kfree(sta32x);
 		return ret;
 	}
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740ebeb5b..1725550c293e 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
 		if (fll) {
+			try_wait_for_completion(&wm8962->fll_lock);
+
 			snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 					    WM8962_FLL_ENA, WM8962_FLL_ENA);
 			if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 					    WM8962_BIAS_ENA | 0x180);
 
 			msleep(5);
-
-			snd_soc_update_bits(codec, WM8962_CLOCKING2,
-					    WM8962_CLKREG_OVD,
-					    WM8962_CLKREG_OVD);
 		}
 
 		/* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
 	snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
 
+	try_wait_for_completion(&wm8962->fll_lock);
+
 	snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 			    WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
 			    WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	 */
 	snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
 
+	/* Ensure we have soft control over all registers */
+	snd_soc_update_bits(codec, WM8962_CLOCKING2,
+			    WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
 	regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 
 	if (pdata) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1aaff0..0cdb9d105671 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
 };
 
 static const struct soc_enum sidetone_hpf =
-	SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+	SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
 
 static const char *hpf_mode_text[] = {
 	"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
 SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
 
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
 
 SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "AIF2RX0", NULL, "AIFCLK" },
 	{ "AIF2RX1", NULL, "AIFCLK" },
 
+	{ "AIF1TX0", NULL, "AIFCLK" },
+	{ "AIF1TX1", NULL, "AIFCLK" },
+	{ "AIF1TX2", NULL, "AIFCLK" },
+	{ "AIF1TX3", NULL, "AIFCLK" },
+	{ "AIF1TX4", NULL, "AIFCLK" },
+	{ "AIF1TX5", NULL, "AIFCLK" },
+
+	{ "AIF2TX0", NULL, "AIFCLK" },
+	{ "AIF2TX1", NULL, "AIFCLK" },
+
 	{ "DSP1RXL", NULL, "SYSDSPCLK" },
 	{ "DSP1RXR", NULL, "SYSDSPCLK" },
 	{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
 
+	/* Clear any pending completions (eg, from failed startups) */
+	try_wait_for_completion(&wm8996->fll_lock);
+
 	snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
 			    WM8996_FLL_ENA, WM8996_FLL_ENA);
 
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c1c9a9..099614e16651 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!res) {
 		err = -ENODEV;
-		goto fail;
+		goto fail_free_info;
 	}
 
 	info->mem = request_mem_region(res->start, resource_size(res),
 				       pdev->name);
 	if (!info->mem) {
 		err = -EBUSY;
-		goto fail;
+		goto fail_free_info;
 	}
 
 	info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
 	iounmap(info->regs);
 fail_release_mem:
 	release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
 	kfree(info);
 fail:
 	return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c8c0b4..cb50598338e9 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
 		 * assume that device_node pointers are a valid comparison.
 		 */
 		np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+		of_node_put(np);
 		if (np == dma_channel_np)
 			return ssi_np;
 
 		np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+		of_node_put(np);
 		if (np == dma_channel_np)
 			return ssi_np;
 	}
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a19297959587..358f0baaf71b 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	}
 
 	machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
-	if (!machine_data)
-		return -ENOMEM;
+	if (!machine_data) {
+		ret = -ENOMEM;
+		goto error_alloc;
+	}
 
 	machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	ret = platform_device_add(sound_device);
 	if (ret) {
 		dev_err(&pdev->dev, "platform device add failed\n");
-		goto error;
+		goto error_sound;
 	}
 	dev_set_drvdata(&pdev->dev, sound_device);
 
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 
 	return 0;
 
+error_sound:
+	platform_device_unregister(sound_device);
 error:
-	of_node_put(codec_np);
-
-	if (sound_device)
-		platform_device_unregister(sound_device);
-
 	kfree(machine_data);
-
+error_alloc:
+	of_node_put(codec_np);
 	return ret;
 }
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f8eda1..fcb862eb0c73 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
 	 * dai->platform name should already point to an allocated buffer.
 	 */
 	ret = of_address_to_resource(dma_channel_np, 0, &res);
-	if (ret)
+	if (ret) {
+		of_node_put(dma_channel_np);
 		return ret;
+	}
 	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
 		 (unsigned long long) res.start, dma_channel_np->name);
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51f363b..8f16cd37c2af 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	if (!priv->mem) {
 		dev_err(&pdev->dev, "request_mem_region failed\n");
 		err = -EBUSY;
-		goto error;
+		goto error_alloc;
 	}
 
 	priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0efe1c..0aa475f92efa 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	/* Set codec bias level */
-	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+	ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
 
 	/* Add hook switch - can be used to control the codec from userspace
 	 * even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
 			ams_delta_hook_switch_gpios);
 
 	/* Keep modem power on */
-	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+	ams_delta_set_bias_level(&ams_delta_audio_card,
+				 &ams_delta_audio_card.rtd[0].codec->dapm,
+				 SND_SOC_BIAS_STANDBY);
 
 	platform_device_unregister(cx20442_platform_device);
 	platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091fc34eb..65f980ef2870 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8996
 	select SND_SOC_WM9081
+	select SND_SOC_WM1250_EV1
 
 config SND_SOC_SPEYSIDE_WM8962
 	tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d00660..c6c65892294e 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5d440d..bc8c1676459f 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
  *
  */
 
+#include <linux/types.h>
 #include <linux/gpio.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 0b9eb5f7ec4c..72535f2daaf2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
 		ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca27490..b085d8e87574 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 
 	if (prefix) {
 		name_len = strlen(long_name) + strlen(prefix) + 2;
-		name = kmalloc(name_len, GFP_ATOMIC);
+		name = kmalloc(name_len, GFP_KERNEL);
 		if (!name)
 			return NULL;
 
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c80589..a62f7dd4ba96 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
 #define snd_soc_16_8_read_i2c NULL
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+		                          unsigned int r)
+{
+	struct spi_device *spi = codec->control_data;
+
+	const u16 reg = cpu_to_be16(r | 0x100);
+	u8 data;
+	int ret;
+
+	ret = spi_write_then_read(spi, &reg, 2, &data, 1);
+	if (ret < 0)
+		return 0;
+	return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
 static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
@@ -295,6 +314,7 @@ static struct {
 	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+	unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
 } io_types[] = {
 	{
 		.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
 		.addr_bits = 16, .data_bits = 8,
 		.write = snd_soc_16_8_write,
 		.i2c_read = snd_soc_16_8_read_i2c,
+		.spi_read = snd_soc_16_8_read_spi,
 	},
 	{
 		.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 #ifdef CONFIG_SPI_MASTER
 		codec->hw_write = do_spi_write;
 #endif
+		if (io_types[i].spi_read)
+			codec->hw_read = io_types[i].spi_read;
 
 		codec->control_data = container_of(codec->dev,
 						   struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98d5846..38b00131b2fe 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 					      IRQF_TRIGGER_FALLING,
 					      gpios[i].name,
 					      &gpios[i]);
-		if (ret)
+		if (ret < 0)
 			goto err;
 
 		if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b5759397afa3..2879c883eebc 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	codec_dai->active--;
 	codec->active--;
 
+	if (!cpu_dai->active && !codec_dai->active)
+		rtd->rate = 0;
+
 	/* Muting the DAC suppresses artifacts caused during digital
 	 * shutdown, for example from stopping clocks.
 	 */
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 661373c2352a..be27f1d229af 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
 
 	/* FIXME: Calculate automatically based on DAPM routes? */
-	if (!machine_is_harmony() && !machine_is_ventana())
+	if (!machine_is_harmony())
 		snd_soc_dapm_nc_pin(dapm, "IN1L");
 	if (!machine_is_seaboard() && !machine_is_aebl())
 		snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (machine_is_harmony() || machine_is_ventana()) {
+	if (machine_is_harmony()) {
 		card->dapm_routes = harmony_audio_map;
 		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
 	} else if (machine_is_seaboard()) {