22 files changed, 189 insertions, 50 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 6e03b465e44e..a2104671f51d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
 		case SNDRV_PCM_STATE_DISCONNECTED:
 			err = -EBADFD;
 			goto _endloop;
+		case SNDRV_PCM_STATE_PAUSED:
+			continue;
 		}
 		if (!tout) {
 			snd_printd("%s write error (DMA or IRQ trouble?)\n",
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index c4671d00babd..c7f6d1cab606 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -474,6 +474,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx)
 	memset(path, 0, sizeof(*path));
 }
 
+/* return a DAC if paired to the given pin by codec driver */
+static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+	struct hda_gen_spec *spec = codec->spec;
+	const hda_nid_t *list = spec->preferred_dacs;
+
+	if (!list)
+		return 0;
+	for (; *list; list += 2)
+		if (*list == pin)
+			return list[1];
+	return 0;
+}
+
 /* look for an empty DAC slot */
 static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin,
 			      bool is_digital)
@@ -1192,7 +1206,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs,
 			continue;
 		}
 
-		dacs[i] = look_for_dac(codec, pin, false);
+		dacs[i] = get_preferred_dac(codec, pin);
+		if (dacs[i]) {
+			if (is_dac_already_used(codec, dacs[i]))
+				badness += bad->shared_primary;
+		}
+
+		if (!dacs[i])
+			dacs[i] = look_for_dac(codec, pin, false);
 		if (!dacs[i] && !i) {
 			/* try to steal the DAC of surrounds for the front */
 			for (j = 1; j < num_outs; j++) {
@@ -4297,6 +4318,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
 	return AC_PWRST_D3;
 }
 
+/* mute all aamix inputs initially; parse up to the first leaves */
+static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
+{
+	int i, nums;
+	const hda_nid_t *conn;
+	bool has_amp;
+
+	nums = snd_hda_get_conn_list(codec, mix, &conn);
+	has_amp = nid_has_mute(codec, mix, HDA_INPUT);
+	for (i = 0; i < nums; i++) {
+		if (has_amp)
+			snd_hda_codec_amp_stereo(codec, mix,
+						 HDA_INPUT, i,
+						 0xff, HDA_AMP_MUTE);
+		else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
+			snd_hda_codec_amp_stereo(codec, conn[i],
+						 HDA_OUTPUT, 0,
+						 0xff, HDA_AMP_MUTE);
+	}
+}
 
 /*
  * Parse the given BIOS configuration and set up the hda_gen_spec
@@ -4435,6 +4476,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 		}
 	}
 
+	/* mute all aamix input initially */
+	if (spec->mixer_nid)
+		mute_all_mixer_nid(codec, spec->mixer_nid);
+
  dig_only:
 	parse_digital(codec);
 
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 7e45cb44d151..0929a06df812 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -249,6 +249,9 @@ struct hda_gen_spec {
 	const struct badness_table *main_out_badness;
 	const struct badness_table *extra_out_badness;
 
+	/* preferred pin/DAC pairs; an array of paired NIDs */
+	const hda_nid_t *preferred_dacs;
+
 	/* loopback mixing mode */
 	bool aamix_mode;
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 27aa14007cbd..956871d8b3d2 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3433,6 +3433,10 @@ static void check_probe_mask(struct azx *chip, int dev)
  * white/black-list for enable_msi
  */
 static struct snd_pci_quirk msi_black_list[] = {
+	SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */
+	SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */
 	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
 	SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
 	SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index cac015be3325..699262a3e07a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -340,6 +340,14 @@ static int patch_ad1986a(struct hda_codec *codec)
 {
 	int err;
 	struct ad198x_spec *spec;
+	static hda_nid_t preferred_pairs[] = {
+		0x1a, 0x03,
+		0x1b, 0x03,
+		0x1c, 0x04,
+		0x1d, 0x05,
+		0x1e, 0x03,
+		0
+	};
 
 	err = alloc_ad_spec(codec);
 	if (err < 0)
@@ -360,6 +368,8 @@ static int patch_ad1986a(struct hda_codec *codec)
 	 * So, let's disable the shared stream.
 	 */
 	spec->gen.multiout.no_share_stream = 1;
+	/* give fixed DAC/pin pairs */
+	spec->gen.preferred_dacs = preferred_pairs;
 
 	/* AD1986A can't manage the dynamic pin on/off smoothly */
 	spec->gen.auto_mute_via_amp = 1;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1f2717f817a0..3fbf2883e06e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
-	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index c4a66ef6cf6f..f281c8068557 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2337,8 +2337,9 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 	int err;
 
 	per_cvt = get_cvt(spec, 0);
-	err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid,
-					    per_cvt->cvt_nid);
+	err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid,
+					  per_cvt->cvt_nid,
+					  HDA_PCM_TYPE_HDMI);
 	if (err < 0)
 		return err;
 	return simple_hdmi_build_jack(codec, 0);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c5ea483d7559..c5646941539a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3849,6 +3849,7 @@ enum {
 	ALC269_FIXUP_ASUS_X101,
 	ALC271_FIXUP_AMIC_MIC2,
 	ALC271_FIXUP_HP_GATE_MIC_JACK,
+	ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572,
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
 	ALC269VB_FIXUP_ASUS_ZENBOOK,
@@ -4111,6 +4112,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC271_FIXUP_AMIC_MIC2,
 	},
+	[ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_limit_int_mic_boost,
+		.chained = true,
+		.chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK,
+	},
 	[ALC269_FIXUP_ACER_AC700] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -4208,6 +4215,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -4239,12 +4247,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -5034,8 +5046,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697cedccd21..1ead3c977a51 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 
 	dma_params = ssc_p->dma_params[dir];
 
-	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
 	ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error);
 
 	pr_debug("%s enabled SSC_SR=0x%08x\n",
@@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int atmel_ssc_trigger(struct snd_pcm_substream *substream,
+			     int cmd, struct snd_soc_dai *dai)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+		break;
+	default:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+		break;
+	}
+
+	return 0;
+}
 
 #ifdef CONFIG_PM
 static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
@@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
 	.startup	= atmel_ssc_startup,
 	.shutdown	= atmel_ssc_shutdown,
 	.prepare	= atmel_ssc_prepare,
+	.trigger	= atmel_ssc_trigger,
 	.hw_params	= atmel_ssc_hw_params,
 	.set_fmt	= atmel_ssc_set_dai_fmt,
 	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 1b372283bd01..7d6a9055874b 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -109,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	dai->stream_name = "WM8731 PCM";
 	dai->codec_dai_name = "wm8731-hifi";
 	dai->init = sam9x5_wm8731_init;
-	dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+	dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
 		| SND_SOC_DAIFMT_CBM_CFM;
 
 	ret = snd_soc_of_parse_card_name(card, "atmel,model");
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 99b359e19d35..0ab2dc296474 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1012,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AEC Loopback", "HPOUT3L", "OUT3L" },
 	{ "AEC Loopback", "HPOUT3R", "OUT3R" },
 	{ "HPOUT3L", NULL, "OUT3L" },
-	{ "HPOUT3R", NULL, "OUT3L" },
+	{ "HPOUT3R", NULL, "OUT3R" },
 
 	{ "AEC Loopback", "SPKOUTL", "OUT4L" },
 	{ "SPKOUTLN", NULL, "OUT4L" },
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1c203e..53bbfac6a83a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
-		aif1 |= WM8904_AIF_LRCLK_INV;
+		aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
 	case SND_SOC_DAIFMT_DSP_A:
 		aif1 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..0f17ed3e29f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
 			    WM8962_SYSCLK_RATE_MASK, clocking4);
 
+	/* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
+	 * So we here provisionally enable it and then disable it afterward
+	 * if current bias_level hasn't reached SND_SOC_BIAS_ON.
+	 */
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
+
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
+
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, 0);
+
 	if (dspclk < 0) {
 		dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
 		return;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..4fbcab63e61f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 		return ret;
 
 	/* Wait for the RAM to start, should be near instantaneous */
-	count = 0;
-	do {
+	for (count = 0; count < 10; ++count) {
 		ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
 				  &val);
 		if (ret != 0)
 			return ret;
-	} while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+		if (val & ADSP2_RAM_RDY)
+			break;
+
+		msleep(1);
+	}
 
 	if (!(val & ADSP2_RAM_RDY)) {
 		adsp_err(dsp, "Failed to start DSP RAM\n");
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e48852b9e8..3fd76bc391de 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 		break;
 	}
 
-	dapm->bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 0b18f654b413..3920a5e8125f 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -473,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
@@ -494,17 +494,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96ce1f4..41949af3baae 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	}
 }
 
+static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
+{
+	unsigned int i;
+
+	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+	     i++) {
+		if (!pcm->chan[i])
+			continue;
+		dma_release_channel(pcm->chan[i]);
+		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+			break;
+	}
+}
+
 /**
  * snd_dmaengine_pcm_register - Register a dmaengine based PCM device
  * @dev: The parent device for the PCM device
@@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	const struct snd_dmaengine_pcm_config *config, unsigned int flags)
 {
 	struct dmaengine_pcm *pcm;
+	int ret;
 
 	pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
 	if (!pcm)
@@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	dmaengine_pcm_request_chan_of(pcm, dev);
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_no_residue_pcm_platform);
 	else
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_pcm_platform);
+	if (ret)
+		goto err_free_dma;
+
+	return 0;
+
+err_free_dma:
+	dmaengine_pcm_release_chan(pcm);
+	kfree(pcm);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
 
@@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 {
 	struct snd_soc_platform *platform;
 	struct dmaengine_pcm *pcm;
-	unsigned int i;
 
 	platform = snd_soc_lookup_platform(dev);
 	if (!platform)
@@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 
 	pcm = soc_platform_to_pcm(platform);
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
-		if (pcm->chan[i]) {
-			dma_release_channel(pcm->chan[i]);
-			if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
-				break;
-		}
-	}
-
 	snd_soc_remove_platform(platform);
+	dmaengine_pcm_release_chan(pcm);
 	kfree(pcm);
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 11a90cd027fa..891b9a9bcbf8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -600,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
 	/* apply codec digital mute */
-	if (!codec->active)
+	if ((playback && codec_dai->playback_active == 1) ||
+	    (!playback && codec_dai->capture_active == 1))
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
 
 	/* free any machine hw params */
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a907e1..8c819f811470 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 08bc6931c7c7..8c7c1028e579 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 	int ret, spdifclock;
 
-	mask = TEGRA20_SPDIF_CTRL_PACK |
-	       TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	mask |= TEGRA20_SPDIF_CTRL_PACK |
+		TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		val = TEGRA20_SPDIF_CTRL_PACK |
-		      TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		val |= TEGRA20_SPDIF_CTRL_PACK |
+		       TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 231a785b3921..02247fee1cf7 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 3454262358b3..f4b12c216f1c 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1603,7 +1603,7 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer)
 			return err;
 	}
 
-	return err;
+	return 0;
 }
 
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)