22 files changed, 151 insertions, 92 deletions

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a41e481e99a3..328b4a43f09a 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -325,7 +325,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			pcm_err(substream->pcm,
-				"BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
+				"invalid position: %s, pos = %ld, buffer size = %ld, period size = %ld\n",
 				name, pos, runtime->buffer_size,
 				runtime->period_size);
 		}
diff --git a/sound/core/timer.c b/sound/core/timer.c
index f24c9fccf008..b982d1b089bd 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1051,8 +1051,8 @@ static int snd_timer_s_start(struct snd_timer * timer)
 		njiff += timer->sticks - priv->correction;
 		priv->correction = 0;
 	}
-	priv->last_expires = priv->tlist.expires = njiff;
-	add_timer(&priv->tlist);
+	priv->last_expires = njiff;
+	mod_timer(&priv->tlist, njiff);
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c1c855a6c0af..a47e8ae0eb30 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec)
 	snd_hda_gen_update_outputs(codec);
 
 	if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) {
-		spec->gpio_data = spec->gen.hp_jack_present ?
-			spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+		if (spec->gen.automute_speaker)
+			spec->gpio_data = spec->gen.hp_jack_present ?
+				spec->gpio_eapd_hp : spec->gpio_eapd_speaker;
+		else
+			spec->gpio_data =
+				spec->gpio_eapd_hp | spec->gpio_eapd_speaker;
 		snd_hda_codec_write(codec, 0x01, 0,
 				    AC_VERB_SET_GPIO_DATA, spec->gpio_data);
 	}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ef198903c0c3..600af5878e75 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -204,8 +204,13 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	if (codec->core.vendor_id != 0x14f150f2)
+	switch (codec->core.vendor_id) {
+	case 0x14f150f2: /* CX20722 */
+	case 0x14f150f4: /* CX20724 */
+		break;
+	default:
 		return;
+	}
 
 	/* Turn the CX20722 codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 70c945603379..f7bcd8dbac14 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2353,6 +2353,10 @@ static void intel_pin_eld_notify(void *audio_ptr, int port)
 	struct hda_codec *codec = audio_ptr;
 	int pin_nid = port + 0x04;
 
+	/* we assume only from port-B to port-D */
+	if (port < 1 || port > 3)
+		return;
+
 	/* skip notification during system suspend (but not in runtime PM);
 	 * the state will be updated at resume
 	 */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c2430b36e1ce..1402ba954b3d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 
 static void alc_headset_mode_default(struct hda_codec *codec)
 {
+	static struct coef_fw coef0225[] = {
+		UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10),
+		{}
+	};
 	static struct coef_fw coef0255[] = {
 		WRITE_COEF(0x45, 0xc089),
 		WRITE_COEF(0x45, 0xc489),
@@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 	};
 
 	switch (codec->core.vendor_id) {
+	case 0x10ec0225:
+		alc_process_coef_fw(codec, coef0225);
+		break;
 	case 0x10ec0255:
 	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0255);
@@ -4750,6 +4757,10 @@ enum {
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
+	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
+	ALC280_FIXUP_HP_HEADSET_MIC,
+	ALC221_FIXUP_HP_FRONT_MIC,
+	ALC292_FIXUP_TPT460,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5375,6 +5386,36 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Disable pass-through path for FRONT 14h */
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x36 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 },
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
+	[ALC280_FIXUP_HP_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_aamix,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC,
+	},
+	[ALC221_FIXUP_HP_FRONT_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x02a19020 }, /* Front Mic */
+			{ }
+		},
+	},
+	[ALC292_FIXUP_TPT460] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_tpt440_dock,
+		.chained = true,
+		.chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5479,6 +5520,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5527,8 +5570,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE),
+	SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460),
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
+	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
@@ -5621,6 +5665,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"},
 	{.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
 	{.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
+	{.id = ALC292_FIXUP_TPT460, .name = "tpt460"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
@@ -5647,10 +5692,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{0x21, 0x03211020}
 
 static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
-	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x14, 0x901701a0}),
-	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
+	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x14, 0x901701b0}),
 	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
@@ -6378,6 +6423,7 @@ enum {
 	ALC668_FIXUP_AUTO_MUTE,
 	ALC668_FIXUP_DELL_DISABLE_AAMIX,
 	ALC668_FIXUP_DELL_XPS13,
+	ALC662_FIXUP_ASUS_Nx50,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -6618,6 +6664,12 @@ static const struct hda_fixup alc662_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_bass_chmap,
 	},
+	[ALC662_FIXUP_ASUS_Nx50] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_auto_mute_via_amp,
+		.chained = true,
+		.chain_id = ALC662_FIXUP_BASS_1A
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -6640,8 +6692,9 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x069f, "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),
+	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
 	SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
+	SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
 	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16),
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 42bcbac801a3..ccdab29a8b66 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2879,6 +2879,7 @@ static void intel8x0_measure_ac97_clock(struct intel8x0 *chip)
 
 static struct snd_pci_quirk intel8x0_clock_list[] = {
 	SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
+	SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000),
 	SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
 	SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
 	SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index e4145509d63c..9c5219392460 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -324,7 +324,7 @@ static const struct snd_soc_component_driver s3c_ac97_component = {
 
 static int s3c_ac97_probe(struct platform_device *pdev)
 {
-	struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res;
+	struct resource *mem_res, *irq_res;
 	struct s3c_audio_pdata *ac97_pdata;
 	int ret;
 
@@ -335,24 +335,6 @@ static int s3c_ac97_probe(struct platform_device *pdev)
 	}
 
 	/* Check for availability of necessary resource */
-	dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dmatx_res) {
-		dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n");
-		return -ENXIO;
-	}
-
-	dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!dmarx_res) {
-		dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n");
-		return -ENXIO;
-	}
-
-	dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2);
-	if (!dmamic_res) {
-		dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n");
-		return -ENXIO;
-	}
-
 	irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
 	if (!irq_res) {
 		dev_err(&pdev->dev, "AC97 IRQ not provided!\n");
@@ -364,11 +346,11 @@ static int s3c_ac97_probe(struct platform_device *pdev)
 	if (IS_ERR(s3c_ac97.regs))
 		return PTR_ERR(s3c_ac97.regs);
 
-	s3c_ac97_pcm_out.channel = dmatx_res->start;
+	s3c_ac97_pcm_out.slave = ac97_pdata->dma_playback;
 	s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA;
-	s3c_ac97_pcm_in.channel = dmarx_res->start;
+	s3c_ac97_pcm_in.slave = ac97_pdata->dma_capture;
 	s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA;
-	s3c_ac97_mic_in.channel = dmamic_res->start;
+	s3c_ac97_mic_in.slave = ac97_pdata->dma_capture_mic;
 	s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA;
 
 	init_completion(&s3c_ac97.done);
diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h
index 0e85dcfec023..085ef30f5ca2 100644
--- a/sound/soc/samsung/dma.h
+++ b/sound/soc/samsung/dma.h
@@ -15,7 +15,7 @@
 #include <sound/dmaengine_pcm.h>
 
 struct s3c_dma_params {
-	int channel;				/* Channel ID */
+	void *slave;				/* Channel ID */
 	dma_addr_t dma_addr;
 	int dma_size;			/* Size of the DMA transfer */
 	char *ch_name;
diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c
index 506f5bf6d082..727008d57d14 100644
--- a/sound/soc/samsung/dmaengine.c
+++ b/sound/soc/samsung/dmaengine.c
@@ -50,14 +50,14 @@ void samsung_asoc_init_dma_data(struct snd_soc_dai *dai,
 
 	if (playback) {
 		playback_data = &playback->dma_data;
-		playback_data->filter_data = (void *)playback->channel;
+		playback_data->filter_data = playback->slave;
 		playback_data->chan_name = playback->ch_name;
 		playback_data->addr = playback->dma_addr;
 		playback_data->addr_width = playback->dma_size;
 	}
 	if (capture) {
 		capture_data = &capture->dma_data;
-		capture_data->filter_data = (void *)capture->channel;
+		capture_data->filter_data = capture->slave;
 		capture_data->chan_name = capture->ch_name;
 		capture_data->addr = capture->dma_addr;
 		capture_data->addr_width = capture->dma_size;
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 7dbf899b2af2..e163b0148c4b 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -1260,27 +1260,14 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 	pri_dai->lock = &pri_dai->spinlock;
 
 	if (!np) {
-		res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-		if (!res) {
-			dev_err(&pdev->dev,
-				"Unable to get I2S-TX dma resource\n");
-			return -ENXIO;
-		}
-		pri_dai->dma_playback.channel = res->start;
-
-		res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-		if (!res) {
-			dev_err(&pdev->dev,
-				"Unable to get I2S-RX dma resource\n");
-			return -ENXIO;
-		}
-		pri_dai->dma_capture.channel = res->start;
-
 		if (i2s_pdata == NULL) {
 			dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n");
 			return -EINVAL;
 		}
 
+		pri_dai->dma_playback.slave = i2s_pdata->dma_playback;
+		pri_dai->dma_capture.slave = i2s_pdata->dma_capture;
+
 		if (&i2s_pdata->type)
 			i2s_cfg = &i2s_pdata->type.i2s;
 
@@ -1341,11 +1328,8 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 		sec_dai->dma_playback.dma_addr = regs_base + I2STXDS;
 		sec_dai->dma_playback.ch_name = "tx-sec";
 
-		if (!np) {
-			res = platform_get_resource(pdev, IORESOURCE_DMA, 2);
-			if (res)
-				sec_dai->dma_playback.channel = res->start;
-		}
+		if (!np)
+			sec_dai->dma_playback.slave = i2s_pdata->dma_play_sec;
 
 		sec_dai->dma_playback.dma_size = 4;
 		sec_dai->addr = pri_dai->addr;
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b320a9d3fbf8..c77f324e0bb8 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -486,7 +486,7 @@ static const struct snd_soc_component_driver s3c_pcm_component = {
 static int s3c_pcm_dev_probe(struct platform_device *pdev)
 {
 	struct s3c_pcm_info *pcm;
-	struct resource *mem_res, *dmatx_res, *dmarx_res;
+	struct resource *mem_res;
 	struct s3c_audio_pdata *pcm_pdata;
 	int ret;
 
@@ -499,18 +499,6 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
 	pcm_pdata = pdev->dev.platform_data;
 
 	/* Check for availability of necessary resource */
-	dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dmatx_res) {
-		dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n");
-		return -ENXIO;
-	}
-
-	dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!dmarx_res) {
-		dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n");
-		return -ENXIO;
-	}
-
 	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!mem_res) {
 		dev_err(&pdev->dev, "Unable to get register resource\n");
@@ -568,8 +556,10 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev)
 	s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start
 							+ S3C_PCM_TXFIFO;
 
-	s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start;
-	s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start;
+	if (pcm_pdata) {
+		s3c_pcm_stereo_in[pdev->id].slave = pcm_pdata->dma_capture;
+		s3c_pcm_stereo_out[pdev->id].slave = pcm_pdata->dma_playback;
+	}
 
 	pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id];
 	pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id];
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 2b766d212ce0..77d27c85a32a 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -34,13 +34,13 @@
 #include "s3c2412-i2s.h"
 
 static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = {
-	.channel	= DMACH_I2S_OUT,
+	.slave		= (void *)(uintptr_t)DMACH_I2S_OUT,
 	.ch_name	= "tx",
 	.dma_size	= 4,
 };
 
 static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = {
-	.channel	= DMACH_I2S_IN,
+	.slave		= (void *)(uintptr_t)DMACH_I2S_IN,
 	.ch_name	= "rx",
 	.dma_size	= 4,
 };
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 5bf723689692..9da3a77ea2c7 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -32,13 +32,13 @@
 #include "s3c24xx-i2s.h"
 
 static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = {
-	.channel	= DMACH_I2S_OUT,
+	.slave		= (void *)(uintptr_t)DMACH_I2S_OUT,
 	.ch_name	= "tx",
 	.dma_size	= 2,
 };
 
 static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = {
-	.channel	= DMACH_I2S_IN,
+	.slave		= (void *)(uintptr_t)DMACH_I2S_IN,
 	.ch_name	= "rx",
 	.dma_size	= 2,
 };
diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c
index 36dbc0e96004..9dd7ee6d03ff 100644
--- a/sound/soc/samsung/spdif.c
+++ b/sound/soc/samsung/spdif.c
@@ -359,7 +359,7 @@ static const struct snd_soc_component_driver samsung_spdif_component = {
 static int spdif_probe(struct platform_device *pdev)
 {
 	struct s3c_audio_pdata *spdif_pdata;
-	struct resource *mem_res, *dma_res;
+	struct resource *mem_res;
 	struct samsung_spdif_info *spdif;
 	int ret;
 
@@ -367,12 +367,6 @@ static int spdif_probe(struct platform_device *pdev)
 
 	dev_dbg(&pdev->dev, "Entered %s\n", __func__);
 
-	dma_res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dma_res) {
-		dev_err(&pdev->dev, "Unable to get dma resource.\n");
-		return -ENXIO;
-	}
-
 	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!mem_res) {
 		dev_err(&pdev->dev, "Unable to get register resource.\n");
@@ -432,7 +426,7 @@ static int spdif_probe(struct platform_device *pdev)
 
 	spdif_stereo_out.dma_size = 2;
 	spdif_stereo_out.dma_addr = mem_res->start + DATA_OUTBUF;
-	spdif_stereo_out.channel = dma_res->start;
+	spdif_stereo_out.slave = spdif_pdata ? spdif_pdata->dma_playback : NULL;
 
 	spdif->dma_playback = &spdif_stereo_out;
 
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 2ed260b10f6d..7ccbcaf6a147 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -285,6 +285,8 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
 	unsigned char data[3];
 	int err, crate;
 
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	/* if endpoint doesn't have sampling rate control, bail out */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 7b1cb365ffab..c07a7eda42a2 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -438,6 +438,9 @@ exit_clear:
  *
  * New endpoints will be added to chip->ep_list and must be freed by
  * calling snd_usb_endpoint_free().
+ *
+ * For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that
+ * bNumEndpoints > 1 beforehand.
  */
 struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 					      struct usb_host_interface *alts,
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index ddca6547399b..1f8fb0d904e0 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -349,6 +349,16 @@ static struct usbmix_name_map bose_companion5_map[] = {
 };
 
 /*
+ * Dell usb dock with ALC4020 codec had a firmware problem where it got
+ * screwed up when zero volume is passed; just skip it as a workaround
+ */
+static const struct usbmix_name_map dell_alc4020_map[] = {
+	{ 16, NULL },
+	{ 19, NULL },
+	{ 0 }
+};
+
+/*
  * Control map entries
  */
 
@@ -431,6 +441,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = aureon_51_2_map,
 	},
 	{
+		.id = USB_ID(0x0bda, 0x4014),
+		.map = dell_alc4020_map,
+	},
+	{
 		.id = USB_ID(0x0dba, 0x1000),
 		.map = mbox1_map,
 	},
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 279025650568..f6c3bf79af9a 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1519,7 +1519,11 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol,
 
 	/* use known values for that card: interface#1 altsetting#1 */
 	iface = usb_ifnum_to_if(chip->dev, 1);
+	if (!iface || iface->num_altsetting < 2)
+		return -EINVAL;
 	alts = &iface->altsetting[1];
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	err = snd_usb_ctl_msg(chip->dev,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 9a9b789a56ef..3033bf7ce309 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -159,6 +159,8 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 	unsigned char data[1];
 	int err;
 
+	if (get_iface_desc(alts)->bNumEndpoints < 1)
+		return -EINVAL;
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
 	data[0] = 1;
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index c458d60d5030..001fb4dc0722 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 		usb_audio_err(chip, "cannot memdup\n");
 		return -ENOMEM;
 	}
+	INIT_LIST_HEAD(&fp->list);
 	if (fp->nr_rates > MAX_NR_RATES) {
 		kfree(fp);
 		return -EINVAL;
@@ -167,19 +168,20 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	stream = (fp->endpoint & USB_DIR_IN)
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 	err = snd_usb_add_audio_stream(chip, stream, fp);
-	if (err < 0) {
-		kfree(fp);
-		kfree(rate_table);
-		return err;
-	}
+	if (err < 0)
+		goto error;
 	if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
 	    fp->altset_idx >= iface->num_altsetting) {
-		kfree(fp);
-		kfree(rate_table);
-		return -EINVAL;
+		err = -EINVAL;
+		goto error;
 	}
 	alts = &iface->altsetting[fp->altset_idx];
 	altsd = get_iface_desc(alts);
+	if (altsd->bNumEndpoints < 1) {
+		err = -EINVAL;
+		goto error;
+	}
+
 	fp->protocol = altsd->bInterfaceProtocol;
 
 	if (fp->datainterval == 0)
@@ -190,6 +192,12 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	snd_usb_init_pitch(chip, fp->iface, alts, fp);
 	snd_usb_init_sample_rate(chip, fp->iface, alts, fp, fp->rate_max);
 	return 0;
+
+ error:
+	list_del(&fp->list); /* unlink for avoiding double-free */
+	kfree(fp);
+	kfree(rate_table);
+	return err;
 }
 
 static int create_auto_pcm_quirk(struct snd_usb_audio *chip,
@@ -462,6 +470,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 	fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
 	fp->datainterval = 0;
 	fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+	INIT_LIST_HEAD(&fp->list);
 
 	switch (fp->maxpacksize) {
 	case 0x120:
@@ -485,6 +494,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 	err = snd_usb_add_audio_stream(chip, stream, fp);
 	if (err < 0) {
+		list_del(&fp->list); /* unlink for avoiding double-free */
 		kfree(fp);
 		return err;
 	}
@@ -1121,12 +1131,15 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	switch (chip->usb_id) {
 	case USB_ID(0x045E, 0x075D): /* MS Lifecam Cinema  */
 	case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */
+	case USB_ID(0x045E, 0x076E): /* MS Lifecam HD-5001 */
 	case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */
 	case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */
 	case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */
+	case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */
 	case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */
 	case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
 	case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
+	case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */
 	case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
 		return true;
 	}
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 81172a395e55..98b2c28ae46b 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -321,7 +321,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
 /*
  * add this endpoint to the chip instance.
  * if a stream with the same endpoint already exists, append to it.
- * if not, create a new pcm stream.
+ * if not, create a new pcm stream. note, fp is added to the substream
+ * fmt_list and will be freed on the chip instance release. do not free
+ * fp or do remove it from the substream fmt_list to avoid double-free.
  */
 int snd_usb_add_audio_stream(struct snd_usb_audio *chip,
 			     int stream,
@@ -682,6 +684,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 					* (fp->maxpacksize & 0x7ff);
 		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
 		fp->clock = clock;
+		INIT_LIST_HEAD(&fp->list);
 
 		/* some quirks for attributes here */
 
@@ -730,6 +733,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
 		dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
 		err = snd_usb_add_audio_stream(chip, stream, fp);
 		if (err < 0) {
+			list_del(&fp->list); /* unlink for avoiding double-free */
 			kfree(fp->rate_table);
 			kfree(fp->chmap);
 			kfree(fp);