Merge branch 'topic/intel-lpe-audio' into for-next

30 files changed, 4103 insertions, 78 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index 5a240e050ae6..ee2e69a9ecd1 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -108,6 +108,8 @@ source "sound/parisc/Kconfig"
 
 source "sound/soc/Kconfig"
 
+source "sound/x86/Kconfig"
+
 endif # SND
 
 menuconfig SOUND_PRIME
diff --git a/sound/Makefile b/sound/Makefile
index c41bdf5fdf24..6de45d2c32f7 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
 obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
-	firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/
+	firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/
 obj-$(CONFIG_SND_AOA) += aoa/
 
 # This one must be compilable even if sound is configured out
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index 8ce93cdc4b0d..00060c4a9deb 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -69,7 +69,7 @@ static void pcm_period_tasklet(unsigned long data);
  * @protocol_size: the size to allocate newly for protocol
  */
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
-		      enum amdtp_stream_direction dir, int flags,
+		      enum amdtp_stream_direction dir, enum cip_flags flags,
 		      unsigned int fmt,
 		      amdtp_stream_process_data_blocks_t process_data_blocks,
 		      unsigned int protocol_size)
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 7be21429cd12..c1bc7fad056e 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -93,7 +93,7 @@ typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
 						unsigned int *syt);
 struct amdtp_stream {
 	struct fw_unit *unit;
-	int flags;
+	enum cip_flags flags;
 	enum amdtp_stream_direction direction;
 	struct mutex mutex;
 
@@ -137,7 +137,7 @@ struct amdtp_stream {
 };
 
 int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
-		      enum amdtp_stream_direction dir, int flags,
+		      enum amdtp_stream_direction dir, enum cip_flags flags,
 		      unsigned int fmt,
 		      amdtp_stream_process_data_blocks_t process_data_blocks,
 		      unsigned int protocol_size);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 06b5a480db8d..89f0b01de30d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2233,6 +2233,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
 	SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
+	SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
 	SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
@@ -7016,6 +7017,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
 	SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8),
 	SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
 	SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c
index efe3a44658d5..4576f987a4a5 100644
--- a/sound/soc/codecs/nau8825.c
+++ b/sound/soc/codecs/nau8825.c
@@ -561,9 +561,9 @@ static void nau8825_xtalk_prepare(struct nau8825 *nau8825)
 	nau8825_xtalk_backup(nau8825);
 	/* Config IIS as master to output signal by codec */
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
-		NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK |
+		NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK |
 		NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER |
-		(0x2 << NAU8825_I2S_DRV_SFT) | 0x1);
+		(0x2 << NAU8825_I2S_LRC_DIV_SFT) | 0x1);
 	/* Ramp up headphone volume to 0dB to get better performance and
 	 * avoid pop noise in headphone.
 	 */
@@ -657,7 +657,7 @@ static void nau8825_xtalk_clean(struct nau8825 *nau8825)
 		NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN);
 	/* Recover default value for IIS */
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2,
-		NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK |
+		NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK |
 		NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_SLAVE);
 	/* Restore value of specific register for cross talk */
 	nau8825_xtalk_restore(nau8825);
@@ -2006,7 +2006,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825,
 			NAU8825_FLL_INTEGER_MASK, fll_param->fll_int);
 	/* FLL pre-scaler */
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4,
-			NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div);
+			NAU8825_FLL_REF_DIV_MASK,
+			fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT);
 	/* select divided VCO input */
 	regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5,
 		NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF);
diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h
index 5d1704e73241..514fd13c2f46 100644
--- a/sound/soc/codecs/nau8825.h
+++ b/sound/soc/codecs/nau8825.h
@@ -137,7 +137,8 @@
 #define NAU8825_FLL_CLK_SRC_FS			(0x3 << NAU8825_FLL_CLK_SRC_SFT)
 
 /* FLL4 (0x07) */
-#define NAU8825_FLL_REF_DIV_MASK		(0x3 << 10)
+#define NAU8825_FLL_REF_DIV_SFT	10
+#define NAU8825_FLL_REF_DIV_MASK	(0x3 << NAU8825_FLL_REF_DIV_SFT)
 
 /* FLL5 (0x08) */
 #define NAU8825_FLL_PDB_DAC_EN		(0x1 << 15)
@@ -247,8 +248,8 @@
 
 /* I2S_PCM_CTRL2 (0x1d) */
 #define NAU8825_I2S_TRISTATE	(1 << 15) /* 0 - normal mode, 1 - Hi-Z output */
-#define NAU8825_I2S_DRV_SFT	12
-#define NAU8825_I2S_DRV_MASK	(0x3 << NAU8825_I2S_DRV_SFT)
+#define NAU8825_I2S_LRC_DIV_SFT	12
+#define NAU8825_I2S_LRC_DIV_MASK	(0x3 << NAU8825_I2S_LRC_DIV_SFT)
 #define NAU8825_I2S_MS_SFT	3
 #define NAU8825_I2S_MS_MASK	(1 << NAU8825_I2S_MS_SFT)
 #define NAU8825_I2S_MS_MASTER	(1 << NAU8825_I2S_MS_SFT)
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 10c2a564a715..1ac96ef9ee20 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3833,6 +3833,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 		}
 	}
 
+	regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1,
+		RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2);
+
 	if (rt5645->pdata.jd_invert) {
 		regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
 			RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 8877b74b0510..bb94d50052d7 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -126,6 +126,16 @@ static const struct reg_default aic3x_reg[] = {
 	{ 108, 0x00 }, { 109, 0x00 },
 };
 
+static bool aic3x_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case AIC3X_RESET:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static const struct regmap_config aic3x_regmap = {
 	.reg_bits = 8,
 	.val_bits = 8,
@@ -133,6 +143,9 @@ static const struct regmap_config aic3x_regmap = {
 	.max_register = DAC_ICC_ADJ,
 	.reg_defaults = aic3x_reg,
 	.num_reg_defaults = ARRAY_SIZE(aic3x_reg),
+
+	.volatile_reg = aic3x_volatile_reg,
+
 	.cache_type = REGCACHE_RBTREE,
 };
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 593b7d1aed46..d72ccef9e238 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1551,7 +1551,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 	const struct wmfw_region *region;
 	const struct wm_adsp_region *mem;
 	const char *region_name;
-	char *file, *text;
+	char *file, *text = NULL;
 	struct wm_adsp_buf *buf;
 	unsigned int reg;
 	int regions = 0;
@@ -1700,10 +1700,21 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 			 regions, le32_to_cpu(region->len), offset,
 			 region_name);
 
+		if ((pos + le32_to_cpu(region->len) + sizeof(*region)) >
+		    firmware->size) {
+			adsp_err(dsp,
+				 "%s.%d: %s region len %d bytes exceeds file length %zu\n",
+				 file, regions, region_name,
+				 le32_to_cpu(region->len), firmware->size);
+			ret = -EINVAL;
+			goto out_fw;
+		}
+
 		if (text) {
 			memcpy(text, region->data, le32_to_cpu(region->len));
 			adsp_info(dsp, "%s: %s\n", file, text);
 			kfree(text);
+			text = NULL;
 		}
 
 		if (reg) {
@@ -1748,6 +1759,7 @@ out_fw:
 	regmap_async_complete(regmap);
 	wm_adsp_buf_free(&buf_list);
 	release_firmware(firmware);
+	kfree(text);
 out:
 	kfree(file);
 
@@ -2233,6 +2245,17 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 		}
 
 		if (reg) {
+			if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) >
+			    firmware->size) {
+				adsp_err(dsp,
+					 "%s.%d: %s region len %d bytes exceeds file length %zu\n",
+					 file, blocks, region_name,
+					 le32_to_cpu(blk->len),
+					 firmware->size);
+				ret = -EINVAL;
+				goto out_fw;
+			}
+
 			buf = wm_adsp_buf_alloc(blk->data,
 						le32_to_cpu(blk->len),
 						&buf_list);
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 2998954a1c74..bdf8398cbc81 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -681,22 +681,19 @@ static int dw_i2s_probe(struct platform_device *pdev)
 	}
 
 	if (!pdata) {
-		ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
-		if (ret == -EPROBE_DEFER) {
-			dev_err(&pdev->dev,
-				"failed to register PCM, deferring probe\n");
-			return ret;
-		} else if (ret) {
-			dev_err(&pdev->dev,
-				"Could not register DMA PCM: %d\n"
-				"falling back to PIO mode\n", ret);
+		if (irq >= 0) {
 			ret = dw_pcm_register(pdev);
-			if (ret) {
-				dev_err(&pdev->dev,
-					"Could not register PIO PCM: %d\n",
+			dev->use_pio = true;
+		} else {
+			ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
+					0);
+			dev->use_pio = false;
+		}
+
+		if (ret) {
+			dev_err(&pdev->dev, "could not register pcm: %d\n",
 					ret);
-				goto err_clk_disable;
-			}
+			goto err_clk_disable;
 		}
 	}
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 50349437d961..fde08660b63b 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -224,6 +224,12 @@ struct fsl_ssi_soc_data {
  * @dbg_stats: Debugging statistics
  *
  * @soc: SoC specific data
+ *
+ * @fifo_watermark: the FIFO watermark setting.  Notifies DMA when
+ *             there are @fifo_watermark or fewer words in TX fifo or
+ *             @fifo_watermark or more empty words in RX fifo.
+ * @dma_maxburst: max number of words to transfer in one go.  So far,
+ *             this is always the same as fifo_watermark.
  */
 struct fsl_ssi_private {
 	struct regmap *regs;
@@ -263,6 +269,9 @@ struct fsl_ssi_private {
 
 	const struct fsl_ssi_soc_data *soc;
 	struct device *dev;
+
+	u32 fifo_watermark;
+	u32 dma_maxburst;
 };
 
 /*
@@ -1051,21 +1060,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 	regmap_write(regs, CCSR_SSI_SRCR, srcr);
 	regmap_write(regs, CCSR_SSI_SCR, scr);
 
-	/*
-	 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't
-	 * use FIFO 1. We program the transmit water to signal a DMA transfer
-	 * if there are only two (or fewer) elements left in the FIFO. Two
-	 * elements equals one frame (left channel, right channel). This value,
-	 * however, depends on the depth of the transmit buffer.
-	 *
-	 * We set the watermark on the same level as the DMA burstsize.  For
-	 * fiq it is probably better to use the biggest possible watermark
-	 * size.
-	 */
-	if (ssi_private->use_dma)
-		wm = ssi_private->fifo_depth - 2;
-	else
-		wm = ssi_private->fifo_depth;
+	wm = ssi_private->fifo_watermark;
 
 	regmap_write(regs, CCSR_SSI_SFCSR,
 			CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) |
@@ -1373,12 +1368,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 		dev_dbg(&pdev->dev, "could not get baud clock: %ld\n",
 			 PTR_ERR(ssi_private->baudclk));
 
-	/*
-	 * We have burstsize be "fifo_depth - 2" to match the SSI
-	 * watermark setting in fsl_ssi_startup().
-	 */
-	ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2;
-	ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2;
+	ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst;
+	ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst;
 	ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
 	ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
 
@@ -1543,6 +1534,47 @@ static int fsl_ssi_probe(struct platform_device *pdev)
                 /* Older 8610 DTs didn't have the fifo-depth property */
 		ssi_private->fifo_depth = 8;
 
+	/*
+	 * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't
+	 * use FIFO 1 but set the watermark appropriately nontheless.
+	 * We program the transmit water to signal a DMA transfer
+	 * if there are N elements left in the FIFO. For chips with 15-deep
+	 * FIFOs, set watermark to 8.  This allows the SSI to operate at a
+	 * high data rate without channel slipping. Behavior is unchanged
+	 * for the older chips with a fifo depth of only 8.  A value of 4
+	 * might be appropriate for the older chips, but is left at
+	 * fifo_depth-2 until sombody has a chance to test.
+	 *
+	 * We set the watermark on the same level as the DMA burstsize.  For
+	 * fiq it is probably better to use the biggest possible watermark
+	 * size.
+	 */
+	switch (ssi_private->fifo_depth) {
+	case 15:
+		/*
+		 * 2 samples is not enough when running at high data
+		 * rates (like 48kHz @ 16 bits/channel, 16 channels)
+		 * 8 seems to split things evenly and leave enough time
+		 * for the DMA to fill the FIFO before it's over/under
+		 * run.
+		 */
+		ssi_private->fifo_watermark = 8;
+		ssi_private->dma_maxburst = 8;
+		break;
+	case 8:
+	default:
+		/*
+		 * maintain old behavior for older chips.
+		 * Keeping it the same because I don't have an older
+		 * board to test with.
+		 * I suspect this could be changed to be something to
+		 * leave some more space in the fifo.
+		 */
+		ssi_private->fifo_watermark = ssi_private->fifo_depth - 2;
+		ssi_private->dma_maxburst = ssi_private->fifo_depth - 2;
+		break;
+	}
+
 	dev_set_drvdata(&pdev->dev, ssi_private);
 
 	if (ssi_private->soc->imx) {
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 507a86a5eafe..8d2fb2d6f532 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -142,7 +142,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
 		 * for Jack detection and button press
 		 */
 		ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_RCCLK,
-					     0,
+					     48000 * 512,
 					     SND_SOC_CLOCK_IN);
 		if (!ret) {
 			if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk)
@@ -825,10 +825,20 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
 	if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && (is_valleyview())) {
 		priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
 		if (IS_ERR(priv->mclk)) {
+			ret_val = PTR_ERR(priv->mclk);
+
 			dev_err(&pdev->dev,
-				"Failed to get MCLK from pmc_plt_clk_3: %ld\n",
-				PTR_ERR(priv->mclk));
-			return PTR_ERR(priv->mclk);
+				"Failed to get MCLK from pmc_plt_clk_3: %d\n",
+				ret_val);
+
+			/*
+			 * Fall back to bit clock usage for -ENOENT (clock not
+			 * available likely due to missing dependencies), bail
+			 * for all other errors, including -EPROBE_DEFER
+			 */
+			if (ret_val != -ENOENT)
+				return ret_val;
+			byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN;
 		}
 	}
 
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 84b5101e6ca6..6c6b63a6b338 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -180,6 +180,9 @@ static int skl_pcm_open(struct snd_pcm_substream *substream,
 	snd_pcm_set_sync(substream);
 
 	mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream);
+	if (!mconfig)
+		return -EINVAL;
+
 	skl_tplg_d0i3_get(skl, mconfig->d0i3_caps);
 
 	return 0;
diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c
index 8fc3178bc79c..b30bd384c8d3 100644
--- a/sound/soc/intel/skylake/skl-sst.c
+++ b/sound/soc/intel/skylake/skl-sst.c
@@ -515,6 +515,9 @@ EXPORT_SYMBOL_GPL(skl_sst_init_fw);
 
 void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx)
 {
+
+	if (ctx->dsp->fw)
+		release_firmware(ctx->dsp->fw);
 	skl_clear_module_table(ctx->dsp);
 	skl_freeup_uuid_list(ctx);
 	skl_ipc_free(&ctx->ipc);
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 4bd68de76130..99b5b0835c1e 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1030,10 +1030,8 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod,
 		return -ENOMEM;
 
 	ret = snd_ctl_add(card, kctrl);
-	if (ret < 0) {
-		snd_ctl_free_one(kctrl);
+	if (ret < 0)
 		return ret;
-	}
 
 	cfg->update = update;
 	cfg->card = card;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f1901bb1466e..baa1afa41e3d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1748,6 +1748,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num)
 
 	component->init = aux_dev->init;
 	component->auxiliary = 1;
+	list_add(&component->card_aux_list, &card->aux_comp_list);
 
 	return 0;
 
@@ -1758,16 +1759,14 @@ err_defer:
 
 static int soc_probe_aux_devices(struct snd_soc_card *card)
 {
-	struct snd_soc_component *comp;
+	struct snd_soc_component *comp, *tmp;
 	int order;
 	int ret;
 
 	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
 		order++) {
-		list_for_each_entry(comp, &card->component_dev_list, card_list) {
-			if (!comp->auxiliary)
-				continue;
-
+		list_for_each_entry_safe(comp, tmp, &card->aux_comp_list,
+					 card_aux_list) {
 			if (comp->driver->probe_order == order) {
 				ret = soc_probe_component(card,	comp);
 				if (ret < 0) {
@@ -1776,6 +1775,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card)
 						comp->name, ret);
 					return ret;
 				}
+				list_del(&comp->card_aux_list);
 			}
 		}
 	}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e7a1eaa2772f..6aba14009c92 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2184,9 +2184,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
 		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED;
+		break;
 	}
 
 out:
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 65670b2b408c..fbfb1fab88d5 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -514,13 +514,12 @@ static void remove_widget(struct snd_soc_component *comp,
 			    == SND_SOC_TPLG_TYPE_MIXER)
 				kfree(kcontrol->tlv.p);
 
-			snd_ctl_remove(card, kcontrol);
-
 			/* Private value is used as struct soc_mixer_control
 			 * for volume mixers or soc_bytes_ext for bytes
 			 * controls.
 			 */
 			kfree((void *)kcontrol->private_value);
+			snd_ctl_remove(card, kcontrol);
 		}
 		kfree(w->kcontrol_news);
 	}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 15d1d5c63c3c..c90607ebe155 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -384,6 +384,9 @@ static void snd_complete_urb(struct urb *urb)
 	if (unlikely(atomic_read(&ep->chip->shutdown)))
 		goto exit_clear;
 
+	if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+		goto exit_clear;
+
 	if (usb_pipeout(ep->pipe)) {
 		retire_outbound_urb(ep, ctx);
 		/* can be stopped during retire callback */
@@ -534,6 +537,11 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep)
 			alive, ep->ep_num);
 	clear_bit(EP_FLAG_STOPPING, &ep->flags);
 
+	ep->data_subs = NULL;
+	ep->sync_slave = NULL;
+	ep->retire_data_urb = NULL;
+	ep->prepare_data_urb = NULL;
+
 	return 0;
 }
 
@@ -912,9 +920,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 /**
  * snd_usb_endpoint_start: start an snd_usb_endpoint
  *
- * @ep:		the endpoint to start
- * @can_sleep:	flag indicating whether the operation is executed in
- * 		non-atomic context
+ * @ep: the endpoint to start
  *
  * A call to this function will increment the use count of the endpoint.
  * In case it is not already running, the URBs for this endpoint will be
@@ -924,7 +930,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
  *
  * Returns an error if the URB submission failed, 0 in all other cases.
  */
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep)
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 {
 	int err;
 	unsigned int i;
@@ -938,8 +944,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep)
 
 	/* just to be sure */
 	deactivate_urbs(ep, false);
-	if (can_sleep)
-		wait_clear_urbs(ep);
 
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
@@ -1020,10 +1024,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep)
 
 	if (--ep->use_count == 0) {
 		deactivate_urbs(ep, false);
-		ep->data_subs = NULL;
-		ep->sync_slave = NULL;
-		ep->retire_data_urb = NULL;
-		ep->prepare_data_urb = NULL;
 		set_bit(EP_FLAG_STOPPING, &ep->flags);
 	}
 }
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 6428392d8f62..584f295d7c77 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -18,7 +18,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 				struct audioformat *fmt,
 				struct snd_usb_endpoint *sync_ep);
 
-int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep);
+int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep);
 int  snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 34c6d4f2c0b6..9aa5b1855481 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -218,7 +218,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 	}
 }
 
-static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep)
+static int start_endpoints(struct snd_usb_substream *subs)
 {
 	int err;
 
@@ -231,7 +231,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep)
 		dev_dbg(&subs->dev->dev, "Starting data EP @%p\n", ep);
 
 		ep->data_subs = subs;
-		err = snd_usb_endpoint_start(ep, can_sleep);
+		err = snd_usb_endpoint_start(ep);
 		if (err < 0) {
 			clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
 			return err;
@@ -260,7 +260,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep)
 		dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep);
 
 		ep->sync_slave = subs->data_endpoint;
-		err = snd_usb_endpoint_start(ep, can_sleep);
+		err = snd_usb_endpoint_start(ep);
 		if (err < 0) {
 			clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
 			return err;
@@ -850,7 +850,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	/* for playback, submit the URBs now; otherwise, the first hwptr_done
 	 * updates for all URBs would happen at the same time when starting */
 	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
-		ret = start_endpoints(subs, true);
+		ret = start_endpoints(subs);
 
  unlock:
 	snd_usb_unlock_shutdown(subs->stream->chip);
@@ -1666,7 +1666,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		err = start_endpoints(subs, false);
+		err = start_endpoints(subs);
 		if (err < 0)
 			return err;
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index f5c68ea0468e..01eff6ce6401 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1135,6 +1135,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
 	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, 0x02F7): /* Plantronics BT-600 */
 	case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */
 	case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */
 	case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
new file mode 100644
index 000000000000..30d066e4885e
--- /dev/null
+++ b/sound/x86/Kconfig
@@ -0,0 +1,15 @@
+menuconfig SND_X86
+	tristate "X86 sound devices"
+	depends on X86
+	---help---
+	  X86 sound devices that don't fall under SoC or PCI categories
+
+if SND_X86
+
+config HDMI_LPE_AUDIO
+	tristate "HDMI audio without HDaudio on Intel Atom platforms"
+	depends on DRM_I915
+	help
+	 Choose this option to support HDMI LPE Audio mode
+
+endif	# SND_X86
diff --git a/sound/x86/Makefile b/sound/x86/Makefile
new file mode 100644
index 000000000000..85ea22a2cf28
--- /dev/null
+++ b/sound/x86/Makefile
@@ -0,0 +1,6 @@
+snd-hdmi-lpe-audio-objs += \
+	intel_hdmi_audio.o \
+	intel_hdmi_audio_if.o \
+	intel_hdmi_lpe_audio.o
+
+obj-$(CONFIG_HDMI_LPE_AUDIO) += snd-hdmi-lpe-audio.o
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
new file mode 100644
index 000000000000..f30155446117
--- /dev/null
+++ b/sound/x86/intel_hdmi_audio.c
@@ -0,0 +1,1870 @@
+/*
+ *   intel_hdmi_audio.c - Intel HDMI audio driver
+ *
+ *  Copyright (C) 2016 Intel Corp
+ *  Authors:	Sailaja Bandarupalli <sailaja.bandarupalli@intel.com>
+ *		Ramesh Babu K V	<ramesh.babu@intel.com>
+ *		Vaibhav Agarwal <vaibhav.agarwal@intel.com>
+ *		Jerome Anand <jerome.anand@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ * ALSA driver for Intel HDMI audio
+ */