Merge tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Lots of small fixes at this time since we've received the ASoC fix
  batch now.

   - Some coverage in ASoC core mostly for minor issues like NULL checks
     for DPCM and proper error handling in DAI instantiation

   - A collection of small device-specific changes in various ASoC codec
     and platform drivers

   - OF-tree refcount fixes in a few ASoC drivers

   - Fixes of memory leaks in the error paths of various ASoC / ALSA
     drivers

   - A workaround for a long-standing issue on AMD HD-audio device

   - Updates of MAINTAINERS, mail addresses, file permission fixups"

* tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (38 commits)
  ALSA: firewire: fix a memory leak bug
  sound: fix a memory leak bug
  ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
  ALSA: hiface: fix multiple memory leak bugs
  ALSA: hda - Don't override global PCM hw info flag
  ALSA: usb-audio: fix a memory leak bug
  ASoC: max98373: Remove executable bits
  ASoC: amd: acp3x: use dma address for acp3x dma driver
  ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
  ASoC: max98373: add 88200 and 96000 sampling rate support
  ASoC: sun4i-i2s: Incorrect SR and WSS computation
  MAINTAINERS: Update Intel ASoC drivers maintainers
  ASoC: ti: davinci-mcasp: Correct slot_width posed constraint
  ASoC: rockchip: Fix mono capture
  ASoC: Intel: Fix some acpi vs apci typo in somme comments
  ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode
  ASoC: Fail card instantiation if DAI format setup fails
  ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread
  ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
  ASoC: dapm: fix a memory leak bug
  ...

MAINTAINERS

@@ -8042,6 +8042,7 @@ S:	Maintained
 F:	drivers/video/fbdev/i810/
 
 INTEL ASoC DRIVERS
+M:	Cezary Rojewski <cezary.rojewski@intel.com>
 M:	Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
 M:	Liam Girdwood <liam.r.girdwood@linux.intel.com>
 M:	Jie Yang <yang.jie@linux.intel.com>
@@ -16084,7 +16085,7 @@ S:	Maintained
 F:	drivers/net/ethernet/ti/netcp*
 
 TI PCM3060 ASoC CODEC DRIVER
-M:	Kirill Marinushkin <kmarinushkin@birdec.tech>
+M:	Kirill Marinushkin <kmarinushkin@birdec.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	Documentation/devicetree/bindings/sound/pcm3060.txt

include/sound/simple_card_utils.h

@@ -141,6 +141,10 @@ inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv,
 {
 	struct device *dev = simple_priv_to_dev(priv);
 
+	/* dai might be NULL */
+	if (!dai)
+		return;
+
 	if (dai->name)
 		dev_dbg(dev, "%s dai name = %s\n",
 			name, dai->name);

include/uapi/sound/sof/fw.h

@@ -13,6 +13,8 @@
 #ifndef __INCLUDE_UAPI_SOF_FW_H__
 #define __INCLUDE_UAPI_SOF_FW_H__
 
+#include <linux/types.h>
+
 #define SND_SOF_FW_SIG_SIZE	4
 #define SND_SOF_FW_ABI		1
 #define SND_SOF_FW_SIG		"Reef"
@@ -46,8 +48,8 @@ enum snd_sof_fw_blk_type {
 
 struct snd_sof_blk_hdr {
 	enum snd_sof_fw_blk_type type;
-	uint32_t size;		/* bytes minus this header */
-	uint32_t offset;	/* offset from base */
+	__u32 size;		/* bytes minus this header */
+	__u32 offset;		/* offset from base */
 } __packed;
 
 /*
@@ -61,8 +63,8 @@ enum snd_sof_fw_mod_type {
 
 struct snd_sof_mod_hdr {
 	enum snd_sof_fw_mod_type type;
-	uint32_t size;		/* bytes minus this header */
-	uint32_t num_blocks;	/* number of blocks */
+	__u32 size;		/* bytes minus this header */
+	__u32 num_blocks;	/* number of blocks */
 } __packed;
 
 /*
@@ -70,9 +72,9 @@ struct snd_sof_mod_hdr {
  */
 struct snd_sof_fw_header {
 	unsigned char sig[SND_SOF_FW_SIG_SIZE]; /* "Reef" */
-	uint32_t file_size;	/* size of file minus this header */
-	uint32_t num_modules;	/* number of modules */
-	uint32_t abi;		/* version of header format */
+	__u32 file_size;	/* size of file minus this header */
+	__u32 num_modules;	/* number of modules */
+	__u32 abi;		/* version of header format */
 } __packed;
 
 #endif

include/uapi/sound/sof/header.h

@@ -9,6 +9,8 @@
 #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
 #define __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
 
+#include <linux/types.h>
+
 /*
  * Header for all non IPC ABI data.
  *
@@ -16,12 +18,12 @@
  * Used by any bespoke component data structures or binary blobs.
  */
 struct sof_abi_hdr {
-	uint32_t magic;		/**< 'S', 'O', 'F', '\0' */
-	uint32_t type;		/**< component specific type */
-	uint32_t size;		/**< size in bytes of data excl. this struct */
-	uint32_t abi;		/**< SOF ABI version */
-	uint32_t reserved[4];	/**< reserved for future use */
-	uint32_t data[0];	/**< Component data - opaque to core */
+	__u32 magic;		/**< 'S', 'O', 'F', '\0' */
+	__u32 type;		/**< component specific type */
+	__u32 size;		/**< size in bytes of data excl. this struct */
+	__u32 abi;		/**< SOF ABI version */
+	__u32 reserved[4];	/**< reserved for future use */
+	__u32 data[0];		/**< Component data - opaque to core */
 }  __packed;
 
 #endif

sound/firewire/packets-buffer.c

@@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
 	packets_per_page = PAGE_SIZE / packet_size;
 	if (WARN_ON(!packets_per_page)) {
 		err = -EINVAL;
-		goto error;
+		goto err_packets;
 	}
 	pages = DIV_ROUND_UP(count, packets_per_page);
 

sound/pci/hda/hda_controller.c

@@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	}
 	runtime->private_data = azx_dev;
 
-	if (chip->gts_present)
-		azx_pcm_hw.info = azx_pcm_hw.info |
-			SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
-
 	runtime->hw = azx_pcm_hw;
+	if (chip->gts_present)
+		runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;
@@ -615,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
+	/* by some reason, the playback stream stalls on PulseAudio with
+	 * tsched=1 when a capture stream triggers.  Until we figure out the
+	 * real cause, disable tsched mode by telling the PCM info flag.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
+		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory

sound/pci/hda/hda_controller.h

@@ -31,7 +31,7 @@
 /* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-/* 17 unused */
+#define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */

sound/pci/hda/hda_intel.c

@@ -64,6 +64,7 @@ enum {
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
 	POS_FIX_SKL,
+	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -332,6 +333,11 @@ enum {
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
 	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
+/* quirks for AMD SB */
+#define AZX_DCAPS_PRESET_AMD_SB \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\
@@ -841,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+#define AMD_FIFO_SIZE	32
+
+/* get the current DMA position with FIFO size correction */
+static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int pos, delay;
+
+	pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev));
+	if (!runtime)
+		return pos;
+
+	runtime->delay = AMD_FIFO_SIZE;
+	delay = frames_to_bytes(runtime, AMD_FIFO_SIZE);
+	if (azx_dev->insufficient) {
+		if (pos < delay) {
+			delay = pos;
+			runtime->delay = bytes_to_frames(runtime, pos);
+		} else {
+			azx_dev->insufficient = 0;
+		}
+	}
+
+	/* correct the DMA position for capture stream */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (pos < delay)
+			pos += azx_dev->core.bufsize;
+		pos -= delay;
+	}
+
+	return pos;
+}
+
+static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev,
+				   unsigned int pos)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+
+	/* just read back the calculated value in the above */
+	return substream->runtime->delay;
+}
+
 static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
 					 struct azx_dev *azx_dev)
 {
@@ -1417,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
 	case POS_FIX_SKL:
+	case POS_FIX_FIFO:
 		return fix;
 	}
 
@@ -1433,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) {
+		dev_dbg(chip->card->dev, "Using FIFO position fix\n");
+		return POS_FIX_FIFO;
+	}
 	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
@@ -1453,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
 		[POS_FIX_SKL] = azx_get_pos_skl,
+		[POS_FIX_FIFO] = azx_get_pos_fifo,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1467,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix)
 			azx_get_delay_from_lpib;
 	}
 
+	if (fix == POS_FIX_FIFO)
+		chip->get_delay[0] = chip->get_delay[1] =
+			azx_get_delay_from_fifo;
 }
 
 /*
@@ -2447,6 +2505,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD, X370 & co */
+	{ PCI_DEVICE(0x1022, 0x1457),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |

sound/soc/amd/raven/acp3x-pcm-dma.c

@@ -31,8 +31,8 @@ struct i2s_stream_instance {
 	u16 num_pages;
 	u16 channels;
 	u32 xfer_resolution;
-	struct page *pg;
 	u64 bytescount;
+	dma_addr_t dma_addr;
 	void __iomem *acp3x_base;
 };
 
@@ -211,9 +211,8 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
 static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 {
 	u16 page_idx;
-	u64 addr;
 	u32 low, high, val, acp_fifo_addr;
-	struct page *pg = rtd->pg;
+	dma_addr_t addr = rtd->dma_addr;
 
 	/* 8 scratch registers used to map one 64 bit address */
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
@@ -229,7 +228,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 
 	for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) {
 		/* Load the low address of page int ACP SRAM through SRBM */
-		addr = page_to_phys(pg);
 		low = lower_32_bits(addr);
 		high = upper_32_bits(addr);
 
@@ -239,7 +237,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 				+ 4);
 		/* Move to next physically contiguos page */
 		val += 8;
-		pg++;
+		addr += PAGE_SIZE;
 	}
 
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -341,7 +339,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	int status;
 	u64 size;
-	struct page *pg;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct i2s_stream_instance *rtd = runtime->private_data;
 
@@ -354,9 +351,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 		return status;
 
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-	pg = virt_to_page(substream->dma_buffer.area);
-	if (pg) {
-		rtd->pg = pg;
+	if (substream->dma_buffer.area) {
+		rtd->dma_addr = substream->dma_buffer.addr;
 		rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 		config_acp3x_dma(rtd, substream->stream);
 		status = 0;
@@ -385,9 +381,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream)
 
 static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd,
+								    DRV_NAME);
+	struct device *parent = component->dev->parent;
 	snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
-					      rtd->pcm->card->dev,
-					      MIN_BUFFER, MAX_BUFFER);
+					      parent, MIN_BUFFER, MAX_BUFFER);
 	return 0;
 }
 

sound/soc/codecs/cs42xx8.c

@@ -47,6 +47,7 @@ struct cs42xx8_priv {
 	unsigned long sysclk;
 	u32 tx_channels;
 	struct gpio_desc *gpiod_reset;
+	u32 rate[2];
 };
 
 /* -127.5dB to 0dB with step of 0.5dB */
@@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
 };
 
 struct cs42xx8_ratios {
-	unsigned int ratio;
-	unsigned char speed;
-	unsigned char mclk;
+	unsigned int mfreq;
+	unsigned int min_mclk;
+	unsigned int max_mclk;
+	unsigned int ratio[3];
 };
 
+/*
+ * According to reference mannual, define the cs42xx8_ratio struct
+ * MFreq2 | MFreq1 | MFreq0 |     Description     | SSM | DSM | QSM |
+ * 0      | 0      | 0      |1.029MHz to 12.8MHz  | 256 | 128 |  64 |
+ * 0      | 0      | 1      |1.536MHz to 19.2MHz  | 384 | 192 |  96 |
+ * 0      | 1      | 0      |2.048MHz to 25.6MHz  | 512 | 256 | 128 |
+ * 0      | 1      | 1      |3.072MHz to 38.4MHz  | 768 | 384 | 192 |
+ * 1      | x      | x      |4.096MHz to 51.2MHz  |1024 | 512 | 256 |
+ */
 static const struct cs42xx8_ratios cs42xx8_ratios[] = {
-	{ 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
-	{ 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
-	{ 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
-	{ 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
-	{ 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
-	{ 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
-	{ 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
-	{ 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
-	{ 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+	{ 0, 1029000, 12800000, {256, 128, 64} },
+	{ 2, 1536000, 19200000, {384, 192, 96} },
+	{ 4, 2048000, 25600000, {512, 256, 128} },
+	{ 6, 3072000, 38400000, {768, 384, 192} },
+	{ 8, 4096000, 51200000, {1024, 512, 256} },
 };
 
 static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai,
@@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	u32 ratio = cs42xx8->sysclk / params_rate(params);
-	u32 i, fm, val, mask;
+	u32 ratio[2];
+	u32 rate[2];
+	u32 fm[2];
+	u32 i, val, mask;
+	bool condition1, condition2;
 
 	if (tx)
 		cs42xx8->tx_channels = params_channels(params);
 
+	rate[tx]  = params_rate(params);
+	rate[!tx] = cs42xx8->rate[!tx];
+
+	ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0;
+	ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0;
+
+	/* Get functional mode for tx and rx according to rate */
+	for (i = 0; i < 2; i++) {
+		if (cs42xx8->slave_mode) {
+			fm[i] = CS42XX8_FM_AUTO;
+		} else {
+			if (rate[i] < 50000) {
+				fm[i] = CS42XX8_FM_SINGLE;
+			} else if (rate[i] > 50000 && rate[i] < 100000) {
+				fm[i] = CS42XX8_FM_DOUBLE;
+			} else if (rate[i] > 100000 && rate[i] < 200000) {
+				fm[i] = CS42XX8_FM_QUAD;
+			} else {
+				dev_err(component->dev,
+					"unsupported sample rate\n");
+				return -EINVAL;
+			}
+		}
+	}
+
 	for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
-		if (cs42xx8_ratios[i].ratio == ratio)
+		/* Is the ratio[tx] valid ? */
+		condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[tx]) :
+			(cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) &&
+			cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk &&
+			cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk;
+
+		if (!ratio[tx])
+			condition1 = true;
+
+		/* Is the ratio[!tx] valid ? */
+		condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[!tx]) :
+			(cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx]));
+
+		if (!ratio[!tx])
+			condition2 = true;
+
+		/*
+		 * Both ratio[tx] and ratio[!tx] is valid, then we get
+		 * a proper MFreq.
+		 */
+		if (condition1 && condition2)
 			break;
 	}
 
@@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
-	val = cs42xx8_ratios[i].mclk;
+	cs42xx8->rate[tx] = params_rate(params);
 
-	fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+	val = cs42xx8_ratios[i].mfreq;
 
 	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
 			   CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
-			   CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+			   CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val);
+
+	return 0;
+}
+
+static int cs42xx8_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
+	/* Clear stored rate */
+	cs42xx8->rate[tx] = 0;
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+			   CS42XX8_FUNCMOD_xC_FM_MASK(tx),
+			   CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO));
 	return 0;
 }
 
@@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
 	.set_fmt	= cs42xx8_set_dai_fmt,
 	.set_sysclk	= cs42xx8_set_dai_sysclk,
 	.hw_params	= cs42xx8_hw_params,
+	.hw_free	= cs42xx8_hw_free,
 	.digital_mute	= cs42xx8_digital_mute,
 };
 

sound/soc/codecs/max98357a.c

@@ -20,20 +20,10 @@
 #include <sound/soc-dapm.h>
 
 struct max98357a_priv {
-	struct delayed_work enable_sdmode_work;
 	struct gpio_desc *sdmode;
 	unsigned int sdmode_delay;
 };
 
-static void max98357a_enable_sdmode_work(struct work_struct *work)
-{
-	struct max98357a_priv *max98357a =
-	container_of(work, struct max98357a_priv,
-			enable_sdmode_work.work);
-
-	gpiod_set_value(max98357a->sdmode, 1);
-}
-
 static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 		int cmd, struct snd_soc_dai *dai)
 {
@@ -46,14 +36,12 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		queue_delayed_work(system_power_efficient_wq,
-				&max98357a->enable_sdmode_work,
-				msecs_to_jiffies(max98357a->sdmode_delay));
+		mdelay(max98357a->sdmode_delay);
+		gpiod_set_value(max98357a->sdmode, 1);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		cancel_delayed_work_sync(&max98357a->enable_sdmode_work);
 		gpiod_set_value(max98357a->sdmode, 0);
 		break;
 	}
@@ -112,30 +100,25 @@ static int max98357a_platform_probe(struct platform_device *pdev)
 	int ret;
 
 	max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL);
-
 	if (!max98357a)
 		return -ENOMEM;
 
 	max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev,
 				"sdmode", GPIOD_OUT_LOW);
-
 	if (IS_ERR(max98357a->sdmode))
 		return PTR_ERR(max98357a->sdmode);
 
 	ret = device_property_read_u32(&pdev->dev, "sdmode-delay",
 					&max98357a->sdmode_delay);
-
 	if (ret) {
 		max98357a->sdmode_delay = 0;
 		dev_dbg(&pdev->dev,
-			"no optional property 'sdmode-delay' found, default: no delay\n");
+			"no optional property 'sdmode-delay' found, "
+			"default: no delay\n");
 	}
 
 	dev_set_drvdata(&pdev->dev, max98357a);
 
-	INIT_DELAYED_WORK(&max98357a->enable_sdmode_work,
-				max98357a_enable_sdmode_work);
-
 	return devm_snd_soc_register_component(&pdev->dev,
 			&max98357a_component_driver,
 			&max98357a_dai_driver, 1);

sound/soc/codecs/max98373.c

@@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream,
 	case 48000:
 		sampling_rate = MAX98373_PCM_SR_SET1_SR_48000;
 		break;
+	case 88200:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_88200;
+		break;
+	case 96000:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_96000;
+		break;
 	default:
 		dev_err(component->dev, "rate %d not supported\n",
 			params_rate(params));

sound/soc/codecs/max98373.h

@@ -130,6 +130,8 @@
 #define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0)
 #define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0)
 #define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0)
+#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0)
+#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0)
 
 /* MAX98373_R2028_PCM_SR_SETUP_2 */
 #define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4)

sound/soc/codecs/pcm3060-i2c.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 I2C driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/i2c.h>
 #include <linux/module.h>
@@ -56,5 +56,5 @@ static struct i2c_driver pcm3060_i2c_driver = {
 module_i2c_driver(pcm3060_i2c_driver);
 
 MODULE_DESCRIPTION("PCM3060 I2C driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060-spi.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 SPI driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <linux/spi/spi.h>
@@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = {
 module_spi_driver(pcm3060_spi_driver);
 
 MODULE_DESCRIPTION("PCM3060 SPI driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.c

@@ -2,7 +2,7 @@
 //
 // PCM3060 codec driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <sound/pcm_params.h>
@@ -342,5 +342,5 @@ int pcm3060_probe(struct device *dev)
 EXPORT_SYMBOL(pcm3060_probe);
 
 MODULE_DESCRIPTION("PCM3060 codec driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");

sound/soc/codecs/pcm3060.h

@@ -2,7 +2,7 @@
 /*
  * PCM3060 codec driver
  *
- * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
  */
 
 #ifndef _SND_SOC_PCM3060_H

sound/soc/codecs/rt1011.c

@@ -39,7 +39,7 @@ static const struct reg_sequence init_list[] = {
 	{ RT1011_POWER_9, 0xa840 },
 
 	{ RT1011_ADC_SET_5, 0x0a20 },
-	{ RT1011_DAC_SET_2, 0xa232 },
+	{ RT1011_DAC_SET_2, 0xa032 },
 	{ RT1011_ADC_SET_1, 0x2925 },
 
 	{ RT1011_SPK_PRO_DC_DET_1, 0xb00c },
@@ -1917,7 +1917,7 @@ static int rt1011_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_2, 0x0000);
 		snd_soc_component_write(component,
-			RT1011_SYSTEM_RESET_3, 0x0000);
+			RT1011_SYSTEM_RESET_3, 0x0001);
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_1, 0x003f);
 		snd_soc_component_write(component,

sound/soc/generic/audio-graph-card.c

@@ -63,6 +63,7 @@ static int graph_get_dai_id(struct device_node *ep)
 	struct device_node *endpoint;
 	struct of_endpoint info;
 	int i, id;
+	const u32 *reg;
 	int ret;
 
 	/* use driver specified DAI ID if exist */
@@ -83,8 +84,9 @@ static int graph_get_dai_id(struct device_node *ep)
 			return info.id;
 
 		node = of_get_parent(ep);
+		reg = of_get_property(node, "reg", NULL);
 		of_node_put(node);
-		if (of_get_property(node, "reg", NULL))
+		if (reg)
 			return info.port;
 	}
 	node = of_graph_get_port_parent(ep);
@@ -208,10 +210,6 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	dev_dbg(dev, "link_of DPCM (%pOF)\n", ep);
 
-	of_node_put(ports);
-	of_node_put(port);
-	of_node_put(node);
-
 	if (li->cpu) {
 		int is_single_links = 0;
 
@@ -229,17 +227,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* card->num_links includes Codec */
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
@@ -263,17 +261,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(ep, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -293,19 +291,23 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(ep, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep,
 				       NULL, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &graph_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(ports);
+	of_node_put(port);
+	of_node_put(node);
+	return ret;
 }
 
 static int graph_dai_link_of(struct asoc_simple_priv *priv,

sound/soc/generic/simple-card-utils.c

@@ -349,6 +349,13 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link)
 	/* Assumes platform == cpu */
 	if (!dai_link->platforms->of_node)
 		dai_link->platforms->of_node = dai_link->cpus->of_node;
+
+	/*
+	 * DPCM BE can be no platform.
+	 * Alloced memory will be waste, but not leak.
+	 */
+	if (!dai_link->platforms->of_node)
+		dai_link->num_platforms = 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform);
 

sound/soc/generic/simple-card.c

@@ -124,8 +124,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	li->link++;
 
-	of_node_put(node);
-
 	/* For single DAI link & old style of DT node */
 	if (is_top)
 		prefix = PREFIX;
@@ -147,17 +145,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
 	} else {
@@ -180,17 +178,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(np, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -208,19 +206,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(np, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, node, codec,
 				       prefix, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &simple_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(node);
+	return ret;
 }
 
 static int simple_dai_link_of(struct asoc_simple_priv *priv,
@@ -364,8 +364,6 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 			goto error;
 		}
 
-		of_node_put(codec);
-
 		/* get convert-xxx property */
 		memset(&adata, 0, sizeof(adata));
 		for_each_child_of_node(node, np)
@@ -387,11 +385,13 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 				ret = func_noml(priv, np, codec, li, is_top);
 
 			if (ret < 0) {
+				of_node_put(codec);
 				of_node_put(np);
 				goto error;
 			}
 		}
 
+		of_node_put(codec);
 		node = of_get_next_child(top, node);
 	} while (!is_top && node);
 

sound/soc/intel/boards/bytcht_es8316.c

@@ -437,6 +437,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = {
 
 /* Please keep this list alphabetically sorted */
 static const struct dmi_system_id byt_cht_es8316_quirk_table[] = {
+	{	/* Irbis NB41 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "NB41"),
+		},
+		.driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP
+					| BYT_CHT_ES8316_JD_INVERTED),
+	},
 	{	/* Teclast X98 Plus II */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"),

sound/soc/intel/common/soc-acpi-intel-bxt-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ * soc-acpi-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/intel/common/soc-acpi-intel-byt-match.c

 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-byt-match.c - tables and support for BYT ACPI enumeration.
+ * soc-acpi-intel-byt-match.c - tables and support for BYT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */

sound/soc/intel/common/soc-acpi-intel-cht-match.c

 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-cht-match.c - tables and support for CHT ACPI enumeration.
+ * soc-acpi-intel-cht-match.c - tables and support for CHT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */

sound/soc/intel/common/soc-acpi-intel-cnl-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ * soc-acpi-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/intel/common/soc-acpi-intel-glk-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ * soc-acpi-intel-glk-match.c - tables and support for GLK ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/intel/common/soc-acpi-intel-hda-match.c

@@ -2,7 +2,7 @@
 // Copyright (c) 2018, Intel Corporation.
 
 /*
- * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ * soc-acpi-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
  *
  */
 

sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c

 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
+ * soc-acpi-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */

sound/soc/intel/common/soc-acpi-intel-icl-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-icl-match.c - tables and support for ICL ACPI enumeration.
+ * soc-acpi-intel-icl-match.c - tables and support for ICL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/intel/common/soc-acpi-intel-kbl-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ * soc-acpi-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/intel/common/soc-acpi-intel-skl-match.c

 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ * soc-acpi-intel-skl-match.c - tables and support for SKL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *

sound/soc/qcom/apq8016_sbc.c

@@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 
 	link = data->dai_link;
 
-	dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-	if (!dlc)
-		return ERR_PTR(-ENOMEM);
+	for_each_child_of_node(node, np) {
+		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
+		if (!dlc)
+			return ERR_PTR(-ENOMEM);
 
-	link->cpus	= &dlc[0];
-	link->platforms	= &dlc[1];
+		link->cpus	= &dlc[0];
+		link->platforms	= &dlc[1];
 
-	link->num_cpus		= 1;
-	link->num_platforms	= 1;
+		link->num_cpus		= 1;
+		link->num_platforms	= 1;
 
-	for_each_child_of_node(node, np) {
 		cpu = of_get_child_by_name(np, "cpu");
 		codec = of_get_child_by_name(np, "codec");
 

sound/soc/rockchip/rockchip_i2s.c

@@ -326,7 +326,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 		val |= I2S_CHN_4;
 		break;
 	case 2:
-	case 1:
 		val |= I2S_CHN_2;
 		break;
 	default:
@@ -459,7 +458,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 	},
 	.capture = {
 		.stream_name = "Capture",
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = (SNDRV_PCM_FMTBIT_S8 |
@@ -659,7 +658,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	}
 
 	if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
-		if (val >= 1 && val <= 8)
+		if (val >= 2 && val <= 8)
 			soc_dai->capture.channels_max = val;
 	}
 

sound/soc/rockchip/rockchip_max98090.c

@@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Speaker"),
 };
 
+static int rk_jack_event(struct notifier_block *nb, unsigned long event,
+			 void *data)
+{
+	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+	if (event & SND_JACK_MICROPHONE)
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+	else
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static struct notifier_block rk_jack_nb = {
+	.notifier_call = rk_jack_event,
+};
+
+static int rk_init(struct snd_soc_pcm_runtime *runtime)
+{
+	/*
+	 * The jack has already been created in the rk_98090_headset_init()
+	 * function.
+	 */
+	snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb);
+
+	return 0;
+}
+
 static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi,
 static struct snd_soc_dai_link rk_dailink = {
 	.name = "max98090",
 	.stream_name = "Audio",
+	.init = rk_init,
 	.ops = &rk_aif1_ops,
 	/* set max98090 as slave */
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |

sound/soc/samsung/odroid.c

@@ -284,9 +284,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 	}
 
 	of_node_put(cpu);
-	of_node_put(codec);
 	if (ret < 0)
-		return ret;
+		goto err_put_node;
 
 	ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link);
 	if (ret < 0)
@@ -309,7 +308,6 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		ret = PTR_ERR(priv->clk_i2s_bus);
 		goto err_put_sclk;
 	}
-	of_node_put(cpu_dai);
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
@@ -317,6 +315,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		goto err_put_clk_i2s;
 	}
 
+	of_node_put(cpu_dai);
+	of_node_put(codec);
 	return 0;
 
 err_put_clk_i2s:
@@ -326,6 +326,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 err_put_cpu_dai:
 	of_node_put(cpu_dai);
 	snd_soc_of_put_dai_link_codecs(codec_link);
+err_put_node:
+	of_node_put(codec);
 	return ret;
 }
 

sound/soc/soc-core.c

@@ -1515,8 +1515,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		}
 	}
 
-	if (dai_link->dai_fmt)
-		snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+	if (dai_link->dai_fmt) {
+		ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+		if (ret)
+			return ret;
+	}
 
 	ret = soc_post_component_init(rtd, dai_link->name);
 	if (ret)

sound/soc/soc-dapm.c

@@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
 		list_add_tail(&widget->work_list, list);
 
 	if (custom_stop_condition && custom_stop_condition(widget, dir)) {
-		widget->endpoints[dir] = 1;
-		return widget->endpoints[dir];
+		list = NULL;
+		custom_stop_condition = NULL;
 	}
 
 	if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
  *
  * Optionally, can be supplied with a function acting as a stopping condition.
  * This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
  */
 static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	struct list_head *list,
@@ -3706,6 +3706,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm,
 		dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
 			w->name, ret);
 
+	kfree_const(w->sname);
+	kfree(w);
 	return ERR_PTR(ret);
 }
 

sound/soc/sof/intel/cnl.c

@@ -101,8 +101,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */

sound/soc/sof/intel/hda-ipc.c

@@ -224,8 +224,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */

sound/soc/sunxi/sun4i-i2s.c

@@ -1002,8 +1002,8 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
 	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
-	.get_sr			= sun8i_i2s_get_sr_wss,
-	.get_wss		= sun8i_i2s_get_sr_wss,
+	.get_sr			= sun4i_i2s_get_sr,
+	.get_wss		= sun4i_i2s_get_wss,
 };
 
 static int sun4i_i2s_init_regmap_fields(struct device *dev,

sound/soc/ti/davinci-mcasp.c

@@ -195,7 +195,7 @@ static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable)
 {
 	u32 bit;
 
-	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) {
+	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) {
 		if (enable)
 			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
 		else
@@ -223,6 +223,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 	if (mcasp_is_synchronous(mcasp)) {
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+		mcasp_set_clk_pdir(mcasp, true);
 	}
 
 	/* Activate serializer(s) */
@@ -1256,6 +1257,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1377,7 +1400,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	struct davinci_mcasp_ruledata *ruledata =
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
-	int i, dir;
+	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
 
 	/* Do not allow more then one stream per direction */
@@ -1406,6 +1429,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
@@ -1431,20 +1455,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width)
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     8, mcasp->slot_width);
+	if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
 	 * set constraints based on what we can provide.
 	 */
 	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
-		int ret;
-
-		ruledata->mcasp = mcasp;
-
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
 					  davinci_mcasp_hw_rule_rate,

sound/sound_core.c

@@ -275,7 +275,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati
 				goto retry;
 			}
 			spin_unlock(&sound_loader_lock);
-			return -EBUSY;
+			r = -EBUSY;
+			goto fail;
 		}
 	}
 

sound/usb/hiface/pcm.c

@@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 		ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP,
 				    hiface_pcm_out_urb_handler);
 		if (ret < 0)
-			return ret;
+			goto error;
 	}
 
 	ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm);
 	if (ret < 0) {
-		kfree(rt);
 		dev_err(&chip->dev->dev, "Cannot create pcm instance\n");
-		return ret;
+		goto error;
 	}
 
 	pcm->private_data = rt;
@@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 
 	chip->pcm = rt;
 	return 0;
+
+error:
+	for (i = 0; i < PCM_N_URBS; i++)
+		kfree(rt->out_urbs[i].buffer);
+	kfree(rt);
+	return ret;
 }

sound/usb/stream.c

@@ -1043,6 +1043,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
 
 		pd = kzalloc(sizeof(*pd), GFP_KERNEL);
 		if (!pd) {
+			kfree(fp->chmap);
 			kfree(fp->rate_table);
 			kfree(fp);
 			return NULL;