Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

24 files changed, 551 insertions, 231 deletions

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index ea7b4787a8af..1c4ca5ec8caf 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -315,6 +315,7 @@ config SND_SOC_FSL_ASOC_CARD
 	depends on OF && I2C
 	# enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m:
 	depends on SND_AC97_CODEC || SND_AC97_CODEC=n
+	select SND_SIMPLE_CARD_UTILS
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_FSL_ESAI
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 4ff2d21bb32f..e13271ea84de 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -30,7 +30,7 @@
 static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 00be73900888..de136c0a497d 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -15,6 +15,8 @@
 #endif
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/simple_card_utils.h>
 
 #include "fsl_esai.h"
 #include "fsl_sai.h"
@@ -33,8 +35,7 @@
 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
 
 /**
- * CODEC private data
- *
+ * struct codec_priv - CODEC private data
  * @mclk_freq: Clock rate of MCLK
  * @mclk_id: MCLK (or main clock) id for set_sysclk()
  * @fll_id: FLL (or secordary clock) id for set_sysclk()
@@ -48,11 +49,10 @@ struct codec_priv {
 };
 
 /**
- * CPU private data
- *
- * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
- * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
- * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ * struct cpu_priv - CPU private data
+ * @sysclk_freq: SYSCLK rates for set_sysclk()
+ * @sysclk_dir: SYSCLK directions for set_sysclk()
+ * @sysclk_id: SYSCLK ids for set_sysclk()
  * @slot_width: Slot width of each frame
  *
  * Note: [1] for tx and [0] for rx
@@ -65,9 +65,10 @@ struct cpu_priv {
 };
 
 /**
- * Freescale Generic ASOC card private data
- *
- * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
+ * @dai_link: DAI link structure including normal one and DPCM link
+ * @hp_jack: Headphone Jack structure
+ * @mic_jack: Microphone Jack structure
  * @pdev: platform device pointer
  * @codec_priv: CODEC private data
  * @cpu_priv: CPU private data
@@ -82,6 +83,8 @@ struct cpu_priv {
 
 struct fsl_asoc_card_priv {
 	struct snd_soc_dai_link dai_link[3];
+	struct asoc_simple_jack hp_jack;
+	struct asoc_simple_jack mic_jack;
 	struct platform_device *pdev;
 	struct codec_priv codec_priv;
 	struct cpu_priv cpu_priv;
@@ -94,8 +97,8 @@ struct fsl_asoc_card_priv {
 	char name[32];
 };
 
-/**
- * This dapm route map exsits for DPCM link only.
+/*
+ * This dapm route map exists for DPCM link only.
  * The other routes shall go through Device Tree.
  *
  * Note: keep all ASRC routes in the second half
@@ -119,6 +122,13 @@ static const struct snd_soc_dapm_route audio_map_ac97[] = {
 	{"ASRC-Capture",  NULL, "AC97 Capture"},
 };
 
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+	/* 1st half -- Normal DAPM routes */
+	{"Playback",  NULL, "CPU-Playback"},
+	/* 2nd half -- ASRC DAPM routes */
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+};
+
 /* Add all possible widgets into here without being redundant */
 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
@@ -138,7 +148,7 @@ static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	struct cpu_priv *cpu_priv = &priv->cpu_priv;
@@ -441,6 +451,44 @@ static int fsl_asoc_card_audmux_init(struct device_node *np,
 	return 0;
 }
 
+static int hp_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_HEADPHONE)
+		/* Disable speaker if headphone is plugged in */
+		snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+
+	return 0;
+}
+
+static struct notifier_block hp_jack_nb = {
+	.notifier_call = hp_jack_event,
+};
+
+static int mic_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)
+		/* Disable dmic if microphone is plugged in */
+		snd_soc_dapm_disable_pin(dapm, "DMIC");
+	else
+		snd_soc_dapm_enable_pin(dapm, "DMIC");
+
+	return 0;
+}
+
+static struct notifier_block mic_jack_nb = {
+	.notifier_call = mic_jack_event,
+};
+
 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
 {
 	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
@@ -483,10 +531,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	struct device_node *cpu_np, *codec_np, *asrc_np;
 	struct device_node *np = pdev->dev.of_node;
 	struct platform_device *asrc_pdev = NULL;
+	struct device_node *bitclkmaster = NULL;
+	struct device_node *framemaster = NULL;
 	struct platform_device *cpu_pdev;
 	struct fsl_asoc_card_priv *priv;
-	struct i2c_client *codec_dev;
+	struct device *codec_dev = NULL;
 	const char *codec_dai_name;
+	const char *codec_dev_name;
+	unsigned int daifmt;
 	u32 width;
 	int ret;
 
@@ -512,10 +564,23 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	}
 
 	codec_np = of_parse_phandle(np, "audio-codec", 0);
-	if (codec_np)
-		codec_dev = of_find_i2c_device_by_node(codec_np);
-	else
-		codec_dev = NULL;
+	if (codec_np) {
+		struct platform_device *codec_pdev;
+		struct i2c_client *codec_i2c;
+
+		codec_i2c = of_find_i2c_device_by_node(codec_np);
+		if (codec_i2c) {
+			codec_dev = &codec_i2c->dev;
+			codec_dev_name = codec_i2c->name;
+		}
+		if (!codec_dev) {
+			codec_pdev = of_find_device_by_node(codec_np);
+			if (codec_pdev) {
+				codec_dev = &codec_pdev->dev;
+				codec_dev_name = codec_pdev->name;
+			}
+		}
+	}
 
 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
 	if (asrc_np)
@@ -523,7 +588,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
 	if (codec_dev) {
-		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+		struct clk *codec_clk = clk_get(codec_dev, NULL);
 
 		if (!IS_ERR(codec_clk)) {
 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
@@ -538,6 +603,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	/* Assign a default DAI format, and allow each card to overwrite it */
 	priv->dai_fmt = DAI_FMT_BASE;
 
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	priv->card.dapm_routes = audio_map;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	/* Diversify the card configurations */
 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
 		codec_dai_name = "cs42888";
@@ -573,12 +643,58 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		codec_dai_name = "ac97-hifi";
 		priv->card.set_bias_level = NULL;
 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
+		priv->card.dapm_routes = audio_map_ac97;
+		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
+		codec_dai_name = "fsl-mqs-dai";
+		priv->card.set_bias_level = NULL;
+		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
+				SND_SOC_DAIFMT_CBS_CFS |
+				SND_SOC_DAIFMT_NB_NF;
+		priv->dai_link[1].dpcm_capture = 0;
+		priv->dai_link[2].dpcm_capture = 0;
+		priv->card.dapm_routes = audio_map_tx;
+		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
+		codec_dai_name = "wm8524-hifi";
+		priv->card.set_bias_level = NULL;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+		priv->dai_link[1].dpcm_capture = 0;
+		priv->dai_link[2].dpcm_capture = 0;
+		priv->cpu_priv.slot_width = 32;
+		priv->card.dapm_routes = audio_map_tx;
+		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
 	} else {
 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
 		ret = -EINVAL;
 		goto asrc_fail;
 	}
 
+	/* Format info from DT is optional. */
+	daifmt = snd_soc_of_parse_daifmt(np, NULL,
+					 &bitclkmaster, &framemaster);
+	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
+	if (bitclkmaster || framemaster) {
+		if (codec_np == bitclkmaster)
+			daifmt |= (codec_np == framemaster) ?
+				SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
+		else
+			daifmt |= (codec_np == framemaster) ?
+				SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
+
+		/* Override dai_fmt with value from DT */
+		priv->dai_fmt = daifmt;
+	}
+
+	/* Change direction according to format */
+	if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
+		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
+		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
+	}
+
+	of_node_put(bitclkmaster);
+	of_node_put(framemaster);
+
 	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
 		dev_err(&pdev->dev, "failed to find codec device\n");
 		ret = -EPROBE_DEFER;
@@ -601,19 +717,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
 	}
 
-	snprintf(priv->name, sizeof(priv->name), "%s-audio",
-		 fsl_asoc_card_is_ac97(priv) ? "ac97" :
-		 codec_dev->name);
-
 	/* Initialize sound card */
 	priv->pdev = pdev;
 	priv->card.dev = &pdev->dev;
-	priv->card.name = priv->name;
+	ret = snd_soc_of_parse_card_name(&priv->card, "model");
+	if (ret) {
+		snprintf(priv->name, sizeof(priv->name), "%s-audio",
+			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
+		priv->card.name = priv->name;
+	}
 	priv->card.dai_link = priv->dai_link;
-	priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
-				 audio_map_ac97 : audio_map;
 	priv->card.late_probe = fsl_asoc_card_late_probe;
-	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
 
@@ -621,13 +735,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	if (!asrc_pdev)
 		priv->card.num_dapm_routes /= 2;
 
-	memcpy(priv->dai_link, fsl_asoc_card_dai,
-	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
-
-	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
-	if (ret) {
-		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
-		goto asrc_fail;
+	if (of_property_read_bool(np, "audio-routing")) {
+		ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+		if (ret) {
+			dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+			goto asrc_fail;
+		}
 	}
 
 	/* Normal DAI Link */
@@ -704,8 +817,37 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	snd_soc_card_set_drvdata(&priv->card, priv);
 
 	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
-	if (ret && ret != -EPROBE_DEFER)
-		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+	if (ret) {
+		if (ret != -EPROBE_DEFER)
+			dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		goto asrc_fail;
+	}
+
+	/*
+	 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
+	 * asoc_simple_init_jack uses these properties for creating
+	 * Headphone Jack and Microphone Jack.
+	 *
+	 * The notifier is initialized in snd_soc_card_jack_new(), then
+	 * snd_soc_jack_notifier_register can be called.
+	 */
+	if (of_property_read_bool(np, "hp-det-gpio")) {
+		ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
+					    1, NULL, "Headphone Jack");
+		if (ret)
+			goto asrc_fail;
+
+		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
+	}
+
+	if (of_property_read_bool(np, "mic-det-gpio")) {
+		ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
+					    0, NULL, "Mic Jack");
+		if (ret)
+			goto asrc_fail;
+
+		snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
+	}
 
 asrc_fail:
 	of_node_put(asrc_np);
@@ -724,6 +866,8 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-sgtl5000", },
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ .compatible = "fsl,imx-audio-wm8960", },
+	{ .compatible = "fsl,imx-audio-mqs", },
+	{ .compatible = "fsl,imx-audio-wm8524", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 95f6a9617b0b..02c81d2e34ad 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -37,7 +37,7 @@ static struct snd_pcm_hw_constraint_list fsl_asrc_rate_constraints = {
 	.list = supported_asrc_rate,
 };
 
-/**
+/*
  * The following tables map the relationship between asrc_inclk/asrc_outclk in
  * fsl_asrc.h and the registers of ASRCSR
  */
@@ -68,7 +68,7 @@ static unsigned char output_clk_map_imx53[ASRC_CLK_MAP_LEN] = {
 	0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7, 0x7,
 };
 
-/**
+/*
  * i.MX8QM/i.MX8QXP uses the same map for input and output.
  * clk_map_imx8qm[0] is for i.MX8QM asrc0
  * clk_map_imx8qm[1] is for i.MX8QM asrc1
@@ -102,16 +102,17 @@ static unsigned char clk_map_imx8qxp[2][ASRC_CLK_MAP_LEN] = {
 };
 
 /**
- * Select the pre-processing and post-processing options
+ * fsl_asrc_sel_proc - Select the pre-processing and post-processing options
+ * @inrate: input sample rate
+ * @outrate: output sample rate
+ * @pre_proc: return value for pre-processing option
+ * @post_proc: return value for post-processing option
+ *
  * Make sure to exclude following unsupported cases before
  * calling this function:
  * 1) inrate > 8.125 * outrate
  * 2) inrate > 16.125 * outrate
  *
- * inrate: input sample rate
- * outrate: output sample rate
- * pre_proc: return value for pre-processing option
- * post_proc: return value for post-processing option
  */
 static void fsl_asrc_sel_proc(int inrate, int outrate,
 			     int *pre_proc, int *post_proc)
@@ -148,7 +149,9 @@ static void fsl_asrc_sel_proc(int inrate, int outrate,
 }
 
 /**
- * Request ASRC pair
+ * fsl_asrc_request_pair - Request ASRC pair
+ * @channels: number of channels
+ * @pair: pointer to pair
  *
  * It assigns pair by the order of A->C->B because allocation of pair B,
  * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A
@@ -193,7 +196,8 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair)
 }
 
 /**
- * Release ASRC pair
+ * fsl_asrc_release_pair - Release ASRC pair
+ * @pair: pair to release
  *
  * It clears the resource from asrc and releases the occupied channels.
  */
@@ -217,7 +221,10 @@ static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair)
 }
 
 /**
- * Configure input and output thresholds
+ * fsl_asrc_set_watermarks- configure input and output thresholds
+ * @pair: pointer to pair
+ * @in: input threshold
+ * @out: output threshold
  */
 static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out)
 {
@@ -234,7 +241,9 @@ static void fsl_asrc_set_watermarks(struct fsl_asrc_pair *pair, u32 in, u32 out)
 }
 
 /**
- * Calculate the total divisor between asrck clock rate and sample rate
+ * fsl_asrc_cal_asrck_divisor - Calculate the total divisor between asrck clock rate and sample rate
+ * @pair: pointer to pair
+ * @div: divider
  *
  * It follows the formula clk_rate = samplerate * (2 ^ prescaler) * divider
  */
@@ -250,7 +259,10 @@ static u32 fsl_asrc_cal_asrck_divisor(struct fsl_asrc_pair *pair, u32 div)
 }
 
 /**
- * Calculate and set the ratio for Ideal Ratio mode only
+ * fsl_asrc_set_ideal_ratio - Calculate and set the ratio for Ideal Ratio mode only
+ * @pair: pointer to pair
+ * @inrate: input rate
+ * @outrate: output rate
  *
  * The ratio is a 32-bit fixed point value with 26 fractional bits.
  */
@@ -293,7 +305,9 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair,
 }
 
 /**
- * Configure the assigned ASRC pair
+ * fsl_asrc_config_pair - Configure the assigned ASRC pair
+ * @pair: pointer to pair
+ * @use_ideal_rate: boolean configuration
  *
  * It configures those ASRC registers according to a configuration instance
  * of struct asrc_config which includes in/output sample rate, width, channel
@@ -508,7 +522,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate)
 }
 
 /**
- * Start the assigned ASRC pair
+ * fsl_asrc_start_pair - Start the assigned ASRC pair
+ * @pair: pointer to pair
  *
  * It enables the assigned pair and makes it stopped at the stall level.
  */
@@ -539,7 +554,8 @@ static void fsl_asrc_start_pair(struct fsl_asrc_pair *pair)
 }
 
 /**
- * Stop the assigned ASRC pair
+ * fsl_asrc_stop_pair - Stop the assigned ASRC pair
+ * @pair: pointer to pair
  */
 static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair)
 {
@@ -552,7 +568,9 @@ static void fsl_asrc_stop_pair(struct fsl_asrc_pair *pair)
 }
 
 /**
- * Get DMA channel according to the pair and direction.
+ * fsl_asrc_get_dma_channel- Get DMA channel according to the pair and direction.
+ * @pair: pointer to pair
+ * @dir: DMA direction
  */
 static struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair,
 						 bool dir)
@@ -582,11 +600,51 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream,
 			SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints);
 }
 
+/* Select proper clock source for internal ratio mode */
+static void fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv,
+				struct fsl_asrc_pair *pair,
+				int in_rate,
+				int out_rate)
+{
+	struct fsl_asrc_pair_priv *pair_priv = pair->private;
+	struct asrc_config *config = pair_priv->config;
+	int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */
+	int clk_rate, clk_index;
+	int i = 0, j = 0;
+
+	rate[IN] = in_rate;
+	rate[OUT] = out_rate;
+
+	/* Select proper clock source for internal ratio mode */
+	for (j = 0; j < 2; j++) {
+		for (i = 0; i < ASRC_CLK_MAP_LEN; i++) {
+			clk_index = asrc_priv->clk_map[j][i];
+			clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]);
+			/* Only match a perfect clock source with no remainder */
+			if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 &&
+			    (clk_rate % rate[j]) == 0)
+				break;
+		}
+
+		select_clk[j] = i;
+	}
+
+	/* Switch to ideal ratio mode if there is no proper clock source */
+	if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) {
+		select_clk[IN] = INCLK_NONE;
+		select_clk[OUT] = OUTCLK_ASRCK1_CLK;
+	}
+
+	config->inclk = select_clk[IN];
+	config->outclk = select_clk[OUT];
+}
+
 static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
 				  struct snd_pcm_hw_params *params,
 				  struct snd_soc_dai *dai)
 {
 	struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai);
+	struct fsl_asrc_priv *asrc_priv = asrc->private;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_asrc_pair *pair = runtime->private_data;
 	struct fsl_asrc_pair_priv *pair_priv = pair->private;
@@ -605,8 +663,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
 
 	config.pair = pair->index;
 	config.channel_num = channels;
-	config.inclk = INCLK_NONE;
-	config.outclk = OUTCLK_ASRCK1_CLK;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		config.input_format   = params_format(params);
@@ -620,6 +676,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,
 		config.output_sample_rate = rate;
 	}
 
+	fsl_asrc_select_clk(asrc_priv, pair,
+			    config.input_sample_rate,
+			    config.output_sample_rate);
+
 	ret = fsl_asrc_config_pair(pair, false);
 	if (ret) {
 		dev_err(dai->dev, "fail to config asrc pair\n");
@@ -854,7 +914,8 @@ static const struct regmap_config fsl_asrc_regmap_config = {
 };
 
 /**
- * Initialize ASRC registers with a default configurations
+ * fsl_asrc_init - Initialize ASRC registers with a default configuration
+ * @asrc: ASRC context
  */
 static int fsl_asrc_init(struct fsl_asrc *asrc)
 {
@@ -888,7 +949,9 @@ static int fsl_asrc_init(struct fsl_asrc *asrc)
 }
 
 /**
- * Interrupt handler for ASRC
+ * fsl_asrc_isr- Interrupt handler for ASRC
+ * @irq: irq number
+ * @dev_id: ASRC context
  */
 static irqreturn_t fsl_asrc_isr(int irq, void *dev_id)
 {
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 5f01a58f422a..29f91cdecbc3 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -129,7 +129,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component,
 				  struct snd_pcm_hw_params *params)
 {
 	enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL;
 	struct snd_dmaengine_dai_dma_data *dma_params_be = NULL;
@@ -313,7 +313,7 @@ static int fsl_asrc_dma_startup(struct snd_soc_component *component,
 				struct snd_pcm_substream *substream)
 {
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	struct device *dev = component->dev;
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index 8b9027f76d8a..a447bafa00d2 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -116,13 +116,9 @@ static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol,
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int *item = ucontrol->value.enumerated.item;
 	unsigned int reg_val, val, mix_clk;
-	int ret;
 
 	/* Get current state */
-	ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val);
-	if (ret)
-		return ret;
-
+	reg_val = snd_soc_component_read(comp, FSL_AUDMIX_CTR);
 	mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK)
 			>> FSL_AUDMIX_CTR_MIXCLK_SHIFT);
 	val = snd_soc_enum_item_to_val(e, item[0]);
@@ -162,9 +158,7 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol,
 	int ret;
 
 	/* Get current state */
-	ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, &reg_val);
-	if (ret)
-		return ret;
+	reg_val = snd_soc_component_read(comp, FSL_AUDMIX_CTR);
 
 	/* "From" state */
 	out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK)
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 13ae089c1911..be021250d6e9 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -200,7 +200,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
 {
 	struct fsl_dma_private *dma_private = dev_id;
 	struct snd_pcm_substream *substream = dma_private->substream;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct device *dev = rtd->dev;
 	struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
 	irqreturn_t ret = IRQ_NONE;
diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c
index c6b5eb2d2af7..60951a8aabd3 100644
--- a/sound/soc/fsl/fsl_easrc.c
+++ b/sound/soc/fsl/fsl_easrc.c
@@ -79,11 +79,8 @@ static int fsl_easrc_get_reg(struct snd_kcontrol *kcontrol,
 	struct soc_mreg_control *mc =
 		(struct soc_mreg_control *)kcontrol->private_value;
 	unsigned int regval;
-	int ret;
 
-	ret = snd_soc_component_read(component, mc->regbase, &regval);
-	if (ret < 0)
-		return ret;
+	regval = snd_soc_component_read(component, mc->regbase);
 
 	ucontrol->value.integer.value[0] = regval;
 
@@ -179,22 +176,21 @@ static int fsl_easrc_set_rs_ratio(struct fsl_asrc_pair *ctx)
 	struct fsl_easrc_ctx_priv *ctx_priv = ctx->private;
 	unsigned int in_rate = ctx_priv->in_params.norm_rate;
 	unsigned int out_rate = ctx_priv->out_params.norm_rate;
-	unsigned int int_bits;
 	unsigned int frac_bits;
 	u64 val;
 	u32 *r;
 
 	switch (easrc_priv->rs_num_taps) {
 	case EASRC_RS_32_TAPS:
-		int_bits = 5;
+		/* integer bits = 5; */
 		frac_bits = 39;
 		break;
 	case EASRC_RS_64_TAPS:
-		int_bits = 6;
+		/* integer bits = 6; */
 		frac_bits = 38;
 		break;
 	case EASRC_RS_128_TAPS:
-		int_bits = 7;
+		/* integer bits = 7; */
 		frac_bits = 37;
 		break;
 	default:
@@ -390,11 +386,11 @@ static int fsl_easrc_resampler_config(struct fsl_asrc *easrc)
  *  For input int[16, 24, 32] -> output float32
  *      scale it by multiplying filter coefficients by 2^-15, 2^-23, 2^-31
  *  input:
- *      asrc:  Structure pointer of fsl_asrc
- *      infilter : Pointer to non-scaled input filter
- *      shift:  The multiply factor
+ *      @easrc:  Structure pointer of fsl_asrc
+ *      @infilter : Pointer to non-scaled input filter
+ *      @shift:  The multiply factor
  *  output:
- *      outfilter: scaled filter
+ *      @outfilter: scaled filter
  */
 static int fsl_easrc_normalize_filter(struct fsl_asrc *easrc,
 				      u64 *infilter,
@@ -964,7 +960,7 @@ static int fsl_easrc_release_slot(struct fsl_asrc *easrc, unsigned int ctx_id)
  *
  * Configure the register relate with context.
  */
-int fsl_easrc_config_context(struct fsl_asrc *easrc, unsigned int ctx_id)
+static int fsl_easrc_config_context(struct fsl_asrc *easrc, unsigned int ctx_id)
 {
 	struct fsl_easrc_ctx_priv *ctx_priv;
 	struct fsl_asrc_pair *ctx;
@@ -1125,15 +1121,15 @@ static int fsl_easrc_process_format(struct fsl_asrc_pair *ctx,
 	return 0;
 }
 
-int fsl_easrc_set_ctx_format(struct fsl_asrc_pair *ctx,
-			     snd_pcm_format_t *in_raw_format,
-			     snd_pcm_format_t *out_raw_format)
+static int fsl_easrc_set_ctx_format(struct fsl_asrc_pair *ctx,
+				    snd_pcm_format_t *in_raw_format,
+				    snd_pcm_format_t *out_raw_format)
 {
 	struct fsl_asrc *easrc = ctx->asrc;
 	struct fsl_easrc_ctx_priv *ctx_priv = ctx->private;
 	struct fsl_easrc_data_fmt *in_fmt = &ctx_priv->in_params.fmt;
 	struct fsl_easrc_data_fmt *out_fmt = &ctx_priv->out_params.fmt;
-	int ret;
+	int ret = 0;
 
 	/* Get the bitfield values for input data format */
 	if (in_raw_format && out_raw_format) {
@@ -1198,10 +1194,9 @@ int fsl_easrc_set_ctx_format(struct fsl_asrc_pair *ctx,
  * to conform with this format. Interleaving parameters are accessed
  * through the ASRC_CTRL_IN_ACCESSa and ASRC_CTRL_OUT_ACCESSa registers
  */
-int fsl_easrc_set_ctx_organziation(struct fsl_asrc_pair *ctx)
+static int fsl_easrc_set_ctx_organziation(struct fsl_asrc_pair *ctx)
 {
 	struct fsl_easrc_ctx_priv *ctx_priv;
-	struct device *dev;
 	struct fsl_asrc *easrc;
 
 	if (!ctx)
@@ -1209,7 +1204,6 @@ int fsl_easrc_set_ctx_organziation(struct fsl_asrc_pair *ctx)
 
 	easrc = ctx->asrc;
 	ctx_priv = ctx->private;
-	dev = &easrc->pdev->dev;
 
 	/* input interleaving parameters */
 	regmap_update_bits(easrc->regmap, REG_EASRC_CIA(ctx->index),
@@ -1242,7 +1236,7 @@ int fsl_easrc_set_ctx_organziation(struct fsl_asrc_pair *ctx)
  * Returns a negative number on error and >=0 as context id
  * on success
  */
-int fsl_easrc_request_context(int channels, struct fsl_asrc_pair *ctx)
+static int fsl_easrc_request_context(int channels, struct fsl_asrc_pair *ctx)
 {
 	enum asrc_pair_index index = ASRC_INVALID_PAIR;
 	struct fsl_asrc *easrc = ctx->asrc;
@@ -1287,17 +1281,15 @@ int fsl_easrc_request_context(int channels, struct fsl_asrc_pair *ctx)
  *
  * This funciton is mainly doing the revert thing in request context
  */
-void fsl_easrc_release_context(struct fsl_asrc_pair *ctx)
+static void fsl_easrc_release_context(struct fsl_asrc_pair *ctx)
 {
 	unsigned long lock_flags;
 	struct fsl_asrc *easrc;
-	struct device *dev;
 
 	if (!ctx)
 		return;
 
 	easrc = ctx->asrc;
-	dev = &easrc->pdev->dev;
 
 	spin_lock_irqsave(&easrc->lock, lock_flags);
 
@@ -1314,7 +1306,7 @@ void fsl_easrc_release_context(struct fsl_asrc_pair *ctx)
  *
  * Enable the DMA request and context
  */
-int fsl_easrc_start_context(struct fsl_asrc_pair *ctx)
+static int fsl_easrc_start_context(struct fsl_asrc_pair *ctx)
 {
 	struct fsl_asrc *easrc = ctx->asrc;
 
@@ -1332,7 +1324,7 @@ int fsl_easrc_start_context(struct fsl_asrc_pair *ctx)
  *
  * Disable the DMA request and context
  */
-int fsl_easrc_stop_context(struct fsl_asrc_pair *ctx)
+static int fsl_easrc_stop_context(struct fsl_asrc_pair *ctx)
 {
 	struct fsl_asrc *easrc = ctx->asrc;
 	int val, i;
@@ -1379,8 +1371,8 @@ int fsl_easrc_stop_context(struct fsl_asrc_pair *ctx)
 	return 0;
 }
 
-struct dma_chan *fsl_easrc_get_dma_channel(struct fsl_asrc_pair *ctx,
-					   bool dir)
+static struct dma_chan *fsl_easrc_get_dma_channel(struct fsl_asrc_pair *ctx,
+						  bool dir)
 {
 	struct fsl_asrc *easrc = ctx->asrc;
 	enum asrc_pair_index index = ctx->index;
@@ -1391,7 +1383,6 @@ struct dma_chan *fsl_easrc_get_dma_channel(struct fsl_asrc_pair *ctx,
 
 	return dma_request_slave_channel(&easrc->pdev->dev, name);
 };
-EXPORT_SYMBOL_GPL(fsl_easrc_get_dma_channel);
 
 static const unsigned int easrc_rates[] = {
 	8000, 11025, 12000, 16000,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index cbcb70d6f8c8..4ae36099ae82 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -22,8 +22,7 @@
 				SNDRV_PCM_FMTBIT_S24_LE)
 
 /**
- * fsl_esai_soc_data: soc specific data
- *
+ * struct fsl_esai_soc_data - soc specific data
  * @imx: for imx platform
  * @reset_at_xrun: flags for enable reset operaton
  */
@@ -33,8 +32,7 @@ struct fsl_esai_soc_data {
 };
 
 /**
- * fsl_esai: ESAI private data
- *
+ * struct fsl_esai - ESAI private data
  * @dma_params_rx: DMA parameters for receive channel
  * @dma_params_tx: DMA parameters for transmit channel
  * @pdev: platform device pointer
@@ -49,6 +47,8 @@ struct fsl_esai_soc_data {
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
  * @slots: number of slots
+ * @tx_mask: slot mask for TX
+ * @rx_mask: slot mask for RX
  * @channels: channel num for tx or rx
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
@@ -157,13 +157,15 @@ static irqreturn_t esai_isr(int irq, void *devid)
 }
 
 /**
- * This function is used to calculate the divisors of psr, pm, fp and it is
- * supposed to be called in set_dai_sysclk() and set_bclk().
+ * fsl_esai_divisor_cal - This function is used to calculate the
+ * divisors of psr, pm, fp and it is supposed to be called in
+ * set_dai_sysclk() and set_bclk().
  *
+ * @dai: pointer to DAI
+ * @tx: current setting is for playback or capture
  * @ratio: desired overall ratio for the paticipating dividers
  * @usefp: for HCK setting, there is no need to set fp divider
  * @fp: bypass other dividers by setting fp directly if fp != 0
- * @tx: current setting is for playback or capture
  */
 static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
 				bool usefp, u32 fp)
@@ -250,13 +252,12 @@ out_fp:
 }
 
 /**
- * This function mainly configures the clock frequency of MCLK (HCKT/HCKR)
- *
- * @Parameters:
- * clk_id: The clock source of HCKT/HCKR
+ * fsl_esai_set_dai_sysclk - configure the clock frequency of MCLK (HCKT/HCKR)
+ * @dai: pointer to DAI
+ * @clk_id: The clock source of HCKT/HCKR
  *	  (Input from outside; output from inside, FSYS or EXTAL)
- * freq: The required clock rate of HCKT/HCKR
- * dir: The clock direction of HCKT/HCKR
+ * @freq: The required clock rate of HCKT/HCKR
+ * @dir: The clock direction of HCKT/HCKR
  *
  * Note: If the direction is input, we do not care about clk_id.
  */
@@ -358,7 +359,10 @@ out:
 }
 
 /**
- * This function configures the related dividers according to the bclk rate
+ * fsl_esai_set_bclk - configure the related dividers according to the bclk rate
+ * @dai: pointer to DAI
+ * @tx: direction boolean
+ * @freq: bclk freq
  */
 static int fsl_esai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 {
@@ -1008,7 +1012,7 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	if (irq < 0)
 		return irq;
 
-	ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0,
+	ret = devm_request_irq(&pdev->dev, irq, esai_isr, IRQF_SHARED,
 			       esai_priv->name, esai_priv);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to claim irq %u\n", irq);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 9d436b0c5718..cdff739924e2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -680,10 +680,11 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
 	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
 
 	regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
-			   FSL_SAI_CR1_RFW_MASK,
+			   FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
 			   sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
 	regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
-			   FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
+			   FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+			   FSL_SAI_MAXBURST_RX - 1);
 
 	snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
 				&sai->dma_params_rx);
@@ -1016,6 +1017,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, sai);
 
 	pm_runtime_enable(&pdev->dev);
+	regcache_cache_only(sai->regmap, true);
 
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
 			&fsl_sai_dai, 1);
@@ -1107,7 +1109,6 @@ static int fsl_sai_runtime_suspend(struct device *dev)
 	clk_disable_unprepare(sai->bus_clk);
 
 	regcache_cache_only(sai->regmap, true);
-	regcache_mark_dirty(sai->regmap);
 
 	return 0;
 }
@@ -1137,6 +1138,7 @@ static int fsl_sai_runtime_resume(struct device *dev)
 	}
 
 	regcache_cache_only(sai->regmap, false);
+	regcache_mark_dirty(sai->regmap);
 	regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR);
 	regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR);
 	usleep_range(1000, 2000);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 76b15deea80c..6aba7d28f5f3 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -94,7 +94,7 @@
 #define FSL_SAI_CSR_FRDE	BIT(0)
 
 /* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK	0x1f
+#define FSL_SAI_CR1_RFW_MASK(x)	((x) - 1)
 
 /* SAI Transmit and Receive Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 1b2e516f9162..455f96908377 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -16,6 +16,7 @@
 #include <linux/of_device.h>
 #include <linux/of_irq.h>
 #include <linux/regmap.h>
+#include <linux/pm_runtime.h>
 
 #include <sound/asoundef.h>
 #include <sound/dmaengine_pcm.h>
@@ -42,6 +43,18 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb };
 
 #define DEFAULT_RXCLK_SRC	1
 
+/**
+ * struct fsl_spdif_soc_data: soc specific data
+ *
+ * @imx: for imx platform
+ * @shared_root_clock: flag of sharing a clock source with others;
+ *                     so the driver shouldn't set root clock rate
+ */
+struct fsl_spdif_soc_data {
+	bool imx;
+	bool shared_root_clock;
+};
+
 /*
  * SPDIF control structure
  * Defines channel status, subcode and Q sub
@@ -68,8 +81,8 @@ struct spdif_mixer_control {
 };
 
 /**
- * fsl_spdif_priv: Freescale SPDIF private data
- *
+ * struct fsl_spdif_priv - Freescale SPDIF private data
+ * @soc: SPDIF soc data
  * @fsl_spdif_control: SPDIF control data
  * @cpu_dai_drv: cpu dai driver
  * @pdev: platform device pointer
@@ -87,8 +100,10 @@ struct spdif_mixer_control {
  * @spbaclk: SPBA clock (optional, depending on SoC design)
  * @dma_params_tx: DMA parameters for transmit channel
  * @dma_params_rx: DMA parameters for receive channel
+ * @regcache_srpc: regcache for SRPC
  */
 struct fsl_spdif_priv {
+	const struct fsl_spdif_soc_data *soc;
 	struct spdif_mixer_control fsl_spdif_control;
 	struct snd_soc_dai_driver cpu_dai_drv;
 	struct platform_device *pdev;
@@ -110,6 +125,27 @@ struct fsl_spdif_priv {
 	u32 regcache_srpc;
 };
 
+static struct fsl_spdif_soc_data fsl_spdif_vf610 = {
+	.imx = false,
+	.shared_root_clock = false,
+};
+
+static struct fsl_spdif_soc_data fsl_spdif_imx35 = {
+	.imx = true,
+	.shared_root_clock = false,
+};
+
+static struct fsl_spdif_soc_data fsl_spdif_imx6sx = {
+	.imx = true,
+	.shared_root_clock = true,
+};
+
+/* Check if clk is a root clock that does not share clock source with others */
+static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk)
+{
+	return (clk == STC_TXCLK_SPDIF_ROOT) && !spdif->soc->shared_root_clock;
+}
+
 /* DPLL locked and lock loss interrupt handler */
 static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv)
 {
@@ -369,7 +405,7 @@ static int spdif_set_rx_clksrc(struct fsl_spdif_priv *spdif_priv,
 static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 				int sample_rate)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	struct regmap *regmap = spdif_priv->regmap;
@@ -420,8 +456,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 
 	sysclk_df = spdif_priv->sysclk_df[rate];
 
-	/* Don't mess up the clocks from other modules */
-	if (clk != STC_TXCLK_SPDIF_ROOT)
+	if (!fsl_spdif_can_set_clk_rate(spdif_priv, clk))
 		goto clk_set_bypass;
 
 	/* The S/PDIF block needs a clock of 64 * fs * txclk_df */
@@ -457,34 +492,19 @@ clk_set_bypass:
 static int fsl_spdif_startup(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct platform_device *pdev = spdif_priv->pdev;
 	struct regmap *regmap = spdif_priv->regmap;
 	u32 scr, mask;
-	int i;
 	int ret;
 
 	/* Reset module and interrupts only for first initialization */
 	if (!snd_soc_dai_active(cpu_dai)) {
-		ret = clk_prepare_enable(spdif_priv->coreclk);
-		if (ret) {
-			dev_err(&pdev->dev, "failed to enable core clock\n");
-			return ret;
-		}
-
-		if (!IS_ERR(spdif_priv->spbaclk)) {
-			ret = clk_prepare_enable(spdif_priv->spbaclk);
-			if (ret) {
-				dev_err(&pdev->dev, "failed to enable spba clock\n");
-				goto err_spbaclk;
-			}
-		}
-
 		ret = spdif_softreset(spdif_priv);
 		if (ret) {
 			dev_err(&pdev->dev, "failed to soft reset\n");
-			goto err;
+			return ret;
 		}
 
 		/* Disable all the interrupts */
@@ -498,18 +518,10 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream,
 		mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
 			SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
 			SCR_TXFIFO_FSEL_MASK;
-		for (i = 0; i < SPDIF_TXRATE_MAX; i++) {
-			ret = clk_prepare_enable(spdif_priv->txclk[i]);
-			if (ret)
-				goto disable_txclk;
-		}
 	} else {
 		scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC;
 		mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
 			SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
-		ret = clk_prepare_enable(spdif_priv->rxclk);
-		if (ret)
-			goto err;
 	}
 	regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
 
@@ -517,39 +529,25 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream,
 	regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0);
 
 	return 0;
-
-disable_txclk:
-	for (i--; i >= 0; i--)
-		clk_disable_unprepare(spdif_priv->txclk[i]);
-err:
-	if (!IS_ERR(spdif_priv->spbaclk))
-		clk_disable_unprepare(spdif_priv->spbaclk);
-err_spbaclk:
-	clk_disable_unprepare(spdif_priv->coreclk);
-
-	return ret;
 }
 
 static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct regmap *regmap = spdif_priv->regmap;
-	u32 scr, mask, i;
+	u32 scr, mask;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		scr = 0;
 		mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK |
 			SCR_TXSEL_MASK | SCR_USRC_SEL_MASK |
 			SCR_TXFIFO_FSEL_MASK;
-		for (i = 0; i < SPDIF_TXRATE_MAX; i++)
-			clk_disable_unprepare(spdif_priv->txclk[i]);
 	} else {
 		scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO;
 		mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK|
 			SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK;
-		clk_disable_unprepare(spdif_priv->rxclk);
 	}
 	regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr);
 
@@ -558,9 +556,6 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream,
 		spdif_intr_status_clear(spdif_priv);
 		regmap_update_bits(regmap, REG_SPDIF_SCR,
 				SCR_LOW_POWER, SCR_LOW_POWER);
-		if (!IS_ERR(spdif_priv->spbaclk))
-			clk_disable_unprepare(spdif_priv->spbaclk);
-		clk_disable_unprepare(spdif_priv->coreclk);
 	}
 }
 
@@ -568,7 +563,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control;
 	struct platform_device *pdev = spdif_priv->pdev;
@@ -596,7 +591,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream,
 static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct regmap *regmap = spdif_priv->regmap;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
@@ -781,8 +776,8 @@ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol,
 }
 
 /* Get valid good bit from interrupt status register */
-static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_value *ucontrol)
+static int fsl_spdif_rx_vbit_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
@@ -796,6 +791,35 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+static int fsl_spdif_tx_vbit_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val;
+
+	regmap_read(regmap, REG_SPDIF_SCR, &val);
+	val = (val & SCR_VAL_MASK) >> SCR_VAL_OFFSET;
+	val = 1 - val;
+	ucontrol->value.integer.value[0] = val;
+
+	return 0;
+}
+
+static int fsl_spdif_tx_vbit_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct regmap *regmap = spdif_priv->regmap;
+	u32 val = (1 - ucontrol->value.integer.value[0]) << SCR_VAL_OFFSET;
+
+	regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_VAL_MASK, val);
+
+	return 0;
+}
+
 /* DPLL lock information */
 static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_info *uinfo)
@@ -953,11 +977,21 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = {
 	/* Valid bit error controller */
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
-		.name = "IEC958 V-Bit Errors",
+		.name = "IEC958 RX V-Bit Errors",
 		.access = SNDRV_CTL_ELEM_ACCESS_READ |
 			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 		.info = fsl_spdif_vbit_info,
-		.get = fsl_spdif_vbit_get,
+		.get = fsl_spdif_rx_vbit_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = "IEC958 TX V-Bit",
+		.access = SNDRV_CTL_ELEM_ACCESS_READ |
+			SNDRV_CTL_ELEM_ACCESS_WRITE |
+			SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+		.info = fsl_spdif_vbit_info,
+		.get = fsl_spdif_tx_vbit_get,
+		.put = fsl_spdif_tx_vbit_put,
 	},
 	/* DPLL lock info get controller */
 	{
@@ -990,6 +1024,10 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 
 	snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
 
+	/*Clear the val bit for Tx*/
+	regmap_update_bits(spdif_private->regmap, REG_SPDIF_SCR,
+			   SCR_VAL_MASK, SCR_VAL_CLEAR);
+
 	return 0;
 }
 
@@ -1186,7 +1224,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			continue;
 
 		ret = fsl_spdif_txclk_caldiv(spdif_priv, clk, savesub, index,
-					     i == STC_TXCLK_SPDIF_ROOT);
+					     fsl_spdif_can_set_clk_rate(spdif_priv, i));
 		if (savesub == ret)
 			continue;
 
@@ -1230,6 +1268,12 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 
 	spdif_priv->pdev = pdev;
 
+	spdif_priv->soc = of_device_get_match_data(&pdev->dev);
+	if (!spdif_priv->soc) {
+		dev_err(&pdev->dev, "failed to get soc data\n");
+		return -ENODEV;
+	}
+
 	/* Initialize this copy of the CPU DAI driver structure */
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
 	spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev);
@@ -1311,6 +1355,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 
 	/* Register with ASoC */
 	dev_set_drvdata(&pdev->dev, spdif_priv);
+	pm_runtime_enable(&pdev->dev);
+	regcache_cache_only(spdif_priv->regmap, true);
 
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
 					      &spdif_priv->cpu_dai_drv, 1);
@@ -1326,41 +1372,96 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	return ret;
 }
 
-#ifdef CONFIG_PM_SLEEP
-static int fsl_spdif_suspend(struct device *dev)
+#ifdef CONFIG_PM
+static int fsl_spdif_runtime_suspend(struct device *dev)
 {
 	struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+	int i;
 
 	regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC,
 			&spdif_priv->regcache_srpc);
-
 	regcache_cache_only(spdif_priv->regmap, true);
-	regcache_mark_dirty(spdif_priv->regmap);
+
+	clk_disable_unprepare(spdif_priv->rxclk);
+
+	for (i = 0; i < SPDIF_TXRATE_MAX; i++)
+		clk_disable_unprepare(spdif_priv->txclk[i]);
+
+	if (!IS_ERR(spdif_priv->spbaclk))
+		clk_disable_unprepare(spdif_priv->spbaclk);
+	clk_disable_unprepare(spdif_priv->coreclk);
 
 	return 0;
 }
 
-static int fsl_spdif_resume(struct device *dev)
+static int fsl_spdif_runtime_resume(struct device *dev)
 {
 	struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev);
+	int ret;
+	int i;
+
+	ret = clk_prepare_enable(spdif_priv->coreclk);
+	if (ret) {
+		dev_err(dev, "failed to enable core clock\n");
+		return ret;
+	}
+
+	if (!IS_ERR(spdif_priv->spbaclk)) {
+		ret = clk_prepare_enable(spdif_priv->spbaclk);
+		if (ret) {
+			dev_err(dev, "failed to enable spba clock\n");
+			goto disable_core_clk;
+		}
+	}
+
+	for (i = 0; i < SPDIF_TXRATE_MAX; i++) {
+		ret = clk_prepare_enable(spdif_priv->txclk[i]);
+		if (ret)
+			goto disable_tx_clk;
+	}
+
+	ret = clk_prepare_enable(spdif_priv->rxclk);
+	if (ret)
+		goto disable_tx_clk;
 
 	regcache_cache_only(spdif_priv->regmap, false);
+	regcache_mark_dirty(spdif_priv->regmap);
 
 	regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC,
 			SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK,
 			spdif_priv->regcache_srpc);
 
-	return regcache_sync(spdif_priv->regmap);
+	ret = regcache_sync(spdif_priv->regmap);
+	if (ret)
+		goto disable_rx_clk;
+
+	return 0;
+
+disable_rx_clk:
+	clk_disable_unprepare(spdif_priv->rxclk);
+disable_tx_clk:
+	for (i--; i >= 0; i--)
+		clk_disable_unprepare(spdif_priv->txclk[i]);
+	if (!IS_ERR(spdif_priv->spbaclk))
+		clk_disable_unprepare(spdif_priv->spbaclk);
+disable_core_clk:
+	clk_disable_unprepare(spdif_priv->coreclk);
+
+	return ret;
 }
-#endif /* CONFIG_PM_SLEEP */
+#endif /* CONFIG_PM */
 
 static const struct dev_pm_ops fsl_spdif_pm = {
-	SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+				pm_runtime_force_resume)
+	SET_RUNTIME_PM_OPS(fsl_spdif_runtime_suspend, fsl_spdif_runtime_resume,
+			   NULL)
 };
 
 static const struct of_device_id fsl_spdif_dt_ids[] = {
-	{ .compatible = "fsl,imx35-spdif", },
-	{ .compatible = "fsl,vf610-spdif", },
+	{ .compatible = "fsl,imx35-spdif", .data = &fsl_spdif_imx35, },
+	{ .compatible = "fsl,vf610-spdif", .data = &fsl_spdif_vf610, },
+	{ .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, },
 	{}
 };
 MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 1a2fa7f18142..d8b9c6547142 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -203,12 +203,10 @@ struct fsl_ssi_soc_data {
 };
 
 /**
- * fsl_ssi: per-SSI private data
- *
+ * struct fsl_ssi - per-SSI private data
  * @regs: Pointer to the regmap registers
  * @irq: IRQ of this SSI
  * @cpu_dai_drv: CPU DAI driver for this device
- *
  * @dai_fmt: DAI configuration this device is currently used with
  * @streams: Mask of current active streams: BIT(TX) and BIT(RX)
  * @i2s_net: I2S and Network mode configurations of SCR register
@@ -221,38 +219,29 @@ struct fsl_ssi_soc_data {
  * @slot_width: Width of each DAI slot
  * @slots: Number of slots
  * @regvals: Specific RX/TX register settings
- *
  * @clk: Clock source to access register
  * @baudclk: Clock source to generate bit and frame-sync clocks
  * @baudclk_streams: Active streams that are using baudclk
- *
  * @regcache_sfcsr: Cache sfcsr register value during suspend and resume
  * @regcache_sacnt: Cache sacnt register value during suspend and resume
- *
  * @dma_params_tx: DMA transmit parameters
  * @dma_params_rx: DMA receive parameters
  * @ssi_phys: physical address of the SSI registers
- *
  * @fiq_params: FIQ stream filtering parameters
- *
  * @card_pdev: Platform_device pointer to register a sound card for PowerPC or
  *             to register a CODEC platform device for AC97
  * @card_name: Platform_device name to register a sound card for PowerPC or
  *             to register a CODEC platform device for AC97
  * @card_idx: The index of SSI to register a sound card for PowerPC or
  *            to register a CODEC platform device for AC97
- *
  * @dbg_stats: Debugging statistics
- *
  * @soc: SoC specific data
  * @dev: Pointer to &pdev->dev
- *
  * @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.
- *
  * @ac97_reg_lock: Mutex lock to serialize AC97 register access operations
  */
 struct fsl_ssi {
@@ -374,7 +363,9 @@ static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi *ssi)
 }
 
 /**
- * Interrupt handler to gather states
+ * fsl_ssi_irq - Interrupt handler to gather states
+ * @irq: irq number
+ * @dev_id: context
  */
 static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 {
@@ -395,7 +386,10 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
 }
 
 /**
- * Set SCR, SIER, STCR and SRCR registers with cached values in regvals
+ * fsl_ssi_config_enable - Set SCR, SIER, STCR and SRCR registers with
+ * cached values in regvals
+ * @ssi: SSI context
+ * @tx: direction
  *
  * Notes:
  * 1) For offline_config SoCs, enable all necessary bits of both streams
@@ -474,7 +468,7 @@ enable_scr:
 	ssi->streams |= BIT(dir);
 }
 
-/**
+/*
  * Exclude bits that are used by the opposite stream
  *
  * When both streams are active, disabling some bits for the current stream
@@ -495,7 +489,10 @@ enable_scr:
 	((vals) & _ssi_xor_shared_bits(vals, avals, aactive))
 
 /**
- * Unset SCR, SIER, STCR and SRCR registers with cached values in regvals
+ * fsl_ssi_config_disable - Unset SCR, SIER, STCR and SRCR registers
+ * with cached values in regvals
+ * @ssi: SSI context
+ * @tx: direction
  *
  * Notes:
  * 1) For offline_config SoCs, to avoid online reconfigurations, disable all
@@ -577,7 +574,9 @@ static void fsl_ssi_tx_ac97_saccst_setup(struct fsl_ssi *ssi)
 }
 
 /**
- * Cache critical bits of SIER, SRCR, STCR and SCR to later set them safely
+ * fsl_ssi_setup_regvals - Cache critical bits of SIER, SRCR, STCR and
+ * SCR to later set them safely
+ * @ssi: SSI context
  */
 static void fsl_ssi_setup_regvals(struct fsl_ssi *ssi)
 {
@@ -630,7 +629,7 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi *ssi)
 static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	int ret;
 
@@ -654,16 +653,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	clk_disable_unprepare(ssi->clk);
 }
 
 /**
- * Configure Digital Audio Interface bit clock
+ * fsl_ssi_set_bclk - Configure Digital Audio Interface bit clock
+ * @substream: ASoC substream
+ * @dai: pointer to DAI
+ * @hw_params: pointers to hw_params
  *
- * Note: This function can be only called when using SSI as DAI master
+ * Notes: This function can be only called when using SSI as DAI master
  *
  * Quick instruction for parameters:
  * freq: Output BCLK frequency = samplerate * slots * slot_width
@@ -782,7 +784,10 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 }
 
 /**
- * Configure SSI based on PCM hardware parameters
+ * fsl_ssi_hw_params - Configure SSI based on PCM hardware parameters
+ * @substream: ASoC substream
+ * @hw_params: pointers to hw_params
+ * @dai: pointer to DAI
  *
  * Notes:
  * 1) SxCCR.WL bits are critical bits that require SSI to be temporarily
@@ -858,7 +863,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 
 	if (fsl_ssi_is_i2s_master(ssi) &&
@@ -997,7 +1002,9 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
 }
 
 /**
- * Configure Digital Audio Interface (DAI) Format
+ * fsl_ssi_set_dai_fmt - Configure Digital Audio Interface (DAI) Format
+ * @dai: pointer to DAI
+ * @fmt: format mask
  */
 static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
@@ -1011,7 +1018,12 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 }
 
 /**
- * Set TDM slot number and slot width
+ * fsl_ssi_set_dai_tdm_slot - Set TDM slot number and slot width
+ * @dai: pointer to DAI
+ * @tx_mask: mask for TX
+ * @rx_mask: mask for RX
+ * @slots: number of slots
+ * @slot_width: number of bits per slot
  */
 static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 				    u32 rx_mask, int slots, int slot_width)
@@ -1055,7 +1067,10 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 }
 
 /**
- * Start or stop SSI and corresponding DMA transaction.
+ * fsl_ssi_trigger - Start or stop SSI and corresponding DMA transaction.
+ * @substream: ASoC substream
+ * @cmd: trigger command
+ * @dai: pointer to DAI
  *
  * The DMA channel is in external master start and pause mode, which
  * means the SSI completely controls the flow of data.
@@ -1063,7 +1078,7 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 			   struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
@@ -1239,7 +1254,8 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
 };
 
 /**
- * Initialize SSI registers
+ * fsl_ssi_hw_init - Initialize SSI registers
+ * @ssi: SSI context
  */
 static int fsl_ssi_hw_init(struct fsl_ssi *ssi)
 {
@@ -1268,7 +1284,8 @@ static int fsl_ssi_hw_init(struct fsl_ssi *ssi)
 }
 
 /**
- * Clear SSI registers
+ * fsl_ssi_hw_clean - Clear SSI registers
+ * @ssi: SSI context
  */
 static void fsl_ssi_hw_clean(struct fsl_ssi *ssi)
 {
@@ -1285,7 +1302,8 @@ static void fsl_ssi_hw_clean(struct fsl_ssi *ssi)
 		regmap_update_bits(ssi->regs, REG_SSI_SCR, SSI_SCR_SSIEN, 0);
 	}
 }
-/**
+
+/*
  * Make every character in a string lower-case
  */
 static void make_lowercase(char *s)
diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c
index 2a20ee23dc52..2c46c55f0a88 100644
--- a/sound/soc/fsl/fsl_ssi_dbg.c
+++ b/sound/soc/fsl/fsl_ssi_dbg.c
@@ -78,7 +78,7 @@ void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *dbg, u32 sisr)
 		dbg->stats.tfe0++;
 }
 
-/**
+/*
  * Show the statistics of a flag only if its interrupt is enabled
  *
  * Compilers will optimize it to a no-op if the interrupt is disabled
@@ -90,7 +90,7 @@ void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *dbg, u32 sisr)
 	} while (0)
 
 
-/**
+/*
  * Display the statistics for the current SSI device
  *
  * To avoid confusion, only show those counts that are enabled
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index e09b45de0efd..202fb8950078 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -6,8 +6,8 @@
  * License. You may obtain a copy of the GNU General Public License
  * Version 2 or later at the following locations:
  *
- * http://www.opensource.org/licenses/gpl-license.html
- * http://www.gnu.org/copyleft/gpl.html
+ * https://www.opensource.org/licenses/gpl-license.html
+ * https://www.gnu.org/copyleft/gpl.html
  */
 
 #include <linux/module.h>
@@ -44,7 +44,7 @@ static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = {
 
 static int imx_audmix_fe_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct device *dev = rtd->card->dev;
@@ -73,7 +73,7 @@ static int imx_audmix_fe_startup(struct snd_pcm_substream *substream)
 static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct device *dev = rtd->card->dev;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
@@ -112,7 +112,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream,
 static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct device *dev = rtd->card->dev;
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF;
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 3ce85a43e08f..25c18b9e348f 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -5,7 +5,7 @@
 // Copyright 2009 Pengutronix, Sascha Hauer <s.hauer@pengutronix.de>
 //
 // Initial development of this code was funded by
-// Phytec Messtechnik GmbH, http://www.phytec.de
+// Phytec Messtechnik GmbH, https://www.phytec.de
 
 #include <linux/clk.h>
 #include <linux/debugfs.h>
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index fab2d6c56653..dd9c1ac81cf5 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -26,7 +26,7 @@
 static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	int ret;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 3b8c796d7829..9e4f66b6b92b 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -114,7 +114,7 @@ static int psc_dma_hw_free(struct snd_soc_component *component,
 static int psc_dma_trigger(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream, int cmd)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
@@ -216,7 +216,7 @@ static int psc_dma_open(struct snd_soc_component *component,
 			struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 	int rc;
@@ -244,7 +244,7 @@ static int psc_dma_open(struct snd_soc_component *component,
 static int psc_dma_close(struct snd_soc_component *component,
 			 struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 
@@ -270,7 +270,7 @@ static snd_pcm_uframes_t
 psc_dma_pointer(struct snd_soc_component *component,
 		struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	struct psc_dma_stream *s;
 	dma_addr_t count;
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 1ab4fbda08cb..3149d59ae968 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -38,7 +38,7 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
 	u32 mode;
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index f7bd90051ce7..eccc833390d4 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -98,7 +98,7 @@ static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card)
  */
 static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct mpc8610_hpcd_data *machine_data =
 		container_of(rtd->card, struct mpc8610_hpcd_data, card);
 	struct device *dev = rtd->card->dev;
@@ -426,9 +426,11 @@ static int __init mpc8610_hpcd_init(void)
 	guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
 	if (of_address_to_resource(guts_np, 0, &res)) {
 		pr_err("mpc8610-hpcd: missing/invalid global utilities node\n");
+		of_node_put(guts_np);
 		return -EINVAL;
 	}
 	guts_phys = res.start;
+	of_node_put(guts_np);
 
 	return platform_driver_register(&mpc8610_hpcd_driver);
 }
diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c
index a36d4e8cd55c..4ead537e090a 100644
--- a/sound/soc/fsl/mx27vis-aic32x4.c
+++ b/sound/soc/fsl/mx27vis-aic32x4.c
@@ -36,7 +36,7 @@ static int mx27vis_amp_muter_gpio;
 static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int ret;
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index fe3091590f20..ac68d2238045 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -121,7 +121,7 @@ static int p1022_ds_machine_probe(struct snd_soc_card *card)
  */
 static int p1022_ds_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct machine_data *mdata =
 		container_of(rtd->card, struct machine_data, card);
 	struct device *dev = rtd->card->dev;
diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c
index f5374fe354ab..714515b8081f 100644
--- a/sound/soc/fsl/p1022_rdk.c
+++ b/sound/soc/fsl/p1022_rdk.c
@@ -127,7 +127,7 @@ static int p1022_rdk_machine_probe(struct snd_soc_card *card)
  */
 static int p1022_rdk_startup(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct machine_data *mdata =
 		container_of(rtd->card, struct machine_data, card);
 	struct device *dev = rtd->card->dev;
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index 8b1551c55452..99611a037ada 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -75,7 +75,7 @@ static const struct _wm8350_audio wm8350_audio[] = {
 static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int i, found = 0;