Merge remote-tracking branch 'asoc/for-5.9' into asoc-next

23 files changed, 547 insertions, 228 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_pai