Merge remote-tracking branches 'asoc/topic/es8328', 'asoc/topic/find-dai', 'asoc/topic/fsl', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-ssi' into asoc-next

Documentation/devicetree/bindings/sound/fsl-sai.txt

@@ -7,8 +7,8 @@ codec/DSP interfaces.
 
 Required properties:
 
-  - compatible		: Compatible list, contains "fsl,vf610-sai" or
+  - compatible		: Compatible list, contains "fsl,vf610-sai",
-			  "fsl,imx6sx-sai".
+			  "fsl,imx6sx-sai" or "fsl,imx6ul-sai"
 
   - reg			: Offset and length of the register set for the device.
 
@@ -48,6 +48,11 @@ Required properties:
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
+Optional properties (for mx6ul):
+
+  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+			 indicates that SAI will output the SAI MCLK clock.
+
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
   default synchronous mode (sync Rx with Tx) will be used, which means both

MAINTAINERS

@@ -4661,6 +4661,7 @@ FREESCALE SOC SOUND DRIVERS
 M:	Timur Tabi <timur@tabi.org>
 M:	Nicolin Chen <nicoleotsuka@gmail.com>
 M:	Xiubo Li <Xiubo.Lee@gmail.com>
+R:	Fabio Estevam <fabio.estevam@nxp.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	linuxppc-dev@lists.ozlabs.org
 S:	Maintained

include/linux/mfd/syscon/imx6q-iomuxc-gpr.h

@@ -447,5 +447,11 @@
 #define IMX6UL_GPR1_ENET2_CLK_OUTPUT		(0x1 << 18)
 #define IMX6UL_GPR1_ENET_CLK_DIR		(0x3 << 17)
 #define IMX6UL_GPR1_ENET_CLK_OUTPUT		(0x3 << 17)
+#define IMX6UL_GPR1_SAI1_MCLK_DIR		(0x1 << 19)
+#define IMX6UL_GPR1_SAI2_MCLK_DIR		(0x1 << 20)
+#define IMX6UL_GPR1_SAI3_MCLK_DIR		(0x1 << 21)
+#define IMX6UL_GPR1_SAI_MCLK_MASK		(0x7 << 19)
+#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \
+		     IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR)
 
 #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */

include/sound/soc.h

@@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 int snd_soc_register_dai(struct snd_soc_component *component,
 	struct snd_soc_dai_driver *dai_drv);
 
+struct snd_soc_dai *snd_soc_find_dai(
+	const struct snd_soc_dai_link_component *dlc);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS

sound/soc/codecs/es8328.c

@@ -26,18 +26,30 @@
 #include <sound/tlv.h>
 #include "es8328.h"
 
-#define ES8328_SYSCLK_RATE_1X 11289600
+static const unsigned int rates_12288[] = {
-#define ES8328_SYSCLK_RATE_2X 22579200
+	8000, 12000, 16000, 24000, 32000, 48000, 96000,
+};
 
-/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
+static const int ratios_12288[] = {
-static struct {
+	10, 7, 6, 4, 3, 2, 0,
-	int rate;
+};
-	u8 ratio;
+
-} mclk_ratios[] = {
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
-	{ 8000, 9 },
+	.count	= ARRAY_SIZE(rates_12288),
-	{11025, 7 },
+	.list	= rates_12288,
-	{22050, 4 },
+};
-	{44100, 2 },
+
+static const unsigned int rates_11289[] = {
+	8018, 11025, 22050, 44100, 88200,
+};
+
+static const int ratios_11289[] = {
+	9, 7, 4, 2, 0,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
+	.count	= ARRAY_SIZE(rates_11289),
+	.list	= rates_11289,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = {
 	"HPVDD",
 };
 
-#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \
+		SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_32000 | \
 		SNDRV_PCM_RATE_22050 | \
-		SNDRV_PCM_RATE_11025)
+		SNDRV_PCM_RATE_16000 | \
-#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+		SNDRV_PCM_RATE_11025 | \
+		SNDRV_PCM_RATE_8000)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | \
+		SNDRV_PCM_FMTBIT_S20_3LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
 
 struct es8328_priv {
 	struct regmap *regmap;
 	struct clk *clk;
 	int playback_fs;
 	bool deemph;
+	int mclkdiv2;
+	const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	const int *mclk_ratios;
 	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
 };
 
@@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute)
 			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
 }
 
+static int es8328_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->sysclk_constraints)
+		snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				es8328->sysclk_constraints);
+
+	return 0;
+}
+
 static int es8328_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
-	int clk_rate;
 	int i;
 	int reg;
-	u8 ratio;
+	int wl;
+	int ratio;
+
+	if (!es8328->sysclk_constraints) {
+		dev_err(codec->dev, "No MCLK configured\n");
+		return -EINVAL;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		reg = ES8328_DACCONTROL2;
 	else
 		reg = ES8328_ADCCONTROL5;
 
-	clk_rate = clk_get_rate(es8328->clk);
+	for (i = 0; i < es8328->sysclk_constraints->count; i++)
+		if (es8328->sysclk_constraints->list[i] == params_rate(params))
+			break;
 
-	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
+	if (i == es8328->sysclk_constraints->count) {
-		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+		dev_err(codec->dev, "LRCLK %d unsupported with current clock\n",
-		dev_err(codec->dev,
+			params_rate(params));
-			"%s: clock is running at %d Hz, not %d or %d Hz\n",
-			 __func__, clk_rate,
-			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
 		return -EINVAL;
 	}
 
-	/* find master mode MCLK to sampling frequency ratio */
+	ratio = es8328->mclk_ratios[i];
-	ratio = mclk_ratios[0].rate;
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
-	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
+			ES8328_MASTERMODE_MCLKDIV2,
-		if (params_rate(params) <= mclk_ratios[i].rate)
+			es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0);
-			ratio = mclk_ratios[i].ratio;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+	switch (params_width(params)) {
-		es8328->playback_fs = params_rate(params);
+	case 16:
-		es8328_set_deemph(codec);
+		wl = 3;
+		break;
+	case 18:
+		wl = 2;
+		break;
+	case 20:
+		wl = 1;
+		break;
+	case 24:
+		wl = 0;
+		break;
+	case 32:
+		wl = 4;
+		break;
+	default:
+		return -EINVAL;
 	}
 
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, ES8328_DACCONTROL1,
+				ES8328_DACCONTROL1_DACWL_MASK,
+				wl << ES8328_DACCONTROL1_DACWL_SHIFT);
+
+		es8328->playback_fs = params_rate(params);
+		es8328_set_deemph(codec);
+	} else
+		snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+				ES8328_ADCCONTROL4_ADCWL_MASK,
+				wl << ES8328_ADCCONTROL4_ADCWL_SHIFT);
+
 	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
 }
 
+static int es8328_set_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int mclkdiv2 = 0;
+
+	switch (freq) {
+	case 0:
+		es8328->sysclk_constraints = NULL;
+		es8328->mclk_ratios = NULL;
+		break;
+	case 22579200:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 11289600:
+		es8328->sysclk_constraints = &constraints_11289;
+		es8328->mclk_ratios = ratios_11289;
+		break;
+	case 24576000:
+		mclkdiv2 = 1;
+		/* fallthru */
+	case 12288000:
+		es8328->sysclk_constraints = &constraints_12288;
+		es8328->mclk_ratios = ratios_12288;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	es8328->mclkdiv2 = mclkdiv2;
+	return 0;
+}
+
 static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	u8 dac_mode = 0;
-	int clk_rate;
+	u8 adc_mode = 0;
-	u8 mode = ES8328_DACCONTROL1_DACWL_16;
 
 	/* set master/slave audio interface */
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
@@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* interface format */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST;
 		break;
 	default:
 		return -EINVAL;
@@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
 		return -EINVAL;
 
-	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
+	snd_soc_update_bits(codec, ES8328_DACCONTROL1,
-	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+			ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode);
+	snd_soc_update_bits(codec, ES8328_ADCCONTROL4,
+			ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode);
 
 	/* Master serial port mode, with BCLK generated automatically */
-	clk_rate = clk_get_rate(es8328->clk);
+	snd_soc_update_bits(codec, ES8328_MASTERMODE,
-	if (clk_rate == ES8328_SYSCLK_RATE_1X)
+			ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC);
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MSC);
-	else
-		snd_soc_write(codec, ES8328_MASTERMODE,
-				ES8328_MASTERMODE_MCLKDIV2 |
-				ES8328_MASTERMODE_MSC);
 
 	return 0;
 }
@@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 }
 
 static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.startup	= es8328_startup,
 	.hw_params	= es8328_hw_params,
 	.digital_mute	= es8328_mute,
+	.set_sysclk	= es8328_set_sysclk,
 	.set_fmt	= es8328_set_dai_fmt,
 };
 
@@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = {
 		.formats = ES8328_FORMATS,
 	},
 	.ops = &es8328_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int es8328_suspend(struct snd_soc_codec *codec)
@@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = {
 	.val_bits	= 8,
 	.max_register	= ES8328_REG_MAX,
 	.cache_type	= REGCACHE_RBTREE,
+	.use_single_rw	= true,
 };
 EXPORT_SYMBOL_GPL(es8328_regmap_config);
 

sound/soc/codecs/es8328.h

@@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
 #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
 #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
-#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0)
 #define ES8328_CONTROL1_ENREF (1 << 2)
 #define ES8328_CONTROL1_SEQEN (1 << 3)
 #define ES8328_CONTROL1_SAMEFS (1 << 4)
@@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL1	0x09
 #define ES8328_ADCCONTROL2	0x0a
 #define ES8328_ADCCONTROL3	0x0b
+
 #define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0)
+#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0)
+#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2
+#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5)
+#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5)
+
 #define ES8328_ADCCONTROL5	0x0d
 #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
 
@@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
 #define ES8328_ADCCONTROL14	0x16
 
 #define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
 #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
-#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
+#define ES8328_DACCONTROL1_DACWL_SHIFT 3
-#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
+#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3)
-#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
-#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
-#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
 #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
 #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)

sound/soc/fsl/fsl_sai.c

@@ -21,6 +21,8 @@
 #include <sound/core.h>
 #include <sound/dmaengine_pcm.h>
 #include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
 
 #include "fsl_sai.h"
 #include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
 	struct fsl_sai *sai;
+	struct regmap *gpr;
 	struct resource *res;
 	void __iomem *base;
 	char tmp[8];
 	int irq, ret, i;
+	int index;
 
 	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
 	if (!sai)
@@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev)
 
 	sai->pdev = pdev;
 
-	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
+	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") ||
+	    of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai"))
 		sai->sai_on_imx = true;
 
 	sai->is_lsb_first = of_property_read_bool(np, "lsb-first");
@@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		fsl_sai_dai.symmetric_samplebits = 0;
 	}
 
+	if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+	    of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+		gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+		if (IS_ERR(gpr)) {
+			dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+			return PTR_ERR(gpr);
+		}
+
+		index = of_alias_get_id(np, "sai");
+		if (index < 0)
+			return index;
+
+		regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+				   MCLK_DIR(index));
+	}
+
 	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
 	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
@@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 static const struct of_device_id fsl_sai_ids[] = {
 	{ .compatible = "fsl,vf610-sai", },
 	{ .compatible = "fsl,imx6sx-sai", },
+	{ .compatible = "fsl,imx6ul-sai", },
 	{ /* sentinel */ }
 };
 MODULE_DEVICE_TABLE(of, fsl_sai_ids);

sound/soc/fsl/fsl_ssi.c

@@ -262,6 +262,7 @@ struct fsl_ssi_private {
 	struct fsl_ssi_dbg dbg_stats;
 
 	const struct fsl_ssi_soc_data *soc;
+	struct device *dev;
 };
 
 /*
@@ -400,6 +401,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private,
 	}
 }
 
+/*
+ * Clear RX or TX FIFO to remove samples from the previous
+ * stream session which may be still present in the FIFO and
+ * may introduce bad samples and/or channel slipping.
+ *
+ * Note: The SOR is not documented in recent IMX datasheet, but
+ * is described in IMX51 reference manual at section 56.3.3.15.
+ */
+static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private,
+		bool is_rx)
+{
+	if (is_rx) {
+		regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+			CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR);
+	} else {
+		regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR,
+			CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR);
+	}
+}
+
 /*
  * Calculate the bits that have to be disabled for the current stream that is
  * getting disabled. This keeps the bits enabled that are necessary for the
@@ -475,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
 	 * (online configuration)
 	 */
 	if (enable) {
-		regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
+		fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE);
+
 		regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr);
 		regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr);
+		regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier);
 	} else {
 		u32 sier;
 		u32 srcr;
@@ -507,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable,
 
 config_done:
 	/* Enabling of subunits is done after configuration */
-	if (enable)
+	if (enable) {
+		if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) {
+			/*
+			 * Be sure the Tx FIFO is filled when TE is set.
+			 * Otherwise, there are some chances to start the
+			 * playback with some void samples inserted first,
+			 * generating a channel slip.
+			 *
+			 * First, SSIEN must be set, to let the FIFO be filled.
+			 *
+			 * Notes:
+			 * - Limit this fix to the DMA case until FIQ cases can
+			 *   be tested.
+			 * - Limit the length of the busy loop to not lock the
+			 *   system too long, even if 1-2 loops are sufficient
+			 *   in general.
+			 */
+			int i;
+			int max_loop = 100;
+			regmap_update_bits(regs, CCSR_SSI_SCR,
+					CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN);
+			for (i = 0; i < max_loop; i++) {
+				u32 sfcsr;
+				regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr);
+				if (CCSR_SSI_SFCSR_TFCNT0(sfcsr))
+					break;
+			}
+			if (i == max_loop) {
+				dev_err(ssi_private->dev,
+					"Timeout waiting TX FIFO filling\n");
+			}
+		}
 		regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr);
+	}
 }
 
 
@@ -671,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	if (IS_ERR(ssi_private->baudclk))
 		return -EINVAL;
 
+	/*
+	 * Hardware limitation: The bclk rate must be
+	 * never greater than 1/5 IPG clock rate
+	 */
+	if (freq * 5 > clk_get_rate(ssi_private->clk)) {
+		dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n");
+		return -EINVAL;
+	}
+
 	baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream));
 
 	/* It should be already enough to divide clock by setting pm alone */
@@ -687,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 		else
 			clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
 
-		/*
-		 * Hardware limitation: The bclk rate must be
-		 * never greater than 1/5 IPG clock rate
-		 */
-		if (clkrate * 5 > clk_get_rate(ssi_private->clk))
-			continue;
-
 		clkrate /= factor;
 		afreq = clkrate / (i + 1);
 
@@ -1159,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 	.playback = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 32,
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
@@ -1403,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi_private->soc = of_id->data;
+	ssi_private->dev = &pdev->dev;
 
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (sprop) {

sound/soc/soc-core.c

@@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component(
 	return NULL;
 }
 
-static struct snd_soc_dai *snd_soc_find_dai(
+/**
+ * snd_soc_find_dai - Find a registered DAI
+ *
+ * @dlc: name of the DAI and optional component info to match
+ *
+ * This function will search all regsitered components and their DAIs to
+ * find the DAI of the same name. The component's of_node and name
+ * should also match if being specified.
+ *
+ * Return: pointer of DAI, or NULL if not found.
+ */
+struct snd_soc_dai *snd_soc_find_dai(
 	const struct snd_soc_dai_link_component *dlc)
 {
 	struct snd_soc_component *component;
@@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
 
 	return NULL;
 }
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
 static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 		struct snd_soc_dai_link *dai_link)