Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.9

The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:

 - Convert users of digital_mute() to mute_stream().
 - Simplify I/O helper functions.
 - Add a helper for getting the RTD from a substream.
 - Many, many fixes and cleanups to the x86 code.
 - New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
   MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
   Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
   of the first phones I worked on!) and TI J721e EVM.

14 files changed, 70 insertions, 54 deletions

diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 8b6295283989..363dc3b1bbe4 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -68,6 +68,7 @@ config SND_MESON_AXG_SOUND_CARD
 	imply SND_MESON_AXG_SPDIFOUT
 	imply SND_MESON_AXG_SPDIFIN
 	imply SND_MESON_AXG_PDM
+	imply SND_MESON_G12A_TOACODEC
 	imply SND_MESON_G12A_TOHDMITX if DRM_MESON_DW_HDMI
 	help
 	  Select Y or M to add support for the AXG SoC sound card
diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c
index 832e22d275fe..932224552146 100644
--- a/sound/soc/meson/aiu-encoder-i2s.c
+++ b/sound/soc/meson/aiu-encoder-i2s.c
@@ -72,11 +72,10 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component,
 {
 	/* Always operate in split (classic interleaved) mode */
 	unsigned int desc = AIU_I2S_SOURCE_DESC_MODE_SPLIT;
-	unsigned int val;
 
 	/* Reset required to update the pipeline */
 	snd_soc_component_write(component, AIU_RST_SOFT, AIU_RST_SOFT_I2S_FAST);
-	snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+	snd_soc_component_read(component, AIU_I2S_SYNC);
 
 	switch (params_physical_width(params)) {
 	case 16: /* Nothing to do */
diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c
index 9a5271ce80fe..d91b0d874342 100644
--- a/sound/soc/meson/aiu-fifo-i2s.c
+++ b/sound/soc/meson/aiu-fifo-i2s.c
@@ -46,7 +46,6 @@ static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 				struct snd_soc_dai *dai)
 {
 	struct snd_soc_component *component = dai->component;
-	unsigned int val;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -54,7 +53,7 @@ static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		snd_soc_component_write(component, AIU_RST_SOFT,
 					AIU_RST_SOFT_I2S_FAST);
-		snd_soc_component_read(component, AIU_I2S_SYNC, &val);
+		snd_soc_component_read(component, AIU_I2S_SYNC);
 		break;
 	}
 
diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c
index d9cede4c33ff..aa88aae8e517 100644
--- a/sound/soc/meson/aiu-fifo.c
+++ b/sound/soc/meson/aiu-fifo.c
@@ -37,8 +37,7 @@ snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int addr;
 
-	snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD,
-			       &addr);
+	addr = snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD);
 
 	return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr);
 }
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 89f7f64747cd..2b77010c2c5c 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -40,7 +40,7 @@ static const struct snd_soc_pcm_stream codec_params = {
 static int axg_card_tdm_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 meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct axg_dai_link_tdm_data *be =
 		(struct axg_dai_link_tdm_data *)priv->link_data[rtd->num];
@@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 
 	lb = &card->dai_link[*index + 1];
 
-	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
 	if (!lb->name)
 		return -ENOMEM;
 
@@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
 		return ret;
 
 	if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
-		ret = meson_card_set_fe_link(card, dai_link, np, true);
+		return meson_card_set_fe_link(card, dai_link, np, true);
 	else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
-		ret = meson_card_set_fe_link(card, dai_link, np, false);
-	else
-		ret = meson_card_set_be_link(card, dai_link, np);
+		return meson_card_set_fe_link(card, dai_link, np, false);
 
+
+	ret = meson_card_set_be_link(card, dai_link, np);
 	if (ret)
 		return ret;
 
-	if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
-		ret = axg_card_parse_tdm(card, np, index);
-	else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
+	if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
 		dai_link->params = &codec_params;
-		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	} else {
+		dai_link->no_pcm = 1;
+		snd_soc_dai_link_set_capabilities(dai_link);
+		if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
+			ret = axg_card_parse_tdm(card, np, index);
 	}
 
 	return ret;
diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
index 7ce6aa97ddf7..e769a5ee6e27 100644
--- a/sound/soc/meson/axg-spdifout.c
+++ b/sound/soc/meson/axg-spdifout.c
@@ -108,7 +108,7 @@ static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd,
 	}
 }
 
-static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute)
+static int axg_spdifout_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
 
@@ -285,10 +285,11 @@ static void axg_spdifout_shutdown(struct snd_pcm_substream *substream,
 
 static const struct snd_soc_dai_ops axg_spdifout_ops = {
 	.trigger	= axg_spdifout_trigger,
-	.digital_mute	= axg_spdifout_digital_mute,
+	.mute_stream	= axg_spdifout_mute,
 	.hw_params	= axg_spdifout_hw_params,
 	.startup	= axg_spdifout_startup,
 	.shutdown	= axg_spdifout_shutdown,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = {
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 358c8c0d861c..f7e8e9da68a0 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
 static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 {
 	struct axg_tdm_stream *ts = formatter->stream;
-	bool invert = formatter->drv->quirks->invert_sclk;
+	bool invert;
 	int ret;
 
 	/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 		return ret;
 
 	/*
-	 * If sclk is inverted, invert it back and provide the inversion
-	 * required by the formatter
+	 * If sclk is inverted, it means the bit should latched on the
+	 * rising edge which is what our HW expects. If not, we need to
+	 * invert it before the formatter.
 	 */
-	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
-	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	invert = axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
index 9ef98e955cb2..a1f0dcc0ff13 100644
--- a/sound/soc/meson/axg-tdm-formatter.h
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -16,7 +16,6 @@ struct snd_kcontrol;
 
 struct axg_tdm_formatter_hw {
 	unsigned int skew_offset;
-	bool invert_sclk;
 };
 
 struct axg_tdm_formatter_ops {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index 6de27238e9df..36df30915378 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -119,18 +119,25 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
 
-	/* These modes are not supported */
-	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (!iface->mclk) {
+			dev_err(dai->dev, "cpu clock master: mclk missing\n");
+			return -ENODEV;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		/* Fall-through */
+	default:
 		return -EINVAL;
 	}
 
-	/* If the TDM interface is the clock master, it requires mclk */
-	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
-		dev_err(dai->dev, "cpu clock master: mclk missing\n");
-		return -ENODEV;
-	}
-
 	iface->fmt = fmt;
 	return 0;
 }
@@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+	if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+	    SND_SOC_DAIFMT_CBS_CFS) {
 		ret = axg_tdm_iface_set_sclk(dai, params);
 		if (ret)
 			return ret;
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 973d4c02ef8d..88ed95ae886b 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
 	.regmap_cfg	= &axg_tdmin_regmap_cfg,
 	.ops		= &axg_tdmin_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk	= false,
 		.skew_offset	= 2,
 	},
 };
 
+static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.quirks		= &(const struct axg_tdm_formatter_hw) {
+		.skew_offset	= 3,
+	},
+};
+
 static const struct of_device_id axg_tdmin_of_match[] = {
 	{
 		.compatible = "amlogic,axg-tdmin",
 		.data = &axg_tdmin_drv,
+	}, {
+		.compatible = "amlogic,g12a-tdmin",
+		.data = &g12a_tdmin_drv,
+	}, {
+		.compatible = "amlogic,sm1-tdmin",
+		.data = &g12a_tdmin_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 418ec314b37d..3ceabddae629 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 1,
 	},
 };
@@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
@@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
index 4abf7efb7eac..5119434a81c4 100644
--- a/sound/soc/meson/gx-card.c
+++ b/sound/soc/meson/gx-card.c
@@ -29,7 +29,7 @@ static const struct snd_soc_pcm_stream codec_params = {
 static int gx_card_i2s_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 meson_card *priv = snd_soc_card_get_drvdata(rtd->card);
 	struct gx_dai_link_i2s_data *be =
 		(struct gx_dai_link_i2s_data *)priv->link_data[rtd->num];
@@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
 		return ret;
 
 	if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
-		ret = meson_card_set_fe_link(card, dai_link, np, true);
-	else
-		ret = meson_card_set_be_link(card, dai_link, np);
+		return  meson_card_set_fe_link(card, dai_link, np, true);
 
+	ret = meson_card_set_be_link(card, dai_link, np);
 	if (ret)
 		return ret;
 
-	/* Check if the cpu is the i2s encoder and parse i2s data */
-	if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
-		ret = gx_card_parse_i2s(card, np, index);
-
 	/* Or apply codec to codec params if necessary */
-	else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
+	if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
 		dai_link->params = &codec_params;
-		dai_link->no_pcm = 0; /* link is not a DPCM BE */
+	} else {
+		dai_link->no_pcm = 1;
+		snd_soc_dai_link_set_capabilities(dai_link);
+		/* Check if the cpu is the i2s encoder and parse i2s data */
+		if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+			ret = gx_card_parse_i2s(card, np, index);
 	}
 
 	return ret;
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index 5a4a91c88734..6a64ac01b5ca 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -13,7 +13,7 @@ int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params,
 			      unsigned int mclk_fs)
 {
-	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;
 	unsigned int mclk;
 	int ret, i;
@@ -119,7 +119,7 @@ unsigned int meson_card_parse_daifmt(struct device_node *node,
 	struct device_node *framemaster = NULL;
 	unsigned int daifmt;
 
-	daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX,
+	daifmt = snd_soc_of_parse_daifmt(node, "",
 					 &bitclkmaster, &framemaster);
 	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
 
@@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card,
 	struct device_node *np;
 	int ret, num_codecs;
 
-	link->no_pcm = 1;
-	link->dpcm_playback = 1;
-	link->dpcm_capture = 1;
-
 	num_codecs = of_get_child_count(node);
 	if (!num_codecs) {
 		dev_err(card->dev, "be link %s has no codec\n",
diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c
index 524a33472337..d07270d17cee 100644
--- a/sound/soc/meson/meson-codec-glue.c
+++ b/sound/soc/meson/meson-codec-glue.c
@@ -98,7 +98,7 @@ EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt);
 int meson_codec_glue_output_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 meson_codec_glue_input *in_data =
 		meson_codec_glue_output_get_input_data(dai->capture_widget);