[PATCH 0/7] ASoC: sti audio fixes and legacy cleaning

[PATCH 0/7] ASoC: sti audio fixes and legacy cleaning

[Arnaud Pouliquen]

Audio fixes and cleaning of code associated to stih416 platform.

Arnaud Pouliquen (7):
  ASoC: sti: fix errors management
  ASoC: sti: fix channel status update after playback start
  ASoC: sti: reset refactoring
  ASoC: sti: clean unused include
  ASoC: codec: clean legacy in sti-sas
  ASoC: sti: sti-sas: add missing return in messages strings
  ASoc: codec: sti enable fast io for regmap

 sound/soc/codecs/sti-sas.c      | 181 +++++-----------------------------------
 sound/soc/sti/sti_uniperif.c    |  43 ++++++++--
 sound/soc/sti/uniperif.h        |   2 +
 sound/soc/sti/uniperif_player.c |  97 +++++++++------------
 sound/soc/sti/uniperif_reader.c |  41 +++------
 5 files changed, 111 insertions(+), 253 deletions(-)

-- 
1.9.1

[PATCH 1/7] ASoC: sti: fix errors management

[Arnaud Pouliquen]

Add missing error messages.
Propagate error of uni_reader_init and uni_reader_init.
Add return at end of dev_err strings.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/sti/sti_uniperif.c    | 20 ++++++++------
 sound/soc/sti/uniperif_player.c | 58 +++++++++++++++++++++++------------------
 sound/soc/sti/uniperif_reader.c | 25 +++++++++---------
 3 files changed, 58 insertions(+), 45 deletions(-)

diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index 549fac3..ee91ae5 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -293,7 +293,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
 
 	/* The uniperipheral should be in stopped state */
 	if (uni->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(uni->dev, "%s: invalid uni state( %d)",
+		dev_err(uni->dev, "%s: invalid uni state( %d)\n",
 			__func__, (int)uni->state);
 		return -EBUSY;
 	}
@@ -301,7 +301,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
 	/* Pinctrl: switch pinstate to sleep */
 	ret = pinctrl_pm_select_sleep_state(uni->dev);
 	if (ret)
-		dev_err(uni->dev, "%s: failed to select pinctrl state",
+		dev_err(uni->dev, "%s: failed to select pinctrl state\n",
 			__func__);
 
 	return ret;
@@ -322,7 +322,7 @@ static int sti_uniperiph_dai_resume(struct snd_soc_dai *dai)
 	/* pinctrl: switch pinstate to default */
 	ret = pinctrl_pm_select_default_state(uni->dev);
 	if (ret)
-		dev_err(uni->dev, "%s: failed to select pinctrl state",
+		dev_err(uni->dev, "%s: failed to select pinctrl state\n",
 			__func__);
 
 	return ret;
@@ -366,11 +366,12 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	const struct of_device_id *of_id;
 	const struct sti_uniperiph_dev_data *dev_data;
 	const char *mode;
+	int ret;
 
 	/* Populate data structure depending on compatibility */
 	of_id = of_match_node(snd_soc_sti_match, node);
 	if (!of_id->data) {
-		dev_err(dev, "data associated to device is missing");
+		dev_err(dev, "data associated to device is missing\n");
 		return -EINVAL;
 	}
 	dev_data = (struct sti_uniperiph_dev_data *)of_id->data;
@@ -389,7 +390,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0);
 
 	if (!uni->mem_region) {
-		dev_err(dev, "Failed to get memory resource");
+		dev_err(dev, "Failed to get memory resource\n");
 		return -ENODEV;
 	}
 
@@ -403,7 +404,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 
 	uni->irq = platform_get_irq(priv->pdev, 0);
 	if (uni->irq < 0) {
-		dev_err(dev, "Failed to get IRQ resource");
+		dev_err(dev, "Failed to get IRQ resource\n");
 		return -ENXIO;
 	}
 
@@ -421,12 +422,15 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	dai_data->stream = dev_data->stream;
 
 	if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		uni_player_init(priv->pdev, uni);
+		ret = uni_player_init(priv->pdev, uni);
 		stream = &dai->playback;
 	} else {
-		uni_reader_init(priv->pdev, uni);
+		ret = uni_reader_init(priv->pdev, uni);
 		stream = &dai->capture;
 	}
+	if (ret < 0)
+		return ret;
+
 	dai->ops = uni->dai_ops;
 
 	stream->stream_name = dai->name;
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 1bc8ebc..c9b4670 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -67,7 +67,7 @@ static inline int reset_player(struct uniperif *player)
 	}
 
 	if (!count) {
-		dev_err(player->dev, "Failed to reset uniperif");
+		dev_err(player->dev, "Failed to reset uniperif\n");
 		return -EIO;
 	}
 
@@ -97,7 +97,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 
 	/* Check for fifo error (underrun) */
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) {
-		dev_err(player->dev, "FIFO underflow error detected");
+		dev_err(player->dev, "FIFO underflow error detected\n");
 
 		/* Interrupt is just for information when underflow recovery */
 		if (player->underflow_enabled) {
@@ -119,7 +119,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 
 	/* Check for dma error (overrun) */
 	if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) {
-		dev_err(player->dev, "DMA error detected");
+		dev_err(player->dev, "DMA error detected\n");
 
 		/* Disable interrupt so doesn't continually fire */
 		SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
@@ -135,11 +135,14 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 	/* Check for underflow recovery done */
 	if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) {
 		if (!player->underflow_enabled) {
-			dev_err(player->dev, "unexpected Underflow recovering");
+			dev_err(player->dev,
+				"unexpected Underflow recovering\n");
 			return -EPERM;
 		}
 		/* Read the underflow recovery duration */
 		tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player);
+		dev_dbg(player->dev, "Underflow recovered (%d LR clocks max)\n",
+			tmp);
 
 		/* Clear the underflow recovery duration */
 		SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player);
@@ -153,7 +156,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 	/* Check if underflow recovery failed */
 	if (unlikely(status &
 		     UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) {
-		dev_err(player->dev, "Underflow recovery failed");
+		dev_err(player->dev, "Underflow recovery failed\n");
 
 		/* Stop the player */
 		snd_pcm_stream_lock(player->substream);
@@ -336,7 +339,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
 
 	/* Oversampling must be multiple of 128 as iec958 frame is 32-bits */
 	if ((clk_div % 128) || (clk_div <= 0)) {
-		dev_err(player->dev, "%s: invalid clk_div %d",
+		dev_err(player->dev, "%s: invalid clk_div %d\n",
 			__func__, clk_div);
 		return -EINVAL;
 	}
@@ -359,7 +362,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player);
 		break;
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -448,12 +451,12 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	 * for 16 bits must be a multiple of 64
 	 */
 	if ((slot_width == 32) && (clk_div % 128)) {
-		dev_err(player->dev, "%s: invalid clk_div", __func__);
+		dev_err(player->dev, "%s: invalid clk_div\n", __func__);
 		return -EINVAL;
 	}
 
 	if ((slot_width == 16) && (clk_div % 64)) {
-		dev_err(player->dev, "%s: invalid clk_div", __func__);
+		dev_err(player->dev, "%s: invalid clk_div\n", __func__);
 		return -EINVAL;
 	}
 
@@ -471,7 +474,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player);
 		break;
 	default:
-		dev_err(player->dev, "subframe format not supported");
+		dev_err(player->dev, "subframe format not supported\n");
 		return -EINVAL;
 	}
 
@@ -491,7 +494,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 		break;
 
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -504,7 +507,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	/* Number of channelsmust be even*/
 	if ((runtime->channels % 2) || (runtime->channels < 2) ||
 	    (runtime->channels > 10)) {
-		dev_err(player->dev, "%s: invalid nb of channels", __func__);
+		dev_err(player->dev, "%s: invalid nb of channels\n", __func__);
 		return -EINVAL;
 	}
 
@@ -758,7 +761,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 
 	/* The player should be stopped */
 	if (player->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state %d", __func__,
+		dev_err(player->dev, "%s: invalid player state %d\n", __func__,
 			player->state);
 		return -EINVAL;
 	}
@@ -787,7 +790,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 	/* Trigger limit must be an even number */
 	if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) ||
 	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) {
-		dev_err(player->dev, "invalid trigger limit %d", trigger_limit);
+		dev_err(player->dev, "invalid trigger limit %d\n",
+			trigger_limit);
 		return -EINVAL;
 	}
 
@@ -808,7 +812,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 		ret = uni_player_prepare_tdm(player, runtime);
 		break;
 	default:
-		dev_err(player->dev, "invalid player type");
+		dev_err(player->dev, "invalid player type\n");
 		return -EINVAL;
 	}
 
@@ -848,7 +852,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 		SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player);
 		break;
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -866,13 +870,13 @@ static int uni_player_start(struct uniperif *player)
 
 	/* The player should be stopped */
 	if (player->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state", __func__);
+		dev_err(player->dev, "%s: invalid player state\n", __func__);
 		return -EINVAL;
 	}
 
 	ret = clk_prepare_enable(player->clk);
 	if (ret) {
-		dev_err(player->dev, "%s: Failed to enable clock", __func__);
+		dev_err(player->dev, "%s: Failed to enable clock\n", __func__);
 		return ret;
 	}
 
@@ -934,7 +938,7 @@ static int uni_player_stop(struct uniperif *player)
 
 	/* The player should not be in stopped state */
 	if (player->state == UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state", __func__);
+		dev_err(player->dev, "%s: invalid player state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -969,7 +973,7 @@ int uni_player_resume(struct uniperif *player)
 		ret = regmap_field_write(player->clk_sel, 1);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: Failed to select freq synth clock",
+				"%s: Failed to select freq synth clock\n",
 				__func__);
 			return ret;
 		}
@@ -1066,7 +1070,7 @@ int uni_player_init(struct platform_device *pdev,
 	ret = uni_player_parse_dt_audio_glue(pdev, player);
 
 	if (ret < 0) {
-		dev_err(player->dev, "Failed to parse DeviceTree");
+		dev_err(player->dev, "Failed to parse DeviceTree\n");
 		return ret;
 	}
 
@@ -1081,15 +1085,17 @@ int uni_player_init(struct platform_device *pdev,
 
 	/* Get uniperif resource */
 	player->clk = of_clk_get(pdev->dev.of_node, 0);
-	if (IS_ERR(player->clk))
+	if (IS_ERR(player->clk)) {
+		dev_err(player->dev, "Failed to get clock\n");
 		ret = PTR_ERR(player->clk);
+	}
 
 	/* Select the frequency synthesizer clock */
 	if (player->clk_sel) {
 		ret = regmap_field_write(player->clk_sel, 1);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: Failed to select freq synth clock",
+				"%s: Failed to select freq synth clock\n",
 				__func__);
 			return ret;
 		}
@@ -1101,7 +1107,7 @@ int uni_player_init(struct platform_device *pdev,
 		ret = regmap_field_write(player->valid_sel, player->id);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: unable to connect to tdm bus", __func__);
+				"%s: unable to connect to tdm bus\n", __func__);
 			return ret;
 		}
 	}
@@ -1109,8 +1115,10 @@ int uni_player_init(struct platform_device *pdev,
 	ret = devm_request_irq(&pdev->dev, player->irq,
 			       uni_player_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), player);
-	if (ret < 0)
+	if (ret < 0) {
+		dev_err(player->dev, "unable to request IRQ %d\n", player->irq);
 		return ret;
+	}
 
 	mutex_init(&player->ctrl_lock);
 
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 0e1c3ee..09314f8 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -52,7 +52,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 
 	if (reader->state == UNIPERIF_STATE_STOPPED) {
 		/* Unexpected IRQ: do nothing */
-		dev_warn(reader->dev, "unexpected IRQ ");
+		dev_warn(reader->dev, "unexpected IRQ\n");
 		return IRQ_HANDLED;
 	}
 
@@ -62,7 +62,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 
 	/* Check for fifo overflow error */
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
-		dev_err(reader->dev, "FIFO error detected");
+		dev_err(reader->dev, "FIFO error detected\n");
 
 		snd_pcm_stream_lock(reader->substream);
 		snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
@@ -105,7 +105,7 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader);
 		break;
 	default:
-		dev_err(reader->dev, "subframe format not supported");
+		dev_err(reader->dev, "subframe format not supported\n");
 		return -EINVAL;
 	}
 
@@ -125,14 +125,14 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
 		break;
 
 	default:
-		dev_err(reader->dev, "format not supported");
+		dev_err(reader->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
 	/* Number of channels must be even */
 	if ((runtime->channels % 2) || (runtime->channels < 2) ||
 	    (runtime->channels > 10)) {
-		dev_err(reader->dev, "%s: invalid nb of channels", __func__);
+		dev_err(reader->dev, "%s: invalid nb of channels\n", __func__);
 		return -EINVAL;
 	}
 
@@ -190,7 +190,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state %d", __func__,
+		dev_err(reader->dev, "%s: invalid reader state %d\n", __func__,
 			reader->state);
 		return -EINVAL;
 	}
@@ -219,7 +219,8 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	if ((!trigger_limit % 2) ||
 	    (trigger_limit != 1 && transfer_size % 2) ||
 	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
-		dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
+		dev_err(reader->dev, "invalid trigger limit %d\n",
+			trigger_limit);
 		return -EINVAL;
 	}
 
@@ -246,7 +247,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader);
 		break;
 	default:
-		dev_err(reader->dev, "format not supported");
+		dev_err(reader->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -294,7 +295,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		count--;
 	}
 	if (!count) {
-		dev_err(reader->dev, "Failed to reset uniperif");
+		dev_err(reader->dev, "Failed to reset uniperif\n");
 		return -EIO;
 	}
 
@@ -305,7 +306,7 @@ static int uni_reader_start(struct uniperif *reader)
 {
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state", __func__);
+		dev_err(reader->dev, "%s: invalid reader state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -325,7 +326,7 @@ static int uni_reader_stop(struct uniperif *reader)
 {
 	/* The reader should not be in stopped state */
 	if (reader->state == UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state", __func__);
+		dev_err(reader->dev, "%s: invalid reader state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -423,7 +424,7 @@ int uni_reader_init(struct platform_device *pdev,
 			       uni_reader_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), reader);
 	if (ret < 0) {
-		dev_err(&pdev->dev, "Failed to request IRQ");
+		dev_err(&pdev->dev, "Failed to request IRQ\n");
 		return -EBUSY;
 	}
 
-- 
1.9.1

Applied "ASoC: sti: fix errors management" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti: fix errors management

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 748abba8f3a93cee13a56350386e59457ffa600d Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:51 +0200
Subject: [PATCH] ASoC: sti: fix errors management

Add missing error messages.
Propagate error of uni_reader_init and uni_reader_init.
Add return at end of dev_err strings.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sti/sti_uniperif.c    | 20 ++++++++------
 sound/soc/sti/uniperif_player.c | 58 +++++++++++++++++++++++------------------
 sound/soc/sti/uniperif_reader.c | 25 +++++++++---------
 3 files changed, 58 insertions(+), 45 deletions(-)

diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index 549fac349fa0..ee91ae5f812a 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -293,7 +293,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
 
 	/* The uniperipheral should be in stopped state */
 	if (uni->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(uni->dev, "%s: invalid uni state( %d)",
+		dev_err(uni->dev, "%s: invalid uni state( %d)\n",
 			__func__, (int)uni->state);
 		return -EBUSY;
 	}
@@ -301,7 +301,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
 	/* Pinctrl: switch pinstate to sleep */
 	ret = pinctrl_pm_select_sleep_state(uni->dev);
 	if (ret)
-		dev_err(uni->dev, "%s: failed to select pinctrl state",
+		dev_err(uni->dev, "%s: failed to select pinctrl state\n",
 			__func__);
 
 	return ret;
@@ -322,7 +322,7 @@ static int sti_uniperiph_dai_resume(struct snd_soc_dai *dai)
 	/* pinctrl: switch pinstate to default */
 	ret = pinctrl_pm_select_default_state(uni->dev);
 	if (ret)
-		dev_err(uni->dev, "%s: failed to select pinctrl state",
+		dev_err(uni->dev, "%s: failed to select pinctrl state\n",
 			__func__);
 
 	return ret;
@@ -366,11 +366,12 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	const struct of_device_id *of_id;
 	const struct sti_uniperiph_dev_data *dev_data;
 	const char *mode;
+	int ret;
 
 	/* Populate data structure depending on compatibility */
 	of_id = of_match_node(snd_soc_sti_match, node);
 	if (!of_id->data) {
-		dev_err(dev, "data associated to device is missing");
+		dev_err(dev, "data associated to device is missing\n");
 		return -EINVAL;
 	}
 	dev_data = (struct sti_uniperiph_dev_data *)of_id->data;
@@ -389,7 +390,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0);
 
 	if (!uni->mem_region) {
-		dev_err(dev, "Failed to get memory resource");
+		dev_err(dev, "Failed to get memory resource\n");
 		return -ENODEV;
 	}
 
@@ -403,7 +404,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 
 	uni->irq = platform_get_irq(priv->pdev, 0);
 	if (uni->irq < 0) {
-		dev_err(dev, "Failed to get IRQ resource");
+		dev_err(dev, "Failed to get IRQ resource\n");
 		return -ENXIO;
 	}
 
@@ -421,12 +422,15 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
 	dai_data->stream = dev_data->stream;
 
 	if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		uni_player_init(priv->pdev, uni);
+		ret = uni_player_init(priv->pdev, uni);
 		stream = &dai->playback;
 	} else {
-		uni_reader_init(priv->pdev, uni);
+		ret = uni_reader_init(priv->pdev, uni);
 		stream = &dai->capture;
 	}
+	if (ret < 0)
+		return ret;
+
 	dai->ops = uni->dai_ops;
 
 	stream->stream_name = dai->name;
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 1bc8ebc2528e..c9b4670b772b 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -67,7 +67,7 @@ static inline int reset_player(struct uniperif *player)
 	}
 
 	if (!count) {
-		dev_err(player->dev, "Failed to reset uniperif");
+		dev_err(player->dev, "Failed to reset uniperif\n");
 		return -EIO;
 	}
 
@@ -97,7 +97,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 
 	/* Check for fifo error (underrun) */
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) {
-		dev_err(player->dev, "FIFO underflow error detected");
+		dev_err(player->dev, "FIFO underflow error detected\n");
 
 		/* Interrupt is just for information when underflow recovery */
 		if (player->underflow_enabled) {
@@ -119,7 +119,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 
 	/* Check for dma error (overrun) */
 	if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) {
-		dev_err(player->dev, "DMA error detected");
+		dev_err(player->dev, "DMA error detected\n");
 
 		/* Disable interrupt so doesn't continually fire */
 		SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
@@ -135,11 +135,14 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 	/* Check for underflow recovery done */
 	if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) {
 		if (!player->underflow_enabled) {
-			dev_err(player->dev, "unexpected Underflow recovering");
+			dev_err(player->dev,
+				"unexpected Underflow recovering\n");
 			return -EPERM;
 		}
 		/* Read the underflow recovery duration */
 		tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player);
+		dev_dbg(player->dev, "Underflow recovered (%d LR clocks max)\n",
+			tmp);
 
 		/* Clear the underflow recovery duration */
 		SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player);
@@ -153,7 +156,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
 	/* Check if underflow recovery failed */
 	if (unlikely(status &
 		     UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) {
-		dev_err(player->dev, "Underflow recovery failed");
+		dev_err(player->dev, "Underflow recovery failed\n");
 
 		/* Stop the player */
 		snd_pcm_stream_lock(player->substream);
@@ -336,7 +339,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
 
 	/* Oversampling must be multiple of 128 as iec958 frame is 32-bits */
 	if ((clk_div % 128) || (clk_div <= 0)) {
-		dev_err(player->dev, "%s: invalid clk_div %d",
+		dev_err(player->dev, "%s: invalid clk_div %d\n",
 			__func__, clk_div);
 		return -EINVAL;
 	}
@@ -359,7 +362,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player);
 		break;
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -448,12 +451,12 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	 * for 16 bits must be a multiple of 64
 	 */
 	if ((slot_width == 32) && (clk_div % 128)) {
-		dev_err(player->dev, "%s: invalid clk_div", __func__);
+		dev_err(player->dev, "%s: invalid clk_div\n", __func__);
 		return -EINVAL;
 	}
 
 	if ((slot_width == 16) && (clk_div % 64)) {
-		dev_err(player->dev, "%s: invalid clk_div", __func__);
+		dev_err(player->dev, "%s: invalid clk_div\n", __func__);
 		return -EINVAL;
 	}
 
@@ -471,7 +474,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player);
 		break;
 	default:
-		dev_err(player->dev, "subframe format not supported");
+		dev_err(player->dev, "subframe format not supported\n");
 		return -EINVAL;
 	}
 
@@ -491,7 +494,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 		break;
 
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -504,7 +507,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
 	/* Number of channelsmust be even*/
 	if ((runtime->channels % 2) || (runtime->channels < 2) ||
 	    (runtime->channels > 10)) {
-		dev_err(player->dev, "%s: invalid nb of channels", __func__);
+		dev_err(player->dev, "%s: invalid nb of channels\n", __func__);
 		return -EINVAL;
 	}
 
@@ -758,7 +761,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 
 	/* The player should be stopped */
 	if (player->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state %d", __func__,
+		dev_err(player->dev, "%s: invalid player state %d\n", __func__,
 			player->state);
 		return -EINVAL;
 	}
@@ -787,7 +790,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 	/* Trigger limit must be an even number */
 	if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) ||
 	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) {
-		dev_err(player->dev, "invalid trigger limit %d", trigger_limit);
+		dev_err(player->dev, "invalid trigger limit %d\n",
+			trigger_limit);
 		return -EINVAL;
 	}
 
@@ -808,7 +812,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 		ret = uni_player_prepare_tdm(player, runtime);
 		break;
 	default:
-		dev_err(player->dev, "invalid player type");
+		dev_err(player->dev, "invalid player type\n");
 		return -EINVAL;
 	}
 
@@ -848,7 +852,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 		SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player);
 		break;
 	default:
-		dev_err(player->dev, "format not supported");
+		dev_err(player->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -866,13 +870,13 @@ static int uni_player_start(struct uniperif *player)
 
 	/* The player should be stopped */
 	if (player->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state", __func__);
+		dev_err(player->dev, "%s: invalid player state\n", __func__);
 		return -EINVAL;
 	}
 
 	ret = clk_prepare_enable(player->clk);
 	if (ret) {
-		dev_err(player->dev, "%s: Failed to enable clock", __func__);
+		dev_err(player->dev, "%s: Failed to enable clock\n", __func__);
 		return ret;
 	}
 
@@ -934,7 +938,7 @@ static int uni_player_stop(struct uniperif *player)
 
 	/* The player should not be in stopped state */
 	if (player->state == UNIPERIF_STATE_STOPPED) {
-		dev_err(player->dev, "%s: invalid player state", __func__);
+		dev_err(player->dev, "%s: invalid player state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -969,7 +973,7 @@ int uni_player_resume(struct uniperif *player)
 		ret = regmap_field_write(player->clk_sel, 1);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: Failed to select freq synth clock",
+				"%s: Failed to select freq synth clock\n",
 				__func__);
 			return ret;
 		}
@@ -1066,7 +1070,7 @@ int uni_player_init(struct platform_device *pdev,
 	ret = uni_player_parse_dt_audio_glue(pdev, player);
 
 	if (ret < 0) {
-		dev_err(player->dev, "Failed to parse DeviceTree");
+		dev_err(player->dev, "Failed to parse DeviceTree\n");
 		return ret;
 	}
 
@@ -1081,15 +1085,17 @@ int uni_player_init(struct platform_device *pdev,
 
 	/* Get uniperif resource */
 	player->clk = of_clk_get(pdev->dev.of_node, 0);
-	if (IS_ERR(player->clk))
+	if (IS_ERR(player->clk)) {
+		dev_err(player->dev, "Failed to get clock\n");
 		ret = PTR_ERR(player->clk);
+	}
 
 	/* Select the frequency synthesizer clock */
 	if (player->clk_sel) {
 		ret = regmap_field_write(player->clk_sel, 1);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: Failed to select freq synth clock",
+				"%s: Failed to select freq synth clock\n",
 				__func__);
 			return ret;
 		}
@@ -1101,7 +1107,7 @@ int uni_player_init(struct platform_device *pdev,
 		ret = regmap_field_write(player->valid_sel, player->id);
 		if (ret) {
 			dev_err(player->dev,
-				"%s: unable to connect to tdm bus", __func__);
+				"%s: unable to connect to tdm bus\n", __func__);
 			return ret;
 		}
 	}
@@ -1109,8 +1115,10 @@ int uni_player_init(struct platform_device *pdev,
 	ret = devm_request_irq(&pdev->dev, player->irq,
 			       uni_player_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), player);
-	if (ret < 0)
+	if (ret < 0) {
+		dev_err(player->dev, "unable to request IRQ %d\n", player->irq);
 		return ret;
+	}
 
 	mutex_init(&player->ctrl_lock);
 
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 0e1c3ee56675..09314f8be841 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -52,7 +52,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 
 	if (reader->state == UNIPERIF_STATE_STOPPED) {
 		/* Unexpected IRQ: do nothing */
-		dev_warn(reader->dev, "unexpected IRQ ");
+		dev_warn(reader->dev, "unexpected IRQ\n");
 		return IRQ_HANDLED;
 	}
 
@@ -62,7 +62,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
 
 	/* Check for fifo overflow error */
 	if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
-		dev_err(reader->dev, "FIFO error detected");
+		dev_err(reader->dev, "FIFO error detected\n");
 
 		snd_pcm_stream_lock(reader->substream);
 		snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
@@ -105,7 +105,7 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
 		SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader);
 		break;
 	default:
-		dev_err(reader->dev, "subframe format not supported");
+		dev_err(reader->dev, "subframe format not supported\n");
 		return -EINVAL;
 	}
 
@@ -125,14 +125,14 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
 		break;
 
 	default:
-		dev_err(reader->dev, "format not supported");
+		dev_err(reader->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
 	/* Number of channels must be even */
 	if ((runtime->channels % 2) || (runtime->channels < 2) ||
 	    (runtime->channels > 10)) {
-		dev_err(reader->dev, "%s: invalid nb of channels", __func__);
+		dev_err(reader->dev, "%s: invalid nb of channels\n", __func__);
 		return -EINVAL;
 	}
 
@@ -190,7 +190,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state %d", __func__,
+		dev_err(reader->dev, "%s: invalid reader state %d\n", __func__,
 			reader->state);
 		return -EINVAL;
 	}
@@ -219,7 +219,8 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	if ((!trigger_limit % 2) ||
 	    (trigger_limit != 1 && transfer_size % 2) ||
 	    (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
-		dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
+		dev_err(reader->dev, "invalid trigger limit %d\n",
+			trigger_limit);
 		return -EINVAL;
 	}
 
@@ -246,7 +247,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader);
 		break;
 	default:
-		dev_err(reader->dev, "format not supported");
+		dev_err(reader->dev, "format not supported\n");
 		return -EINVAL;
 	}
 
@@ -294,7 +295,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 		count--;
 	}
 	if (!count) {
-		dev_err(reader->dev, "Failed to reset uniperif");
+		dev_err(reader->dev, "Failed to reset uniperif\n");
 		return -EIO;
 	}
 
@@ -305,7 +306,7 @@ static int uni_reader_start(struct uniperif *reader)
 {
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state", __func__);
+		dev_err(reader->dev, "%s: invalid reader state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -325,7 +326,7 @@ static int uni_reader_stop(struct uniperif *reader)
 {
 	/* The reader should not be in stopped state */
 	if (reader->state == UNIPERIF_STATE_STOPPED) {
-		dev_err(reader->dev, "%s: invalid reader state", __func__);
+		dev_err(reader->dev, "%s: invalid reader state\n", __func__);
 		return -EINVAL;
 	}
 
@@ -423,7 +424,7 @@ int uni_reader_init(struct platform_device *pdev,
 			       uni_reader_irq_handler, IRQF_SHARED,
 			       dev_name(&pdev->dev), reader);
 	if (ret < 0) {
-		dev_err(&pdev->dev, "Failed to request IRQ");
+		dev_err(&pdev->dev, "Failed to request IRQ\n");
 		return -EBUSY;
 	}
 
-- 
2.8.1

[PATCH 2/7] ASoC: sti: fix channel status update after playback start

[Arnaud Pouliquen]

If 'IEC958 Playback Default' control is updated during playback,
Channel status needs to be set according to the runtime structure.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/sti/uniperif_player.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index c9b4670..00a0fb3 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -617,7 +617,11 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol,
 	iec958->status[3] = ucontrol->value.iec958.status[3];
 	mutex_unlock(&player->ctrl_lock);
 
-	uni_player_set_channel_status(player, NULL);
+	if (player->substream && player->substream->runtime)
+		uni_player_set_channel_status(player,
+					      player->substream->runtime);
+	else
+		uni_player_set_channel_status(player, NULL);
 
 	return 0;
 }
-- 
1.9.1

Applied "ASoC: sti: fix channel status update after playback start" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti: fix channel status update after playback start

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 1e6d304431958929b601b013687b73293ba27b88 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:52 +0200
Subject: [PATCH] ASoC: sti: fix channel status update after playback start

If 'IEC958 Playback Default' control is updated during playback,
Channel status needs to be set according to the runtime structure.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sti/uniperif_player.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 1bc8ebc2528e..ad54d4cf58ad 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -614,7 +614,11 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol,
 	iec958->status[3] = ucontrol->value.iec958.status[3];
 	mutex_unlock(&player->ctrl_lock);
 
-	uni_player_set_channel_status(player, NULL);
+	if (player->substream && player->substream->runtime)
+		uni_player_set_channel_status(player,
+					      player->substream->runtime);
+	else
+		uni_player_set_channel_status(player, NULL);
 
 	return 0;
 }
-- 
2.8.1

[PATCH 3/7] ASoC: sti: reset refactoring

[Arnaud Pouliquen]

Reset is common to player and reader, migrate function in sti_uniperif.c

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/sti/sti_uniperif.c    | 23 +++++++++++++++++++++++
 sound/soc/sti/uniperif.h        |  2 ++
 sound/soc/sti/uniperif_player.c | 34 +++-------------------------------
 sound/soc/sti/uniperif_reader.c | 15 +--------------
 4 files changed, 29 insertions(+), 45 deletions(-)

diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index ee91ae5..98eb205 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -7,6 +7,7 @@
 
 #include <linux/module.h>
 #include <linux/pinctrl/consumer.h>
+#include <linux/delay.h>
 
 #include "uniperif.h"
 
@@ -97,6 +98,28 @@ struct sti_uniperiph_dev_data {
 	{},
 };
 
+int  sti_uniperiph_reset(struct uniperif *uni)
+{
+	int count = 10;
+
+	/* Reset uniperipheral uni */
+	SET_UNIPERIF_SOFT_RST_SOFT_RST(uni);
+
+	if (uni->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
+		while (GET_UNIPERIF_SOFT_RST_SOFT_RST(uni) && count) {
+			udelay(5);
+			count--;
+		}
+	}
+
+	if (!count) {
+		dev_err(uni->dev, "Failed to reset uniperif\n");
+		return -EIO;
+	}
+
+	return 0;
+}
+
 int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots,
 			       int slot_width)
diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h
index 1993c65..d487dd2 100644
--- a/sound/soc/sti/uniperif.h
+++ b/sound/soc/sti/uniperif.h
@@ -1397,6 +1397,8 @@ static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
 	return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
 }
 
+int  sti_uniperiph_reset(struct uniperif *uni);
+
 int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots,
 			       int slot_width);
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 00a0fb3..d59ec90 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/delay.h>
 #include <linux/io.h>
 #include <linux/mfd/syscon.h>
 
@@ -55,25 +54,6 @@
 	.buffer_bytes_max = 256 * PAGE_SIZE
 };
 
-static inline int reset_player(struct uniperif *player)
-{
-	int count = 10;
-
-	if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
-		while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) {
-			udelay(5);
-			count--;
-		}
-	}
-
-	if (!count) {
-		dev_err(player->dev, "Failed to reset uniperif\n");
-		return -EIO;
-	}
-
-	return 0;
-}
-
 /*
  * uni_player_irq_handler
  * In case of error audio stream is stopped; stop action is protected via PCM
@@ -862,10 +842,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 
 	SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0);
 
-	/* Reset uniperipheral player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
 
-	return reset_player(player);
+	return sti_uniperiph_reset(player);
 }
 
 static int uni_player_start(struct uniperif *player)
@@ -897,10 +875,7 @@ static int uni_player_start(struct uniperif *player)
 		SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player);
 	}
 
-	/* Reset uniperipheral player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
-
-	ret = reset_player(player);
+	ret = sti_uniperiph_reset(player);
 	if (ret < 0) {
 		clk_disable_unprepare(player->clk);
 		return ret;
@@ -949,10 +924,7 @@ static int uni_player_stop(struct uniperif *player)
 	/* Turn the player off */
 	SET_UNIPERIF_CTRL_OPERATION_OFF(player);
 
-	/* Soft reset the player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
-
-	ret = reset_player(player);
+	ret = sti_uniperiph_reset(player);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 09314f8..59043f7 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/delay.h>
 #include <linux/io.h>
 
 #include <sound/soc.h>
@@ -186,7 +185,6 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	struct uniperif *reader = priv->dai_data.uni;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int transfer_size, trigger_limit, ret;
-	int count = 10;
 
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
@@ -288,18 +286,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	}
 
 	/* Reset uniperipheral reader */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(reader);
-
-	while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) {
-		udelay(5);
-		count--;
-	}
-	if (!count) {
-		dev_err(reader->dev, "Failed to reset uniperif\n");
-		return -EIO;
-	}
-
-	return 0;
+	return sti_uniperiph_reset(reader);
 }
 
 static int uni_reader_start(struct uniperif *reader)
-- 
1.9.1

Applied "ASoC: sti: reset refactoring" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti: reset refactoring

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 4c88f89f9c255d0a754e38ff1a55a6f8cef362e8 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:53 +0200
Subject: [PATCH] ASoC: sti: reset refactoring

Reset is common to player and reader, migrate function in sti_uniperif.c

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sti/sti_uniperif.c    | 23 +++++++++++++++++++++++
 sound/soc/sti/uniperif.h        |  2 ++
 sound/soc/sti/uniperif_player.c | 34 +++-------------------------------
 sound/soc/sti/uniperif_reader.c | 15 +--------------
 4 files changed, 29 insertions(+), 45 deletions(-)

diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c
index ee91ae5f812a..98eb205a0b62 100644
--- a/sound/soc/sti/sti_uniperif.c
+++ b/sound/soc/sti/sti_uniperif.c
@@ -7,6 +7,7 @@
 
 #include <linux/module.h>
 #include <linux/pinctrl/consumer.h>
+#include <linux/delay.h>
 
 #include "uniperif.h"
 
@@ -97,6 +98,28 @@ static const struct of_device_id snd_soc_sti_match[] = {
 	{},
 };
 
+int  sti_uniperiph_reset(struct uniperif *uni)
+{
+	int count = 10;
+
+	/* Reset uniperipheral uni */
+	SET_UNIPERIF_SOFT_RST_SOFT_RST(uni);
+
+	if (uni->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
+		while (GET_UNIPERIF_SOFT_RST_SOFT_RST(uni) && count) {
+			udelay(5);
+			count--;
+		}
+	}
+
+	if (!count) {
+		dev_err(uni->dev, "Failed to reset uniperif\n");
+		return -EIO;
+	}
+
+	return 0;
+}
+
 int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots,
 			       int slot_width)
diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h
index 1993c655fb79..d487dd2ef016 100644
--- a/sound/soc/sti/uniperif.h
+++ b/sound/soc/sti/uniperif.h
@@ -1397,6 +1397,8 @@ static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
 	return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
 }
 
+int  sti_uniperiph_reset(struct uniperif *uni);
+
 int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots,
 			       int slot_width);
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index c9b4670b772b..00022aa48280 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/delay.h>
 #include <linux/io.h>
 #include <linux/mfd/syscon.h>
 
@@ -55,25 +54,6 @@ static const struct snd_pcm_hardware uni_player_pcm_hw = {
 	.buffer_bytes_max = 256 * PAGE_SIZE
 };
 
-static inline int reset_player(struct uniperif *player)
-{
-	int count = 10;
-
-	if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
-		while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) {
-			udelay(5);
-			count--;
-		}
-	}
-
-	if (!count) {
-		dev_err(player->dev, "Failed to reset uniperif\n");
-		return -EIO;
-	}
-
-	return 0;
-}
-
 /*
  * uni_player_irq_handler
  * In case of error audio stream is stopped; stop action is protected via PCM
@@ -858,10 +838,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
 
 	SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0);
 
-	/* Reset uniperipheral player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
 
-	return reset_player(player);
+	return sti_uniperiph_reset(player);
 }
 
 static int uni_player_start(struct uniperif *player)
@@ -893,10 +871,7 @@ static int uni_player_start(struct uniperif *player)
 		SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player);
 	}
 
-	/* Reset uniperipheral player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
-
-	ret = reset_player(player);
+	ret = sti_uniperiph_reset(player);
 	if (ret < 0) {
 		clk_disable_unprepare(player->clk);
 		return ret;
@@ -945,10 +920,7 @@ static int uni_player_stop(struct uniperif *player)
 	/* Turn the player off */
 	SET_UNIPERIF_CTRL_OPERATION_OFF(player);
 
-	/* Soft reset the player */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
-
-	ret = reset_player(player);
+	ret = sti_uniperiph_reset(player);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 09314f8be841..59043f7a0e5c 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/delay.h>
 #include <linux/io.h>
 
 #include <sound/soc.h>
@@ -186,7 +185,6 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	struct uniperif *reader = priv->dai_data.uni;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int transfer_size, trigger_limit, ret;
-	int count = 10;
 
 	/* The reader should be stopped */
 	if (reader->state != UNIPERIF_STATE_STOPPED) {
@@ -288,18 +286,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
 	}
 
 	/* Reset uniperipheral reader */
-	SET_UNIPERIF_SOFT_RST_SOFT_RST(reader);
-
-	while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) {
-		udelay(5);
-		count--;
-	}
-	if (!count) {
-		dev_err(reader->dev, "Failed to reset uniperif\n");
-		return -EIO;
-	}
-
-	return 0;
+	return sti_uniperiph_reset(reader);
 }
 
 static int uni_reader_start(struct uniperif *reader)
-- 
2.8.1

[PATCH 4/7] ASoC: sti: clean unused include

[Arnaud Pouliquen]

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/sti/uniperif_player.c | 1 -
 sound/soc/sti/uniperif_reader.c | 3 ---
 2 files changed, 4 deletions(-)

diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index d59ec90..60ae31a 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/io.h>
 #include <linux/mfd/syscon.h>
 
 #include <sound/asoundef.h>
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 59043f7..5992c6a 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -5,9 +5,6 @@
  * License terms:  GNU General Public License (GPL), version 2
  */
 
-#include <linux/clk.h>
-#include <linux/io.h>
-
 #include <sound/soc.h>
 
 #include "uniperif.h"
-- 
1.9.1

Applied "ASoC: sti: clean unused include" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti: clean unused include

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 4db61af068f50948a41b32a32fc3361f7ad152df Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:54 +0200
Subject: [PATCH] ASoC: sti: clean unused include

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sti/uniperif_player.c | 1 -
 sound/soc/sti/uniperif_reader.c | 3 ---
 2 files changed, 4 deletions(-)

diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 00022aa48280..bea352a1504e 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -6,7 +6,6 @@
  */
 
 #include <linux/clk.h>
-#include <linux/io.h>
 #include <linux/mfd/syscon.h>
 
 #include <sound/asoundef.h>
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 59043f7a0e5c..5992c6ab3833 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -5,9 +5,6 @@
  * License terms:  GNU General Public License (GPL), version 2
  */
 
-#include <linux/clk.h>
-#include <linux/io.h>
-
 #include <sound/soc.h>
 
 #include "uniperif.h"
-- 
2.8.1

[PATCH 5/7] ASoC: codec: clean legacy in sti-sas

[Arnaud Pouliquen]

stih416 is no more supported, clean associated code.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/codecs/sti-sas.c | 169 ++++-----------------------------------------
 1 file changed, 15 insertions(+), 154 deletions(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 7b31ee9..1488f4f 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -14,28 +14,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-/* chipID supported */
-#define CHIPID_STIH416 0
-#define CHIPID_STIH407 1
-
 /* DAC definitions */
 
-/* stih416 DAC registers */
-/* sysconf 2517: Audio-DAC-Control */
-#define STIH416_AUDIO_DAC_CTRL 0x00000814
-/* sysconf 2519: Audio-Gue-Control */
-#define STIH416_AUDIO_GLUE_CTRL 0x0000081C
-
-#define STIH416_DAC_NOT_STANDBY	0x3
-#define STIH416_DAC_SOFTMUTE	0x4
-#define STIH416_DAC_ANA_NOT_PWR	0x5
-#define STIH416_DAC_NOT_PNDBG	0x6
-
-#define STIH416_DAC_NOT_STANDBY_MASK	BIT(STIH416_DAC_NOT_STANDBY)
-#define STIH416_DAC_SOFTMUTE_MASK	BIT(STIH416_DAC_SOFTMUTE)
-#define STIH416_DAC_ANA_NOT_PWR_MASK	BIT(STIH416_DAC_ANA_NOT_PWR)
-#define STIH416_DAC_NOT_PNDBG_MASK	BIT(STIH416_DAC_NOT_PNDBG)
-
 /* stih407 DAC registers */
 /* sysconf 5041: Audio-Gue-Control */
 #define STIH407_AUDIO_GLUE_CTRL 0x000000A4
@@ -63,14 +43,9 @@ enum {
 	STI_SAS_DAI_ANALOG_OUT,
 };
 
-static const struct reg_default stih416_sas_reg_defaults[] = {
-	{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
-	{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
-};
-
 static const struct reg_default stih407_sas_reg_defaults[] = {
-	{ STIH416_AUDIO_DAC_CTRL, 0x000000000 },
-	{ STIH416_AUDIO_GLUE_CTRL, 0x00000040 },
+	{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
+	{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
 };
 
 struct sti_dac_audio {
@@ -89,7 +64,6 @@ struct sti_spdif_audio {
 
 /* device data structure */
 struct sti_sas_dev_data {
-	const int chipid; /* IC version */
 	const struct regmap_config *regmap;
 	const struct snd_soc_dai_ops *dac_ops;  /* DAC function callbacks */
 	const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */
@@ -155,43 +129,19 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 	}
 
 	/* Init DAC configuration */
-	switch (data->dev_data->chipid) {
-	case CHIPID_STIH407:
-		/* init configuration */
-		ret =  snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-					   STIH407_DAC_STANDBY_MASK,
-					   STIH407_DAC_STANDBY_MASK);
-
-		if (!ret)
-			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-						  STIH407_DAC_STANDBY_ANA_MASK,
-						  STIH407_DAC_STANDBY_ANA_MASK);
-		if (!ret)
-			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-						  STIH407_DAC_SOFTMUTE_MASK,
-						  STIH407_DAC_SOFTMUTE_MASK);
-		break;
-	case CHIPID_STIH416:
-		ret =  snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					   STIH416_DAC_NOT_STANDBY_MASK, 0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_ANA_NOT_PWR, 0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_NOT_PNDBG_MASK,
-						   0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_SOFTMUTE_MASK,
-						   STIH416_DAC_SOFTMUTE_MASK);
-		break;
-	default:
-		return -EINVAL;
-	}
+	/* init configuration */
+	ret =  snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+				   STIH407_DAC_STANDBY_MASK,
+				   STIH407_DAC_STANDBY_MASK);
+
+	if (!ret)
+		ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					  STIH407_DAC_STANDBY_ANA_MASK,
+					  STIH407_DAC_STANDBY_ANA_MASK);
+	if (!ret)
+		ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					  STIH407_DAC_SOFTMUTE_MASK,
+					  STIH407_DAC_SOFTMUTE_MASK);
 
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to update DAC registers");
@@ -217,37 +167,6 @@ static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int stih416_dac_probe(struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
-	struct sti_dac_audio *dac = &drvdata->dac;
-
-	/* Get reset control */
-	dac->rst = devm_reset_control_get(codec->dev, "dac_rst");
-	if (IS_ERR(dac->rst)) {
-		dev_err(dai->codec->dev,
-			"%s: ERROR: DAC reset control not defined !\n",
-			__func__);
-		dac->rst = NULL;
-		return -EFAULT;
-	}
-	/* Put the DAC into reset */
-	reset_control_assert(dac->rst);
-
-	return 0;
-}
-
-static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = {
-	SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL,
-			 STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0),
-	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL,
-			     STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0),
-	SND_SOC_DAPM_DAC("DAC standby",  "dac_p", STIH416_AUDIO_DAC_CTRL,
-			 STIH416_DAC_NOT_STANDBY, 0),
-	SND_SOC_DAPM_OUTPUT("DAC Output"),
-};
-
 static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
 	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL,
 			     STIH407_DAC_STANDBY_ANA, 1, NULL, 0),
@@ -256,30 +175,11 @@ static int stih416_dac_probe(struct snd_soc_dai *dai)
 	SND_SOC_DAPM_OUTPUT("DAC Output"),
 };
 
-static const struct snd_soc_dapm_route stih416_sas_route[] = {
-	{"DAC Output", NULL, "DAC bandgap"},
-	{"DAC Output", NULL, "DAC standby ana"},
-	{"DAC standby ana", NULL, "DAC standby"},
-};
-
 static const struct snd_soc_dapm_route stih407_sas_route[] = {
 	{"DAC Output", NULL, "DAC standby ana"},
 	{"DAC standby ana", NULL, "DAC standby"},
 };
 
-static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	if (mute) {
-		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					    STIH416_DAC_SOFTMUTE_MASK,
-					    STIH416_DAC_SOFTMUTE_MASK);
-	} else {
-		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					    STIH416_DAC_SOFTMUTE_MASK, 0);
-	}
-}
 
 static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
@@ -407,13 +307,6 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops stih416_dac_ops = {
-	.set_fmt = sti_sas_dac_set_fmt,
-	.mute_stream = stih416_sas_dac_mute,
-	.prepare = sti_sas_prepare,
-	.set_sysclk = sti_sas_set_sysclk,
-};
-
 static const struct snd_soc_dai_ops stih407_dac_ops = {
 	.set_fmt = sti_sas_dac_set_fmt,
 	.mute_stream = stih407_sas_dac_mute,
@@ -434,31 +327,7 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 	.reg_write = sti_sas_write_reg,
 };
 
-static const struct regmap_config stih416_sas_regmap = {
-	.reg_bits = 32,
-	.val_bits = 32,
-
-	.max_register = STIH416_AUDIO_DAC_CTRL,
-	.reg_defaults = stih416_sas_reg_defaults,
-	.num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults),
-	.volatile_reg = sti_sas_volatile_register,
-	.cache_type = REGCACHE_RBTREE,
-	.reg_read = sti_sas_read_reg,
-	.reg_write = sti_sas_write_reg,
-};
-
-static const struct sti_sas_dev_data stih416_data = {
-	.chipid = CHIPID_STIH416,
-	.regmap = &stih416_sas_regmap,
-	.dac_ops = &stih416_dac_ops,
-	.dapm_widgets = stih416_sas_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets),
-	.dapm_routes =	stih416_sas_route,
-	.num_dapm_routes = ARRAY_SIZE(stih416_sas_route),
-};
-
 static const struct sti_sas_dev_data stih407_data = {
-	.chipid = CHIPID_STIH407,
 	.regmap = &stih407_sas_regmap,
 	.dac_ops = &stih407_dac_ops,
 	.dapm_widgets = stih407_sas_dapm_widgets,
@@ -533,10 +402,6 @@ static int sti_sas_codec_probe(struct snd_soc_codec *codec)
 
 static const struct of_device_id sti_sas_dev_match[] = {
 	{
-		.compatible = "st,stih416-sas-codec",
-		.data = &stih416_data,
-	},
-	{
 		.compatible = "st,stih407-sas-codec",
 		.data = &stih407_data,
 	},
@@ -584,10 +449,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
 	}
 	drvdata->spdif.regmap = drvdata->dac.regmap;
 
-	/* Set DAC dai probe */
-	if (drvdata->dev_data->chipid == CHIPID_STIH416)
-		sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe;
-
 	sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops;
 
 	/* Set dapms*/
-- 
1.9.1

Applied "ASoC: sti-sas: clean legacy in sti-sas" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti-sas: clean legacy in sti-sas

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 165a57a3df0206b5609502d37e907944d8eb06ee Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:55 +0200
Subject: [PATCH] ASoC: sti-sas: clean legacy in sti-sas

stih416 is no more supported, clean associated code.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sti-sas.c | 169 ++++-----------------------------------------
 1 file changed, 15 insertions(+), 154 deletions(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 7b31ee9b82bc..1488f4fb1c5e 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -14,28 +14,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-/* chipID supported */
-#define CHIPID_STIH416 0
-#define CHIPID_STIH407 1
-
 /* DAC definitions */
 
-/* stih416 DAC registers */
-/* sysconf 2517: Audio-DAC-Control */
-#define STIH416_AUDIO_DAC_CTRL 0x00000814
-/* sysconf 2519: Audio-Gue-Control */
-#define STIH416_AUDIO_GLUE_CTRL 0x0000081C
-
-#define STIH416_DAC_NOT_STANDBY	0x3
-#define STIH416_DAC_SOFTMUTE	0x4
-#define STIH416_DAC_ANA_NOT_PWR	0x5
-#define STIH416_DAC_NOT_PNDBG	0x6
-
-#define STIH416_DAC_NOT_STANDBY_MASK	BIT(STIH416_DAC_NOT_STANDBY)
-#define STIH416_DAC_SOFTMUTE_MASK	BIT(STIH416_DAC_SOFTMUTE)
-#define STIH416_DAC_ANA_NOT_PWR_MASK	BIT(STIH416_DAC_ANA_NOT_PWR)
-#define STIH416_DAC_NOT_PNDBG_MASK	BIT(STIH416_DAC_NOT_PNDBG)
-
 /* stih407 DAC registers */
 /* sysconf 5041: Audio-Gue-Control */
 #define STIH407_AUDIO_GLUE_CTRL 0x000000A4
@@ -63,14 +43,9 @@ enum {
 	STI_SAS_DAI_ANALOG_OUT,
 };
 
-static const struct reg_default stih416_sas_reg_defaults[] = {
-	{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
-	{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
-};
-
 static const struct reg_default stih407_sas_reg_defaults[] = {
-	{ STIH416_AUDIO_DAC_CTRL, 0x000000000 },
-	{ STIH416_AUDIO_GLUE_CTRL, 0x00000040 },
+	{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
+	{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
 };
 
 struct sti_dac_audio {
@@ -89,7 +64,6 @@ struct sti_spdif_audio {
 
 /* device data structure */
 struct sti_sas_dev_data {
-	const int chipid; /* IC version */
 	const struct regmap_config *regmap;
 	const struct snd_soc_dai_ops *dac_ops;  /* DAC function callbacks */
 	const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */
@@ -155,43 +129,19 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 	}
 
 	/* Init DAC configuration */
-	switch (data->dev_data->chipid) {
-	case CHIPID_STIH407:
-		/* init configuration */
-		ret =  snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-					   STIH407_DAC_STANDBY_MASK,
-					   STIH407_DAC_STANDBY_MASK);
-
-		if (!ret)
-			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-						  STIH407_DAC_STANDBY_ANA_MASK,
-						  STIH407_DAC_STANDBY_ANA_MASK);
-		if (!ret)
-			ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
-						  STIH407_DAC_SOFTMUTE_MASK,
-						  STIH407_DAC_SOFTMUTE_MASK);
-		break;
-	case CHIPID_STIH416:
-		ret =  snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					   STIH416_DAC_NOT_STANDBY_MASK, 0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_ANA_NOT_PWR, 0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_NOT_PNDBG_MASK,
-						   0);
-		if (!ret)
-			ret =  snd_soc_update_bits(codec,
-						   STIH416_AUDIO_DAC_CTRL,
-						   STIH416_DAC_SOFTMUTE_MASK,
-						   STIH416_DAC_SOFTMUTE_MASK);
-		break;
-	default:
-		return -EINVAL;
-	}
+	/* init configuration */
+	ret =  snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+				   STIH407_DAC_STANDBY_MASK,
+				   STIH407_DAC_STANDBY_MASK);
+
+	if (!ret)
+		ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					  STIH407_DAC_STANDBY_ANA_MASK,
+					  STIH407_DAC_STANDBY_ANA_MASK);
+	if (!ret)
+		ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
+					  STIH407_DAC_SOFTMUTE_MASK,
+					  STIH407_DAC_SOFTMUTE_MASK);
 
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to update DAC registers");
@@ -217,37 +167,6 @@ static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int stih416_dac_probe(struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
-	struct sti_dac_audio *dac = &drvdata->dac;
-
-	/* Get reset control */
-	dac->rst = devm_reset_control_get(codec->dev, "dac_rst");
-	if (IS_ERR(dac->rst)) {
-		dev_err(dai->codec->dev,
-			"%s: ERROR: DAC reset control not defined !\n",
-			__func__);
-		dac->rst = NULL;
-		return -EFAULT;
-	}
-	/* Put the DAC into reset */
-	reset_control_assert(dac->rst);
-
-	return 0;
-}
-
-static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = {
-	SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL,
-			 STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0),
-	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL,
-			     STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0),
-	SND_SOC_DAPM_DAC("DAC standby",  "dac_p", STIH416_AUDIO_DAC_CTRL,
-			 STIH416_DAC_NOT_STANDBY, 0),
-	SND_SOC_DAPM_OUTPUT("DAC Output"),
-};
-
 static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
 	SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL,
 			     STIH407_DAC_STANDBY_ANA, 1, NULL, 0),
@@ -256,30 +175,11 @@ static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("DAC Output"),
 };
 
-static const struct snd_soc_dapm_route stih416_sas_route[] = {
-	{"DAC Output", NULL, "DAC bandgap"},
-	{"DAC Output", NULL, "DAC standby ana"},
-	{"DAC standby ana", NULL, "DAC standby"},
-};
-
 static const struct snd_soc_dapm_route stih407_sas_route[] = {
 	{"DAC Output", NULL, "DAC standby ana"},
 	{"DAC standby ana", NULL, "DAC standby"},
 };
 
-static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	if (mute) {
-		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					    STIH416_DAC_SOFTMUTE_MASK,
-					    STIH416_DAC_SOFTMUTE_MASK);
-	} else {
-		return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
-					    STIH416_DAC_SOFTMUTE_MASK, 0);
-	}
-}
 
 static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
@@ -407,13 +307,6 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static const struct snd_soc_dai_ops stih416_dac_ops = {
-	.set_fmt = sti_sas_dac_set_fmt,
-	.mute_stream = stih416_sas_dac_mute,
-	.prepare = sti_sas_prepare,
-	.set_sysclk = sti_sas_set_sysclk,
-};
-
 static const struct snd_soc_dai_ops stih407_dac_ops = {
 	.set_fmt = sti_sas_dac_set_fmt,
 	.mute_stream = stih407_sas_dac_mute,
@@ -434,31 +327,7 @@ static const struct regmap_config stih407_sas_regmap = {
 	.reg_write = sti_sas_write_reg,
 };
 
-static const struct regmap_config stih416_sas_regmap = {
-	.reg_bits = 32,
-	.val_bits = 32,
-
-	.max_register = STIH416_AUDIO_DAC_CTRL,
-	.reg_defaults = stih416_sas_reg_defaults,
-	.num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults),
-	.volatile_reg = sti_sas_volatile_register,
-	.cache_type = REGCACHE_RBTREE,
-	.reg_read = sti_sas_read_reg,
-	.reg_write = sti_sas_write_reg,
-};
-
-static const struct sti_sas_dev_data stih416_data = {
-	.chipid = CHIPID_STIH416,
-	.regmap = &stih416_sas_regmap,
-	.dac_ops = &stih416_dac_ops,
-	.dapm_widgets = stih416_sas_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets),
-	.dapm_routes =	stih416_sas_route,
-	.num_dapm_routes = ARRAY_SIZE(stih416_sas_route),
-};
-
 static const struct sti_sas_dev_data stih407_data = {
-	.chipid = CHIPID_STIH407,
 	.regmap = &stih407_sas_regmap,
 	.dac_ops = &stih407_dac_ops,
 	.dapm_widgets = stih407_sas_dapm_widgets,
@@ -533,10 +402,6 @@ static struct snd_soc_codec_driver sti_sas_driver = {
 
 static const struct of_device_id sti_sas_dev_match[] = {
 	{
-		.compatible = "st,stih416-sas-codec",
-		.data = &stih416_data,
-	},
-	{
 		.compatible = "st,stih407-sas-codec",
 		.data = &stih407_data,
 	},
@@ -584,10 +449,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
 	}
 	drvdata->spdif.regmap = drvdata->dac.regmap;
 
-	/* Set DAC dai probe */
-	if (drvdata->dev_data->chipid == CHIPID_STIH416)
-		sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe;
-
 	sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops;
 
 	/* Set dapms*/
-- 
2.8.1

[PATCH 6/7] ASoC: sti: sti-sas: add missing return in messages strings

[Arnaud Pouliquen]

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/codecs/sti-sas.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 1488f4f..21d087b 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -124,7 +124,7 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 		ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
 					  SPDIF_BIPHASE_IDLE_MASK, 0);
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to update SPDIF registers");
+		dev_err(codec->dev, "Failed to update SPDIF registers\n");
 		return ret;
 	}
 
@@ -144,7 +144,7 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 					  STIH407_DAC_SOFTMUTE_MASK);
 
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to update DAC registers");
+		dev_err(codec->dev, "Failed to update DAC registers\n");
 		return ret;
 	}
 
@@ -292,13 +292,13 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 	switch (dai->id) {
 	case STI_SAS_DAI_SPDIF_OUT:
 		if ((drvdata->spdif.mclk / runtime->rate) != 128) {
-			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			dev_err(codec->dev, "unexpected mclk-fs ratio\n");
 			return -EINVAL;
 		}
 		break;
 	case STI_SAS_DAI_ANALOG_OUT:
 		if ((drvdata->dac.mclk / runtime->rate) != 256) {
-			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			dev_err(codec->dev, "unexpected mclk-fs ratio\n");
 			return -EINVAL;
 		}
 		break;
@@ -423,7 +423,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
 	/* Populate data structure depending on compatibility */
 	of_id = of_match_node(sti_sas_dev_match, pnode);
 	if (!of_id->data) {
-		dev_err(&pdev->dev, "data associated to device is missing");
+		dev_err(&pdev->dev, "data associated to device is missing\n");
 		return -EINVAL;
 	}
 
-- 
1.9.1

Applied "ASoC: sti-sas: add missing return in messages strings" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti-sas: add missing return in messages strings

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 92591efabc013fa791f96df881aafcc104ba759d Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:56 +0200
Subject: [PATCH] ASoC: sti-sas: add missing return in messages strings

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sti-sas.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 1488f4fb1c5e..21d087be2e93 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -124,7 +124,7 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 		ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
 					  SPDIF_BIPHASE_IDLE_MASK, 0);
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to update SPDIF registers");
+		dev_err(codec->dev, "Failed to update SPDIF registers\n");
 		return ret;
 	}
 
@@ -144,7 +144,7 @@ static int  sti_sas_init_sas_registers(struct snd_soc_codec *codec,
 					  STIH407_DAC_SOFTMUTE_MASK);
 
 	if (ret < 0) {
-		dev_err(codec->dev, "Failed to update DAC registers");
+		dev_err(codec->dev, "Failed to update DAC registers\n");
 		return ret;
 	}
 
@@ -292,13 +292,13 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 	switch (dai->id) {
 	case STI_SAS_DAI_SPDIF_OUT:
 		if ((drvdata->spdif.mclk / runtime->rate) != 128) {
-			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			dev_err(codec->dev, "unexpected mclk-fs ratio\n");
 			return -EINVAL;
 		}
 		break;
 	case STI_SAS_DAI_ANALOG_OUT:
 		if ((drvdata->dac.mclk / runtime->rate) != 256) {
-			dev_err(codec->dev, "unexpected mclk-fs ratio");
+			dev_err(codec->dev, "unexpected mclk-fs ratio\n");
 			return -EINVAL;
 		}
 		break;
@@ -423,7 +423,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
 	/* Populate data structure depending on compatibility */
 	of_id = of_match_node(sti_sas_dev_match, pnode);
 	if (!of_id->data) {
-		dev_err(&pdev->dev, "data associated to device is missing");
+		dev_err(&pdev->dev, "data associated to device is missing\n");
 		return -EINVAL;
 	}
 
-- 
2.8.1

[PATCH 7/7] ASoc: codec: sti enable fast io for regmap

[Arnaud Pouliquen]

Some registers accesses are done in atomic context.
Enable fast io to use spinlock instead of mutex to protect access.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
---
 sound/soc/codecs/sti-sas.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 21d087b..62c6187 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -317,7 +317,7 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
 static const struct regmap_config stih407_sas_regmap = {
 	.reg_bits = 32,
 	.val_bits = 32,
-
+	.fast_io = true,
 	.max_register = STIH407_AUDIO_DAC_CTRL,
 	.reg_defaults = stih407_sas_reg_defaults,
 	.num_reg_defaults = ARRAY_SIZE(stih407_sas_reg_defaults),
-- 
1.9.1

Applied "ASoC: sti-sas: enable fast io for regmap" to the asoc tree

[Mark Brown]

The patch

   ASoC: sti-sas: enable fast io for regmap

has been applied to the asoc tree at

   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.  

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark

>From 7e235deb69dc7b1c4b5e1ac63a3157ef98ceeff3 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 24 Oct 2016 16:42:57 +0200
Subject: [PATCH] ASoC: sti-sas: enable fast io for regmap

Some registers accesses are done in atomic context.
Enable fast io to use spinlock instead of mutex to protect access.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/sti-sas.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index 7b31ee9b82bc..d6e00c77edcd 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -424,7 +424,7 @@ static const struct snd_soc_dai_ops stih407_dac_ops = {
 static const struct regmap_config stih407_sas_regmap = {
 	.reg_bits = 32,
 	.val_bits = 32,
-
+	.fast_io = true,
 	.max_register = STIH407_AUDIO_DAC_CTRL,
 	.reg_defaults = stih407_sas_reg_defaults,
 	.num_reg_defaults = ARRAY_SIZE(stih407_sas_reg_defaults),
-- 
2.8.1

Re: [PATCH 0/7] ASoC: sti audio fixes and legacy cleaning

[Mark Brown]

[-- Attachment #1.1: Type: text/plain, Size: 450 bytes --]

On Mon, Oct 24, 2016 at 04:42:50PM +0200, Arnaud Pouliquen wrote:
> Audio fixes and cleaning of code associated to stih416 platform.

Please don't mix fixes and cleanups like this, bug fixes ought to get
sent to Linus so they get into the first release they can while cleanups
can wait for the next merge window so you should make sure that any
fixes come at the start of the series in order to ensure that there are
no dependencies on the cleanups.

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