From 6cd249cfad68a231336983e2216d75b3ddfde1d6 Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Mon, 8 Jul 2019 22:19:01 +0800
Subject: ASoC: max98357a: use mdelay for sdmode-delay

max98357a_daiops_trigger() is possible to be called in atomic context if
the .nonatomic flag is equal to 0 in the DAI links.

When cancel_delayed_work_sync() in max98357a_daiops_trigger() is called
in atomic context, kernel emits the following message: "BUG: sleeping
function called from invalid context".

According to the DT binding document, value less than or equal to 5ms of
sdmod-delay should be sufficient to avoid the pop noise.  Use mdelay
(i.e. busy loop) for such low delay should be acceptable.

Fixes: cec5b01f8f1c ("ASoC: max98357a: avoid speaker pop when playback
startup")

Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190708141901.68797-1-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98357a.c | 25 ++++---------------------
 1 file changed, 4 insertions(+), 21 deletions(-)

diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 6f0e28f903bf..16313b973eaa 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -20,20 +20,10 @@
 #include <sound/soc-dapm.h>
 
 struct max98357a_priv {
-	struct delayed_work enable_sdmode_work;
 	struct gpio_desc *sdmode;
 	unsigned int sdmode_delay;
 };
 
-static void max98357a_enable_sdmode_work(struct work_struct *work)
-{
-	struct max98357a_priv *max98357a =
-	container_of(work, struct max98357a_priv,
-			enable_sdmode_work.work);
-
-	gpiod_set_value(max98357a->sdmode, 1);
-}
-
 static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 		int cmd, struct snd_soc_dai *dai)
 {
@@ -46,14 +36,12 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		queue_delayed_work(system_power_efficient_wq,
-				&max98357a->enable_sdmode_work,
-				msecs_to_jiffies(max98357a->sdmode_delay));
+		mdelay(max98357a->sdmode_delay);
+		gpiod_set_value(max98357a->sdmode, 1);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		cancel_delayed_work_sync(&max98357a->enable_sdmode_work);
 		gpiod_set_value(max98357a->sdmode, 0);
 		break;
 	}
@@ -112,30 +100,25 @@ static int max98357a_platform_probe(struct platform_device *pdev)
 	int ret;
 
 	max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL);
-
 	if (!max98357a)
 		return -ENOMEM;
 
 	max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev,
 				"sdmode", GPIOD_OUT_LOW);
-
 	if (IS_ERR(max98357a->sdmode))
 		return PTR_ERR(max98357a->sdmode);
 
 	ret = device_property_read_u32(&pdev->dev, "sdmode-delay",
 					&max98357a->sdmode_delay);
-
 	if (ret) {
 		max98357a->sdmode_delay = 0;
 		dev_dbg(&pdev->dev,
-			"no optional property 'sdmode-delay' found, default: no delay\n");
+			"no optional property 'sdmode-delay' found, "
+			"default: no delay\n");
 	}
 
 	dev_set_drvdata(&pdev->dev, max98357a);
 
-	INIT_DELAYED_WORK(&max98357a->enable_sdmode_work,
-				max98357a_enable_sdmode_work);
-
 	return devm_snd_soc_register_component(&pdev->dev,
 			&max98357a_component_driver,
 			&max98357a_dai_driver, 1);
-- 
cgit v1.2.3


From 72365164cbefe3afa7a146d27d502ed688bf7323 Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Tue, 9 Jul 2019 10:22:16 -0700
Subject: ASoC: rt1308: Remove executable attribute from source files

These are source files not executable.

Signed-off-by: Joe Perches <joe@perches.com>
Link: https://lore.kernel.org/r/d198a3e6ed3a0e9070afeb6aca69903c3e985149.camel@perches.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1308.c | 0
 sound/soc/codecs/rt1308.h | 0
 2 files changed, 0 insertions(+), 0 deletions(-)
 mode change 100755 => 100644 sound/soc/codecs/rt1308.c
 mode change 100755 => 100644 sound/soc/codecs/rt1308.h

diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
old mode 100755
new mode 100644
diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h
old mode 100755
new mode 100644
-- 
cgit v1.2.3


From 9e944c9be2456159fb8c36b0ba3170b2f01c3887 Mon Sep 17 00:00:00 2001
From: Kirill Marinushkin <kmarinushkin@birdec.com>
Date: Wed, 10 Jul 2019 07:51:35 +0200
Subject: ASoC: Relocate my e-mail to .com domain zone

Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.com>
Link: https://lore.kernel.org/r/20190710055135.21377-1-kmarinushkin@birdec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 MAINTAINERS                    | 2 +-
 sound/soc/codecs/pcm3060-i2c.c | 4 ++--
 sound/soc/codecs/pcm3060-spi.c | 4 ++--
 sound/soc/codecs/pcm3060.c     | 4 ++--
 sound/soc/codecs/pcm3060.h     | 2 +-
 5 files changed, 8 insertions(+), 8 deletions(-)

diff --git a/MAINTAINERS b/MAINTAINERS
index 3e75361f9b3b..11db05b56744 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -15795,7 +15795,7 @@ S:	Maintained
 F:	drivers/net/ethernet/ti/netcp*
 
 TI PCM3060 ASoC CODEC DRIVER
-M:	Kirill Marinushkin <kmarinushkin@birdec.tech>
+M:	Kirill Marinushkin <kmarinushkin@birdec.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	Documentation/devicetree/bindings/sound/pcm3060.txt
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
index cdc8314882bc..abcdeb922201 100644
--- a/sound/soc/codecs/pcm3060-i2c.c
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 I2C driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/i2c.h>
 #include <linux/module.h>
@@ -56,5 +56,5 @@ static struct i2c_driver pcm3060_i2c_driver = {
 module_i2c_driver(pcm3060_i2c_driver);
 
 MODULE_DESCRIPTION("PCM3060 I2C driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
index f6f19fa80932..3b79734b832b 100644
--- a/sound/soc/codecs/pcm3060-spi.c
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 SPI driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <linux/spi/spi.h>
@@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = {
 module_spi_driver(pcm3060_spi_driver);
 
 MODULE_DESCRIPTION("PCM3060 SPI driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
index 32b26f1c2282..b2358069cf9b 100644
--- a/sound/soc/codecs/pcm3060.c
+++ b/sound/soc/codecs/pcm3060.c
@@ -2,7 +2,7 @@
 //
 // PCM3060 codec driver
 //
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
 
 #include <linux/module.h>
 #include <sound/pcm_params.h>
@@ -342,5 +342,5 @@ int pcm3060_probe(struct device *dev)
 EXPORT_SYMBOL(pcm3060_probe);
 
 MODULE_DESCRIPTION("PCM3060 codec driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
index 75931c9a9d85..18d51e5dac2c 100644
--- a/sound/soc/codecs/pcm3060.h
+++ b/sound/soc/codecs/pcm3060.h
@@ -2,7 +2,7 @@
 /*
  * PCM3060 codec driver
  *
- * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
  */
 
 #ifndef _SND_SOC_PCM3060_H
-- 
cgit v1.2.3


From 52db6685932e326ed607644ab7ebdae8c194adda Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 10 Jul 2019 16:59:55 +0900
Subject: ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai()

props->xxx_dai might be NULL when DPCM.
This patch cares it for debug.

Fixes: commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o922gw4u.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/simple_card_utils.h | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 954563ee2277..985a5f583de4 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -141,6 +141,10 @@ inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv,
 {
 	struct device *dev = simple_priv_to_dev(priv);
 
+	/* dai might be NULL */
+	if (!dai)
+		return;
+
 	if (dai->name)
 		dev_dbg(dev, "%s dai name = %s\n",
 			name, dai->name);
-- 
cgit v1.2.3


From 794fcee8da3c0c8a01b08ecad1c181cb0a622868 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 10 Jul 2019 17:01:12 +0900
Subject: ASoC: simple-card-utils: care no Platform for DPCM

commit 34614739988ad ("ASoC: soc-core: support dai_link with
platforms_num != 1") supports multi Platform, and
commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform
possibility") removed no Platform from simple-card.

Multi Platform is now checking both Platform name/of_node are NULL case.
But in normal case, DPCM be doesn't have Platform.

asoc_simple_canonicalize_platform() try to use CPU of_node
to Platform (This is needed for DMAEngine platform case),
but it still might be NULL at DPCM be.

This patch try to use no Platform after that if Platform of_node
is still NULL. It can't probe without this patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87muhmgw2o.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card-utils.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ac8678fe55ff..556b1a789629 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -349,6 +349,13 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link)
 	/* Assumes platform == cpu */
 	if (!dai_link->platforms->of_node)
 		dai_link->platforms->of_node = dai_link->cpus->of_node;
+
+	/*
+	 * DPCM BE can be no platform.
+	 * Alloced memory will be waste, but not leak.
+	 */
+	if (!dai_link->platforms->of_node)
+		dai_link->num_platforms = 0;
 }
 EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform);
 
-- 
cgit v1.2.3


From 724808ad556c15e9473418d082f8aae81dd267f6 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Wed, 10 Jul 2019 15:25:06 +0800
Subject: ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm()

The node variable is still being used after the of_node_put() call,
which may result in use-after-free.

Fixes: cfc652a73331 ("ASoC: simple-card: tidyup prefix for snd_soc_codec_conf")
Link: https://lore.kernel.org/r/1562743509-30496-2-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index e5cde0d5e63c..4117e54884e5 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -124,8 +124,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	li->link++;
 
-	of_node_put(node);
-
 	/* For single DAI link & old style of DT node */
 	if (is_top)
 		prefix = PREFIX;
@@ -147,17 +145,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
 	} else {
@@ -180,17 +178,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(np, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -208,19 +206,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(np, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, node, codec,
 				       prefix, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &simple_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(node);
+	return ret;
 }
 
 static int simple_dai_link_of(struct asoc_simple_priv *priv,
-- 
cgit v1.2.3


From 27862d5a3325bc531ec15e3c607e44aa0fd57f6f Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Wed, 10 Jul 2019 15:25:07 +0800
Subject: ASoC: simple-card: fix an use-after-free in simple_for_each_link()

The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.

Fixes: d947cdfd4be2 ("ASoC: simple-card: cleanup DAI link loop method - step1")
Link: https://lore.kernel.org/r/1562743509-30496-3-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 4117e54884e5..ef849151ba56 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -364,8 +364,6 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 			goto error;
 		}
 
-		of_node_put(codec);
-
 		/* get convert-xxx property */
 		memset(&adata, 0, sizeof(adata));
 		for_each_child_of_node(node, np)
@@ -387,11 +385,13 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 				ret = func_noml(priv, np, codec, li, is_top);
 
 			if (ret < 0) {
+				of_node_put(codec);
 				of_node_put(np);
 				goto error;
 			}
 		}
 
+		of_node_put(codec);
 		node = of_get_next_child(top, node);
 	} while (!is_top && node);
 
-- 
cgit v1.2.3


From aa2e362cb6b3f5ca88093ada01e1a0ace8a517b2 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Wed, 10 Jul 2019 15:25:08 +0800
Subject: ASoC: audio-graph-card: fix use-after-free in
 graph_dai_link_of_dpcm()

After calling of_node_put() on the ports, port, and node variables,
they are still being used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.

Fixes: dd98fbc558a0 ("ASoC: audio-graph-card: cleanup DAI link loop method - step1")
Link: https://lore.kernel.org/r/1562743509-30496-4-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index e438011f5e45..bddfcfd7bedf 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -208,10 +208,6 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	dev_dbg(dev, "link_of DPCM (%pOF)\n", ep);
 
-	of_node_put(ports);
-	of_node_put(port);
-	of_node_put(node);
-
 	if (li->cpu) {
 		int is_single_links = 0;
 
@@ -229,17 +225,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links);
 		if (ret)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "fe.%s",
 						   cpus->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* card->num_links includes Codec */
 		asoc_simple_canonicalize_cpu(dai_link, is_single_links);
@@ -263,17 +259,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 		ret = asoc_simple_parse_codec(ep, dai_link);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		ret = asoc_simple_set_dailink_name(dev, dai_link,
 						   "be.%s",
 						   codecs->dai_name);
 		if (ret < 0)
-			return ret;
+			goto out_put_node;
 
 		/* check "prefix" from top node */
 		snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -293,19 +289,23 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 
 	ret = asoc_simple_parse_tdm(ep, dai);
 	if (ret)
-		return ret;
+		goto out_put_node;
 
 	ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep,
 				       NULL, &dai_link->dai_fmt);
 	if (ret < 0)
-		return ret;
+		goto out_put_node;
 
 	dai_link->dpcm_playback		= 1;
 	dai_link->dpcm_capture		= 1;
 	dai_link->ops			= &graph_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
-	return 0;
+out_put_node:
+	of_node_put(ports);
+	of_node_put(port);
+	of_node_put(node);
+	return ret;
 }
 
 static int graph_dai_link_of(struct asoc_simple_priv *priv,
-- 
cgit v1.2.3


From c152f8491a8d9a4b25afd65a86eb5e55e2a8c380 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Wed, 10 Jul 2019 15:25:09 +0800
Subject: ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()

After calling of_node_put() on the node variable, it is still being
used, which may result in use-after-free.
Fix this issue by calling of_node_put() after the last usage.

Fixes: a0c426fe1433 ("ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id()")
Link: https://lore.kernel.org/r/1562743509-30496-5-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index bddfcfd7bedf..343ede8042c3 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -63,6 +63,7 @@ static int graph_get_dai_id(struct device_node *ep)
 	struct device_node *endpoint;
 	struct of_endpoint info;
 	int i, id;
+	u32 *reg;
 	int ret;
 
 	/* use driver specified DAI ID if exist */
@@ -83,8 +84,9 @@ static int graph_get_dai_id(struct device_node *ep)
 			return info.id;
 
 		node = of_get_parent(ep);
+		reg = of_get_property(node, "reg", NULL);
 		of_node_put(node);
-		if (of_get_property(node, "reg", NULL))
+		if (reg)
 			return info.port;
 	}
 	node = of_graph_get_port_parent(ep);
-- 
cgit v1.2.3


From 09297c2f7a5428776369ba3b9904718a358e5559 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Thu, 11 Jul 2019 16:22:14 +0800
Subject: ASoC: rt1011: fix DC calibration offset not applying

There are two issues to fix:
- DC offset calibration data will be reset after stopping playback.
- DC offset calibration data should be applied in the initial setting.

Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20190711082214.8142-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt1011.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 5605b660f4bf..0a6ff13d76e1 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -39,7 +39,7 @@ static const struct reg_sequence init_list[] = {
 	{ RT1011_POWER_9, 0xa840 },
 
 	{ RT1011_ADC_SET_5, 0x0a20 },
-	{ RT1011_DAC_SET_2, 0xa232 },
+	{ RT1011_DAC_SET_2, 0xa032 },
 	{ RT1011_ADC_SET_1, 0x2925 },
 
 	{ RT1011_SPK_PRO_DC_DET_1, 0xb00c },
@@ -1917,7 +1917,7 @@ static int rt1011_set_bias_level(struct snd_soc_component *component,
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_2, 0x0000);
 		snd_soc_component_write(component,
-			RT1011_SYSTEM_RESET_3, 0x0000);
+			RT1011_SYSTEM_RESET_3, 0x0001);
 		snd_soc_component_write(component,
 			RT1011_SYSTEM_RESET_1, 0x003f);
 		snd_soc_component_write(component,
-- 
cgit v1.2.3


From ec3042ad39d4e2ddbc3a3344f90bb10d8feb53bc Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 11 Jul 2019 13:10:45 +0900
Subject: ASoC: audio-graph-card: add missing const at graph_get_dai_id()

commit c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in
graph_get_dai_id()") fixups use-after-free issue,
but, it need to use "const" for reg. This patch adds it.

We will have below without this patch

LINUX/sound/soc/generic/audio-graph-card.c: In function 'graph_get_dai_id':
LINUX/sound/soc/generic/audio-graph-card.c:87:7: warning: assignment discards\
 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
   reg = of_get_property(node, "reg", NULL);

Fixes: c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Wen Yang <wen.yang99@zte.com.cn>
Link: https://lore.kernel.org/r/87sgrd43ja.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/audio-graph-card.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 343ede8042c3..ebf2ca3249cb 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -63,7 +63,7 @@ static int graph_get_dai_id(struct device_node *ep)
 	struct device_node *endpoint;
 	struct of_endpoint info;
 	int i, id;
-	u32 *reg;
+	const u32 *reg;
 	int ret;
 
 	/* use driver specified DAI ID if exist */
-- 
cgit v1.2.3


From 9b6d104a6b150bd4d3e5b039340e1f6b20c2e3c1 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Sat, 13 Jul 2019 11:46:14 +0800
Subject: ASoC: samsung: odroid: fix an use-after-free issue for codec

The codec variable is still being used after the of_node_put() call,
which may result in use-after-free.

Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI")
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Krzysztof Kozlowski <krzk@kernel.org>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Sylwester Nawrocki <s.nawrocki@samsung.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/odroid.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index dfb6e460e7eb..64ebe895cdd7 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -284,9 +284,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
 	}
 
 	of_node_put(cpu);
-	of_node_put(codec);
 	if (ret < 0)
-		return ret;
+		goto err_put_node;
 
 	ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link);
 	if (ret < 0)
@@ -317,6 +316,7 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		goto err_put_clk_i2s;
 	}
 
+	of_node_put(codec);
 	return 0;
 
 err_put_clk_i2s:
@@ -326,6 +326,8 @@ err_put_sclk:
 err_put_cpu_dai:
 	of_node_put(cpu_dai);
 	snd_soc_of_put_dai_link_codecs(codec_link);
+err_put_node:
+	of_node_put(codec);
 	return ret;
 }
 
-- 
cgit v1.2.3


From 2abee12c0ab1924a69993d2c063a39a952e7d836 Mon Sep 17 00:00:00 2001
From: Wen Yang <wen.yang99@zte.com.cn>
Date: Sat, 13 Jul 2019 11:46:15 +0800
Subject: ASoC: samsung: odroid: fix a double-free issue for cpu_dai

The cpu_dai variable is still being used after the of_node_put() call,
which may result in double-free:

        of_node_put(cpu_dai);            ---> released here

        ret = devm_snd_soc_register_card(dev, card);
        if (ret < 0) {
...
                goto err_put_clk_i2s;    --> jump to err_put_clk_i2s
...

err_put_clk_i2s:
        clk_put(priv->clk_i2s_bus);
err_put_sclk:
        clk_put(priv->sclk_i2s);
err_put_cpu_dai:
        of_node_put(cpu_dai);            --> double-free here

Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance")
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Krzysztof Kozlowski <krzk@kernel.org>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Sylwester Nawrocki <s.nawrocki@samsung.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.com.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/samsung/odroid.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index 64ebe895cdd7..f0f5fa9c27d3 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -308,7 +308,6 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		ret = PTR_ERR(priv->clk_i2s_bus);
 		goto err_put_sclk;
 	}
-	of_node_put(cpu_dai);
 
 	ret = devm_snd_soc_register_card(dev, card);
 	if (ret < 0) {
@@ -316,6 +315,7 @@ static int odroid_audio_probe(struct platform_device *pdev)
 		goto err_put_clk_i2s;
 	}
 
+	of_node_put(cpu_dai);
 	of_node_put(codec);
 	return 0;
 
-- 
cgit v1.2.3


From aa2ba991c4206d5b778dcaa7b4997396e79f8e90 Mon Sep 17 00:00:00 2001
From: Hans de Goede <hdegoede@redhat.com>
Date: Fri, 12 Jul 2019 13:27:08 +0200
Subject: ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook

The Irbis NB41 netbook has its internal mic on IN2, inverted jack-detect
and stereo speakers, add a quirk for this.

Cc: russianneuromancer@ya.ru
Reported-and-tested-by: russianneuromancer@ya.ru
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20190712112708.25327-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/boards/bytcht_es8316.c | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index fac09be3cade..46612331f5ea 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -437,6 +437,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = {
 
 /* Please keep this list alphabetically sorted */
 static const struct dmi_system_id byt_cht_es8316_quirk_table[] = {
+	{	/* Irbis NB41 */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "NB41"),
+		},
+		.driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP
+					| BYT_CHT_ES8316_JD_INVERTED),
+	},
 	{	/* Teclast X98 Plus II */
 		.matches = {
 			DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"),
-- 
cgit v1.2.3


From 8dd26dff00c0636b1d8621acaeef3f6f3a39dd77 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Thu, 18 Jul 2019 09:43:33 +0100
Subject: ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks

DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a
list of the widgets connected to a specific front end DAI so it
can search through this list for available back end DAIs. The
custom_stop_condition was added to is_connected_ep to facilitate this
list not containing more widgets than is necessary. Doing so both
speeds up the DPCM handling as less widgets need to be searched and
avoids issues with CODEC to CODEC links as these would be confused
with back end DAIs if they appeared in the list of available widgets.

custom_stop_condition was implemented by aborting the graph walk
when the condition is triggered, however there is an issue with this
approach. Whilst walking the graph is_connected_ep should update the
endpoints cache on each widget, if the walk is aborted the number
of attached end points is unknown for that sub-graph. When the stop
condition triggered, the original patch ignored the triggering widget
and returned zero connected end points; a later patch updated this
to set the triggering widget's cache to 1 and return that. Both of
these approaches result in inaccurate values being stored in various
end point caches as the values propagate back through the graph,
which can result in later issues with widgets powering/not powering
unexpectedly.

As the original goal was to reduce the size of the widget list passed
to the DPCM code, the simplest solution is to limit the functionality
of the custom_stop_condition to the widget list. This means the rest
of the graph will still be processed resulting in correct end point
caches, but only widgets up to the stop condition will be added to the
returned widget list.

Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets")
Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links")
Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 6b44b4a78b8e..9cd87e47ee8f 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
 		list_add_tail(&widget->work_list, list);
 
 	if (custom_stop_condition && custom_stop_condition(widget, dir)) {
-		widget->endpoints[dir] = 1;
-		return widget->endpoints[dir];
+		list = NULL;
+		custom_stop_condition = NULL;
 	}
 
 	if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
  *
  * Optionally, can be supplied with a function acting as a stopping condition.
  * This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
  */
 static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	struct list_head *list,
-- 
cgit v1.2.3


From 48dfd37a0f85400610153101c72222bf01523699 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@nxp.com>
Date: Tue, 16 Jul 2019 17:45:47 +0800
Subject: ASoC: cs42xx8: Fix MFREQ selection issue for async mode

When sample rate of TX is different with sample rate of RX in
async mode, the MFreq selection will be wrong.

For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz.
Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX
and RX instance, the correct value of MFreq is 2 for both TX and RX.

But original method will cause MFreq = 0 for TX, MFreq = 2 for RX.
If TX is started after RX, RX will be impacted, RX work abnormal with
MFreq = 0.

This patch is to select proper MFreq value according to TX rate and
RX rate.

Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/20190716094547.46787-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++++++++++++++++++++++--------
 1 file changed, 97 insertions(+), 19 deletions(-)

diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 6203f54d9f25..5b049fcdba20 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -47,6 +47,7 @@ struct cs42xx8_priv {
 	unsigned long sysclk;
 	u32 tx_channels;
 	struct gpio_desc *gpiod_reset;
+	u32 rate[2];
 };
 
 /* -127.5dB to 0dB with step of 0.5dB */
@@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
 };
 
 struct cs42xx8_ratios {
-	unsigned int ratio;
-	unsigned char speed;
-	unsigned char mclk;
+	unsigned int mfreq;
+	unsigned int min_mclk;
+	unsigned int max_mclk;
+	unsigned int ratio[3];
 };
 
+/*
+ * According to reference mannual, define the cs42xx8_ratio struct
+ * MFreq2 | MFreq1 | MFreq0 |     Description     | SSM | DSM | QSM |
+ * 0      | 0      | 0      |1.029MHz to 12.8MHz  | 256 | 128 |  64 |
+ * 0      | 0      | 1      |1.536MHz to 19.2MHz  | 384 | 192 |  96 |
+ * 0      | 1      | 0      |2.048MHz to 25.6MHz  | 512 | 256 | 128 |
+ * 0      | 1      | 1      |3.072MHz to 38.4MHz  | 768 | 384 | 192 |
+ * 1      | x      | x      |4.096MHz to 51.2MHz  |1024 | 512 | 256 |
+ */
 static const struct cs42xx8_ratios cs42xx8_ratios[] = {
-	{ 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
-	{ 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
-	{ 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
-	{ 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
-	{ 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
-	{ 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
-	{ 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
-	{ 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
-	{ 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+	{ 0, 1029000, 12800000, {256, 128, 64} },
+	{ 2, 1536000, 19200000, {384, 192, 96} },
+	{ 4, 2048000, 25600000, {512, 256, 128} },
+	{ 6, 3072000, 38400000, {768, 384, 192} },
+	{ 8, 4096000, 51200000, {1024, 512, 256} },
 };
 
 static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai,
@@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_component *component = dai->component;
 	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	u32 ratio = cs42xx8->sysclk / params_rate(params);
-	u32 i, fm, val, mask;
+	u32 ratio[2];
+	u32 rate[2];
+	u32 fm[2];
+	u32 i, val, mask;
+	bool condition1, condition2;
 
 	if (tx)
 		cs42xx8->tx_channels = params_channels(params);
 
+	rate[tx]  = params_rate(params);
+	rate[!tx] = cs42xx8->rate[!tx];
+
+	ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0;
+	ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0;
+
+	/* Get functional mode for tx and rx according to rate */
+	for (i = 0; i < 2; i++) {
+		if (cs42xx8->slave_mode) {
+			fm[i] = CS42XX8_FM_AUTO;
+		} else {
+			if (rate[i] < 50000) {
+				fm[i] = CS42XX8_FM_SINGLE;
+			} else if (rate[i] > 50000 && rate[i] < 100000) {
+				fm[i] = CS42XX8_FM_DOUBLE;
+			} else if (rate[i] > 100000 && rate[i] < 200000) {
+				fm[i] = CS42XX8_FM_QUAD;
+			} else {
+				dev_err(component->dev,
+					"unsupported sample rate\n");
+				return -EINVAL;
+			}
+		}
+	}
+
 	for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
-		if (cs42xx8_ratios[i].ratio == ratio)
+		/* Is the ratio[tx] valid ? */
+		condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[tx]) :
+			(cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) &&
+			cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk &&
+			cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk;
+
+		if (!ratio[tx])
+			condition1 = true;
+
+		/* Is the ratio[!tx] valid ? */
+		condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ?
+			(cs42xx8_ratios[i].ratio[0] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[1] == ratio[!tx] ||
+			cs42xx8_ratios[i].ratio[2] == ratio[!tx]) :
+			(cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx]));
+
+		if (!ratio[!tx])
+			condition2 = true;
+
+		/*
+		 * Both ratio[tx] and ratio[!tx] is valid, then we get
+		 * a proper MFreq.
+		 */
+		if (condition1 && condition2)
 			break;
 	}
 
@@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
-	val = cs42xx8_ratios[i].mclk;
+	cs42xx8->rate[tx] = params_rate(params);
 
-	fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+	mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+	val = cs42xx8_ratios[i].mfreq;
 
 	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
 			   CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
-			   CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+			   CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val);
+
+	return 0;
+}
+
+static int cs42xx8_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
+	/* Clear stored rate */
+	cs42xx8->rate[tx] = 0;
+
+	regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+			   CS42XX8_FUNCMOD_xC_FM_MASK(tx),
+			   CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO));
 	return 0;
 }
 
@@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
 	.set_fmt	= cs42xx8_set_dai_fmt,
 	.set_sysclk	= cs42xx8_set_dai_sysclk,
 	.hw_params	= cs42xx8_hw_params,
+	.hw_free	= cs42xx8_hw_free,
 	.digital_mute	= cs42xx8_digital_mute,
 };
 
-- 
cgit v1.2.3


From f86621cd6c6f54edfdd62da347b2bbb8d7fddc8d Mon Sep 17 00:00:00 2001
From: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Date: Fri, 19 Jul 2019 19:39:29 +0200
Subject: SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress
 detection

The TS3A227E says that the headset keypress detection needs the MICBIAS
power in order to report the key events to ensure proper operation
The headset keypress detection needs the MICBIAS power in order to report
the key events all the time as long as MIC is present. So MICBIAS pin
is forced on when a MICROPHONE is detected.

On Veyron Minnie I observed that if the MICBIAS power is not present and
the key press detection is activated (just because it is enabled when you
insert a headset), it randomly reports a keypress on insert.
E.g. (KEY_PLAYPAUSE)

 Event: (SW_HEADPHONE_INSERT), value 1
 Event: (SW_MICROPHONE_INSERT), value 1
 Event: -------------- SYN_REPORT ------------
 Event: (KEY_PLAYPAUSE), value 1

Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying
effect that the media player starts a play/pause loop.

Note that, although most of the time the key reported is the one
associated with BTN_0, not always this is true. On my tests I also saw
different keys reported

Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20190719173929.24065-1-enric.balletbo@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_max98090.c | 32 ++++++++++++++++++++++++++++++++
 1 file changed, 32 insertions(+)

diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index c5fc24675a33..782e534d4c0d 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Speaker"),
 };
 
+static int rk_jack_event(struct notifier_block *nb, unsigned long event,
+			 void *data)
+{
+	struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+	struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+	if (event & SND_JACK_MICROPHONE)
+		snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+	else
+		snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static struct notifier_block rk_jack_nb = {
+	.notifier_call = rk_jack_event,
+};
+
+static int rk_init(struct snd_soc_pcm_runtime *runtime)
+{
+	/*
+	 * The jack has already been created in the rk_98090_headset_init()
+	 * function.
+	 */
+	snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb);
+
+	return 0;
+}
+
 static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi,
 static struct snd_soc_dai_link rk_dailink = {
 	.name = "max98090",
 	.stream_name = "Audio",
+	.init = rk_init,
 	.ops = &rk_aif1_ops,
 	/* set max98090 as slave */
 	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-- 
cgit v1.2.3


From 62ec3d13601bd626ca7a0edef6d45dbb753d94e8 Mon Sep 17 00:00:00 2001
From: Masahiro Yamada <yamada.masahiro@socionext.com>
Date: Sun, 21 Jul 2019 23:23:08 +0900
Subject: ASoC: SOF: use __u32 instead of uint32_t in uapi headers
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to
make sure they can be included from user-space.

Currently, header.h and fw.h are excluded from the test coverage.
To make them join the compile-test, we need to fix the build errors
attached below.

For a case like this, we decided to use __u{8,16,32,64} variable types
in this discussion:

  https://lkml.org/lkml/2019/6/5/18

Build log:

  CC      usr/include/sound/sof/header.h.s
  CC      usr/include/sound/sof/fw.h.s
In file included from <command-line>:32:0:
./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’
  uint32_t magic;  /**< 'S', 'O', 'F', '\0' */
  ^~~~~~~~
./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’
  uint32_t type;  /**< component specific type */
  ^~~~~~~~
./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’
  uint32_t size;  /**< size in bytes of data excl. this struct */
  ^~~~~~~~
./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’
  uint32_t abi;  /**< SOF ABI version */
  ^~~~~~~~
./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’
  uint32_t reserved[4]; /**< reserved for future use */
  ^~~~~~~~
./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’
  uint32_t data[0]; /**< Component data - opaque to core */
  ^~~~~~~~
In file included from <command-line>:32:0:
./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’
  uint32_t size;  /* bytes minus this header */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’
  uint32_t offset; /* offset from base */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’
  uint32_t size;  /* bytes minus this header */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’
  uint32_t num_blocks; /* number of blocks */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’
  uint32_t file_size; /* size of file minus this header */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’
  uint32_t num_modules; /* number of modules */
  ^~~~~~~~
./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’
  uint32_t abi;  /* version of header format */
  ^~~~~~~~

Signed-off-by: Masahiro Yamada <yamada.masahiro@socionext.com>
Link: https://lore.kernel.org/r/20190721142308.30306-1-yamada.masahiro@socionext.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/uapi/sound/sof/fw.h     | 16 +++++++++-------
 include/uapi/sound/sof/header.h | 14 ++++++++------
 2 files changed, 17 insertions(+), 13 deletions(-)

diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h
index 1afca973eb09..e9f697467a86 100644
--- a/include/uapi/sound/sof/fw.h
+++ b/include/uapi/sound/sof/fw.h
@@ -13,6 +13,8 @@
 #ifndef __INCLUDE_UAPI_SOF_FW_H__
 #define __INCLUDE_UAPI_SOF_FW_H__
 
+#include <linux/types.h>
+
 #define SND_SOF_FW_SIG_SIZE	4
 #define SND_SOF_FW_ABI		1
 #define SND_SOF_FW_SIG		"Reef"
@@ -46,8 +48,8 @@ enum snd_sof_fw_blk_type {
 
 struct snd_sof_blk_hdr {
 	enum snd_sof_fw_blk_type type;
-	uint32_t size;		/* bytes minus this header */
-	uint32_t offset;	/* offset from base */
+	__u32 size;		/* bytes minus this header */
+	__u32 offset;		/* offset from base */
 } __packed;
 
 /*
@@ -61,8 +63,8 @@ enum snd_sof_fw_mod_type {
 
 struct snd_sof_mod_hdr {
 	enum snd_sof_fw_mod_type type;
-	uint32_t size;		/* bytes minus this header */
-	uint32_t num_blocks;	/* number of blocks */
+	__u32 size;		/* bytes minus this header */
+	__u32 num_blocks;	/* number of blocks */
 } __packed;
 
 /*
@@ -70,9 +72,9 @@ struct snd_sof_mod_hdr {
  */
 struct snd_sof_fw_header {
 	unsigned char sig[SND_SOF_FW_SIG_SIZE]; /* "Reef" */
-	uint32_t file_size;	/* size of file minus this header */
-	uint32_t num_modules;	/* number of modules */
-	uint32_t abi;		/* version of header format */
+	__u32 file_size;	/* size of file minus this header */
+	__u32 num_modules;	/* number of modules */
+	__u32 abi;		/* version of header format */
 } __packed;
 
 #endif
diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h
index 7868990b0d6f..5f4518e7a972 100644
--- a/include/uapi/sound/sof/header.h
+++ b/include/uapi/sound/sof/header.h
@@ -9,6 +9,8 @@
 #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
 #define __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__
 
+#include <linux/types.h>
+
 /*
  * Header for all non IPC ABI data.
  *
@@ -16,12 +18,12 @@
  * Used by any bespoke component data structures or binary blobs.
  */
 struct sof_abi_hdr {
-	uint32_t magic;		/**< 'S', 'O', 'F', '\0' */
-	uint32_t type;		/**< component specific type */
-	uint32_t size;		/**< size in bytes of data excl. this struct */
-	uint32_t abi;		/**< SOF ABI version */
-	uint32_t reserved[4];	/**< reserved for future use */
-	uint32_t data[0];	/**< Component data - opaque to core */
+	__u32 magic;		/**< 'S', 'O', 'F', '\0' */
+	__u32 type;		/**< component specific type */
+	__u32 size;		/**< size in bytes of data excl. this struct */
+	__u32 abi;		/**< SOF ABI version */
+	__u32 reserved[4];	/**< reserved for future use */
+	__u32 data[0];		/**< Component data - opaque to core */
 }  __packed;
 
 #endif
-- 
cgit v1.2.3


From 45004d66f2a28d78f543fb2ffbc133e31dc2d162 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Mon, 22 Jul 2019 08:57:44 -0500
Subject: ASoC: dapm: fix a memory leak bug

In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in
dapm_cnew_widget() to hold the new dapm widget. Then, different actions are
taken according to the id of the widget, i.e., 'w->id'. If any failure
occurs during this process, snd_soc_dapm_new_control_unlocked() should be
terminated by going to the 'request_failed' label. However, the allocated
kernel buffer is not freed on this code path, leading to a memory leak bug.

To fix the above issue, free the buffer before returning from
snd_soc_dapm_new_control_unlocked() through the 'request_failed' label.

Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-dapm.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 9cd87e47ee8f..656cb5cd9cd8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3704,6 +3704,8 @@ request_failed:
 		dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
 			w->name, ret);
 
+	kfree_const(w->sname);
+	kfree(w);
 	return ERR_PTR(ret);
 }
 
-- 
cgit v1.2.3


From 8201f11a1f75e3aa7d5327d0b1d8cb544aeaa62f Mon Sep 17 00:00:00 2001
From: Stephan Gerhold <stephan@gerhold.net>
Date: Mon, 22 Jul 2019 15:03:52 +0200
Subject: ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links

apq8016_sbc_parse_of() sets up multiple DAI links, depending on the
number of nodes in the device tree. However, at the moment
CPU and platform components are only allocated for the first link.
This causes an oops when more than one link is defined:

	Internal error: Oops: 96000044 [#1] SMP
	CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4
	Call trace:
	 apq8016_sbc_platform_probe+0x1a8/0x3f0
	 platform_drv_probe+0x50/0xa0
	...

Move the allocation inside the loop to ensure that each link is
properly initialized.

Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/apq8016_sbc.c | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index f60a71990f66..ac75838bbfab 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
 
 	link = data->dai_link;
 
-	dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
-	if (!dlc)
-		return ERR_PTR(-ENOMEM);
+	for_each_child_of_node(node, np) {
+		dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
+		if (!dlc)
+			return ERR_PTR(-ENOMEM);
 
-	link->cpus	= &dlc[0];
-	link->platforms	= &dlc[1];
+		link->cpus	= &dlc[0];
+		link->platforms	= &dlc[1];
 
-	link->num_cpus		= 1;
-	link->num_platforms	= 1;
+		link->num_cpus		= 1;
+		link->num_platforms	= 1;
 
-	for_each_child_of_node(node, np) {
 		cpu = of_get_child_by_name(np, "cpu");
 		codec = of_get_child_by_name(np, "codec");
 
-- 
cgit v1.2.3


From 717dedb1dcee92788b81233aa0a221573c95daff Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 22 Jul 2019 09:14:01 -0500
Subject: ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread

Downgrade "nothing to do in IRQ thread" message from error to a debug
message in the IPC interrupt handler thread.

The spurious wake-up can happen if a HDA stream interrupt is
raised while the IPC interrupt thread is running. IPC functionality
is not impacted by this condition, so debug is a more appropriate
trace level.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sof/intel/cnl.c     | 4 ++--
 sound/soc/sof/intel/hda-ipc.c | 4 ++--
 2 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index f2b392998f20..ffd8d4394537 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -101,8 +101,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 50244b82600c..2ecba91f5219 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -224,8 +224,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
 		/*
 		 * This interrupt is not shared so no need to return IRQ_NONE.
 		 */
-		dev_err_ratelimited(sdev->dev,
-				    "error: nothing to do in IRQ thread\n");
+		dev_dbg_ratelimited(sdev->dev,
+				    "nothing to do in IPC IRQ thread\n");
 	}
 
 	/* re-enable IPC interrupt */
-- 
cgit v1.2.3


From 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b Mon Sep 17 00:00:00 2001
From: Ricard Wanderlof <ricard.wanderlof@axis.com>
Date: Wed, 24 Jul 2019 11:38:44 +0200
Subject: ASoC: Fail card instantiation if DAI format setup fails

If the DAI format setup fails, there is no valid communication format
between CPU and CODEC, so fail card instantiation, rather than continue
with a card that will most likely not function properly.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c0a774d0a5ff..1486fb2eb921 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1511,8 +1511,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
 		}
 	}
 
-	if (dai_link->dai_fmt)
-		snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+	if (dai_link->dai_fmt) {
+		ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+		if (ret)
+			return ret;
+	}
 
 	ret = soc_post_component_init(rtd, dai_link->name);
 	if (ret)
-- 
cgit v1.2.3


From 34a2a80ff30b5d2330abfa8980c7f0cc15a8158a Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Thu, 25 Jul 2019 11:34:23 +0300
Subject: ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode

When running McASP as master capture alone will not record any audio unless
a parallel playback stream is running. As soon as the playback stops the
captured data is going to be silent again.

In McASP master mode we need to set the PDIR for the clock pins and fix
the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above
AMUTE.

This went unnoticed as most of the boards uses McASP as slave and neither
of these issues are visible (audible) in those setups.

Fixes: ca3d9433349e ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-mcasp.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 2c518088b64d..4d611565375b 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -195,7 +195,7 @@ static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable)
 {
 	u32 bit;
 
-	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) {
+	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) {
 		if (enable)
 			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
 		else
@@ -223,6 +223,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 	if (mcasp_is_synchronous(mcasp)) {
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+		mcasp_set_clk_pdir(mcasp, true);
 	}
 
 	/* Activate serializer(s) */
-- 
cgit v1.2.3


From e51b69808b7ec06fc61f5a332f338d94b64b0537 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Thu, 25 Jul 2019 07:35:23 +0200
Subject: ASoC: Intel: Fix some acpi vs apci typo in somme comments

Fix some typo to have the filaname given in a comment match the real name
of the file.
Some 'acpi' have erroneously been written 'apci'

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/intel/common/soc-acpi-intel-bxt-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-byt-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-cht-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-cnl-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-glk-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-hda-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +-
 sound/soc/intel/common/soc-acpi-intel-icl-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-kbl-match.c     | 2 +-
 sound/soc/intel/common/soc-acpi-intel-skl-match.c     | 2 +-
 10 files changed, 10 insertions(+), 10 deletions(-)

diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
index 229e39586868..4a5adae1d785 100644
--- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ * soc-acpi-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index b94b482ac34f..1cc801ba92eb 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-byt-match.c - tables and support for BYT ACPI enumeration.
+ * soc-acpi-intel-byt-match.c - tables and support for BYT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index b7f11f6be1cf..d0fb43c2b9f6 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-cht-match.c - tables and support for CHT ACPI enumeration.
+ * soc-acpi-intel-cht-match.c - tables and support for CHT ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
index c36c0aa4f683..771b0ef21051 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ * soc-acpi-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
index 616eb09e78a0..60dea358fa04 100644
--- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ * soc-acpi-intel-glk-match.c - tables and support for GLK ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
index 68ae43f7b4b2..cc972d2ac691 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hda-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -2,7 +2,7 @@
 // Copyright (c) 2018, Intel Corporation.
 
 /*
- * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ * soc-acpi-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
  *
  */
 
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index d27853e7a369..34eb0baaa951 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0-only
 /*
- * soc-apci-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
+ * soc-acpi-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
  *
  * Copyright (c) 2017, Intel Corporation.
  */
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index 0b430b9b3673..38977669b576 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-icl-match.c - tables and support for ICL ACPI enumeration.
+ * soc-acpi-intel-icl-match.c - tables and support for ICL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 4b331058e807..e200baa11011 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ * soc-acpi-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
index 0c9c0edd35b3..42fa40a8d932 100644
--- a/sound/soc/intel/common/soc-acpi-intel-skl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -1,6 +1,6 @@
 // SPDX-License-Identifier: GPL-2.0
 /*
- * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ * soc-acpi-intel-skl-match.c - tables and support for SKL ACPI enumeration.
  *
  * Copyright (c) 2018, Intel Corporation.
  *
-- 
cgit v1.2.3


From 789e162a6255325325bd321ab0cd51dc7e285054 Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang <cychiang@chromium.org>
Date: Fri, 26 Jul 2019 12:42:02 +0800
Subject: ASoC: rockchip: Fix mono capture

This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0.
Revert "ASoC: rockchip: i2s: Support mono capture"

Previous discussion in

https://patchwork.kernel.org/patch/10147153/

explains the issue of the patch.
While device is configured as 1-ch, hardware is still
generating a 2-ch stream.
When user space reads the data and assumes it is a 1-ch stream,
the rate will be slower by 2x.

Revert the change so 1-ch is not supported.
User space can selectively take one channel data out of two channel
if 1-ch is preferred.
Currently, both channels record identical data.

Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/rockchip/rockchip_i2s.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 0a34d0eb8dba..88ebaf6e1880 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -326,7 +326,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
 		val |= I2S_CHN_4;
 		break;
 	case 2:
-	case 1:
 		val |= I2S_CHN_2;
 		break;
 	default:
@@ -459,7 +458,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 	},
 	.capture = {
 		.stream_name = "Capture",
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = (SNDRV_PCM_FMTBIT_S8 |
@@ -659,7 +658,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
 	}
 
 	if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
-		if (val >= 1 && val <= 8)
+		if (val >= 2 && val <= 8)
 			soc_dai->capture.channels_max = val;
 	}
 
-- 
cgit v1.2.3


From 1e112c35e3c96db7c8ca6ddaa96574f00c06e7db Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 26 Jul 2019 09:42:43 +0300
Subject: ASoC: ti: davinci-mcasp: Correct slot_width posed constraint

The slot_width is a property for the bus while the constraint for
SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format.

Applying slot_width constraint to sample_bits works most of the time, but
it will blacklist valid formats in some cases.

With slot_width 24 we can support S24_3LE and S24_LE formats as they both
look the same on the bus, but a a 24 constraint on sample_bits would not
allow S24_LE as it is stored in 32bits in memory.

Implement a simple hw_rule function to allow all formats which require less
or equal number of bits on the bus as slot_width (if configured).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/ti/davinci-mcasp.c | 43 ++++++++++++++++++++++++++++++++++---------
 1 file changed, 34 insertions(+), 9 deletions(-)

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 4d611565375b..44708c8f90d6 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1257,6 +1257,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
 static const unsigned int davinci_mcasp_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000,
@@ -1378,7 +1400,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	struct davinci_mcasp_ruledata *ruledata =
 					&mcasp->ruledata[substream->stream];
 	u32 max_channels = 0;
-	int i, dir;
+	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
 
 	/* Do not allow more then one stream per direction */
@@ -1407,6 +1429,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			max_channels++;
 	}
 	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
 	max_channels *= tdm_slots;
 	/*
 	 * If the already active stream has less channels than the calculated
@@ -1432,20 +1455,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
 				   &mcasp->chconstr[substream->stream]);
 
-	if (mcasp->slot_width)
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     8, mcasp->slot_width);
+	if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
 
 	/*
 	 * If we rely on implicit BCLK divider setting we should
 	 * set constraints based on what we can provide.
 	 */
 	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
-		int ret;
-
-		ruledata->mcasp = mcasp;
-
 		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
 					  SNDRV_PCM_HW_PARAM_RATE,
 					  davinci_mcasp_hw_rule_rate,
-- 
cgit v1.2.3