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(-)

(limited to 'sound')

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

(limited to 'sound')

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(-)

(limited to 'sound')

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 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(+)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(+)

(limited to 'sound')

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 59d81c1e3cade953a0cb3f66ce9a3f2398fdfac3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 16 Jul 2019 11:52:00 +0200
Subject: ALSA: hda - Optimize resume for codecs without jack detection

The codecs without jack detection also don't have to be resumed
forcibly because, obviously, they have no jack.  Skip the forced
resume in such a case as optimization as well.

Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e30e86ca6b72..51f10ed9bc43 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2942,7 +2942,7 @@ static int hda_codec_runtime_resume(struct device *dev)
 static int hda_codec_force_resume(struct device *dev)
 {
 	struct hda_codec *codec = dev_to_hda_codec(dev);
-	bool forced_resume = !codec->relaxed_resume;
+	bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used;
 	int ret;
 
 	/* The get/put pair below enforces the runtime resume even if the
-- 
cgit v1.2.3


From 70256b42caaf3e13c2932c2be7903a73fbe8bb8b Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Thu, 18 Jul 2019 17:53:13 +0800
Subject: ALSA: line6: Fix wrong altsetting for LINE6_PODHD500_1

Commit 7b9584fa1c0b ("staging: line6: Move altsetting to properties")
set a wrong altsetting for LINE6_PODHD500_1 during refactoring.

Set the correct altsetting number to fix the issue.

BugLink: https://bugs.launchpad.net/bugs/1790595
Fixes: 7b9584fa1c0b ("staging: line6: Move altsetting to properties")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/line6/podhd.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index f0662bd4e50f..27bf61c177c0 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = {
 		.name = "POD HD500",
 		.capabilities	= LINE6_CAP_PCM
 				| LINE6_CAP_HWMON,
-		.altsetting = 1,
+		.altsetting = 0,
 		.ep_ctrl_r = 0x81,
 		.ep_ctrl_w = 0x01,
 		.ep_audio_r = 0x86,
-- 
cgit v1.2.3


From 0e279dcea0ec897af1c979ebee4ec92b461793f5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 19 Jul 2019 10:55:05 +0200
Subject: ALSA: pcm: Fix refcount_inc() on zero usage

The recent rewrite of PCM link lock management introduced the refcount
in snd_pcm_group object, managed by the kernel refcount_t API.  This
caused unexpected kernel warnings when the kernel is built with
CONFIG_REFCOUNT_FULL=y.  As the warning line indicates, the problem is
obviously that we start with refcount=0 and do refcount_inc() for
adding each PCM link, while refcount_t API doesn't like refcount_inc()
performed on zero.

For adapting the proper refcount_t usage, this patch changes the logic
slightly:
- The initial refcount is 1, assuming the single list entry
- The refcount is incremented / decremented at each PCM link addition
  and deletion
- ... which allows us concentrating only on the refcount as a release
  condition

Fixes: f57f3df03a8e ("ALSA: pcm: More fine-grained PCM link locking")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204221
Reported-and-tested-by: Duncan Overbruck <kernel@duncano.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_native.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 860543a4c840..12dd9b318db1 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group)
 	spin_lock_init(&group->lock);
 	mutex_init(&group->mutex);
 	INIT_LIST_HEAD(&group->substreams);
-	refcount_set(&group->refs, 0);
+	refcount_set(&group->refs, 1);
 }
 
 /* define group lock helpers */
@@ -1096,8 +1096,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group,
 
 	if (!group)
 		return;
-	do_free = refcount_dec_and_test(&group->refs) &&
-		list_empty(&group->substreams);
+	do_free = refcount_dec_and_test(&group->refs);
 	snd_pcm_group_unlock(group, substream->pcm->nonatomic);
 	if (do_free)
 		kfree(group);
@@ -2020,6 +2019,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
 	snd_pcm_group_lock_irq(target_group, nonatomic);
 	snd_pcm_stream_lock(substream1);
 	snd_pcm_group_assign(substream1, target_group);
+	refcount_inc(&target_group->refs);
 	snd_pcm_stream_unlock(substream1);
 	snd_pcm_group_unlock_irq(target_group, nonatomic);
  _end:
@@ -2056,13 +2056,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream)
 	snd_pcm_group_lock_irq(group, nonatomic);
 
 	relink_to_local(substream);
+	refcount_dec(&group->refs);
 
 	/* detach the last stream, too */
 	if (list_is_singular(&group->substreams)) {
 		relink_to_local(list_first_entry(&group->substreams,
 						 struct snd_pcm_substream,
 						 link_list));
-		do_free = !refcount_read(&group->refs);
+		do_free = refcount_dec_and_test(&group->refs);
 	}
 
 	snd_pcm_group_unlock_irq(group, nonatomic);
-- 
cgit v1.2.3


From e4091bdd2fd957793a10449a8682c767578b0430 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Sun, 21 Jul 2019 12:25:58 +0200
Subject: ALSA: line6: Fix a typo

s/Vairax/Variax/

Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/line6/variax.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c
index 0d24c72c155f..ed158f04de80 100644
--- a/sound/usb/line6/variax.c
+++ b/sound/usb/line6/variax.c
@@ -244,5 +244,5 @@ static struct usb_driver variax_driver = {
 
 module_usb_driver(variax_driver);
 
-MODULE_DESCRIPTION("Vairax Workbench USB driver");
+MODULE_DESCRIPTION("Variax Workbench USB driver");
 MODULE_LICENSE("GPL");
-- 
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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(+)

(limited to 'sound')

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 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(+)

(limited to 'sound')

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 4475f8c4ab7b248991a60d9c02808dbb813d6be8 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 10:24:33 +0100
Subject: ALSA: compress: Fix regression on compressed capture streams

A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.

To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.

Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/compress_driver.h |  5 +----
 sound/core/compress_offload.c   | 16 +++++++++++-----
 2 files changed, 12 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index c5188ff724d1..bc88d6f964da 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -173,10 +173,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream)
 	if (snd_BUG_ON(!stream))
 		return;
 
-	if (stream->direction == SND_COMPRESS_PLAYBACK)
-		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
-	else
-		stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+	stream->runtime->state = SNDRV_PCM_STATE_SETUP;
 
 	wake_up(&stream->runtime->sleep);
 }
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 99b882158705..d79aee6b9edd 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
 		stream->metadata_set = false;
 		stream->next_track = false;
 
-		if (stream->direction == SND_COMPRESS_PLAYBACK)
-			stream->runtime->state = SNDRV_PCM_STATE_SETUP;
-		else
-			stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+		stream->runtime->state = SNDRV_PCM_STATE_SETUP;
 	} else {
 		return -EPERM;
 	}
@@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_SETUP:
+		if (stream->direction != SND_COMPRESS_CAPTURE)
+			return -EPERM;
+		break;
+	case SNDRV_PCM_STATE_PREPARED:
+		break;
+	default:
 		return -EPERM;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
 	if (!retval)
 		stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
-- 
cgit v1.2.3


From 26c3f1542f5064310ad26794c09321780d00c57d Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 10:24:34 +0100
Subject: ALSA: compress: Prevent bypasses of set_params

Currently, whilst in SNDRV_PCM_STATE_OPEN it is possible to call
snd_compr_stop, snd_compr_drain and snd_compr_partial_drain, which
allow a transition to SNDRV_PCM_STATE_SETUP. The stream should
only be able to move to the setup state once it has received a
SNDRV_COMPRESS_SET_PARAMS ioctl. Fix this issue by not allowing
those ioctls whilst in the open state.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 30 ++++++++++++++++++++++++------
 1 file changed, 24 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index d79aee6b9edd..40dae723c59d 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -711,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
 		return -EPERM;
+	default:
+		break;
+	}
+
 	retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
 	if (!retval) {
 		snd_compr_drain_notify(stream);
@@ -801,9 +807,14 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
 {
 	int retval;
 
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
 		return -EPERM;
+	default:
+		break;
+	}
 
 	retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
 	if (retval) {
@@ -840,9 +851,16 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 {
 	int retval;
-	if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
-			stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+
+	switch (stream->runtime->state) {
+	case SNDRV_PCM_STATE_OPEN:
+	case SNDRV_PCM_STATE_SETUP:
+	case SNDRV_PCM_STATE_PREPARED:
 		return -EPERM;
+	default:
+		break;
+	}
+
 	/* stream can be drained only when next track has been signalled */
 	if (stream->next_track == false)
 		return -EPERM;
-- 
cgit v1.2.3


From a70ab8a8645083f3700814e757f2940a88b7ef88 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 10:24:35 +0100
Subject: ALSA: compress: Don't allow paritial drain operations on capture
 streams

Partial drain and next track are intended for gapless playback and
don't really have an obvious interpretation for a capture stream, so
makes sense to not allow those operations on capture streams.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound')

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 40dae723c59d..6cf5b8440cf3 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -834,6 +834,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
 	if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
 		return -EPERM;
 
+	/* next track doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
+		return -EPERM;
+
 	/* you can signal next track if this is intended to be a gapless stream
 	 * and current track metadata is set
 	 */
@@ -861,6 +865,10 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 		break;
 	}
 
+	/* partial drain doesn't have any meaning for capture streams */
+	if (stream->direction == SND_COMPRESS_CAPTURE)
+		return -EPERM;
+
 	/* stream can be drained only when next track has been signalled */
 	if (stream->next_track == false)
 		return -EPERM;
-- 
cgit v1.2.3


From 3b8179944cb0dd53e5223996966746cdc8a60657 Mon Sep 17 00:00:00 2001
From: Charles Keepax <ckeepax@opensource.cirrus.com>
Date: Mon, 22 Jul 2019 10:24:36 +0100
Subject: ALSA: compress: Be more restrictive about when a drain is allowed

Draining makes little sense in the situation of hardware overrun, as the
hardware will have consumed all its available samples. Additionally,
draining whilst the stream is paused would presumably get stuck as no
data is being consumed on the DSP side.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/compress_offload.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 6cf5b8440cf3..41905afada63 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -811,7 +811,10 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
 	case SNDRV_PCM_STATE_OPEN:
 	case SNDRV_PCM_STATE_SETUP:
 	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
 		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
 	default:
 		break;
 	}
@@ -860,7 +863,10 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
 	case SNDRV_PCM_STATE_OPEN:
 	case SNDRV_PCM_STATE_SETUP:
 	case SNDRV_PCM_STATE_PREPARED:
+	case SNDRV_PCM_STATE_PAUSED:
 		return -EPERM;
+	case SNDRV_PCM_STATE_XRUN:
+		return -EPIPE;
 	default:
 		break;
 	}
-- 
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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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 607975b30db41aad6edc846ed567191aa6b7d893 Mon Sep 17 00:00:00 2001
From: Ding Xiang <dingxiang@cmss.chinamobile.com>
Date: Tue, 23 Jul 2019 15:44:41 +0800
Subject: ALSA: ac97: Fix double free of ac97_codec_device

put_device will call ac97_codec_release to free
ac97_codec_device and other resources, so remove the kfree
and other redundant code.

Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus")
Signed-off-by: Ding Xiang <dingxiang@cmss.chinamobile.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/ac97/bus.c | 13 ++++---------
 1 file changed, 4 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 7b977b753a03..7985dd8198b6 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
 						      vendor_id);
 
 	ret = device_add(&codec->dev);
-	if (ret)
-		goto err_free_codec;
+	if (ret) {
+		put_device(&codec->dev);
+		return ret;
+	}
 
 	return 0;
-err_free_codec:
-	of_node_put(codec->dev.of_node);
-	put_device(&codec->dev);
-	kfree(codec);
-	ac97_ctrl->codecs[idx] = NULL;
-
-	return ret;
 }
 
 unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
-- 
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(-)

(limited to 'sound')

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 2756d9143aa517b97961e85412882b8ce31371a6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 19 Jul 2019 10:27:54 +0200
Subject: ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips

It turned out that the recent Intel HD-audio controller chips show a
significant stall during the system PM resume intermittently.  It
doesn't happen so often and usually it may read back successfully
after one or more seconds, but in some rare worst cases the driver
went into fallback mode.

After trial-and-error, we found out that the communication stall seems
covered by issuing the sync after each verb write, as already done for
AMD and other chipsets.  So this patch enables the write-sync flag for
the recent Intel chips, Skylake and onward, as a workaround.

Also, since Broxton and co have the very same driver flags as Skylake,
refer to the Skylake driver flags instead of defining the same
contents again for simplification.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901
Reported-and-tested-by: Todd Brandt <todd.e.brandt@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index cb8b0945547c..1e14d7270adf 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -313,11 +313,10 @@ enum {
 
 #define AZX_DCAPS_INTEL_SKYLAKE \
 	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+	 AZX_DCAPS_SYNC_WRITE |\
 	 AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
 
-#define AZX_DCAPS_INTEL_BROXTON \
-	(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
-	 AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
+#define AZX_DCAPS_INTEL_BROXTON		AZX_DCAPS_INTEL_SKYLAKE
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
-- 
cgit v1.2.3


From 3f8809499bf02ef7874254c5e23fc764a47a21a0 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Thu, 25 Jul 2019 14:57:37 +0800
Subject: ALSA: hda - Add a conexant codec entry to let mute led work

This conexant codec isn't in the supported codec list yet, the hda
generic driver can drive this codec well, but on a Lenovo machine
with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI
to make the leds work. After adding this codec to the list, the
driver patch_conexant.c will apply THINKPAD_ACPI to this machine.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4f8d0845ee1e..f299f137eaea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1083,6 +1083,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
  */
 
 static const struct hda_device_id snd_hda_id_conexant[] = {
+	HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
 	HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
-- 
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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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(-)

(limited to 'sound')

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


From 74bf71ed792ab0f64631cc65ccdb54c356c36d45 Mon Sep 17 00:00:00 2001
From: Samuel Thibault <samuel.thibault@ens-lyon.org>
Date: Fri, 26 Jul 2019 23:47:02 +0200
Subject: ALSA: hda: Fix 1-minute detection delay when i915 module is not
 available

Distribution installation images such as Debian include different sets
of modules which can be downloaded dynamically.  Such images may notably
include the hda sound modules but not the i915 DRM module, even if the
latter was enabled at build time, as reported on
https://bugs.debian.org/931507

In such a case hdac_i915 would be linked in and try to load the i915
module, fail since it is not there, but still wait for a whole minute
before giving up binding with it.

This fixes such as case by only waiting for the binding if the module
was properly loaded (or module support is disabled, in which case i915
is already compiled-in anyway).

Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Signed-off-by: Samuel Thibault <samuel.thibault@ens-lyon.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/hdac_i915.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 1192c7561d62..3c2db3816029 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -136,10 +136,12 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 	if (!acomp)
 		return -ENODEV;
 	if (!acomp->ops) {
-		request_module("i915");
-		/* 60s timeout */
-		wait_for_completion_timeout(&bind_complete,
-					    msecs_to_jiffies(60 * 1000));
+		if (!IS_ENABLED(CONFIG_MODULES) ||
+		    !request_module("i915")) {
+			/* 60s timeout */
+			wait_for_completion_timeout(&bind_complete,
+						   msecs_to_jiffies(60 * 1000));
+		}
 	}
 	if (!acomp->ops) {
 		dev_info(bus->dev, "couldn't bind with audio component\n");
-- 
cgit v1.2.3


From 37151a41df800493cfcbbef4f7208ffe04feb959 Mon Sep 17 00:00:00 2001
From: Yuki Tsunashima <ytsunashima@jp.adit-jv.com>
Date: Mon, 29 Jul 2019 17:10:36 +0200
Subject: ALSA: pcm: fix lost wakeup event scenarios in snd_pcm_drain

lost wakeup can occur after enabling irq, therefore put task
into interruptible before enabling interrupts,

without this change, task can be put to sleep and snd_pcm_drain
will delay

Fixes: f2b3614cefb6 ("ALSA: PCM - Don't check DMA time-out too shortly")
Signed-off-by: Yuki Tsunashima <ytsunashima@jp.adit-jv.com>
Signed-off-by: Suresh Udipi <sudipi@jp.adit-jv.com>
[ported from 4.9]
Signed-off-by: Adam Miartus <amiartus@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_native.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 12dd9b318db1..703857aab00f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1873,6 +1873,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 		if (!to_check)
 			break; /* all drained */
 		init_waitqueue_entry(&wait, current);
+		set_current_state(TASK_INTERRUPTIBLE);
 		add_wait_queue(&to_check->sleep, &wait);
 		snd_pcm_stream_unlock_irq(substream);
 		if (runtime->no_period_wakeup)
@@ -1885,7 +1886,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
 			}
 			tout = msecs_to_jiffies(tout * 1000);
 		}
-		tout = schedule_timeout_interruptible(tout);
+		tout = schedule_timeout(tout);
 
 		snd_pcm_stream_lock_irq(substream);
 		group = snd_pcm_stream_group_ref(substream);
-- 
cgit v1.2.3


From 5d78e1c2b7f4be00bbe62141603a631dc7812f35 Mon Sep 17 00:00:00 2001
From: Hillf Danton <hdanton@sina.com>
Date: Tue, 30 Jul 2019 17:24:36 +0800
Subject: ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check

syzbot found the following crash on:

  general protection fault: 0000 [#1] SMP KASAN
  RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75
  Call Trace:
    snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb  sound/usb/quirks.c:1007
    snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline]
    snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280
    usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576
    usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361
    really_probe+0x281/0x650 drivers/base/dd.c:548
    ....

It was introduced in commit 801ebf1043ae for checking pipe and endpoint
types. It is fixed by adding a check of the ep pointer in question.

BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2
Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2@syzkaller.appspotmail.com>
Fixes: 801ebf1043ae ("ALSA: usb-audio: Sanity checks for each pipe and EP types")
Cc: Andrey Konovalov <andreyknvl@google.com>
Signed-off-by: Hillf Danton <hdanton@sina.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/helper.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index 71d5f540334a..4c12cc5b53fd 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -72,7 +72,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe)
 	struct usb_host_endpoint *ep;
 
 	ep = usb_pipe_endpoint(dev, pipe);
-	if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
+	if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
 		return -EINVAL;
 	return 0;
 }
-- 
cgit v1.2.3


From 52f87f3ca251f5e43b42e78ab9816b2b07718bfe Mon Sep 17 00:00:00 2001
From: Marcus Cooper <codekipper@gmail.com>
Date: Mon, 29 Jul 2019 17:21:30 +0200
Subject: ASoC: sun4i-i2s: Incorrect SR and WSS computation

The A64 audio codec uses the original I2S block but the SR and
WSS computation currently assigned is for the newer block.

Fixes: 619c15f7fac9 (ASoC: sun4i-i2s: Change SR and WSS computation)
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/sunxi/sun4i-i2s.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9b2232908b65..7fa5c61169db 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1002,8 +1002,8 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
 	.field_rxchanmap	= REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
 	.field_txchansel	= REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
 	.field_rxchansel	= REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
-	.get_sr			= sun8i_i2s_get_sr_wss,
-	.get_wss		= sun8i_i2s_get_sr_wss,
+	.get_sr			= sun4i_i2s_get_sr,
+	.get_wss		= sun4i_i2s_get_wss,
 };
 
 static int sun4i_i2s_init_regmap_fields(struct device *dev,
-- 
cgit v1.2.3


From b9da500bde81ad820b5d95c6bf52fc33e1f490ee Mon Sep 17 00:00:00 2001
From: fengchunguo <chunguo.feng@amlogic.com>
Date: Wed, 31 Jul 2019 15:41:56 +0800
Subject: ASoC: max98373: add 88200 and 96000 sampling rate support
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

88200 and 96000 sampling rate was not enabled on driver, so can't be played.

The error information:
max98373 3-0031:rate 96000 not supported
max98373 3-0031:ASoC: can't set max98373-aif1 hw params: -22

Signed-off-by: fengchunguo <chunguo.feng@amlogic.com>
Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98373.c | 6 ++++++
 sound/soc/codecs/max98373.h | 2 ++
 2 files changed, 8 insertions(+)
 mode change 100644 => 100755 sound/soc/codecs/max98373.c
 mode change 100644 => 100755 sound/soc/codecs/max98373.h

(limited to 'sound')

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
old mode 100644
new mode 100755
index 528695cd6a1c..8c601a3ebc27
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream,
 	case 48000:
 		sampling_rate = MAX98373_PCM_SR_SET1_SR_48000;
 		break;
+	case 88200:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_88200;
+		break;
+	case 96000:
+		sampling_rate = MAX98373_PCM_SR_SET1_SR_96000;
+		break;
 	default:
 		dev_err(component->dev, "rate %d not supported\n",
 			params_rate(params));
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
old mode 100644
new mode 100755
index f6a37aa02f26..a59e51355a84
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -130,6 +130,8 @@
 #define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0)
 #define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0)
 #define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0)
+#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0)
+#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0)
 
 /* MAX98373_R2028_PCM_SR_SETUP_2 */
 #define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4)
-- 
cgit v1.2.3


From 88639051017fb61a414b636dd0fc490da2b62b64 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Fri, 2 Aug 2019 19:21:23 +0530
Subject: ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver

AMD platform device acp3x_rv_i2s created by parent PCI device
driver. Pass struct device of the parent to
snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use
correct dma_ops. Otherwise, it will use default dma_ops which
is nommu_dma_ops on x86_64 even when IOMMU is enabled and
set to non passthrough mode.

Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index a4ade6bb5beb..905ed2f1861b 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -385,9 +385,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream)
 
 static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd,
+								    DRV_NAME);
+	struct device *parent = component->dev->parent;
 	snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
-					      rtd->pcm->card->dev,
-					      MIN_BUFFER, MAX_BUFFER);
+					      parent, MIN_BUFFER, MAX_BUFFER);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 30c21734d853dae99d05a5295a59b7e26ccd5135 Mon Sep 17 00:00:00 2001
From: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Date: Fri, 2 Aug 2019 19:21:24 +0530
Subject: ASoC: amd: acp3x: use dma address for acp3x dma driver

We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.

This patch fixes page faults when IOMMU is enabled.

Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/raven/acp3x-pcm-dma.c | 14 +++++---------
 1 file changed, 5 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index 905ed2f1861b..bc4dfafdfcd1 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -31,8 +31,8 @@ struct i2s_stream_instance {
 	u16 num_pages;
 	u16 channels;
 	u32 xfer_resolution;
-	struct page *pg;
 	u64 bytescount;
+	dma_addr_t dma_addr;
 	void __iomem *acp3x_base;
 };
 
@@ -211,9 +211,8 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
 static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 {
 	u16 page_idx;
-	u64 addr;
 	u32 low, high, val, acp_fifo_addr;
-	struct page *pg = rtd->pg;
+	dma_addr_t addr = rtd->dma_addr;
 
 	/* 8 scratch registers used to map one 64 bit address */
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
@@ -229,7 +228,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 
 	for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) {
 		/* Load the low address of page int ACP SRAM through SRBM */
-		addr = page_to_phys(pg);
 		low = lower_32_bits(addr);
 		high = upper_32_bits(addr);
 
@@ -239,7 +237,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
 				+ 4);
 		/* Move to next physically contiguos page */
 		val += 8;
-		pg++;
+		addr += PAGE_SIZE;
 	}
 
 	if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -341,7 +339,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	int status;
 	u64 size;
-	struct page *pg;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct i2s_stream_instance *rtd = runtime->private_data;
 
@@ -354,9 +351,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
 		return status;
 
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
-	pg = virt_to_page(substream->dma_buffer.area);
-	if (pg) {
-		rtd->pg = pg;
+	if (substream->dma_buffer.area) {
+		rtd->dma_addr = substream->dma_buffer.addr;
 		rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
 		config_acp3x_dma(rtd, substream->stream);
 		status = 0;
-- 
cgit v1.2.3


From 7c0767643f3b6b0dd2cda923ae37a18590d431cf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Tue, 6 Aug 2019 11:15:06 +0100
Subject: ASoC: max98373: Remove executable bits

Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max98373.c | 0
 sound/soc/codecs/max98373.h | 0
 2 files changed, 0 insertions(+), 0 deletions(-)
 mode change 100755 => 100644 sound/soc/codecs/max98373.c
 mode change 100755 => 100644 sound/soc/codecs/max98373.h

(limited to 'sound')

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
old mode 100755
new mode 100644
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
old mode 100755
new mode 100644
-- 
cgit v1.2.3


From a67060201b746a308b1674f66bf289c9faef6d09 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Tue, 6 Aug 2019 03:00:27 -0400
Subject: ALSA: usb-audio: fix a memory leak bug

In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.

To fix the above issue, free 'fp->chmap' before returning NULL.

Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/stream.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 7ee9d17d0143..e852c7fd6109 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1043,6 +1043,7 @@ found_clock:
 
 		pd = kzalloc(sizeof(*pd), GFP_KERNEL);
 		if (!pd) {
+			kfree(fp->chmap);
 			kfree(fp->rate_table);
 			kfree(fp);
 			return NULL;
-- 
cgit v1.2.3


From c1c6c877b0c79fd7e05c931435aa42211eaeebaf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Aug 2019 14:03:56 +0200
Subject: ALSA: hda - Don't override global PCM hw info flag

The commit bfcba288b97f ("ALSA - hda: Add support for link audio time
reporting") introduced the conditional PCM hw info setup, but it
overwrites the global azx_pcm_hw object.  This will cause a problem if
any other HD-audio controller, as it'll inherit the same bit flag
although another controller doesn't support that feature.

Fix the bug by setting the PCM hw info flag locally.

Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index c8d1b4316245..2fbdde239936 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	}
 	runtime->private_data = azx_dev;
 
-	if (chip->gts_present)
-		azx_pcm_hw.info = azx_pcm_hw.info |
-			SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
-
 	runtime->hw = azx_pcm_hw;
+	if (chip->gts_present)
+		runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME;
 	runtime->hw.channels_min = hinfo->channels_min;
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;
-- 
cgit v1.2.3


From 3d92aa45fbfd7319e3a19f4ec59fd32b3862b723 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Wed, 7 Aug 2019 04:08:51 -0500
Subject: ALSA: hiface: fix multiple memory leak bugs

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later
on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In
hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through
kzalloc().  However, if hiface_pcm_init_urb() fails, both 'rt' and
'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs.
Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails.

To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'.

Fixes: a91c3fb2f842 ("Add M2Tech hiFace USB-SPDIF driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/hiface/pcm.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c
index 14fc1e1d5d13..c406497c5919 100644
--- a/sound/usb/hiface/pcm.c
+++ b/sound/usb/hiface/pcm.c
@@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 		ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP,
 				    hiface_pcm_out_urb_handler);
 		if (ret < 0)
-			return ret;
+			goto error;
 	}
 
 	ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm);
 	if (ret < 0) {
-		kfree(rt);
 		dev_err(&chip->dev->dev, "Cannot create pcm instance\n");
-		return ret;
+		goto error;
 	}
 
 	pcm->private_data = rt;
@@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq)
 
 	chip->pcm = rt;
 	return 0;
+
+error:
+	for (i = 0; i < PCM_N_URBS; i++)
+		kfree(rt->out_urbs[i].buffer);
+	kfree(rt);
+	return ret;
 }
-- 
cgit v1.2.3


From c02f77d32d2c45cfb1b2bb99eabd8a78f5ecc7db Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 6 Aug 2019 17:31:48 +0200
Subject: ALSA: hda - Workaround for crackled sound on AMD controller
 (1022:1457)

A long-time problem on the recent AMD chip (X370, X470, B450, etc with
PCI ID 1022:1457) with Realtek codecs is the crackled or distorted
sound for capture streams, as well as occasional playback hiccups.
After lengthy debugging sessions, the workarounds we've found are like
the following:

- Set up the proper driver caps for this controller, similar as the
  other AMD controller.

- Correct the DMA position reporting with the fixed FIFO size, which
  is similar like as workaround used for VIA chip set.

- Even after the position correction, PulseAudio still shows
  mysterious stalls of playback streams when a capture is triggered in
  timer-scheduled mode.  Since we have no clear way to eliminate the
  stall, pass the BATCH PCM flag for PA to suppress the tsched mode as
  a temporary workaround.

This patch implements the workarounds.  For the driver caps, it
defines a new preset, AXZ_DCAPS_PRESET_AMD_SB.  It enables the FIFO-
corrected position reporting (corresponding to the new position_fix=6)
and enforces the SNDRV_PCM_INFO_BATCH flag.

Note that the current implementation is merely a workaround.
Hopefully we'll find a better alternative in future, especially about
removing the BATCH flag hack again.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_controller.c |  7 +++++
 sound/pci/hda/hda_controller.h |  2 +-
 sound/pci/hda/hda_intel.c      | 63 +++++++++++++++++++++++++++++++++++++++++-
 3 files changed, 70 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2fbdde239936..48d863736b3c 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -613,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				     20,
 				     178000000);
 
+	/* by some reason, the playback stream stalls on PulseAudio with
+	 * tsched=1 when a capture stream triggers.  Until we figure out the
+	 * real cause, disable tsched mode by telling the PCM info flag.
+	 */
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
+		runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
+
 	if (chip->align_buffer_size)
 		/* constrain buffer sizes to be multiple of 128
 		   bytes. This is more efficient in terms of memory
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index baa15374fbcb..f2a6df5e6bcb 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -31,7 +31,7 @@
 /* 14 unused */
 #define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
 #define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-/* 17 unused */
+#define AZX_DCAPS_AMD_WORKAROUND (1 << 17)	/* AMD-specific workaround */
 #define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
 #define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
 #define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1e14d7270adf..a6d8c0d77b84 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -64,6 +64,7 @@ enum {
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
 	POS_FIX_SKL,
+	POS_FIX_FIFO,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -332,6 +333,11 @@ enum {
 #define AZX_DCAPS_PRESET_ATI_HDMI_NS \
 	(AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF)
 
+/* quirks for AMD SB */
+#define AZX_DCAPS_PRESET_AMD_SB \
+	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
 	(AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\
@@ -841,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+#define AMD_FIFO_SIZE	32
+
+/* get the current DMA position with FIFO size correction */
+static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int pos, delay;
+
+	pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev));
+	if (!runtime)
+		return pos;
+
+	runtime->delay = AMD_FIFO_SIZE;
+	delay = frames_to_bytes(runtime, AMD_FIFO_SIZE);
+	if (azx_dev->insufficient) {
+		if (pos < delay) {
+			delay = pos;
+			runtime->delay = bytes_to_frames(runtime, pos);
+		} else {
+			azx_dev->insufficient = 0;
+		}
+	}
+
+	/* correct the DMA position for capture stream */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		if (pos < delay)
+			pos += azx_dev->core.bufsize;
+		pos -= delay;
+	}
+
+	return pos;
+}
+
+static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev,
+				   unsigned int pos)
+{
+	struct snd_pcm_substream *substream = azx_dev->core.substream;
+
+	/* just read back the calculated value in the above */
+	return substream->runtime->delay;
+}
+
 static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
 					 struct azx_dev *azx_dev)
 {
@@ -1417,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
 	case POS_FIX_SKL:
+	case POS_FIX_FIFO:
 		return fix;
 	}
 
@@ -1433,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n");
 		return POS_FIX_VIACOMBO;
 	}
+	if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) {
+		dev_dbg(chip->card->dev, "Using FIFO position fix\n");
+		return POS_FIX_FIFO;
+	}
 	if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) {
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
@@ -1453,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
 		[POS_FIX_SKL] = azx_get_pos_skl,
+		[POS_FIX_FIFO] = azx_get_pos_fifo,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1467,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix)
 			azx_get_delay_from_lpib;
 	}
 
+	if (fix == POS_FIX_FIFO)
+		chip->get_delay[0] = chip->get_delay[1] =
+			azx_get_delay_from_fifo;
 }
 
 /*
@@ -2447,6 +2505,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD Hudson */
 	{ PCI_DEVICE(0x1022, 0x780d),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB },
+	/* AMD, X370 & co */
+	{ PCI_DEVICE(0x1022, 0x1457),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
-- 
cgit v1.2.3


From c7cd7c748a3250ca33509f9235efab9c803aca09 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Thu, 8 Aug 2019 00:15:21 -0500
Subject: sound: fix a memory leak bug

In sound_insert_unit(), the controlling structure 's' is allocated through
kmalloc(). Then it is added to the sound driver list by invoking
__sound_insert_unit(). Later on, if __register_chrdev() fails, 's' is
removed from the list through __sound_remove_unit(). If 'index' is not less
than 0, -EBUSY is returned to indicate the error. However, 's' is not
deallocated on this execution path, leading to a memory leak bug.

To fix the above issue, free 's' before -EBUSY is returned.

Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/sound_core.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/sound_core.c b/sound/sound_core.c
index b730d97c4de6..90d118cd9164 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -275,7 +275,8 @@ retry:
 				goto retry;
 			}
 			spin_unlock(&sound_loader_lock);
-			return -EBUSY;
+			r = -EBUSY;
+			goto fail;
 		}
 	}
 
-- 
cgit v1.2.3


From 1be3c1fae6c1e1f5bb982b255d2034034454527a Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Thu, 8 Aug 2019 00:50:58 -0500
Subject: ALSA: firewire: fix a memory leak bug

In iso_packets_buffer_init(), 'b->packets' is allocated through
kmalloc_array(). Then, the aligned packet size is checked. If it is
larger than PAGE_SIZE, -EINVAL will be returned to indicate the error.
However, the allocated 'b->packets' is not deallocated on this path,
leading to a memory leak.

To fix the above issue, free 'b->packets' before returning the error code.

Fixes: 31ef9134eb52 ("ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v2.6.39+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/firewire/packets-buffer.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c
index 0d35359d25cd..0ecafd0c6722 100644
--- a/sound/firewire/packets-buffer.c
+++ b/sound/firewire/packets-buffer.c
@@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit,
 	packets_per_page = PAGE_SIZE / packet_size;
 	if (WARN_ON(!packets_per_page)) {
 		err = -EINVAL;
-		goto error;
+		goto err_packets;
 	}
 	pages = DIV_ROUND_UP(count, packets_per_page);
 
-- 
cgit v1.2.3


From de768ce45466f3009809719eb7b1f6f5277d9373 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 9 Aug 2019 11:23:00 +0200
Subject: ALSA: hda - Apply workaround for another AMD chip 1022:1487

MSI MPG X570 board is with another AMD HD-audio controller (PCI ID
1022:1487) and it requires the same workaround applied for X370, etc
(PCI ID 1022:1457).

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a6d8c0d77b84..99fc0917339b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2508,6 +2508,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* AMD, X370 & co */
 	{ PCI_DEVICE(0x1022, 0x1457),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
+	/* AMD, X570 & co */
+	{ PCI_DEVICE(0x1022, 0x1487),
+	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* AMD Stoney */
 	{ PCI_DEVICE(0x1022, 0x157a),
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB |
-- 
cgit v1.2.3


From cfef67f016e4c00a2f423256fc678a6967a9fc09 Mon Sep 17 00:00:00 2001
From: Wenwen Wang <wenwen@cs.uga.edu>
Date: Fri, 9 Aug 2019 23:29:48 -0500
Subject: ALSA: hda - Fix a memory leak bug

In snd_hda_parse_generic_codec(), 'spec' is allocated through kzalloc().
Then, the pin widgets in 'codec' are parsed. However, if the parsing
process fails, 'spec' is not deallocated, leading to a memory leak.

To fix the above issue, free 'spec' before returning the error.

Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 485edaba0037..8f2beb1f3ae4 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6100,7 +6100,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec)
 
 	err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0);
 	if (err < 0)
-		return err;
+		goto error;
 
 	err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg);
 	if (err < 0)
-- 
cgit v1.2.3


From 190d03814eb3b49d4f87ff38fef26d36f3568a60 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Aug 2019 17:39:56 +0200
Subject: ALSA: hda/realtek - Add quirk for HP Envy x360

HP Envy x360 (AMD Ryzen-based model) with 103c:8497 needs the same
quirk like HP Spectre x360 for enabling the mute LED over Mic3 pin.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204373
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index de224cbea7a0..8aaf1d9c55cf 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6987,6 +6987,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+	SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-- 
cgit v1.2.3


From 401714d9534aad8c24196b32600da683116bbe09 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 14 Aug 2019 12:09:07 +0800
Subject: ALSA: hda - Let all conexant codec enter D3 when rebooting

We have 3 new lenovo laptops which have conexant codec 0x14f11f86,
these 3 laptops also have the noise issue when rebooting, after
letting the codec enter D3 before rebooting or poweroff, the noise
disappers.

Instead of adding a new ID again in the reboot_notify(), let us make
this function apply to all conexant codec. In theory make codec enter
D3 before rebooting or poweroff is harmless, and I tested this change
on a couple of other Lenovo laptops which have different conexant
codecs, there is no side effect so far.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f299f137eaea..93a303676aea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -163,15 +163,6 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
 
-	switch (codec->core.vendor_id) {
-	case 0x14f12008: /* CX8200 */
-	case 0x14f150f2: /* CX20722 */
-	case 0x14f150f4: /* CX20724 */
-		break;
-	default:
-		return;
-	}
-
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
-- 
cgit v1.2.3


From 871b9066027702e6e6589da0e1edd3b7dede7205 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 14 Aug 2019 12:09:08 +0800
Subject: ALSA: hda - Add a generic reboot_notify

Make codec enter D3 before rebooting or poweroff can fix the noise
issue on some laptops. And in theory it is harmless for all codecs
to enter D3 before rebooting or poweroff, let us add a generic
reboot_notify, then realtek and conexant drivers can call this
function.

Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c    | 19 +++++++++++++++++++
 sound/pci/hda/hda_generic.h    |  1 +
 sound/pci/hda/patch_conexant.c |  6 +-----
 sound/pci/hda/patch_realtek.c  | 11 +----------
 4 files changed, 22 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8f2beb1f3ae4..5bf24fb819d2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -6051,6 +6051,24 @@ void snd_hda_gen_free(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_GPL(snd_hda_gen_free);
 
+/**
+ * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting
+ * @codec: the HDA codec
+ *
+ * This can be put as patch_ops reboot_notify function.
+ */
+void snd_hda_gen_reboot_notify(struct hda_codec *codec)
+{
+	/* Make the codec enter D3 to avoid spurious noises from the internal
+	 * speaker during (and after) reboot
+	 */
+	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
+	snd_hda_codec_write(codec, codec->core.afg, 0,
+			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+	msleep(10);
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify);
+
 #ifdef CONFIG_PM
 /**
  * snd_hda_gen_check_power_status - check the loopback power save state
@@ -6078,6 +6096,7 @@ static const struct hda_codec_ops generic_patch_ops = {
 	.init = snd_hda_gen_init,
 	.free = snd_hda_gen_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+	.reboot_notify = snd_hda_gen_reboot_notify,
 #ifdef CONFIG_PM
 	.check_power_status = snd_hda_gen_check_power_status,
 #endif
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 35a670a71c42..5f199dcb0d18 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -332,6 +332,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 				  struct auto_pin_cfg *cfg);
 int snd_hda_gen_build_controls(struct hda_codec *codec);
 int snd_hda_gen_build_pcms(struct hda_codec *codec);
+void snd_hda_gen_reboot_notify(struct hda_codec *codec);
 
 /* standard jack event callbacks */
 void snd_hda_gen_hp_automute(struct hda_codec *codec,
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 93a303676aea..14298ef45b21 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -166,11 +166,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec)
 	/* Turn the problematic codec into D3 to avoid spurious noises
 	   from the internal speaker during (and after) reboot */
 	cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false);
-
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
+	snd_hda_gen_reboot_notify(codec);
 }
 
 static void cx_auto_free(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8aaf1d9c55cf..e333b3e30e31 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -869,15 +869,6 @@ static void alc_reboot_notify(struct hda_codec *codec)
 		alc_shutup(codec);
 }
 
-/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */
-static void alc_d3_at_reboot(struct hda_codec *codec)
-{
-	snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3);
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
-	msleep(10);
-}
-
 #define alc_free	snd_hda_gen_free
 
 #ifdef CONFIG_PM
@@ -5152,7 +5143,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
-		spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */
+		spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */
 		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 		codec->power_save_node = 0; /* avoid click noises */
 		snd_hda_apply_pincfgs(codec, pincfgs);
-- 
cgit v1.2.3


From daac07156b330b18eb5071aec4b3ddca1c377f2c Mon Sep 17 00:00:00 2001
From: Hui Peng <benquike@gmail.com>
Date: Tue, 13 Aug 2019 22:34:04 -0400
Subject: ALSA: usb-audio: Fix an OOB bug in parse_audio_mixer_unit

The `uac_mixer_unit_descriptor` shown as below is read from the
device side. In `parse_audio_mixer_unit`, `baSourceID` field is
accessed from index 0 to `bNrInPins` - 1, the current implementation
assumes that descriptor is always valid (the length  of descriptor
is no shorter than 5 + `bNrInPins`). If a descriptor read from
the device side is invalid, it may trigger out-of-bound memory
access.

```
struct uac_mixer_unit_descriptor {
	__u8 bLength;
	__u8 bDescriptorType;
	__u8 bDescriptorSubtype;
	__u8 bUnitID;
	__u8 bNrInPins;
	__u8 baSourceID[];
}
```

This patch fixes the bug by add a sanity check on the length of
the descriptor.

Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Peng <benquike@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7498b5191b68..ea487378be17 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -744,6 +744,8 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
 		return -EINVAL;
 	if (!desc->bNrInPins)
 		return -EINVAL;
+	if (desc->bLength < sizeof(*desc) + desc->bNrInPins)
+		return -EINVAL;
 
 	switch (state->mixer->protocol) {
 	case UAC_VERSION_1:
-- 
cgit v1.2.3


From 19bce474c45be69a284ecee660aa12d8f1e88f18 Mon Sep 17 00:00:00 2001
From: Hui Peng <benquike@gmail.com>
Date: Thu, 15 Aug 2019 00:31:34 -0400
Subject: ALSA: usb-audio: Fix a stack buffer overflow bug in check_input_term

`check_input_term` recursively calls itself with input from
device side (e.g., uac_input_terminal_descriptor.bCSourceID)
as argument (id). In `check_input_term`, if `check_input_term`
is called with the same `id` argument as the caller, it triggers
endless recursive call, resulting kernel space stack overflow.

This patch fixes the bug by adding a bitmap to `struct mixer_build`
to keep track of the checked ids and stop the execution if some id
has been checked (similar to how parse_audio_unit handles unitid
argument).

Reported-by: Hui Peng <benquike@gmail.com>
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Signed-off-by: Hui Peng <benquike@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 35 +++++++++++++++++++++++++++--------
 1 file changed, 27 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ea487378be17..b5927c3d5bc0 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -68,6 +68,7 @@ struct mixer_build {
 	unsigned char *buffer;
 	unsigned int buflen;
 	DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS);
+	DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS);
 	struct usb_audio_term oterm;
 	const struct usbmix_name_map *map;
 	const struct usbmix_selector_map *selector_map;
@@ -775,16 +776,25 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state,
  * parse the source unit recursively until it reaches to a terminal
  * or a branched unit.
  */
-static int check_input_term(struct mixer_build *state, int id,
+static int __check_input_term(struct mixer_build *state, int id,
 			    struct usb_audio_term *term)
 {
 	int protocol = state->mixer->protocol;
 	int err;
 	void *p1;
+	unsigned char *hdr;
 
 	memset(term, 0, sizeof(*term));
-	while ((p1 = find_audio_control_unit(state, id)) != NULL) {
-		unsigned char *hdr = p1;
+	for (;;) {
+		/* a loop in the terminal chain? */
+		if (test_and_set_bit(id, state->termbitmap))
+			return -EINVAL;
+
+		p1 = find_audio_control_unit(state, id);
+		if (!p1)
+			break;
+
+		hdr = p1;
 		term->id = id;
 
 		if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
@@ -802,7 +812,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 					/* call recursively to verify that the
 					 * referenced clock entity is valid */
-					err = check_input_term(state, d->bCSourceID, term);
+					err = __check_input_term(state, d->bCSourceID, term);
 					if (err < 0)
 						return err;
 
@@ -836,7 +846,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC2_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -899,7 +909,7 @@ static int check_input_term(struct mixer_build *state, int id,
 
 				/* call recursively to verify that the
 				 * referenced clock entity is valid */
-				err = check_input_term(state, d->bCSourceID, term);
+				err = __check_input_term(state, d->bCSourceID, term);
 				if (err < 0)
 					return err;
 
@@ -950,7 +960,7 @@ static int check_input_term(struct mixer_build *state, int id,
 			case UAC3_CLOCK_SELECTOR: {
 				struct uac_selector_unit_descriptor *d = p1;
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 				term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */
@@ -966,7 +976,7 @@ static int check_input_term(struct mixer_build *state, int id,
 					return -EINVAL;
 
 				/* call recursively to retrieve the channel info */
-				err = check_input_term(state, d->baSourceID[0], term);
+				err = __check_input_term(state, d->baSourceID[0], term);
 				if (err < 0)
 					return err;
 
@@ -984,6 +994,15 @@ static int check_input_term(struct mixer_build *state, int id,
 	return -ENODEV;
 }
 
+
+static int check_input_term(struct mixer_build *state, int id,
+			    struct usb_audio_term *term)
+{
+	memset(term, 0, sizeof(*term));
+	memset(state->termbitmap, 0, sizeof(state->termbitmap));
+	return __check_input_term(state, id, term);
+}
+
 /*
  * Feature Unit
  */
-- 
cgit v1.2.3