From e958b5884725dac86d36c1e7afe5a55f31feb0b2 Mon Sep 17 00:00:00 2001
From: Robert Hancock <robert.hancock@calian.com>
Date: Fri, 7 Jan 2022 15:47:06 -0600
Subject: [PATCH 01/31] ASoC: xilinx: xlnx_formatter_pcm: Make buffer bytes
 multiple of period bytes

This patch is based on one in the Xilinx kernel tree, "ASoc: xlnx: Make
buffer bytes multiple of period bytes" by Devarsh Thakkar. The same
issue exists in the mainline version of the driver. The original
patch description is as follows:

"The Xilinx Audio Formatter IP has a constraint on period
bytes to be multiple of 64. This leads to driver changing
the period size to suitable frames such that period bytes
are multiple of 64.

Now since period bytes and period size are updated but not
the buffer bytes, this may make the buffer bytes unaligned
and not multiple of period bytes.

When this happens we hear popping noise as while DMA is being
done the buffer bytes are not enough to complete DMA access
for last period of frame within the application buffer boundary.

To avoid this, align buffer bytes too as multiple of 64, and
set another constraint to always enforce number of periods as
integer. Now since, there is already a rule in alsa core
to enforce Buffer size = Number of Periods * Period Size
this automatically aligns buffer bytes as multiple of period
bytes."

Fixes: 6f6c3c36f091 ("ASoC: xlnx: add pcm formatter platform driver")
Cc: Devarsh Thakkar <devarsh.thakkar@xilinx.com>
Signed-off-by: Robert Hancock <robert.hancock@calian.com>
Link: https://lore.kernel.org/r/20220107214711.1100162-2-robert.hancock@calian.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/xilinx/xlnx_formatter_pcm.c | 27 ++++++++++++++++++++++++---
 1 file changed, 24 insertions(+), 3 deletions(-)

diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index 91afea9d5de6..ce19a6058b27 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -37,6 +37,7 @@
 #define XLNX_AUD_XFER_COUNT	0x28
 #define XLNX_AUD_CH_STS_START	0x2C
 #define XLNX_BYTES_PER_CH	0x44
+#define XLNX_AUD_ALIGN_BYTES	64
 
 #define AUD_STS_IOC_IRQ_MASK	BIT(31)
 #define AUD_STS_CH_STS_MASK	BIT(29)
@@ -368,12 +369,32 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware);
 	runtime->private_data = stream_data;
 
-	/* Resize the period size divisible by 64 */
+	/* Resize the period bytes as divisible by 64 */
 	err = snd_pcm_hw_constraint_step(runtime, 0,
-					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					 XLNX_AUD_ALIGN_BYTES);
 	if (err) {
 		dev_err(component->dev,
-			"unable to set constraint on period bytes\n");
+			"Unable to set constraint on period bytes\n");
+		return err;
+	}
+
+	/* Resize the buffer bytes as divisible by 64 */
+	err = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					 XLNX_AUD_ALIGN_BYTES);
+	if (err) {
+		dev_err(component->dev,
+			"Unable to set constraint on buffer bytes\n");
+		return err;
+	}
+
+	/* Set periods as integer multiple */
+	err = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0) {
+		dev_err(component->dev,
+			"Unable to set constraint on periods to be integer\n");
 		return err;
 	}
 

From a64067f4cecaaa4deed8e33d3266bc0bcc189142 Mon Sep 17 00:00:00 2001
From: Robert Hancock <robert.hancock@calian.com>
Date: Fri, 7 Jan 2022 15:47:10 -0600
Subject: [PATCH 02/31] ASoC: simple-card: fix probe failure on platform
 component

A previous change to simple-card resulted in asoc_simple_parse_dai
attempting to retrieve the dai_name for platform components, which are
unlikely to have a valid DAI name. This caused simple-card to fail to
probe when using the xlnx_formatter_pcm as the platform component, since
it does not register any DAI components.

Since the dai_name is not used for platform components, just skip trying
to retrieve it for those.

Fixes: f107294c6422 ("ASoC: simple-card: support snd_soc_dai_link_component style for cpu")
Signed-off-by: Robert Hancock <robert.hancock@calian.com>
Link: https://lore.kernel.org/r/20220107214711.1100162-6-robert.hancock@calian.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/generic/simple-card.c | 26 +++++++++++++++++++++++++-
 1 file changed, 25 insertions(+), 1 deletion(-)

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index a89d1cfdda32..78419e18717d 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -28,6 +28,30 @@ static const struct snd_soc_ops simple_ops = {
 	.hw_params	= asoc_simple_hw_params,
 };
 
+static int asoc_simple_parse_platform(struct device_node *node,
+				      struct snd_soc_dai_link_component *dlc)
+{
+	struct of_phandle_args args;
+	int ret;
+
+	if (!node)
+		return 0;
+
+	/*
+	 * Get node via "sound-dai = <&phandle port>"
+	 * it will be used as xxx_of_node on soc_bind_dai_link()
+	 */
+	ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args);
+	if (ret)
+		return ret;
+
+	/* dai_name is not required and may not exist for plat component */
+
+	dlc->of_node = args.np;
+
+	return 0;
+}
+
 static int asoc_simple_parse_dai(struct device_node *node,
 				 struct snd_soc_dai_link_component *dlc,
 				 int *is_single_link)
@@ -289,7 +313,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = asoc_simple_parse_dai(plat, platforms, NULL);
+	ret = asoc_simple_parse_platform(plat, platforms);
 	if (ret < 0)
 		goto dai_link_of_err;
 

From f7a6021aaf02088870559f82fc13c58cda7fea1a Mon Sep 17 00:00:00 2001
From: Jiasheng Jiang <jiasheng@iscas.ac.cn>
Date: Tue, 11 Jan 2022 10:50:48 +0800
Subject: [PATCH 03/31] ASoC: cpcap: Check for NULL pointer after calling
 of_get_child_by_name

If the device does not exist, of_get_child_by_name() will return NULL
pointer.
And devm_snd_soc_register_component() does not check it.
Also, I have noticed that cpcap_codec_driver has not been used yet.
Therefore, it should be better to check it in order to avoid the future
dereference of the NULL pointer.

Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec")
Signed-off-by: Jiasheng Jiang <jiasheng@iscas.ac.cn>
Link: https://lore.kernel.org/r/20220111025048.524134-1-jiasheng@iscas.ac.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/cpcap.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index 598e09024e23..ffdf8b615efa 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -1667,6 +1667,8 @@ static int cpcap_codec_probe(struct platform_device *pdev)
 {
 	struct device_node *codec_node =
 		of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec");
+	if (!codec_node)
+		return -ENODEV;
 
 	pdev->dev.of_node = codec_node;
 

From 4c907bcd9dcd233da6707059d777ab389dcbd964 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Wed, 19 Jan 2022 15:31:01 +0300
Subject: [PATCH 04/31] ASoC: max9759: fix underflow in
 speaker_gain_control_put()

Check for negative values of "priv->gain" to prevent an out of bounds
access.  The concern is that these might come from the user via:
  -> snd_ctl_elem_write_user()
    -> snd_ctl_elem_write()
      -> kctl->put()

Fixes: fa8d915172b8 ("ASoC: max9759: Add Amplifier Driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20220119123101.GA9509@kili
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/max9759.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c
index d75fd61b9032..bc57d7687f16 100644
--- a/sound/soc/codecs/max9759.c
+++ b/sound/soc/codecs/max9759.c
@@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol);
 	struct max9759 *priv = snd_soc_component_get_drvdata(c);
 
-	if (ucontrol->value.integer.value[0] > 3)
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] > 3)
 		return -EINVAL;
 
 	priv->gain = ucontrol->value.integer.value[0];

From 579b2c8f72d974f27d85bbd53846f34675ee3b01 Mon Sep 17 00:00:00 2001
From: Julian Braha <julianbraha@gmail.com>
Date: Mon, 17 Jan 2022 00:03:24 -0500
Subject: [PATCH 05/31] ASoC: mediatek: fix unmet dependency on GPIOLIB for
 SND_SOC_DMIC

When SND_SOC_MT8195_MT6359_RT1011_RT5682 is selected,
and GPIOLIB is not selected,
Kbuild gives the following warning:

WARNING: unmet direct dependencies detected for SND_SOC_DMIC
  Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n]
  Selected by [y]:
  - SND_SOC_MT8195_MT6359_RT1011_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8195 [=y] && MTK_PMIC_WRAP [=y]

This is because SND_SOC_MT8195_MT6359_RT1011_RT5682
selects SND_SOC_DMIC without selecting or depending on
GPIOLIB, depsite SND_SOC_DMIC depending on GPIOLIB.

This unmet dependency bug was detected by Kismet,
a static analysis tool for Kconfig. Please advise
if this is not the appropriate solution.

Signed-off-by: Julian Braha <julianbraha@gmail.com>
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20220117050324.68371-1-julianbraha@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/mediatek/Kconfig | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 9306b7ca2644..0d154350f180 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -216,7 +216,7 @@ config SND_SOC_MT8195_MT6359_RT1019_RT5682
 
 config SND_SOC_MT8195_MT6359_RT1011_RT5682
 	tristate "ASoC Audio driver for MT8195 with MT6359 RT1011 RT5682 codec"
-	depends on I2C
+	depends on I2C && GPIOLIB
 	depends on SND_SOC_MT8195 && MTK_PMIC_WRAP
 	select SND_SOC_MT6359
 	select SND_SOC_RT1011

From 248be352bbae1a0f14d0d3511a5b0bb9665097f5 Mon Sep 17 00:00:00 2001
From: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Date: Thu, 20 Jan 2022 19:06:01 +0530
Subject: [PATCH 06/31] ASoC: amd: acp-mach: Fix Left and Right rt1019 amp
 devices

We're setting wrong card codec conf for rt1019 amp devices in our
machine driver. Due to this left and right amp channels data are
reversed in our machines as wrong device prefix results in wrong
value for "Mono LR Select" rt1019 mixer control. Reverse dev ids
in codec conf with Left and Right name_prefix to fix such issue.

Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Link: https://lore.kernel.org/r/20220120133605.476138-1-AjitKumar.Pandey@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/amd/acp/acp-mach-common.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c
index c9caade5cb74..cd05ee2802c9 100644
--- a/sound/soc/amd/acp/acp-mach-common.c
+++ b/sound/soc/amd/acp/acp-mach-common.c
@@ -303,11 +303,11 @@ static const struct snd_soc_dapm_route rt1019_map_lr[] = {
 
 static struct snd_soc_codec_conf rt1019_conf[] = {
 	{
-		 .dlc = COMP_CODEC_CONF("i2c-10EC1019:00"),
+		 .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"),
 		 .name_prefix = "Left",
 	},
 	{
-		 .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"),
+		 .dlc = COMP_CODEC_CONF("i2c-10EC1019:00"),
 		 .name_prefix = "Right",
 	},
 };

From 817f7c9335ec01e0f5e8caffc4f1dcd5e458a4c0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Mon, 24 Jan 2022 15:32:51 +0000
Subject: [PATCH 07/31] ASoC: ops: Reject out of bounds values in
 snd_soc_put_volsw()

We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.

Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20220124153253.3548853-2-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-ops.c | 18 ++++++++++++++++--
 1 file changed, 16 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 08eaa9ddf191..fbe5d326b0f2 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -316,13 +316,27 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	if (sign_bit)
 		mask = BIT(sign_bit + 1) - 1;
 
-	val = ((ucontrol->value.integer.value[0] + min) & mask);
+	val = ucontrol->value.integer.value[0];
+	if (mc->platform_max && val > mc->platform_max)
+		return -EINVAL;
+	if (val > max - min)
+		return -EINVAL;
+	if (val < 0)
+		return -EINVAL;
+	val = (val + min) & mask;
 	if (invert)
 		val = max - val;
 	val_mask = mask << shift;
 	val = val << shift;
 	if (snd_soc_volsw_is_stereo(mc)) {
-		val2 = ((ucontrol->value.integer.value[1] + min) & mask);
+		val2 = ucontrol->value.integer.value[1];
+		if (mc->platform_max && val2 > mc->platform_max)
+			return -EINVAL;
+		if (val2 > max - min)
+			return -EINVAL;
+		if (val2 < 0)
+			return -EINVAL;
+		val2 = (val2 + min) & mask;
 		if (invert)
 			val2 = max - val2;
 		if (reg == reg2) {

From 4f1e50d6a9cf9c1b8c859d449b5031cacfa8404e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Mon, 24 Jan 2022 15:32:52 +0000
Subject: [PATCH 08/31] ASoC: ops: Reject out of bounds values in
 snd_soc_put_volsw_sx()

We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.

Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20220124153253.3548853-3-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-ops.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index fbe5d326b0f2..c31e63b27193 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -423,8 +423,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
 	int err = 0;
 	unsigned int val, val_mask;
 
+	val = ucontrol->value.integer.value[0];
+	if (mc->platform_max && val > mc->platform_max)
+		return -EINVAL;
+	if (val > max - min)
+		return -EINVAL;
+	if (val < 0)
+		return -EINVAL;
 	val_mask = mask << shift;
-	val = (ucontrol->value.integer.value[0] + min) & mask;
+	val = (val + min) & mask;
 	val = val << shift;
 
 	err = snd_soc_component_update_bits(component, reg, val_mask, val);

From 4cf28e9ae6e2e11a044be1bcbcfa1b0d8675fe4d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Mon, 24 Jan 2022 15:32:53 +0000
Subject: [PATCH 09/31] ASoC: ops: Reject out of bounds values in
 snd_soc_put_xr_sx()

We don't currently validate that the values being set are within the range
we advertised to userspace as being valid, do so and reject any values
that are out of range.

Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20220124153253.3548853-4-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-ops.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index c31e63b27193..dc0e7c8d31f3 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -879,6 +879,8 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
 	long val = ucontrol->value.integer.value[0];
 	unsigned int i;
 
+	if (val < mc->min || val > mc->max)
+		return -EINVAL;
 	if (invert)
 		val = max - val;
 	val &= mask;

From c5c1546a654f613e291a7c5d6f3660fc1eb6d0c7 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 26 Jan 2022 11:35:46 +0000
Subject: [PATCH 10/31] ASoC: codecs: wcd938x: fix incorrect used of portid

Mixer controls have the channel id in mixer->reg, which is not same
as port id. port id should be derived from chan_info array.
So fix this. Without this, its possible that we could corrupt
struct wcd938x_sdw_priv by accessing port_map array out of range
with channel id instead of port id.

Fixes: e8ba1e05bdc0 ("ASoC: codecs: wcd938x: add basic controls")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220126113549.8853-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd938x.c | 25 ++++++++++++++-----------
 1 file changed, 14 insertions(+), 11 deletions(-)

diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index eff200a07d9f..5994644c8702 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -1432,14 +1432,10 @@ static int wcd938x_sdw_connect_port(struct wcd938x_sdw_ch_info *ch_info,
 	return 0;
 }
 
-static int wcd938x_connect_port(struct wcd938x_sdw_priv *wcd, u8 ch_id, u8 enable)
+static int wcd938x_connect_port(struct wcd938x_sdw_priv *wcd, u8 port_num, u8 ch_id, u8 enable)
 {
-	u8 port_num;
-
-	port_num = wcd->ch_info[ch_id].port_num;
-
 	return wcd938x_sdw_connect_port(&wcd->ch_info[ch_id],
-					&wcd->port_config[port_num],
+					&wcd->port_config[port_num - 1],
 					enable);
 }
 
@@ -2593,6 +2589,7 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol,
 	struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component);
 	struct wcd938x_sdw_priv *wcd;
 	int value = ucontrol->value.integer.value[0];
+	int portidx;
 	struct soc_mixer_control *mc;
 	bool hphr;
 
@@ -2606,10 +2603,12 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol,
 	else
 		wcd938x->comp1_enable = value;
 
+	portidx = wcd->ch_info[mc->reg].port_num;
+
 	if (value)
-		wcd938x_connect_port(wcd, mc->reg, true);
+		wcd938x_connect_port(wcd, portidx, mc->reg, true);
 	else
-		wcd938x_connect_port(wcd, mc->reg, false);
+		wcd938x_connect_port(wcd, portidx, mc->reg, false);
 
 	return 0;
 }
@@ -2882,9 +2881,11 @@ static int wcd938x_get_swr_port(struct snd_kcontrol *kcontrol,
 	struct wcd938x_sdw_priv *wcd;
 	struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value;
 	int dai_id = mixer->shift;
-	int portidx = mixer->reg;
+	int portidx, ch_idx = mixer->reg;
+
 
 	wcd = wcd938x->sdw_priv[dai_id];
+	portidx = wcd->ch_info[ch_idx].port_num;
 
 	ucontrol->value.integer.value[0] = wcd->port_enable[portidx];
 
@@ -2899,12 +2900,14 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol,
 	struct wcd938x_sdw_priv *wcd;
 	struct soc_mixer_control *mixer =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	int portidx = mixer->reg;
+	int ch_idx = mixer->reg;
+	int portidx;
 	int dai_id = mixer->shift;
 	bool enable;
 
 	wcd = wcd938x->sdw_priv[dai_id];
 
+	portidx = wcd->ch_info[ch_idx].port_num;
 	if (ucontrol->value.integer.value[0])
 		enable = true;
 	else
@@ -2912,7 +2915,7 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol,
 
 	wcd->port_enable[portidx] = enable;
 
-	wcd938x_connect_port(wcd, portidx, enable);
+	wcd938x_connect_port(wcd, portidx, ch_idx, enable);
 
 	return 0;
 

From fca041a3ab70a099a6d5519ecb689b6279bd04f3 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 26 Jan 2022 11:35:47 +0000
Subject: [PATCH 11/31] ASoC: codecs: lpass-rx-macro: fix sidetone register
 offsets

For some reason we ended up with incorrect register offfset calcuations
for sidetone. regmap clearly throw errors when accessing these incorrect
registers as these do not belong to any read/write ranges.
so fix them to point to correct register offsets.

Fixes: f3ce6f3c9a99 ("ASoC: codecs: lpass-rx-macro: add iir widgets")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220126113549.8853-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lpass-rx-macro.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c
index aec5127260fd..6ffe88345de5 100644
--- a/sound/soc/codecs/lpass-rx-macro.c
+++ b/sound/soc/codecs/lpass-rx-macro.c
@@ -2688,8 +2688,8 @@ static uint32_t get_iir_band_coeff(struct snd_soc_component *component,
 	int reg, b2_reg;
 
 	/* Address does not automatically update if reading */
-	reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx;
-	b2_reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx;
+	reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 0x80 * iir_idx;
+	b2_reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 0x80 * iir_idx;
 
 	snd_soc_component_write(component, reg,
 				((band_idx * BAND_MAX + coeff_idx) *
@@ -2718,7 +2718,7 @@ static uint32_t get_iir_band_coeff(struct snd_soc_component *component,
 static void set_iir_band_coeff(struct snd_soc_component *component,
 			       int iir_idx, int band_idx, uint32_t value)
 {
-	int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx;
+	int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 0x80 * iir_idx;
 
 	snd_soc_component_write(component, reg, (value & 0xFF));
 	snd_soc_component_write(component, reg, (value >> 8) & 0xFF);
@@ -2739,7 +2739,7 @@ static int rx_macro_put_iir_band_audio_mixer(
 	int iir_idx = ctl->iir_idx;
 	int band_idx = ctl->band_idx;
 	u32 coeff[BAND_MAX];
-	int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx;
+	int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 0x80 * iir_idx;
 
 	memcpy(&coeff[0], ucontrol->value.bytes.data, params->max);
 

From bd2347fd67d8da0fa76296507cc556da0a233bcb Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 26 Jan 2022 11:35:48 +0000
Subject: [PATCH 12/31] ASoC: codecs: wcd938x: fix return value of mixer put
 function

wcd938x_ear_pa_put_gain, wcd938x_set_swr_port and  wcd938x_set_compander
currently returns zero eventhough it changes the value.
Fix this, so that change notifications are sent correctly.

Fixes: e8ba1e05bdc01 ("ASoC: codecs: wcd938x: add basic controls")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220126113549.8853-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wcd938x.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
index 5994644c8702..36cbc66914f9 100644
--- a/sound/soc/codecs/wcd938x.c
+++ b/sound/soc/codecs/wcd938x.c
@@ -2559,7 +2559,7 @@ static int wcd938x_ear_pa_put_gain(struct snd_kcontrol *kcontrol,
 				      WCD938X_EAR_GAIN_MASK,
 				      ucontrol->value.integer.value[0]);
 
-	return 0;
+	return 1;
 }
 
 static int wcd938x_get_compander(struct snd_kcontrol *kcontrol,
@@ -2610,7 +2610,7 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol,
 	else
 		wcd938x_connect_port(wcd, portidx, mc->reg, false);
 
-	return 0;
+	return 1;
 }
 
 static int wcd938x_ldoh_get(struct snd_kcontrol *kcontrol,
@@ -2917,7 +2917,7 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol,
 
 	wcd938x_connect_port(wcd, portidx, ch_idx, enable);
 
-	return 0;
+	return 1;
 
 }
 

From 8f2e5c65ec7534cce6d315fccf2c3aef023f68f0 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 26 Jan 2022 11:35:49 +0000
Subject: [PATCH 13/31] ASoC: qdsp6: q6apm-dai: only stop graphs that are
 started

Its possible that the sound card is just opened and closed without actually
playing stream, ex: if the audio file itself is missing.

Even in such cases we do call stop on graphs that are not yet started.
DSP can throw errors in such cases, so add a check to see if the graph
was started before stopping it.

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220126113549.8853-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index eb1c3aec479b..19c4a90ec1ea 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -308,8 +308,11 @@ static int q6apm_dai_close(struct snd_soc_component *component,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct q6apm_dai_rtd *prtd = runtime->private_data;
 
-	q6apm_graph_stop(prtd->graph);
-	q6apm_unmap_memory_regions(prtd->graph, substream->stream);
+	if (prtd->state) { /* only stop graph that is started */
+		q6apm_graph_stop(prtd->graph);
+		q6apm_unmap_memory_regions(prtd->graph, substream->stream);
+	}
+
 	q6apm_graph_close(prtd->graph);
 	prtd->graph = NULL;
 	kfree(prtd);

From 549f8ffc7b2f7561bea7f90930b6c5104318e87b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 26 Jan 2022 15:50:11 +0100
Subject: [PATCH 14/31] ALSA: hda: Fix UAF of leds class devs at unbinding

The LED class devices that are created by HD-audio codec drivers are
registered via devm_led_classdev_register() and associated with the
HD-audio codec device.  Unfortunately, it turned out that the devres
release doesn't work for this case; namely, since the codec resource
release happens before the devm call chain, it triggers a NULL
dereference or a UAF for a stale set_brightness_delay callback.

For fixing the bug, this patch changes the LED class device register
and unregister in a manual manner without devres, keeping the
instances in hda_gen_spec.

Reported-by: Alexander Sergeyev <sergeev917@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdomain
Link: https://lore.kernel.org/r/20220126145011.16728-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 17 +++++++++++++++--
 sound/pci/hda/hda_generic.h |  3 +++
 2 files changed, 18 insertions(+), 2 deletions(-)

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 3bf5e3410703..fc114e522480 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -91,6 +91,12 @@ static void snd_hda_gen_spec_free(struct hda_gen_spec *spec)
 	free_kctls(spec);
 	snd_array_free(&spec->paths);
 	snd_array_free(&spec->loopback_list);
+#ifdef CONFIG_SND_HDA_GENERIC_LEDS
+	if (spec->led_cdevs[LED_AUDIO_MUTE])
+		led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MUTE]);
+	if (spec->led_cdevs[LED_AUDIO_MICMUTE])
+		led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MICMUTE]);
+#endif
 }
 
 /*
@@ -3922,7 +3928,10 @@ static int create_mute_led_cdev(struct hda_codec *codec,
 						enum led_brightness),
 				bool micmute)
 {
+	struct hda_gen_spec *spec = codec->spec;
 	struct led_classdev *cdev;
+	int idx = micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE;
+	int err;
 
 	cdev = devm_kzalloc(&codec->core.dev, sizeof(*cdev), GFP_KERNEL);
 	if (!cdev)
@@ -3932,10 +3941,14 @@ static int create_mute_led_cdev(struct hda_codec *codec,
 	cdev->max_brightness = 1;
 	cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute";
 	cdev->brightness_set_blocking = callback;
-	cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE);
+	cdev->brightness = ledtrig_audio_get(idx);
 	cdev->flags = LED_CORE_SUSPENDRESUME;
 
-	return devm_led_classdev_register(&codec->core.dev, cdev);
+	err = led_classdev_register(&codec->core.dev, cdev);
+	if (err < 0)
+		return err;
+	spec->led_cdevs[idx] = cdev;
+	return 0;
 }
 
 /**
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index 8e1bc8ea74fc..34eba40cc6e6 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -294,6 +294,9 @@ struct hda_gen_spec {
 				   struct hda_jack_callback *cb);
 	void (*mic_autoswitch_hook)(struct hda_codec *codec,
 				    struct hda_jack_callback *cb);
+
+	/* leds */
+	struct led_classdev *led_cdevs[NUM_AUDIO_LEDS];
 };
 
 /* values for add_stereo_mix_input flag */

From 3da4b7403db87d39bc2613cfd790de1de99a70ab Mon Sep 17 00:00:00 2001
From: Tom Rix <trix@redhat.com>
Date: Wed, 26 Jan 2022 10:21:42 -0800
Subject: [PATCH 15/31] ALSA: usb-audio: initialize variables that could ignore
 errors

clang static analysis reports this representative issue
mixer.c:1548:35: warning: Assigned value is garbage or undefined
        ucontrol->value.integer.value[0] = val;
                                         ^ ~~~

The filter_error() macro allows errors to be ignored.
If errors can be ignored, initialize variables
so garbage will not be used.

Fixes: 48cc42973509 ("ALSA: usb-audio: Filter error from connector kctl ops, too")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20220126182142.1184819-1-trix@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index e8f3f8d622ec..630766ba259f 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1527,6 +1527,10 @@ error:
 		usb_audio_err(chip,
 			"cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
 			UAC_GET_CUR, validx, idx, cval->val_type);
+
+		if (val)
+			*val = 0;
+
 		return filter_error(cval, ret);
 	}
 

From 0444f82766f0b5b9c8302ad802dafa5dd0e722d0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 27 Jan 2022 14:57:17 +0100
Subject: [PATCH 16/31] ALSA: hda: Fix signedness of sscanf() arguments

The %x format of sscanf() takes an unsigned int pointer, while we pass
a signed int pointer.  Practically it's OK, but this may result in a
compile warning.  Let's fix it.

Fixes: a235d5b8e550 ("ALSA: hda: Allow model option to specify PCI SSID alias")
Reported-by: kernel test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20220127135717.31751-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 82c492b05667..cd1db943b7e0 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -981,7 +981,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 	int id = HDA_FIXUP_ID_NOT_SET;
 	const char *name = NULL;
 	const char *type = NULL;
-	int vendor, device;
+	unsigned int vendor, device;
 
 	if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET)
 		return;

From fb25621da5702c104ce0a48de5b174ced09e5b4e Mon Sep 17 00:00:00 2001
From: Miaoqian Lin <linmq006@gmail.com>
Date: Thu, 27 Jan 2022 13:13:34 +0000
Subject: [PATCH 17/31] ASoC: fsl: Add missing error handling in
 pcm030_fabric_probe

Add the missing platform_device_put() and platform_device_del()
before return from pcm030_fabric_probe in the error handling case.

Fixes: c912fa913446 ("ASoC: fsl: register the wm9712-codec")
Signed-off-by: Miaoqian Lin <linmq006@gmail.com>
Link: https://lore.kernel.org/r/20220127131336.30214-1-linmq006@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/pcm030-audio-fabric.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index af3c3b90c0ac..83b4a22bf15a 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -93,16 +93,21 @@ static int pcm030_fabric_probe(struct platform_device *op)
 		dev_err(&op->dev, "platform_device_alloc() failed\n");
 
 	ret = platform_device_add(pdata->codec_device);
-	if (ret)
+	if (ret) {
 		dev_err(&op->dev, "platform_device_add() failed: %d\n", ret);
+		platform_device_put(pdata->codec_device);
+	}
 
 	ret = snd_soc_register_card(card);
-	if (ret)
+	if (ret) {
 		dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
+		platform_device_del(pdata->codec_device);
+		platform_device_put(pdata->codec_device);
+	}
 
 	platform_set_drvdata(op, pdata);
-
 	return ret;
+
 }
 
 static int pcm030_fabric_remove(struct platform_device *op)

From 3c75c0ea5da749bd1efebd1387f2e5011b8c7d78 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Jan 2022 16:52:48 +0100
Subject: [PATCH 18/31] ASoC: soc-pcm: Fix DPCM lockdep warning due to nested
 stream locks

The recent change for DPCM locking caused spurious lockdep warnings.
Actually the warnings are false-positive, as those are triggered due
to the nested stream locks for FE and BE.  Since both locks belong to
the same lock class, lockdep sees it as if a deadlock.

For fixing this, we need to take PCM stream locks for BE with the
nested lock primitives.  Since currently snd_pcm_stream_lock*() helper
assumes only the top-level single locking, a new helper function
snd_pcm_stream_lock_irqsave_nested() is defined for a single-depth
nested lock, which is now used in the BE DAI trigger that is always
performed inside a FE stream lock.

Fixes: b2ae80663008 ("ASoC: soc-pcm: serialize BE triggers")
Reported-and-tested-by: Hans de Goede <hdegoede@redhat.com>
Reported-and-tested-by: Marek Szyprowski <m.szyprowski@samsung.com>
Link: https://lore.kernel.org/r/73018f3c-9769-72ea-0325-b3f8e2381e30@redhat.com
Link: https://lore.kernel.org/alsa-devel/9a0abddd-49e9-872d-2f00-a1697340f786@samsung.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20220119155249.26754-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/pcm.h     | 15 +++++++++++++++
 sound/core/pcm_native.c | 13 +++++++++++++
 sound/soc/soc-pcm.c     |  6 +++---
 3 files changed, 31 insertions(+), 3 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 33451f8ff755..524220fe1af6 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -614,6 +614,7 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream);
 void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream);
 void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream);
 unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream);
+unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream);
 
 /**
  * snd_pcm_stream_lock_irqsave - Lock the PCM stream
@@ -632,6 +633,20 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream);
 void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
 				      unsigned long flags);
 
+/**
+ * snd_pcm_stream_lock_irqsave_nested - Single-nested PCM stream locking
+ * @substream: PCM substream
+ * @flags: irq flags
+ *
+ * This locks the PCM stream like snd_pcm_stream_lock_irqsave() but with
+ * the single-depth lockdep subclass.
+ */
+#define snd_pcm_stream_lock_irqsave_nested(substream, flags)		\
+	do {								\
+		typecheck(unsigned long, flags);			\
+		flags = _snd_pcm_stream_lock_irqsave_nested(substream); \
+	} while (0)
+
 /**
  * snd_pcm_group_for_each_entry - iterate over the linked substreams
  * @s: the iterator
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 621883e71194..a056b3ef3c84 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -172,6 +172,19 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream)
 }
 EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
 
+unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream)
+{
+	unsigned long flags = 0;
+	if (substream->pcm->nonatomic)
+		mutex_lock_nested(&substream->self_group.mutex,
+				  SINGLE_DEPTH_NESTING);
+	else
+		spin_lock_irqsave_nested(&substream->self_group.lock, flags,
+					 SINGLE_DEPTH_NESTING);
+	return flags;
+}
+EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested);
+
 /**
  * snd_pcm_stream_unlock_irqrestore - Unlock the PCM stream
  * @substream: PCM substream
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 7abfc48b26ca..e8876e65c649 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -46,8 +46,8 @@ static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd,
 	snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream));
 }
 
-#define snd_soc_dpcm_stream_lock_irqsave(rtd, stream, flags) \
-	snd_pcm_stream_lock_irqsave(snd_soc_dpcm_get_substream(rtd, stream), flags)
+#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \
+	snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags)
 
 static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd,
 						  int stream)
@@ -2094,7 +2094,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
 		be = dpcm->be;
 		be_substream = snd_soc_dpcm_get_substream(be, stream);
 
-		snd_soc_dpcm_stream_lock_irqsave(be, stream, flags);
+		snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags);
 
 		/* is this op for this BE ? */
 		if (!snd_soc_dpcm_be_can_update(fe, be, stream))

From 9f620684c1ef5a002b6622ecc7b5818e81252f48 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Jan 2022 16:52:49 +0100
Subject: [PATCH 19/31] ASoC: soc-pcm: Move debugfs removal out of spinlock

The recent fix for DPCM locking also covered the loop in
dpcm_be_disconnect() with the FE stream lock.  This caused an
unexpected side effect, thought: calling debugfs_remove_recursive() in
the spinlock may lead to lockdep splats as the code there assumes the
SOFTIRQ-safe context.

For avoiding the problem, this patch changes the disconnection
procedure to two phases: at first, the matching entries are removed
from the linked list, then the resources are freed outside the lock.

Fixes: b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking")
Reported-and-tested-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20220119155249.26754-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-pcm.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e8876e65c649..9a954680d492 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1268,6 +1268,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe,
 void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 {
 	struct snd_soc_dpcm *dpcm, *d;
+	LIST_HEAD(deleted_dpcms);
 
 	snd_soc_dpcm_mutex_assert_held(fe);
 
@@ -1287,13 +1288,18 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
 		/* BEs still alive need new FE */
 		dpcm_be_reparent(fe, dpcm->be, stream);
 
-		dpcm_remove_debugfs_state(dpcm);
-
 		list_del(&dpcm->list_be);
-		list_del(&dpcm->list_fe);
-		kfree(dpcm);
+		list_move(&dpcm->list_fe, &deleted_dpcms);
 	}
 	snd_soc_dpcm_stream_unlock_irq(fe, stream);
+
+	while (!list_empty(&deleted_dpcms)) {
+		dpcm = list_first_entry(&deleted_dpcms, struct snd_soc_dpcm,
+					list_fe);
+		list_del(&dpcm->list_fe);
+		dpcm_remove_debugfs_state(dpcm);
+		kfree(dpcm);
+	}
 }
 
 /* get BE for DAI widget and stream */

From 06feec6005c9d9500cd286ec440aabf8b2ddd94d Mon Sep 17 00:00:00 2001
From: Dmitry Osipenko <digetx@gmail.com>
Date: Wed, 12 Jan 2022 22:50:39 +0300
Subject: [PATCH 20/31] ASoC: hdmi-codec: Fix OOB memory accesses

Correct size of iec_status array by changing it to the size of status
array of the struct snd_aes_iec958. This fixes out-of-bounds slab
read accesses made by memcpy() of the hdmi-codec driver. This problem
is reported by KASAN.

Cc: stable@vger.kernel.org
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20220112195039.1329-1-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/uapi/sound/asound.h   | 4 +++-
 sound/soc/codecs/hdmi-codec.c | 2 +-
 2 files changed, 4 insertions(+), 2 deletions(-)

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index ff7e638221c5..228279ea0670 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -56,8 +56,10 @@
  *                                                                          *
  ****************************************************************************/
 
+#define AES_IEC958_STATUS_SIZE		24
+
 struct snd_aes_iec958 {
-	unsigned char status[24];	/* AES/IEC958 channel status bits */
+	unsigned char status[AES_IEC958_STATUS_SIZE]; /* AES/IEC958 channel status bits */
 	unsigned char subcode[147];	/* AES/IEC958 subcode bits */
 	unsigned char pad;		/* nothing */
 	unsigned char dig_subframe[4];	/* AES/IEC958 subframe bits */
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index b61f980cabdc..b07607a9ecea 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -277,7 +277,7 @@ struct hdmi_codec_priv {
 	bool busy;
 	struct snd_soc_jack *jack;
 	unsigned int jack_status;
-	u8 iec_status[5];
+	u8 iec_status[AES_IEC958_STATUS_SIZE];
 };
 
 static const struct snd_soc_dapm_widget hdmi_widgets[] = {

From 4045daf0fa87846a27f56329fddad2deeb5ca354 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Wed, 26 Jan 2022 12:03:25 +0200
Subject: [PATCH 21/31] ASoC: rt5682: Fix deadlock on resume

On resume from suspend the following chain of events can happen:
A rt5682_resume() -> mod_delayed_work() for jack_detect_work
B DAPM sequence starts ( DAPM is locked now)

A1. rt5682_jack_detect_handler() scheduled
 - Takes both jdet_mutex and calibrate_mutex
 - Calls in to rt5682_headset_detect() which tries to take DAPM lock, it
   starts to wait for it as B path took it already.
B1. DAPM sequence reaches the "HP Amp", rt5682_hp_event() tries to take
    the jdet_mutex, but it is locked in A1, so it waits.

Deadlock.

To solve the deadlock, drop the jdet_mutex, use the jack_detect_work to do
the jack removal handling, move the dapm lock up one level to protect the
most of the rt5682_jack_detect_handler(), but not the jack reporting as it
might trigger a DAPM sequence.
The rt5682_headset_detect() can be changed to static as well.

Fixes: 8deb34a90f063 ("ASoC: rt5682: fix the wrong jack type detected")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20220126100325.16513-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5682-i2c.c | 15 ++++-----------
 sound/soc/codecs/rt5682.c     | 24 ++++++++----------------
 sound/soc/codecs/rt5682.h     |  2 --
 3 files changed, 12 insertions(+), 29 deletions(-)

diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c
index 20e0f90ea498..20fc0f3766de 100644
--- a/sound/soc/codecs/rt5682-i2c.c
+++ b/sound/soc/codecs/rt5682-i2c.c
@@ -59,18 +59,12 @@ static void rt5682_jd_check_handler(struct work_struct *work)
 	struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv,
 		jd_check_work.work);
 
-	if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL)
-		& RT5682_JDH_RS_MASK) {
+	if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) & RT5682_JDH_RS_MASK)
 		/* jack out */
-		rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0);
-
-		snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
-			SND_JACK_HEADSET |
-			SND_JACK_BTN_0 | SND_JACK_BTN_1 |
-			SND_JACK_BTN_2 | SND_JACK_BTN_3);
-	} else {
+		mod_delayed_work(system_power_efficient_wq,
+				 &rt5682->jack_detect_work, 0);
+	else
 		schedule_delayed_work(&rt5682->jd_check_work, 500);
-	}
 }
 
 static irqreturn_t rt5682_irq(int irq, void *data)
@@ -198,7 +192,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c,
 	}
 
 	mutex_init(&rt5682->calibrate_mutex);
-	mutex_init(&rt5682->jdet_mutex);
 	rt5682_calibrate(rt5682);
 
 	rt5682_apply_patch_list(rt5682, &i2c->dev);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 415ec564c82e..0a0ec4a021e1 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -922,15 +922,13 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component,
  *
  * Returns detect status.
  */
-int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
+static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dapm_context *dapm = &component->dapm;
 	unsigned int val, count;
 
 	if (jack_insert) {
-		snd_soc_dapm_mutex_lock(dapm);
-
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
 			RT5682_PWR_VREF2 | RT5682_PWR_MB,
 			RT5682_PWR_VREF2 | RT5682_PWR_MB);
@@ -981,8 +979,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 		snd_soc_component_update_bits(component, RT5682_MICBIAS_2,
 			RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK,
 			RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU);
-
-		snd_soc_dapm_mutex_unlock(dapm);
 	} else {
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
@@ -1011,7 +1007,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 	dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type);
 	return rt5682->jack_type;
 }
-EXPORT_SYMBOL_GPL(rt5682_headset_detect);
 
 static int rt5682_set_jack_detect(struct snd_soc_component *component,
 		struct snd_soc_jack *hs_jack, void *data)
@@ -1094,6 +1089,7 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 {
 	struct rt5682_priv *rt5682 =
 		container_of(work, struct rt5682_priv, jack_detect_work.work);
+	struct snd_soc_dapm_context *dapm;
 	int val, btn_type;
 
 	while (!rt5682->component)
@@ -1102,7 +1098,9 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 	while (!rt5682->component->card->instantiated)
 		usleep_range(10000, 15000);
 
-	mutex_lock(&rt5682->jdet_mutex);
+	dapm = snd_soc_component_get_dapm(rt5682->component);
+
+	snd_soc_dapm_mutex_lock(dapm);
 	mutex_lock(&rt5682->calibrate_mutex);
 
 	val = snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL)
@@ -1162,6 +1160,9 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 		rt5682->irq_work_delay_time = 50;
 	}
 
+	mutex_unlock(&rt5682->calibrate_mutex);
+	snd_soc_dapm_mutex_unlock(dapm);
+
 	snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type,
 		SND_JACK_HEADSET |
 		SND_JACK_BTN_0 | SND_JACK_BTN_1 |
@@ -1174,9 +1175,6 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 		else
 			cancel_delayed_work_sync(&rt5682->jd_check_work);
 	}
-
-	mutex_unlock(&rt5682->calibrate_mutex);
-	mutex_unlock(&rt5682->jdet_mutex);
 }
 EXPORT_SYMBOL_GPL(rt5682_jack_detect_handler);
 
@@ -1526,7 +1524,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_component *component =
 		snd_soc_dapm_to_component(w->dapm);
-	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -1538,17 +1535,12 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w,
 			RT5682_DEPOP_1, 0x60, 0x60);
 		snd_soc_component_update_bits(component,
 			RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080);
-
-		mutex_lock(&rt5682->jdet_mutex);
-
 		snd_soc_component_update_bits(component, RT5682_HP_CTRL_2,
 			RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN,
 			RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN);
 		usleep_range(5000, 10000);
 		snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1,
 			RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L);
-
-		mutex_unlock(&rt5682->jdet_mutex);
 		break;
 
 	case SND_SOC_DAPM_POST_PMD:
diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h
index c917c76200ea..52ff0d9c36c5 100644
--- a/sound/soc/codecs/rt5682.h
+++ b/sound/soc/codecs/rt5682.h
@@ -1463,7 +1463,6 @@ struct rt5682_priv {
 
 	int jack_type;
 	int irq_work_delay_time;
-	struct mutex jdet_mutex;
 };
 
 extern const char *rt5682_supply_names[RT5682_NUM_SUPPLIES];
@@ -1473,7 +1472,6 @@ int rt5682_sel_asrc_clk_src(struct snd_soc_component *component,
 
 void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev);
 
-int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert);
 void rt5682_jack_detect_handler(struct work_struct *work);
 
 bool rt5682_volatile_register(struct device *dev, unsigned int reg);

From 1601033da2dd2052e0489137f7788a46a8fcd82f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Fri, 28 Jan 2022 19:24:43 +0000
Subject: [PATCH 22/31] ASoC: ops: Check for negative values before reading
 them

The controls allow inputs to be specified as negative but our manipulating
them into register fields need to be done on unsigned variables so the
checks for negative numbers weren't taking effect properly. Do the checks
for negative values on the variable in the ABI struct rather than on our
local unsigned copy.

Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20220128192443.3504823-1-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-ops.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index dc0e7c8d31f3..9833611b83d1 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -316,26 +316,26 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	if (sign_bit)
 		mask = BIT(sign_bit + 1) - 1;
 
+	if (ucontrol->value.integer.value[0] < 0)
+		return -EINVAL;
 	val = ucontrol->value.integer.value[0];
 	if (mc->platform_max && val > mc->platform_max)
 		return -EINVAL;
 	if (val > max - min)
 		return -EINVAL;
-	if (val < 0)
-		return -EINVAL;
 	val = (val + min) & mask;
 	if (invert)
 		val = max - val;
 	val_mask = mask << shift;
 	val = val << shift;
 	if (snd_soc_volsw_is_stereo(mc)) {
+		if (ucontrol->value.integer.value[1] < 0)
+			return -EINVAL;
 		val2 = ucontrol->value.integer.value[1];
 		if (mc->platform_max && val2 > mc->platform_max)
 			return -EINVAL;
 		if (val2 > max - min)
 			return -EINVAL;
-		if (val2 < 0)
-			return -EINVAL;
 		val2 = (val2 + min) & mask;
 		if (invert)
 			val2 = max - val2;
@@ -423,13 +423,13 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
 	int err = 0;
 	unsigned int val, val_mask;
 
+	if (ucontrol->value.integer.value[0] < 0)
+		return -EINVAL;
 	val = ucontrol->value.integer.value[0];
 	if (mc->platform_max && val > mc->platform_max)
 		return -EINVAL;
 	if (val > max - min)
 		return -EINVAL;
-	if (val < 0)
-		return -EINVAL;
 	val_mask = mask << shift;
 	val = (val + min) & mask;
 	val = val << shift;

From b837a9f5ab3bdfab9233c9f98a6bef717673a3e5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 31 Jan 2022 08:57:38 +0100
Subject: [PATCH 23/31] ALSA: hda: realtek: Fix race at concurrent COEF updates

The COEF access is done with two steps: setting the index then read or
write the data.  When multiple COEF accesses are performed
concurrently, the index and data might be paired unexpectedly.
In most cases, this isn't a big problem as the COEF setup is done at
the initialization, but some dynamic changes like the mute LED may hit
such a race.

For avoiding the racy COEF accesses, this patch introduces a new
mutex coef_mutex to alc_spec, and wrap the COEF accessing functions
with it.

Reported-by: Alexander Sergeyev <sergeev917@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdomain
Link: https://lore.kernel.org/r/20220131075738.24323-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 61 ++++++++++++++++++++++++++++-------
 1 file changed, 50 insertions(+), 11 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 668274e52674..a5677be0a405 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -98,6 +98,7 @@ struct alc_spec {
 	unsigned int gpio_mic_led_mask;
 	struct alc_coef_led mute_led_coef;
 	struct alc_coef_led mic_led_coef;
+	struct mutex coef_mutex;
 
 	hda_nid_t headset_mic_pin;
 	hda_nid_t headphone_mic_pin;
@@ -137,8 +138,8 @@ struct alc_spec {
  * COEF access helper functions
  */
 
-static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
-			       unsigned int coef_idx)
+static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
+				 unsigned int coef_idx)
 {
 	unsigned int val;
 
@@ -147,28 +148,61 @@ static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
 	return val;
 }
 
+static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
+			       unsigned int coef_idx)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
+
+	mutex_lock(&spec->coef_mutex);
+	val = __alc_read_coefex_idx(codec, nid, coef_idx);
+	mutex_unlock(&spec->coef_mutex);
+	return val;
+}
+
 #define alc_read_coef_idx(codec, coef_idx) \
 	alc_read_coefex_idx(codec, 0x20, coef_idx)
 
-static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
-				 unsigned int coef_idx, unsigned int coef_val)
+static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
+				   unsigned int coef_idx, unsigned int coef_val)
 {
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, coef_idx);
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PROC_COEF, coef_val);
 }
 
+static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
+				 unsigned int coef_idx, unsigned int coef_val)
+{
+	struct alc_spec *spec = codec->spec;
+
+	mutex_lock(&spec->coef_mutex);
+	__alc_write_coefex_idx(codec, nid, coef_idx, coef_val);
+	mutex_unlock(&spec->coef_mutex);
+}
+
 #define alc_write_coef_idx(codec, coef_idx, coef_val) \
 	alc_write_coefex_idx(codec, 0x20, coef_idx, coef_val)
 
+static void __alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
+				    unsigned int coef_idx, unsigned int mask,
+				    unsigned int bits_set)
+{
+	unsigned int val = __alc_read_coefex_idx(codec, nid, coef_idx);
+
+	if (val != -1)
+		__alc_write_coefex_idx(codec, nid, coef_idx,
+				       (val & ~mask) | bits_set);
+}
+
 static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid,
 				  unsigned int coef_idx, unsigned int mask,
 				  unsigned int bits_set)
 {
-	unsigned int val = alc_read_coefex_idx(codec, nid, coef_idx);
+	struct alc_spec *spec = codec->spec;
 
-	if (val != -1)
-		alc_write_coefex_idx(codec, nid, coef_idx,
-				     (val & ~mask) | bits_set);
+	mutex_lock(&spec->coef_mutex);
+	__alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set);
+	mutex_unlock(&spec->coef_mutex);
 }
 
 #define alc_update_coef_idx(codec, coef_idx, mask, bits_set)	\
@@ -201,13 +235,17 @@ struct coef_fw {
 static void alc_process_coef_fw(struct hda_codec *codec,
 				const struct coef_fw *fw)
 {
+	struct alc_spec *spec = codec->spec;
+
+	mutex_lock(&spec->coef_mutex);
 	for (; fw->nid; fw++) {
 		if (fw->mask == (unsigned short)-1)
-			alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val);
+			__alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val);
 		else
-			alc_update_coefex_idx(codec, fw->nid, fw->idx,
-					      fw->mask, fw->val);
+			__alc_update_coefex_idx(codec, fw->nid, fw->idx,
+						fw->mask, fw->val);
 	}
+	mutex_unlock(&spec->coef_mutex);
 }
 
 /*
@@ -1153,6 +1191,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
 	codec->spdif_status_reset = 1;
 	codec->forced_resume = 1;
 	codec->patch_ops = alc_patch_ops;
+	mutex_init(&spec->coef_mutex);
 
 	err = alc_codec_rename_from_preset(codec);
 	if (err < 0) {

From 63394a16086fc2152869d7902621e2525e14bc40 Mon Sep 17 00:00:00 2001
From: Christian Lachner <gladiac@gmail.com>
Date: Sat, 29 Jan 2022 12:32:41 +0100
Subject: [PATCH 24/31] ALSA: hda/realtek: Add missing fixup-model entry for
 Gigabyte X570 ALC1220 quirks

The initial commit of the new Gigabyte X570 ALC1220 quirks lacked the
fixup-model entry in alc882_fixup_models[]. It seemed not to cause any ill
effects but for completeness sake this commit makes up for that.

Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220129113243.93068-2-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a5677be0a405..d662ad805960 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2665,6 +2665,7 @@ static const struct hda_model_fixup alc882_fixup_models[] = {
 	{.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"},
 	{.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"},
 	{.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"},
+	{.id = ALC1220_FIXUP_GB_X570, .name = "gb-x570"},
 	{.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"},
 	{}
 };

From 41a8601302ecbe704ac970552c33dc942300fc37 Mon Sep 17 00:00:00 2001
From: Christian Lachner <gladiac@gmail.com>
Date: Sat, 29 Jan 2022 12:32:42 +0100
Subject: [PATCH 25/31] ALSA: hda/realtek: Fix silent output on Gigabyte X570S
 Aorus Master (newer chipset)

Newer versions of the X570 Master come with a newer revision of the
mainboard chipset - the X570S. These boards have the same ALC1220 codec
but seem to initialize the codec with a different parameter in Coef 0x7
which causes the output audio to be very low. We therefore write a
known-good value to Coef 0x7 to fix that. As the value is the exact same
as on the other X570(non-S) boards the same quirk-function can be shared
between both generations.

This commit adds the Gigabyte X570S Aorus Master to the list of boards
using the ALC1220_FIXUP_GB_X570 quirk. This fixes both, the silent output
and the no-audio after reboot from windows problems.

This work has been tested by the folks over at the level1techs forum here:
https://forum.level1techs.com/t/has-anybody-gotten-audio-working-in-linux-on-aorus-x570-master/154072

Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220129113243.93068-3-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d662ad805960..54301d208d2b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2164,6 +2164,7 @@ static void alc1220_fixup_gb_x570(struct hda_codec *codec,
 {
 	static const hda_nid_t conn1[] = { 0x0c };
 	static const struct coef_fw gb_x570_coefs[] = {
+		WRITE_COEF(0x07, 0x03c0),
 		WRITE_COEF(0x1a, 0x01c1),
 		WRITE_COEF(0x1b, 0x0202),
 		WRITE_COEF(0x43, 0x3005),
@@ -2591,6 +2592,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570),
 	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570),
 	SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950),

From ea3541961376f733373839cc90493aafa8a7f733 Mon Sep 17 00:00:00 2001
From: Christian Lachner <gladiac@gmail.com>
Date: Sat, 29 Jan 2022 12:32:43 +0100
Subject: [PATCH 26/31] ALSA: hda/realtek: Fix silent output on Gigabyte X570
 Aorus Xtreme after reboot from Windows

This commit switches the Gigabyte X570 Aorus Xtreme from using the
ALC1220_FIXUP_CLEVO_P950 to the ALC1220_FIXUP_GB_X570 quirk. This fixes
the no-audio after reboot from windows problem.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220129113243.93068-4-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 54301d208d2b..0d52eca57bbc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2591,7 +2591,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570),
-	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+	SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_GB_X570),
 	SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570),
 	SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),

From 94db9cc8f8fa2d5426ce79ec4ca16028f7084224 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Albert=20Geant=C4=83?= <albertgeanta@gmail.com>
Date: Mon, 31 Jan 2022 03:05:23 +0200
Subject: [PATCH 27/31] ALSA: hda/realtek: Add quirk for ASUS GU603
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The ASUS GU603 (Zephyrus M16 - SSID 1043:16b2) requires a quirk similar to
other ASUS devices for correctly routing the 4 integrated speakers. This
fixes it by adding a corresponding quirk entry, which connects the bass
speakers to the proper DAC.

Signed-off-by: Albert Geantă <albertgeanta@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20220131010523.546386-1-albertgeanta@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0d52eca57bbc..8315bf7d4c38 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9011,6 +9011,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),

From a4f399a1416f645ac701064a55b0cb5203707ac9 Mon Sep 17 00:00:00 2001
From: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Date: Sun, 30 Jan 2022 09:06:36 +0100
Subject: [PATCH 28/31] Input: wm97xx: Simplify resource management

Since the commit in the Fixes tag below, 'wm->input_dev' is a managed
resource that doesn't need to be explicitly unregistered or freed (see
devm_input_allocate_device() documentation)

So, remove some unless line of code to slightly simplify it.

Fixes: c72f61e74073 ("Input: wm97xx: split out touchscreen registering")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87dce7e80ea9b191843fa22415ca3aef5f3cc2e6.1643529968.git.christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 drivers/input/touchscreen/wm97xx-core.c | 12 +++---------
 1 file changed, 3 insertions(+), 9 deletions(-)

diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c
index 78d2ee99f37a..1b58611c8084 100644
--- a/drivers/input/touchscreen/wm97xx-core.c
+++ b/drivers/input/touchscreen/wm97xx-core.c
@@ -615,10 +615,9 @@ static int wm97xx_register_touch(struct wm97xx *wm)
 	 * extensions)
 	 */
 	wm->touch_dev = platform_device_alloc("wm97xx-touch", -1);
-	if (!wm->touch_dev) {
-		ret = -ENOMEM;
-		goto touch_err;
-	}
+	if (!wm->touch_dev)
+		return -ENOMEM;
+
 	platform_set_drvdata(wm->touch_dev, wm);
 	wm->touch_dev->dev.parent = wm->dev;
 	wm->touch_dev->dev.platform_data = pdata;
@@ -629,9 +628,6 @@ static int wm97xx_register_touch(struct wm97xx *wm)
 	return 0;
 touch_reg_err:
 	platform_device_put(wm->touch_dev);
-touch_err:
-	input_unregister_device(wm->input_dev);
-	wm->input_dev = NULL;
 
 	return ret;
 }
@@ -639,8 +635,6 @@ touch_err:
 static void wm97xx_unregister_touch(struct wm97xx *wm)
 {
 	platform_device_unregister(wm->touch_dev);
-	input_unregister_device(wm->input_dev);
-	wm->input_dev = NULL;
 }
 
 static int _wm97xx_probe(struct wm97xx *wm)

From ff4865b3c8cd746ef72f59bdd485848b4cebd43d Mon Sep 17 00:00:00 2001
From: "Rafael J. Wysocki" <rafael.j.wysocki@intel.com>
Date: Wed, 26 Jan 2022 20:48:49 +0100
Subject: [PATCH 29/31] ALSA: Replace acpi_bus_get_device()

Replace acpi_bus_get_device() that is going to be dropped with
acpi_fetch_acpi_dev().

No intentional functional impact.

Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Link: https://lore.kernel.org/r/2828205.e9J7NaK4W3@kreacher
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/hda/intel-sdw-acpi.c | 7 +++----
 sound/soc/soc-acpi.c       | 7 ++-----
 2 files changed, 5 insertions(+), 9 deletions(-)

diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c
index b7758dbe2371..5cb92f7ccbca 100644
--- a/sound/hda/intel-sdw-acpi.c
+++ b/sound/hda/intel-sdw-acpi.c
@@ -50,11 +50,11 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, int i)
 static int
 sdw_intel_scan_controller(struct sdw_intel_acpi_info *info)
 {
-	struct acpi_device *adev;
+	struct acpi_device *adev = acpi_fetch_acpi_dev(info->handle);
 	int ret, i;
 	u8 count;
 
-	if (acpi_bus_get_device(info->handle, &adev))
+	if (!adev)
 		return -EINVAL;
 
 	/* Found controller, find links supported */
@@ -119,7 +119,6 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level,
 				     void *cdata, void **return_value)
 {
 	struct sdw_intel_acpi_info *info = cdata;
-	struct acpi_device *adev;
 	acpi_status status;
 	u64 adr;
 
@@ -127,7 +126,7 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level,
 	if (ACPI_FAILURE(status))
 		return AE_OK; /* keep going */
 
-	if (acpi_bus_get_device(handle, &adev)) {
+	if (!acpi_fetch_acpi_dev(handle)) {
 		pr_err("%s: Couldn't find ACPI handle\n", __func__);
 		return AE_NOT_FOUND;
 	}
diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c
index cbd7ea48837b..142476f1396f 100644
--- a/sound/soc/soc-acpi.c
+++ b/sound/soc/soc-acpi.c
@@ -55,16 +55,13 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_find_machine);
 static acpi_status snd_soc_acpi_find_package(acpi_handle handle, u32 level,
 					     void *context, void **ret)
 {
-	struct acpi_device *adev;
+	struct acpi_device *adev = acpi_fetch_acpi_dev(handle);
 	acpi_status status;
 	struct snd_soc_acpi_package_context *pkg_ctx = context;
 
 	pkg_ctx->data_valid = false;
 
-	if (acpi_bus_get_device(handle, &adev))
-		return AE_OK;
-
-	if (adev->status.present && adev->status.functional) {
+	if (adev && adev->status.present && adev->status.functional) {
 		struct acpi_buffer buffer = {ACPI_ALLOCATE_BUFFER, NULL};
 		union acpi_object  *myobj = NULL;
 

From 4ee02e20893d2f9e951c7888f2284fa608ddaa35 Mon Sep 17 00:00:00 2001
From: Jonas Hahnfeld <hahnjo@hahnjo.de>
Date: Mon, 31 Jan 2022 19:35:16 +0100
Subject: [PATCH 30/31] ALSA: usb-audio: Correct quirk for VF0770
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This device provides both audio and video. The original quirk added in
commit 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") used
USB_DEVICE to match the vendor and product ID. Depending on module order,
if snd-usb-audio was asked first, it would match the entire device and
uvcvideo wouldn't get to see it. Change the matching to USB_AUDIO_DEVICE
to restore uvcvideo matching in all cases.

Fixes: 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770")
Reported-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Tested-by: Jukka Heikintalo <heikintalo.jukka@gmail.com>
Reported-by: Paweł Susicki <pawel.susicki@gmail.com>
Tested-by: Paweł Susicki <pawel.susicki@gmail.com>
Cc: <stable@vger.kernel.org> # 5.4, 5.10, 5.14, 5.15
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Link: https://lore.kernel.org/r/20220131183516.61191-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/quirks-table.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index b1522e43173e..0ea39565e623 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -84,7 +84,7 @@
  * combination.
  */
 {
-	USB_DEVICE(0x041e, 0x4095),
+	USB_AUDIO_DEVICE(0x041e, 0x4095),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_COMPOSITE,

From 1c7f0e349aa5f8f80b1cac3d4917405332e14cdf Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Date: Tue, 1 Feb 2022 13:21:44 +0200
Subject: [PATCH 31/31] ALSA: hda: Skip codec shutdown in case the codec is not
 registered

If the codec->registered is not set then it means that pm_runtime is
not yet enabled and the codec->pcm_list_head has not been initialized.

The access to the not initialized pcm_list_head will lead a kernel crash
during shutdown.

Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Tested-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Fixes: b98444ed597d ("ALSA: hda: Suspend codec at shutdown")
Link: https://lore.kernel.org/r/20220201112144.29411-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 4 ++++
 1 file changed, 4 insertions(+)

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 7016b48227bf..f552785d301e 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3000,6 +3000,10 @@ void snd_hda_codec_shutdown(struct hda_codec *codec)
 {
 	struct hda_pcm *cpcm;
 
+	/* Skip the shutdown if codec is not registered */
+	if (!codec->registered)
+		return;
+
 	list_for_each_entry(cpcm, &codec->pcm_list_head, list)
 		snd_pcm_suspend_all(cpcm->pcm);