[PATCH AUTOSEL 6.1 054/212] ASoC: mediatek: mt6359: Add stub for mt6359_accdet_enable_jack_detect

[PATCH AUTOSEL 6.1 054/212] ASoC: mediatek: mt6359: Add stub for mt6359_accdet_enable_jack_detect

[Sasha Levin]

From: Nícolas F. R. A. Prado <nfraprado@collabora.com>

[ Upstream commit 0116a7d84b32537a10d9bea1fd1bfc06577ef527 ]

Add a stub for mt6359_accdet_enable_jack_detect() to prevent linker
failures in the machine sound drivers calling it when
CONFIG_SND_SOC_MT6359_ACCDET is not enabled.

Suggested-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Signed-off-by: Nícolas F. R. A. Prado <nfraprado@collabora.com>
Link: https://patch.msgid.link/20250306-mt8188-accdet-v3-3-7828e835ff4b@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/mt6359-accdet.h | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/sound/soc/codecs/mt6359-accdet.h b/sound/soc/codecs/mt6359-accdet.h
index c234f2f4276a1..78ada3a5bfae5 100644
--- a/sound/soc/codecs/mt6359-accdet.h
+++ b/sound/soc/codecs/mt6359-accdet.h
@@ -123,6 +123,15 @@ struct mt6359_accdet {
 	struct workqueue_struct *jd_workqueue;
 };
 
+#if IS_ENABLED(CONFIG_SND_SOC_MT6359_ACCDET)
 int mt6359_accdet_enable_jack_detect(struct snd_soc_component *component,
 				     struct snd_soc_jack *jack);
+#else
+static inline int
+mt6359_accdet_enable_jack_detect(struct snd_soc_component *component,
+				 struct snd_soc_jack *jack)
+{
+	return -EOPNOTSUPP;
+}
+#endif
 #endif
-- 
2.39.5

[PATCH AUTOSEL 6.1 061/212] ASoC: sun4i-codec: support hp-det-gpios property

[Sasha Levin]

From: Ryan Walklin <ryan@testtoast.com>

[ Upstream commit a149377c033afe6557c50892ebbfc0e8b7e2e253 ]

Add support for GPIO headphone detection with the hp-det-gpios
property. In order for this to properly disable the path upon
removal of headphones, the output must be labelled Headphone which
is a common sink in the driver.

Describe a headphone jack and detection GPIO in the driver, check for
a corresponding device tree node, and enable jack detection in a new
machine init function if described.

Signed-off-by: Chris Morgan <macromorgan@hotmail.com>
Signed-off-by: Ryan Walklin <ryan@testtoast.com>

--
Changelog v1..v2:
- Separate DAPM changes into separate patch and add rationale.

Tested-by: Philippe Simons <simons.philippe@gmail.com>
Link: https://patch.msgid.link/20250214220247.10810-4-ryan@testtoast.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/sunxi/sun4i-codec.c | 53 +++++++++++++++++++++++++++++++++++
 1 file changed, 53 insertions(+)

diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index 835dc34043670..1e310958f8c08 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -25,6 +25,7 @@
 #include <linux/gpio/consumer.h>
 
 #include <sound/core.h>
+#include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
@@ -239,6 +240,7 @@ struct sun4i_codec {
 	struct clk	*clk_module;
 	struct reset_control *rst;
 	struct gpio_desc *gpio_pa;
+	struct gpio_desc *gpio_hp;
 
 	/* ADC_FIFOC register is at different offset on different SoCs */
 	struct regmap_field *reg_adc_fifoc;
@@ -1273,6 +1275,49 @@ static struct snd_soc_dai_driver dummy_cpu_dai = {
 	 },
 };
 
+static struct snd_soc_jack sun4i_headphone_jack;
+
+static struct snd_soc_jack_pin sun4i_headphone_jack_pins[] = {
+	{ .pin = "Headphone", .mask = SND_JACK_HEADPHONE },
+};
+
+static struct snd_soc_jack_gpio sun4i_headphone_jack_gpio = {
+	.name = "hp-det",
+	.report = SND_JACK_HEADPHONE,
+	.debounce_time = 150,
+};
+
+static int sun4i_codec_machine_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	if (scodec->gpio_hp) {
+		ret = snd_soc_card_jack_new_pins(card, "Headphone Jack",
+						 SND_JACK_HEADPHONE,
+						 &sun4i_headphone_jack,
+						 sun4i_headphone_jack_pins,
+						 ARRAY_SIZE(sun4i_headphone_jack_pins));
+		if (ret) {
+			dev_err(rtd->dev,
+				"Headphone jack creation failed: %d\n", ret);
+			return ret;
+		}
+
+		sun4i_headphone_jack_gpio.desc = scodec->gpio_hp;
+		ret = snd_soc_jack_add_gpios(&sun4i_headphone_jack, 1,
+					     &sun4i_headphone_jack_gpio);
+
+		if (ret) {
+			dev_err(rtd->dev, "Headphone GPIO not added: %d\n", ret);
+			return ret;
+		}
+	}
+
+	return 0;
+}
+
 static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
 							int *num_links)
 {
@@ -1298,6 +1343,7 @@ static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev,
 	link->codecs->name	= dev_name(dev);
 	link->platforms->name	= dev_name(dev);
 	link->dai_fmt		= SND_SOC_DAIFMT_I2S;
+	link->init		= sun4i_codec_machine_init;
 
 	*num_links = 1;
 
@@ -1738,6 +1784,13 @@ static int sun4i_codec_probe(struct platform_device *pdev)
 		return ret;
 	}
 
+	scodec->gpio_hp = devm_gpiod_get_optional(&pdev->dev, "hp-det", GPIOD_IN);
+	if (IS_ERR(scodec->gpio_hp)) {
+		ret = PTR_ERR(scodec->gpio_hp);
+		dev_err_probe(&pdev->dev, ret, "Failed to get hp-det gpio\n");
+		return ret;
+	}
+
 	/* reg_field setup */
 	scodec->reg_adc_fifoc = devm_regmap_field_alloc(&pdev->dev,
 							scodec->regmap,
-- 
2.39.5

[PATCH AUTOSEL 6.1 076/212] ASoC: qcom: sm8250: explicitly set format in sm8250_be_hw_params_fixup()

[Sasha Levin]

From: Alexey Klimov <alexey.klimov@linaro.org>

[ Upstream commit 89be3c15a58b2ccf31e969223c8ac93ca8932d81 ]

Setting format to s16le is required for compressed playback on compatible
soundcards.

Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Link: https://patch.msgid.link/20250228161430.373961-1-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/qcom/sm8250.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c
index 41be09a07ca71..65e51b6b46ff9 100644
--- a/sound/soc/qcom/sm8250.c
+++ b/sound/soc/qcom/sm8250.c
@@ -7,6 +7,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/pcm.h>
+#include <sound/pcm_params.h>
 #include <linux/soundwire/sdw.h>
 #include <sound/jack.h>
 #include <linux/input-event-codes.h>
@@ -39,9 +40,11 @@ static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 					SNDRV_PCM_HW_PARAM_RATE);
 	struct snd_interval *channels = hw_param_interval(params,
 					SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 
 	rate->min = rate->max = 48000;
 	channels->min = channels->max = 2;
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
 }
-- 
2.39.5

[PATCH AUTOSEL 6.1 082/212] ALSA: seq: Improve data consistency at polling

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit e3cd33ab17c33bd8f1a9df66ec83a15dd8f7afbb ]

snd_seq_poll() calls snd_seq_write_pool_allocated() that reads out a
field in client->pool object, while it can be updated concurrently via
ioctls, as reported by syzbot.  The data race itself is harmless, as
it's merely a poll() call, and the state is volatile.  OTOH, the read
out of poll object info from the caller side is fragile, and we can
leave it better in snd_seq_pool_poll_wait() alone.

A similar pattern is seen in snd_seq_kernel_client_write_poll(), too,
which is called from the OSS sequencer.

This patch drops the pool checks from the caller side and add the
pool->lock in snd_seq_pool_poll_wait() for better data consistency.

Reported-by: syzbot+2d373c9936c00d7e120c@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/67c88903.050a0220.15b4b9.0028.GAE@google.com
Link: https://patch.msgid.link/20250307084246.29271-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/core/seq/seq_clientmgr.c | 5 +----
 sound/core/seq/seq_memory.c    | 1 +
 2 files changed, 2 insertions(+), 4 deletions(-)

diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index 2d707afa1ef1c..1252ea7ad55ea 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -1140,8 +1140,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait)
 	if (snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT) {
 
 		/* check if data is available in the pool */
-		if (!snd_seq_write_pool_allocated(client) ||
-		    snd_seq_pool_poll_wait(client->pool, file, wait))
+		if (snd_seq_pool_poll_wait(client->pool, file, wait))
 			mask |= EPOLLOUT | EPOLLWRNORM;
 	}
 
@@ -2382,8 +2381,6 @@ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table
 	if (client == NULL)
 		return -ENXIO;
 
-	if (! snd_seq_write_pool_allocated(client))
-		return 1;
 	if (snd_seq_pool_poll_wait(client->pool, file, wait))
 		return 1;
 	return 0;
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index 47ef6bc30c0ee..e30b92d85079b 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -366,6 +366,7 @@ int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file,
 			   poll_table *wait)
 {
 	poll_wait(file, &pool->output_sleep, wait);
+	guard(spinlock_irq)(&pool->lock);
 	return snd_seq_output_ok(pool);
 }
 
-- 
2.39.5

[PATCH AUTOSEL 6.1 098/212] ALSA: hda/realtek: Enable PC beep passthrough for HP EliteBook 855 G7

[Sasha Levin]

From: "Maciej S. Szmigiero" <mail@maciej.szmigiero.name>

[ Upstream commit aa85822c611aef7cd4dc17d27121d43e21bb82f0 ]

PC speaker works well on this platform in BIOS and in Linux until sound
card drivers are loaded. Then it stops working.

There seems to be a beep generator node at 0x1a in this CODEC
(ALC269_TYPE_ALC215) but it seems to be only connected to capture mixers
at nodes 0x22 and 0x23.
If I unmute the mixer input for 0x1a at node 0x23 and start recording
from its "ALC285 Analog" capture device I can clearly hear beeps in that
recording.

So the beep generator is indeed working properly, however I wasn't able to
figure out any way to connect it to speakers.

However, the bits in the "Passthrough Control" register (0x36) seems to
work at least partially: by zeroing "B" and "h" and setting "S" I can at
least make the PIT PC speaker output appear either in this laptop speakers
or headphones (depending on whether they are connected or not).

There are some caveats, however:
* If the CODEC gets runtime-suspended the beeps stop so it needs HDA beep
device for keeping it awake during beeping.

* If the beep generator node is generating any beep the PC beep passthrough
seems to be temporarily inhibited, so the HDA beep device has to be
prevented from using the actual beep generator node - but the beep device
is still necessary due to the previous point.

* In contrast with other platforms here beep amplification has to be
disabled otherwise the beeps output are WAY louder than they were on pure
BIOS setup.

Unless someone (from Realtek probably) knows how to make the beep generator
node output appear in speakers / headphones using PC beep passthrough seems
to be the only way to make PC speaker beeping actually work on this
platform.

Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Acked-by: kailang@realtek.com
Link: https://patch.msgid.link/7461f695b4daed80f2fc4b1463ead47f04f9ad05.1739741254.git.mail@maciej.szmigiero.name
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 include/sound/hda_codec.h     |  1 +
 sound/pci/hda/hda_beep.c      | 15 +++++++++------
 sound/pci/hda/patch_realtek.c | 34 +++++++++++++++++++++++++++++++++-
 3 files changed, 43 insertions(+), 7 deletions(-)

diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index bbb7805e85d8e..4ca45d5895dfd 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -199,6 +199,7 @@ struct hda_codec {
 	/* beep device */
 	struct hda_beep *beep;
 	unsigned int beep_mode;
+	bool beep_just_power_on;
 
 	/* widget capabilities cache */
 	u32 *wcaps;
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index e63621bcb2142..1a684e47d4d18 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -31,8 +31,9 @@ static void generate_tone(struct hda_beep *beep, int tone)
 			beep->power_hook(beep, true);
 		beep->playing = 1;
 	}
-	snd_hda_codec_write(codec, beep->nid, 0,
-			    AC_VERB_SET_BEEP_CONTROL, tone);
+	if (!codec->beep_just_power_on)
+		snd_hda_codec_write(codec, beep->nid, 0,
+				    AC_VERB_SET_BEEP_CONTROL, tone);
 	if (!tone && beep->playing) {
 		beep->playing = 0;
 		if (beep->power_hook)
@@ -212,10 +213,12 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	struct hda_beep *beep;
 	int err;
 
-	if (!snd_hda_get_bool_hint(codec, "beep"))
-		return 0; /* disabled explicitly by hints */
-	if (codec->beep_mode == HDA_BEEP_MODE_OFF)
-		return 0; /* disabled by module option */
+	if (!codec->beep_just_power_on) {
+		if (!snd_hda_get_bool_hint(codec, "beep"))
+			return 0; /* disabled explicitly by hints */
+		if (codec->beep_mode == HDA_BEEP_MODE_OFF)
+			return 0; /* disabled by module option */
+	}
 
 	beep = kzalloc(sizeof(*beep), GFP_KERNEL);
 	if (beep == NULL)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 61b48f2418bf0..2f67cd955d651 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -24,6 +24,7 @@
 #include <sound/hda_codec.h>
 #include "hda_local.h"
 #include "hda_auto_parser.h"
+#include "hda_beep.h"
 #include "hda_jack.h"
 #include "hda_generic.h"
 #include "hda_component.h"
@@ -6858,6 +6859,30 @@ static void alc285_fixup_hp_envy_x360(struct hda_codec *codec,
 	}
 }
 
+static void alc285_fixup_hp_beep(struct hda_codec *codec,
+				 const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		codec->beep_just_power_on = true;
+	} else  if (action == HDA_FIXUP_ACT_INIT) {
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+		/*
+		 * Just enable loopback to internal speaker and headphone jack.
+		 * Disable amplification to get about the same beep volume as
+		 * was on pure BIOS setup before loading the driver.
+		 */
+		alc_update_coef_idx(codec, 0x36, 0x7070, BIT(13));
+
+		snd_hda_enable_beep_device(codec, 1);
+
+#if !IS_ENABLED(CONFIG_INPUT_PCSPKR)
+		dev_warn_once(hda_codec_dev(codec),
+			      "enable CONFIG_INPUT_PCSPKR to get PC beeps\n");
+#endif
+#endif
+	}
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -7400,6 +7425,7 @@ enum {
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED,
+	ALC285_FIXUP_HP_BEEP_MICMUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED_COEFBIT2,
 	ALC236_FIXUP_HP_GPIO_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
@@ -8947,6 +8973,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_spectre_x360_mute_led,
 	},
+	[ALC285_FIXUP_HP_BEEP_MICMUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_beep,
+		.chained = true,
+		.chain_id = ALC285_FIXUP_HP_MUTE_LED,
+	},
 	[ALC236_FIXUP_HP_MUTE_LED_COEFBIT2] = {
 	    .type = HDA_FIXUP_FUNC,
 	    .v.func = alc236_fixup_hp_mute_led_coefbit2,
@@ -9860,7 +9892,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
-	SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x8760, "HP EliteBook 8{4,5}5 G7", ALC285_FIXUP_HP_BEEP_MICMUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x876e, "HP ENVY x360 Convertible 13-ay0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
-- 
2.39.5

[PATCH AUTOSEL 6.1 155/212] ASoC: ops: Enforce platform maximum on initial value

[Sasha Levin]

From: Martin Povišer <povik+lin@cutebit.org>

[ Upstream commit 783db6851c1821d8b983ffb12b99c279ff64f2ee ]

Lower the volume if it is violating the platform maximum at its initial
value (i.e. at the time of the 'snd_soc_limit_volume' call).

Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
[Cherry picked from the Asahi kernel with fixups -- broonie]
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250208-asoc-volume-limit-v1-1-b98fcf4cdbad@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-ops.c | 29 ++++++++++++++++++++++++++++-
 1 file changed, 28 insertions(+), 1 deletion(-)

diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index b4cfc34d00ee6..eff1355cc3df0 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -638,6 +638,33 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
 
+static int snd_soc_clip_to_platform_max(struct snd_kcontrol *kctl)
+{
+	struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
+	struct snd_ctl_elem_value uctl;
+	int ret;
+
+	if (!mc->platform_max)
+		return 0;
+
+	ret = kctl->get(kctl, &uctl);
+	if (ret < 0)
+		return ret;
+
+	if (uctl.value.integer.value[0] > mc->platform_max)
+		uctl.value.integer.value[0] = mc->platform_max;
+
+	if (snd_soc_volsw_is_stereo(mc) &&
+	    uctl.value.integer.value[1] > mc->platform_max)
+		uctl.value.integer.value[1] = mc->platform_max;
+
+	ret = kctl->put(kctl, &uctl);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 /**
  * snd_soc_limit_volume - Set new limit to an existing volume control.
  *
@@ -662,7 +689,7 @@ int snd_soc_limit_volume(struct snd_soc_card *card,
 		struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value;
 		if (max <= mc->max - mc->min) {
 			mc->platform_max = max;
-			ret = 0;
+			ret = snd_soc_clip_to_platform_max(kctl);
 		}
 	}
 	return ret;
-- 
2.39.5

[PATCH AUTOSEL 6.1 156/212] ASoC: tas2764: Add reg defaults for TAS2764_INT_CLK_CFG

[Sasha Levin]

From: Hector Martin <marcan@marcan.st>

[ Upstream commit d64c4c3d1c578f98d70db1c5e2535b47adce9d07 ]

Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250208-asoc-tas2764-v1-4-dbab892a69b5@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/tas2764.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index fc8479d3d2852..72db361ac3611 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -636,6 +636,7 @@ static const struct reg_default tas2764_reg_defaults[] = {
 	{ TAS2764_TDM_CFG2, 0x0a },
 	{ TAS2764_TDM_CFG3, 0x10 },
 	{ TAS2764_TDM_CFG5, 0x42 },
+	{ TAS2764_INT_CLK_CFG, 0x19 },
 };
 
 static const struct regmap_range_cfg tas2764_regmap_ranges[] = {
-- 
2.39.5

[PATCH AUTOSEL 6.1 157/212] ASoC: tas2764: Mark SW_RESET as volatile

[Sasha Levin]

From: Hector Martin <marcan@marcan.st>

[ Upstream commit f37f1748564ac51d32f7588bd7bfc99913ccab8e ]

Since the bit is self-clearing.

Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250208-asoc-tas2764-v1-3-dbab892a69b5@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/tas2764.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 72db361ac3611..94428487a8855 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -654,6 +654,7 @@ static const struct regmap_range_cfg tas2764_regmap_ranges[] = {
 static bool tas2764_volatile_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
+	case TAS2764_SW_RST:
 	case TAS2764_INT_LTCH0 ... TAS2764_INT_LTCH4:
 	case TAS2764_INT_CLK_CFG:
 		return true;
-- 
2.39.5

[PATCH AUTOSEL 6.1 158/212] ASoC: tas2764: Power up/down amp on mute ops

[Sasha Levin]

From: Hector Martin <marcan@marcan.st>

[ Upstream commit 1c3b5f37409682184669457a5bdf761268eafbe5 ]

The ASoC convention is that clocks are removed after codec mute, and
power up/down is more about top level power management. For these chips,
the "mute" state still expects a TDM clock, and yanking the clock in
this state will trigger clock errors. So, do the full
shutdown<->mute<->active transition on the mute operation, so the amp is
in software shutdown by the time the clocks are removed.

This fixes TDM clock errors when streams are stopped.

Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250208-asoc-tas2764-v1-1-dbab892a69b5@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/tas2764.c | 51 ++++++++++++++++----------------------
 1 file changed, 21 insertions(+), 30 deletions(-)

diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c
index 94428487a8855..10f0f07b90ff2 100644
--- a/sound/soc/codecs/tas2764.c
+++ b/sound/soc/codecs/tas2764.c
@@ -182,33 +182,6 @@ static SOC_ENUM_SINGLE_DECL(
 static const struct snd_kcontrol_new tas2764_asi1_mux =
 	SOC_DAPM_ENUM("ASI1 Source", tas2764_ASI1_src_enum);
 
-static int tas2764_dac_event(struct snd_soc_dapm_widget *w,
-			     struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component);
-	int ret;
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		tas2764->dac_powered = true;
-		ret = tas2764_update_pwr_ctrl(tas2764);
-		break;
-	case SND_SOC_DAPM_PRE_PMD:
-		tas2764->dac_powered = false;
-		ret = tas2764_update_pwr_ctrl(tas2764);
-		break;
-	default:
-		dev_err(tas2764->dev, "Unsupported event\n");
-		return -EINVAL;
-	}
-
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new isense_switch =
 	SOC_DAPM_SINGLE("Switch", TAS2764_PWR_CTRL, TAS2764_ISENSE_POWER_EN, 1, 1);
 static const struct snd_kcontrol_new vsense_switch =
@@ -221,8 +194,7 @@ static const struct snd_soc_dapm_widget tas2764_dapm_widgets[] = {
 			    1, &isense_switch),
 	SND_SOC_DAPM_SWITCH("VSENSE", TAS2764_PWR_CTRL, TAS2764_VSENSE_POWER_EN,
 			    1, &vsense_switch),
-	SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2764_dac_event,
-			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_OUTPUT("OUT"),
 	SND_SOC_DAPM_SIGGEN("VMON"),
 	SND_SOC_DAPM_SIGGEN("IMON")
@@ -243,9 +215,28 @@ static int tas2764_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct tas2764_priv *tas2764 =
 			snd_soc_component_get_drvdata(dai->component);
+	int ret;
+
+	if (!mute) {
+		tas2764->dac_powered = true;
+		ret = tas2764_update_pwr_ctrl(tas2764);
+		if (ret)
+			return ret;
+	}
 
 	tas2764->unmuted = !mute;
-	return tas2764_update_pwr_ctrl(tas2764);
+	ret = tas2764_update_pwr_ctrl(tas2764);
+	if (ret)
+		return ret;
+
+	if (mute) {
+		tas2764->dac_powered = false;
+		ret = tas2764_update_pwr_ctrl(tas2764);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
 }
 
 static int tas2764_set_bitwidth(struct tas2764_priv *tas2764, int bitwidth)
-- 
2.39.5

[PATCH AUTOSEL 6.1 159/212] ASoC: soc-dai: check return value at snd_soc_dai_set_tdm_slot()

[Sasha Levin]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

[ Upstream commit 7f1186a8d738661b941b298fd6d1d5725ed71428 ]

snd_soc_dai_set_tdm_slot() calls .xlate_tdm_slot_mask() or
snd_soc_xlate_tdm_slot_mask(), but didn't check its return value.
Let's check it.

This patch might break existing driver. In such case, let's makes
each func to void instead of int.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87o6z7yk61.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/soc-dai.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 49752af0e205d..ba38b6e6b2649 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -270,10 +270,11 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 
 	if (dai->driver->ops &&
 	    dai->driver->ops->xlate_tdm_slot_mask)
-		dai->driver->ops->xlate_tdm_slot_mask(slots,
-						      &tx_mask, &rx_mask);
+		ret = dai->driver->ops->xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
 	else
-		snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+		ret = snd_soc_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+	if (ret)
+		goto err;
 
 	dai->tx_mask = tx_mask;
 	dai->rx_mask = rx_mask;
@@ -282,6 +283,7 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	    dai->driver->ops->set_tdm_slot)
 		ret = dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
 						      slots, slot_width);
+err:
 	return soc_dai_ret(dai, ret);
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
-- 
2.39.5

[PATCH AUTOSEL 6.1 200/212] ASoC: codecs: pcm3168a: Allow for 24-bit in provider mode

[Sasha Levin]

From: Cezary Rojewski <cezary.rojewski@intel.com>

[ Upstream commit 7d92a38d67e5d937b64b20aa4fd14451ee1772f3 ]

As per codec device specification, 24-bit is allowed in provider mode.
Update the code to reflect that.

Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250203141051.2361323-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/pcm3168a.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 9d6431338fb71..329549936bd5c 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -494,9 +494,9 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream,
 		}
 		break;
 	case 24:
-		if (provider_mode || (format == SND_SOC_DAIFMT_DSP_A) ||
-		    		     (format == SND_SOC_DAIFMT_DSP_B)) {
-			dev_err(component->dev, "24-bit slots not supported in provider mode, or consumer mode using DSP\n");
+		if (!provider_mode && ((format == SND_SOC_DAIFMT_DSP_A) ||
+				       (format == SND_SOC_DAIFMT_DSP_B))) {
+			dev_err(component->dev, "24-bit slots not supported in consumer mode using DSP\n");
 			return -EINVAL;
 		}
 		break;
-- 
2.39.5