June 2025 - Alsa-devel - mailman.alsa-project.org

Call to snd_pcm_hw_params_get_buffer_size causes memory corruption in calling program
by GitHub issues - opened 23 Jun '25

23 Jun '25

alsa-project/alsa-lib issue #461 was opened from harryrowe01: Calling snd_pcm_hw_params_get_buffer_size is causing memory corruption in the calling program. The following C program demonstrates the issue. /* This example opens the default PCM device, sets some parameters, and then displays the value of some of the hardware parameters. It does not perform any sound playback or recording. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API /* All of the ALSA library API is defined * in this header */ #include <alsa/asoundlib.h> int main() { int rc; snd_pcm_t *handle; snd_pcm_hw_params_t *params, *save_params; unsigned int val, val2; int dir; snd_pcm_uframes_t frames; /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; // val = 8000; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Display information about the PCM interface */ // Before calling snd_pcm_hw_params_get_buffer_size, save // a copy of params which points to the parameter block. save_params = params; // This call to snd_pcm_hw_params_get_buffer_size causes // the value of params in main to be corrupted. // Since the params argument to the function is passed // by value, the function itself should not have access to // the original variable in main. snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *)&val); // Then if the pointer has been modified, flag the problem and // restore the pointer from the saved value. if( params != save_params ) { printf( "The pointer to the parameter block has been corrupted!\n" ); printf( "was %lx \n", save_params ); printf( "is now %lx \n", params ); printf( "So I'm fixing it.\n" ); params = save_params; } printf("buffer size = %u frames\n", val); snd_pcm_close(handle); return 0; } Compiled and run as follows: harry@idun:~/Documents/morse$ cc bad_alsa.c -lasound -o bad_alsa harry@idun:~/Documents/morse$ ./bad_alsa The pointer to the parameter block has been corrupted! was 7ffcd2011610 is now 7ffc00000000 So I'm fixing it. buffer size = 1048576 frames harry@idun:~/Documents/morse$ cc bad_alsa.c -lasound -o bad_alsa harry@idun:~/Documents/morse$ ./bad_alsa The pointer to the parameter block has been corrupted! was 7fffc7483030 is now 7fff00000000 So I'm fixing it. buffer size = 1048576 frames harry@idun:~/Documents/morse$ Host is an AMD64 version of Debian 12. Installed version of ALSA library is libasound2 1.2.8-1+b1 Issue URL : https://github.com/alsa-project/alsa-lib/issues/461 Repository URL: https://github.com/alsa-project/alsa-lib

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Add UCM support for MediaTek mt8365-evk
by GitHub pull_request - opened 23 Jun '25

23 Jun '25

alsa-project/alsa-ucm-conf pull request #583 was opened from zoran055: Add support for MT8365 EVK with SOF enabled. When SOF is disabled, the "HiFi" use case is used; when SOF is enabled, the "SOF" use case is used. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/583 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/583.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

From: Seungbin Lee <seungbin.lee(a)samsung.com> Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <seungbin.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

FW: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

Add recipient addition and modification patch to the body of the transmission. From: 이승빈 <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > Sent: Thursday, June 19, 2025 10:25 AM To: 'tiwai(a)suse.de' <tiwai(a)suse.de <mailto:tiwai@suse.de> > Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Hi I am sending this email to report an issue found in sound/soc/soc- compress.c. Scenario: Playing a mix of MP3 and WAV files repeatedly from YouTube Music by pressing "next". Issue occurs when snd_soc_dpcm_runtime_update is called and soc_compr_trigger_fe is invoked, causing the value of “fe- >dpcm[stream].runtime_update” to change mid-process. This results in a problem where, during the closing of the BE connected to the FE in snd_soc_dpcm_runtime_update, only the entry is removed from the list without actually being closed. The attached patch resolves this issue as confirmed through testing. Additionally, while performing DPCM related operations in soc-compress.c, snd_soc_dpcm_mutex_lock is used; however, soc_compr_trigger_fe does not have this lock. Please review the attached patch regarding this matter. Thank you. From 72a0340bf7dccc6f53dc2d02e04bb3f894d796e3 Mon Sep 17 00:00:00 2001 From: Seungbin Lee <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <seungbin.lee(a)samsung.com <mailto:seungbin.lee@samsung.com> > --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

RE: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 23 Jun '25

23 Jun '25

I am sending it again after changing it to plain text mail. Thanks From: 이승빈 <seungbin.lee(a)samsung.com> Sent: Monday, June 23, 2025 10:31 AM To: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org>; 'linux-sound(a)vger.kernel.org' <linux-sound(a)vger.kernel.org>; 'gregkh(a)linuxfoundation.org' <gregkh(a)linuxfoundation.org>; 'pierre-louis.bossart(a)linux.intel.com' <pierre-louis.bossart(a)linux.intel.com>; 'perex(a)perex.cz' <perex(a)perex.cz>; 'tiwai(a)suse.de' <tiwai(a)suse.de> Cc: 'pilsun.jang(a)samsung.com' <pilsun.jang(a)samsung.com>; 's47.kang(a)samsung.com' <s47.kang(a)samsung.com>; 'donghee.moon(a)samsung.com' <donghee.moon(a)samsung.com> Subject: FW: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Add recipient addition and modification patch to the body of the transmission. From: 이승빈 <mailto:seungbin.lee@samsung.com> Sent: Thursday, June 19, 2025 10:25 AM To: 'tiwai(a)suse.de' <mailto:tiwai@suse.de> Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe Hi I am sending this email to report an issue found in sound/soc/soc-compress.c. Scenario: Playing a mix of MP3 and WAV files repeatedly from YouTube Music by pressing "next". Issue occurs when snd_soc_dpcm_runtime_update is called and soc_compr_trigger_fe is invoked, causing the value of “fe->dpcm[stream].runtime_update” to change mid-process. This results in a problem where, during the closing of the BE connected to the FE in snd_soc_dpcm_runtime_update, only the entry is removed from the list without actually being closed. The attached patch resolves this issue as confirmed through testing. Additionally, while performing DPCM related operations in soc-compress.c, snd_soc_dpcm_mutex_lock is used; however, soc_compr_trigger_fe does not have this lock. Please review the attached patch regarding this matter. Thank you. From 72a0340bf7dccc6f53dc2d02e04bb3f894d796e3 Mon Sep 17 00:00:00 2001 From: Seungbin Lee <mailto:seungbin.lee@samsung.com> Date: Wed, 18 Jun 2025 13:33:02 +0900 Subject: [PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe If the soc_compr_trigger_fe function is called during snd_soc_dpcm_runtime_update execution, the FE's DPCM runtime update variable may be changed. Then the BE list connected to the FE flies away, and the BE cannot be closed. This needs to be protected by the snd_soc_dpcm_mutex_lock. Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <mailto:seungbin.lee@samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

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[PATCH] ALSA: hda/realtek: add fixup for HP Spectre x360 14t-ea100 (ALC245)
by Kaden Berger 22 Jun '25

22 Jun '25

This patch adds a quirk to enable proper speaker output on the HP Spectre x360 14t-ea100 with the Realtek ALC245 code. SIgned-off-by: Kaden Berger <kadenb816(a)gmail.com> diff -ura linux-6.15.1.orig/sound/pci/hda/patch_realtek.c linux-6.15.1.new/sound/pci/hda/patch_realtek.c --- linux-6.15.1.orig/sound/pci/hda/patch_realtek.c 2025-06-09 08:21:07.178183524 -0500 +++ linux-6.15.1.new/sound/pci/hda/patch_realtek.c 2025-06-09 08:26:16.859486062 -0500 @@ -10792,6 +10793,7 @@ SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), + SND_PCI_QUIRK(0x103c, 0x89da, "HP Spectre x360 14t-ea100", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS),

2 1

[PATCH v2] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 22 Jun '25

22 Jun '25

Add audio quirks to fix speaker output and headset detection on the following Clevo models: - V350ENC - V350WNPQ - V540TU - X560WNR - X580WNS Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfcc204264aa..13bea7863bef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2657,6 +2657,7 @@ static const struct hda_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1558, 0x3702, "Clevo X370SN[VW]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x5802, "Clevo X58[05]WN[RST]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -11131,6 +11132,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x35a1, "Clevo V3[56]0EN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x35b1, "Clevo V3[57]0WN[MNP]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11158,6 +11161,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5700, "Clevo X560WN[RST]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11197,6 +11201,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), + SND_PCI_QUIRK(0x1558, 0xa743, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.49.0

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ucm2: USB-Audio: Add Teufel CAGE PRO
by GitHub pull_request - opened 21 Jun '25

21 Jun '25

alsa-project/alsa-ucm-conf pull request #582 was opened from hsitter: two stereo outputs: one for "game" and one for "chat". one mono input Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/582 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/582.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: amd: ps: fix for soundwire failures during hibernation exit sequence
by Vijendar Mukunda 21 Jun '25

21 Jun '25

During the hibernate entry sequence, ACP registers will be reset to default values and acp ip will be completely powered off including acp SoundWire pads. During resume sequence, if acp SoundWire pad keeper enable register is not restored along with pad pulldown control register value, then SoundWire manager links won't be powered on correctly results in peripheral register access failures and completely audio function is broken. Add code to store the acp SoundWire pad keeper enable register and acp pad pulldown ctrl register values before entering into suspend state and restore the register values during resume sequence based on condition check for acp SoundWire pad keeper enable register for ACP6.3, ACP7.0 & ACP7.1 platforms. Fixes: 491628388005 ("ASoC: amd: ps: add callback functions for acp pci driver pm ops") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/acp63.h | 5 +++++ sound/soc/amd/ps/ps-common.c | 18 ++++++++++++++++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 85feae45c44c..babf2fa7ea27 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -82,6 +82,7 @@ #define ACP63_SDW0_DMA_MAX_STREAMS 6 #define ACP63_SDW1_DMA_MAX_STREAMS 2 #define ACP63_P1_AUDIO_TX_THRESHOLD 6 +#define ACP_SW_PAD_KEEPER_EN 0x0001454 /* * Below entries describes SDW0 instance DMA stream id and DMA irq bit mapping @@ -334,6 +335,8 @@ struct acp_hw_ops { * @addr: pci ioremap address * @reg_range: ACP reigister range * @acp_rev: ACP PCI revision id + * @acp_sw_pad_keeper_en: store acp SoundWire pad keeper enable register value + * @acp_pad_pulldown_ctrl: store acp pad pulldown control register value * @acp63_sdw0-dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire * manager-SW0 instance * @acp63_sdw_dma_intr_stat: DMA interrupt status array for ACP6.3 platform SoundWire @@ -367,6 +370,8 @@ struct acp63_dev_data { u32 addr; u32 reg_range; u32 acp_rev; + u32 acp_sw_pad_keeper_en; + u32 acp_pad_pulldown_ctrl; u16 acp63_sdw0_dma_intr_stat[ACP63_SDW0_DMA_MAX_STREAMS]; u16 acp63_sdw1_dma_intr_stat[ACP63_SDW1_DMA_MAX_STREAMS]; u16 acp70_sdw0_dma_intr_stat[ACP70_SDW0_DMA_MAX_STREAMS]; diff --git a/sound/soc/amd/ps/ps-common.c b/sound/soc/amd/ps/ps-common.c index 1c89fb5fe1da..f18d2a0d83aa 100644 --- a/sound/soc/amd/ps/ps-common.c +++ b/sound/soc/amd/ps/ps-common.c @@ -160,6 +160,8 @@ static int __maybe_unused snd_acp63_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -197,6 +199,7 @@ static int __maybe_unused snd_acp63_runtime_resume(struct device *dev) static int __maybe_unused snd_acp63_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -209,6 +212,12 @@ static int __maybe_unused snd_acp63_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } @@ -408,6 +417,8 @@ static int __maybe_unused snd_acp70_suspend(struct device *dev) adata = dev_get_drvdata(dev); if (adata->is_sdw_dev) { + adata->acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + adata->acp_pad_pulldown_ctrl = readl(adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); adata->sdw_en_stat = check_acp_sdw_enable_status(adata); if (adata->sdw_en_stat) { writel(1, adata->acp63_base + ACP_ZSC_DSP_CTRL); @@ -445,6 +456,7 @@ static int __maybe_unused snd_acp70_runtime_resume(struct device *dev) static int __maybe_unused snd_acp70_resume(struct device *dev) { struct acp63_dev_data *adata; + u32 acp_sw_pad_keeper_en; int ret; adata = dev_get_drvdata(dev); @@ -459,6 +471,12 @@ static int __maybe_unused snd_acp70_resume(struct device *dev) if (ret) dev_err(dev, "ACP init failed\n"); + acp_sw_pad_keeper_en = readl(adata->acp63_base + ACP_SW_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW_PAD_KEEPER_EN:0x%x\n", acp_sw_pad_keeper_en); + if (!acp_sw_pad_keeper_en) { + writel(adata->acp_sw_pad_keeper_en, adata->acp63_base + ACP_SW0_PAD_KEEPER_EN); + writel(adata->acp_pad_pulldown_ctrl, adata->acp63_base + ACP_PAD_PULLDOWN_CTRL); + } return ret; } -- 2.45.2

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[PATCH] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 20 Jun '25

20 Jun '25

Add audio quirks to fix speaker output and headset detection on the following Clevo models: - V350ENC - V350WNPQ - V540TU - X560WNR - X580WNS Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfcc204264aa..54d6edf203a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11131,6 +11131,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x35a1, "Clevo V3[56]0EN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x35b1, "Clevo V3[57]0WN[MNP]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11158,6 +11160,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5700, "Clevo X560WN[RST]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -11197,6 +11200,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), + SND_PCI_QUIRK(0x1558, 0xa743, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.49.0

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