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

[PATCH] ASoC: amd: yc: add quirk for Acer Nitro ANV15-41 internal mic
by Yuzuru10 23 Jun '25

23 Jun '25

This patch adds DMI-based quirk for the Acer Nitro ANV15-41, allowing the internal microphone to be detected correctly on machines with "RB" as board vendor. --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 98022e5fd428..499f9f7c76ee 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -353,6 +353,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "RB"), + DMI_MATCH(DMI_PRODUCT_NAME, "Nitro ANV15-41"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.50.0

1 0

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

1 0

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

1 0

[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

1 0

[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

2 1

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

1 0

[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

2 1

[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

1 0

aplay.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 20 Jun '25

20 Jun '25

Package: alsa-utils Version: 1.2.14-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use grep -n -e ' $' -e '\\~$' -e ' \\f.$' -e ' \\"' <file> to find (most) trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff:<stdin>:258: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: 13.0 APT prefers testing-security APT policy: (500, 'testing-security'), (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.32-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34.2-2 ii libasound2t64 1.2.14-1 ii libatopology2t64 1.2.14-1 ii libc6 2.41-8 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

alsactl.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 20 Jun '25

20 Jun '25

Package: alsa-utils Version: 1.2.14-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use grep -n -e ' $' -e '\\~$' -e ' \\f.$' -e ' \\"' <file> to find (most) trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff:<stdin>:224: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: 13.0 APT prefers testing-security APT policy: (500, 'testing-security'), (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.32-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 34.2-2 ii libasound2t64 1.2.14-1 ii libatopology2t64 1.2.14-1 ii libc6 2.41-8 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

Handling Lost Audio IRQs in ALSA: Triggering XRUN on Timer Detection?
by Verma, Sandeep 20 Jun '25

20 Jun '25

Hi all, I’m currently investigating a sporadic issue in our system where an audio interrupt is occasionally lost. When this happens, ALSA detects the anomaly (presumably through its internal timestamp or delay detection), but it doesn’t appear to take corrective action—leading to persistent audio jitter afterward. To address this, I’m considering implementing a mechanism in our driver or platform layer that uses a timer to check if the expected audio interrupt hasn’t arrived within a certain timeframe (e.g., 2x the period size). If this condition is met, I plan to explicitly trigger an XRUN to reset the pipeline and recover cleanly. My questions are: a. Is this an acceptable and “ALSA-friendly” way to handle lost IRQs? b. If this is a reasonable approach, why doesn’t ALSA do this by default? c. Is there a better or recommended way within the ALSA framework to detect and recover from such missed interrupts? I’d appreciate any guidance or suggestions. If this kind of timer-based recovery would be broadly useful, I’d also be happy to explore whether it could be proposed upstream. Thanks, Sandeep

1 0

[PATCH v2] ALSA: hda/realtek: Fixup ft alc257 rename alc3328
by wangdicheng 20 Jun '25

20 Jun '25

Audio ALC3328 recognized as ALC257, updated PCI ID0x10EC12F0 to rename it to 3328 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- 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 cd0d7ba7320e..2a08115d8c17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1088,6 +1088,7 @@ static const struct alc_codec_rename_pci_table rename_pci_tbl[] = { { 0x10ec0668, 0x103c, 0, "ALC3662" }, { 0x10ec0283, 0x17aa, 0, "ALC3239" }, { 0x10ec0292, 0x17aa, 0, "ALC3232" }, + { 0x10ec0257, 0x12f0, 0, "ALC3328" }, { } /* terminator */ }; -- 2.25.1

2 1

[GIT PULL] ASoC fixes for v6.16-rc2
by Mark Brown 20 Jun '25

20 Jun '25

The following changes since commit 19272b37aa4f83ca52bdf9c16d5d81bdd1354494: Linux 6.16-rc1 (2025-06-08 13:44:43 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc2 for you to fetch changes up to 16ea4666bbb7f5bd1130fa2d75631ccf8b62362e: ASoC: Intel: sof-function-topology-lib: Print out the unsupported dmic count (2025-06-19 15:44:42 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 A relatively large collection of fixes and updates that came in since the merge window. Of note are a couple of Cirrus ones which change the firmware naming for some newly added devices, and a fix from Laurentiu for issues booting firmwares on the DSPs on i.MX8 SoCs. ---------------------------------------------------------------- Gabriel Santese (1): ASoC: amd: yc: Add quirk for MSI Bravo 17 D7VF internal mic Harshit Mogalapalli (1): ASoC: cs48l32: Fix a signedness bug in cs48l32_hw_params() Huacai Chen (1): ASoC: loongson: Fix build warnings about export.h Laurentiu Mihalcea (1): ASoC: SOF: imx8: add core shutdown operation for imx8/imx8x Mark Brown (1): ASoC: cs35l56: Change firmware filenames for SoundWire Peter Ujfalusi (1): ASoC: Intel: sof-function-topology-lib: Print out the unsupported dmic count Raven Black (1): ASoC: amd: yc: update quirk data for HP Victus Richard Fitzgerald (4): ASoC: cs35l56: Use SoundWire address as firmware name suffix for new silicon ASoC: cs35l56: Use SoundWire address as alternate firmware suffix on L56 B0 ASoC: doc: cs35l56: Update to add new SoundWire firmware filename suffix ASoC: doc: cs35l56: Add CS35L63 to the list of supported devices Sven Peter (1): ASoC: apple: mca: Drop default ARCH_APPLE in Kconfig Thorsten Blum (1): ASoC: sdw_utils: Fix potential NULL pointer deref in is_sdca_endpoint_present() Zhang Yi (1): ASoC: codecs: ES8326: Modify initialization configuration Documentation/sound/codecs/cs35l56.rst | 24 ++++++-- sound/soc/amd/yc/acp6x-mach.c | 14 +++++ sound/soc/apple/Kconfig | 1 - sound/soc/codecs/cs35l56-sdw.c | 18 +++--- sound/soc/codecs/cs35l56.c | 72 +++++++++++++++++++--- sound/soc/codecs/cs35l56.h | 3 + sound/soc/codecs/cs48l32.c | 4 ++ sound/soc/codecs/es8326.c | 3 +- sound/soc/codecs/wm_adsp.c | 27 +++++--- sound/soc/codecs/wm_adsp.h | 2 + sound/soc/intel/common/sof-function-topology-lib.c | 3 +- sound/soc/loongson/loongson_i2s.c | 1 + sound/soc/sdw_utils/soc_sdw_utils.c | 2 + sound/soc/sof/imx/imx8.c | 15 +++++ 14 files changed, 154 insertions(+), 35 deletions(-)

2 1

Unable to see my Focusrite Scarlett Solo 4th gen
by GitHub issues - opened 18 Jun '25

18 Jun '25

alsa-project/alsa-lib issue #460 was opened from thereticle: Hi, a few days ago I bought a Focusrite Scarlett Solo 4th gen. It worked for a while, but since I had problems with Davinci Resolve (crackling sound) I tried to fix it without success until I could not see the Scarlett anymore. Actually lsusb sees the card Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 001 Device 002: ID 0db0:0076 Micro Star International MYSTIC LIGHT Bus 001 Device 007: ID 1235:8218 Focusrite-Novation Scarlett Solo 4th Gen Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub [...] but alsa does not see it. After many attempts with the help of ChatGPT and other IA, I thought I had solved it by installing the alsa-ucm-conf package, but after the next reboot the card disappeared again. I use Xubuntu 25.04 with Pipewire (but I also tried with pulseaudio without success). Any suggestions? Issue URL : https://github.com/alsa-project/alsa-lib/issues/460 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ALSA: hda/conexant: Renaming the codec with device ID 0x1f86 and 0x1f87
by wangdicheng 18 Jun '25

18 Jun '25

From: wangdicheng <wangdicheng(a)kylinos.cn> Due to changes in the manufacturer's plan, all 0x14f11f86 will be named CX11880, and 0x14f11f87 will be named SN6140 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/hda/patch_conexant.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4985e72b9094..1366a33de8d3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -42,7 +42,7 @@ struct conexant_spec { unsigned int gpio_led; unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; - bool is_cx8070_sn6140; + bool is_cx11880_sn6140; }; @@ -195,7 +195,7 @@ static int cx_auto_init(struct hda_codec *codec) cxt_init_gpio_led(codec); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); - if (spec->is_cx8070_sn6140) + if (spec->is_cx11880_sn6140) cx_fixup_headset_recog(codec); return 0; @@ -247,7 +247,7 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + /* In cx11880 and sn6140, the node 16 can only be configured to headphone or disabled, * the node 19 can only be configured to microphone or disabled. * Check hp&mic tag to process headset plugin & plugout. */ @@ -1191,11 +1191,11 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = cx_auto_patch_ops; - /* init cx8070/sn6140 flag and reset headset_present_flag */ + /* init cx11880/sn6140 flag and reset headset_present_flag */ switch (codec->core.vendor_id) { case 0x14f11f86: case 0x14f11f87: - spec->is_cx8070_sn6140 = true; + spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); break; } @@ -1283,7 +1283,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { - HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f11f86, "CX11880", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f11f87, "SN6140", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto), -- 2.25.1

2 1

[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

2 2

[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

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[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jun '25

17 Jun '25

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

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[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 17 Jun '25

17 Jun '25

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

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[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 17 Jun '25

17 Jun '25

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

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[PATCH v1] ASoC: tas2781: Move the "include linux/debugfs.h" into tas2781.h
by Shenghao Ding 16 Jun '25

16 Jun '25

Move the include linux/debugfs.h into tas2781.h for code clean. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2781.h | 4 ++++ sound/soc/codecs/tas2781-i2c.c | 3 --- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 40cd3bd079b5..3875e92f1ec5 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -17,6 +17,10 @@ #ifndef __TAS2781_H__ #define __TAS2781_H__ +#ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C +#include <linux/debugfs.h> +#endif + #include "tas2781-dsp.h" /* version number */ diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index c40d8f754d89..9f4d965a1335 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -14,9 +14,6 @@ // #include <linux/crc8.h> -#ifdef CONFIG_SND_SOC_TAS2781_ACOUST_I2C -#include <linux/debugfs.h> -#endif #include <linux/firmware.h> #include <linux/gpio/consumer.h> #include <linux/i2c.h> -- 2.43.0

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add UCM2 support for MSI MAG B850M Mortar Wifi (ALC4080)
by GitHub pull_request - opened 16 Jun '25

16 Jun '25

alsa-project/alsa-ucm-conf pull request #581 was opened from DanielDecker: Solves problems with front panel audio and S/PDIF sound output for MSI MAG B850M Mortar Wifi, which uses the Realtek ALC4080 audio chip. I also tested line in (back side) and microphone input (front panel). [alsa-info.txt](https://alsa-project.org/db/?f=70ed25ea247c3ac8c253755a5c70e… Thanks in advance. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/581 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/581.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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