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Improvements to Steinberg UR-44 device support
by GitHub pull_request - opened 24 Jun '25

24 Jun '25

alsa-project/alsa-ucm-conf pull request #584 was opened from peterix: - Relabeled the inputs and outputs to make sense considering the device's physical layout and internals - Added one more USB device ID for the UR44 (0x1704 is a non-C variant) - Added stereo inputs in sensible combinations for the various inputs on the device - 1 for inputs 5 & 6 in the back - 3 'adjacent ones' for the 4 inputs in the front - 1 & 2 - 2 & 3 - 3 & 4 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/584 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/584.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: rt721-sdca: fix boost gain calculation error
by Jack Yu 24 Jun '25

24 Jun '25

Fix the boost gain calculation error in rt721_sdca_set_gain_get. This patch is specific for "FU33 Boost Volume". Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt721-sdca.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index 1c9f32e405cf..ba080957e933 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -430,6 +430,7 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; unsigned int adc_vol_flag = 0; const unsigned int interval_offset = 0xc0; + const unsigned int tendA = 0x200; const unsigned int tendB = 0xa00; if (strstr(ucontrol->id.name, "FU1E Capture Volume") || @@ -439,9 +440,16 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, regmap_read(rt721->mbq_regmap, mc->reg, &read_l); regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); - if (mc->shift == 8) /* boost gain */ + if (mc->shift == 8) { + /* boost gain */ ctl_l = read_l / tendB; - else { + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (read_l == 0x8000 || read_l == 0xfe00) + ctl_l = 0; + else + ctl_l = read_l / tendA + 1; + } else { if (adc_vol_flag) ctl_l = mc->max - (((0x1e00 - read_l) & 0xffff) / interval_offset); else @@ -449,9 +457,16 @@ static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, } if (read_l != read_r) { - if (mc->shift == 8) /* boost gain */ + if (mc->shift == 8) { + /* boost gain */ ctl_r = read_r / tendB; - else { /* ADC/DAC gain */ + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (read_r == 0x8000 || read_r == 0xfe00) + ctl_r = 0; + else + ctl_r = read_r / tendA + 1; + } else { /* ADC/DAC gain */ if (adc_vol_flag) ctl_r = mc->max - (((0x1e00 - read_r) & 0xffff) / interval_offset); else -- 2.34.1

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[PATCH V2] ASoC: amd: ps: fix for soundwire failures during hibernation exit sequence
by Vijendar Mukunda 24 Jun '25

24 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> --- Changes since v1: removed extra macro ACP_SW_PAD_KEEPER_EN and used existing macro ACP_SW0_PAD_KEEPER_EN in the code. sound/soc/amd/ps/acp63.h | 4 ++++ sound/soc/amd/ps/ps-common.c | 18 ++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index 85feae45c44c..d7c994e26e4d 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -334,6 +334,8 @@ 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 +369,8 @@ 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..7b4966b75dc6 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_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); @@ -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_SW0_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW0_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; } @@ -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_SW0_PAD_KEEPER_EN); + dev_dbg(dev, "ACP_SW0_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] 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. Signed-off-by: Yuzuru <yuzuru_10(a)proton.me> --- 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

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[PATCH] ASoC: amd: yc: Add DMI quirk for Lenovo IdeaPad Slim 5 15
by Oliver Schramm 23 Jun '25

23 Jun '25

It's smaller brother has already received the patch to enable the microphone, now add it too to the DMI quirk table. Cc: stable(a)vger.kernel.org Signed-off-by: Oliver Schramm <oliver.schramm97(a)gmail.com> --- 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..c9e1b777b70e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -360,6 +360,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83J2"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83J3"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.49.0

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[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

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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

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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

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