From 95b2e0361c88753afa030c10698be0a1a5b67650 Mon Sep 17 00:00:00 2001 From: Kyle Zeng Date: Thu, 17 Sep 2026 12:56:43 +0200 Subject: [PATCH 01/53] ALSA: seq: Serialize compat port-info ioctls The native sequencer ioctl path serializes handler calls with client->ioctl_mutex, but the translated port-info compat path invokes the same handlers through snd_seq_kernel_client_ctl() without taking that mutex. This lets concurrent compat CREATE_PORT requests pass the port-count check before any request reaches the serialized insertion. The computed integer port index can then exceed the address field range and wrap to an existing index. Subsequent subscriber teardown can resolve the duplicate address to the wrong port and access a freed subscriber. Take ioctl_mutex while dispatching converted port-info requests, matching the native ioctl path. All translated port-info commands share this helper, so their accesses to the client port state are serialized as well. Fixes: b3defb791b26 ("ALSA: seq: Make ioctls race-free") Cc: stable@vger.kernel.org Assisted-by: Codex:gpt-5.6-sol gpt-6-astra Signed-off-by: Kyle Zeng Signed-off-by: Bruno Produit Link: https://patch.msgid.link/20260917105643.90102-1-bruno.produit@trailofbits.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_compat.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c index 22679dca9aae..80110501da6f 100644 --- a/sound/core/seq/seq_compat.c +++ b/sound/core/seq/seq_compat.c @@ -44,7 +44,9 @@ static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned return -EFAULT; data->kernel = NULL; - err = snd_seq_kernel_client_ctl(client->number, cmd, data); + scoped_guard(mutex, &client->ioctl_mutex) { + err = snd_seq_kernel_client_ctl(client->number, cmd, data); + } if (err < 0) return err; From 4cae4ca34992d210d115e8e891f1e88c9df5f24c Mon Sep 17 00:00:00 2001 From: Seiji Adachi Date: Thu, 17 Sep 2026 22:18:51 +0900 Subject: [PATCH 02/53] ASoC: amd: yc: Add Lenovo ThinkPad E16 Gen 2 (21JN) to quirks Signed-off-by: Seiji Adachi Link: https://patch.msgid.link/20260917131851.2122144-1-seiji_adachi@yahoo.co.jp Signed-off-by: Mark Brown --- 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 385fede6d77f..ea173bdab1d5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21JN"), + } + }, { .driver_data = &acp6x_card, .matches = { From 8d78e4f906cf0a1984cf3c852653727835e1a9e0 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 17 Sep 2026 15:58:51 +0200 Subject: [PATCH 03/53] ASoC: rt766: fix HID dependency for SND_SOC_SDCA_HID Selecting SND_SOC_SDCA_HID only works if HID is enabled: WARNING: unmet direct dependencies detected for SND_SOC_SDCA_HID Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_SDCA [=y] && (HID [=n]=y [=y] || HID [=n]=SND_SOC_SDCA [=y]) Selected by [y]: - SND_SOC_RT766_SDCA_SDW [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SOUNDWIRE [=y] && SND_SOC_SDCA [=y] Change the dependency to require SND_SOC_SDCA_CLASS instead, which solves the problem without extra selects. Fixes: b50ecf5873df ("ASoC: rt766: add RT766/RT767 SDCA driver") Signed-off-by: Arnd Bergmann Link: https://patch.msgid.link/20260917135950.405423-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d88593c2bac8..8e251e1d84b4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1989,9 +1989,7 @@ config SND_SOC_RT715_SDCA_SDW config SND_SOC_RT766_SDCA_SDW tristate "Realtek RT766 SDCA Codec - SDW" depends on SOUNDWIRE - depends on SND_SOC_SDCA - select SND_SOC_SDCA_HID - select SND_SOC_SDCA_IRQ + depends on SND_SOC_SDCA_CLASS select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ From 28114992dd2f03724eb3d024839b3f9f299656d3 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Thu, 17 Sep 2026 22:21:49 +0530 Subject: [PATCH 04/53] ASoC: amd: acp: enable TAS2783 and add RT712-VB SoundWire machine The ACP7.0 SoundWire machine table did not cover boards that combine TI TAS2783 smart amplifiers on SoundWire link 0 with a Realtek RT712-VB codec on link 1. In addition, the TAS2783 SoundWire codec driver was not selected by the AMD legacy SoundWire machine config, so it was not built for these platforms. Select SND_SOC_TAS2783_SDW from SND_SOC_AMD_LEGACY_SDW_MACH so the TAS2783 SoundWire codec driver is built together with the AMD legacy (no-DSP) SoundWire machine. Add a new link/address map, acp70_tas2783a_l0_rt712_vb_l1, describing the TAS2783 amplifiers on link 0 and the RT712-VB codec on link 1, and register a corresponding entry in snd_soc_acpi_amd_acp70_sdw_machines[]. The entry uses snd_soc_acpi_amd_sdca_is_device_rt712_vb() as its machine_check callback so it only matches platforms that actually expose the RT712-VB device. Signed-off-by: Syed Saba Kareem Link: https://patch.msgid.link/20260917165314.1880855-1-syed.sabakareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 20 ++++++++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index a04c301df4d6..e880e44b9763 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_AMD_LEGACY_SDW_MACH select SND_SOC_RT715_SDW select SND_SOC_RT715_SDCA_SDW select SND_SOC_RT722_SDCA_SDW + select SND_SOC_TAS2783_SDW help This option enables Legacy(No DSP) sound card support for SoundWire enabled AMD platforms along with ACP PDM controller. diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 815e088b437d..2a9da6de7bbc 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -748,7 +748,27 @@ static const struct snd_soc_acpi_link_adr acp70_rt721_only[] = { {} }; +static const struct snd_soc_acpi_link_adr acp70_tas2783a_l0_rt712_vb_l1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(tas2783_2_adr), + .adr_d = tas2783_2_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt712_vb_1_group1_adr), + .adr_d = rt712_vb_1_group1_adr, + }, + {} +}; + struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { + { + .link_mask = BIT(0) | BIT(1), + .links = acp70_tas2783a_l0_rt712_vb_l1, + .machine_check = snd_soc_acpi_amd_sdca_is_device_rt712_vb, + .drv_name = "amd_sdw", + }, { .link_mask = BIT(0), .links = acp70_tas2783_2, From bcd61aa247c28b721ac21dda6e4135f75a2de29f Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Mon, 14 Sep 2026 18:44:56 +0800 Subject: [PATCH 05/53] ASoC: codecs: max98363: fix uninitialized stream_config->type stream_config is not initialized before being passed to sdw_stream_add_slave(). The type field may contain garbage and is later copied to stream->type by sdw_config_stream(). Zero-initialize stream_config so type defaults to SDW_STREAM_PCM. Fixes: 18c0af945fa3 ("ASoC: max98363: add soundwire amplifier driver") Cc: stable@vger.kernel.org Signed-off-by: Jiangshan Yi Link: https://patch.msgid.link/20260914104456.376666-1-yijiangshan@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/codecs/max98363.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index 0e32b7a94cb0..4a6d7b163acf 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -200,7 +200,7 @@ static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream, struct max98363_priv *max98363 = snd_soc_component_get_drvdata(component); - struct sdw_stream_config stream_config; + struct sdw_stream_config stream_config = {0}; struct sdw_port_config port_config; enum sdw_data_direction direction; struct sdw_stream_runtime *stream; From 174d160884199cad97413f33fe1b56027dc0d3e1 Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Mon, 14 Sep 2026 18:45:51 +0800 Subject: [PATCH 06/53] ASoC: codecs: rt1017-sdca-sdw: fix uninitialized stream_config->type stream_config is not initialized before being passed to sdw_stream_add_slave(). The type field may contain garbage and is later copied to stream->type by sdw_config_stream(). Zero-initialize stream_config so type defaults to SDW_STREAM_PCM. While at it, use snd_sdw_params_to_config() helper instead of open-coding the same logic. Fixes: 2b7aecd58528 ("ASoC: rt1017: Add RT1017 SDCA amplifier driver") Cc: stable@vger.kernel.org Signed-off-by: Jiangshan Yi Link: https://patch.msgid.link/20260914104551.377880-1-yijiangshan@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt1017-sdca-sdw.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index caf75e5657ef..01cc2511a7d0 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include @@ -573,11 +574,10 @@ static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); - struct sdw_stream_config stream_config; + struct sdw_stream_config stream_config = {0}; struct sdw_port_config port_config; - enum sdw_data_direction direction; struct sdw_stream_runtime *sdw_stream; - int retval, port, num_channels, ch_mask; + int retval, port; unsigned int sampling_rate; dev_dbg(dai->dev, "%s %s", __func__, dai->name); @@ -591,28 +591,17 @@ static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ /* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - direction = SDW_DATA_DIR_RX; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) port = 1; - } else { - direction = SDW_DATA_DIR_TX; + else port = 2; - } - num_channels = params_channels(params); - ch_mask = (1 << num_channels) - 1; - - stream_config.frame_rate = params_rate(params); - stream_config.ch_count = num_channels; - stream_config.bps = snd_pcm_format_width(params_format(params)); - stream_config.direction = direction; - - port_config.ch_mask = ch_mask; + snd_sdw_params_to_config(substream, params, &stream_config, &port_config); port_config.num = port; dev_dbg(dai->dev, "frame_rate %d, ch_count %d, bps %d, direction %d, ch_mask %d, port: %d\n", - params_rate(params), num_channels, snd_pcm_format_width(params_format(params)), - direction, ch_mask, port); + stream_config.frame_rate, stream_config.ch_count, stream_config.bps, + stream_config.direction, port_config.ch_mask, port); retval = sdw_stream_add_slave(rt1017->sdw_slave, &stream_config, &port_config, 1, sdw_stream); From d395e1b62d9694f2e4add9122f8d473eacd374aa Mon Sep 17 00:00:00 2001 From: Daniel Ortiz Date: Sat, 19 Sep 2026 20:19:20 -0500 Subject: [PATCH 07/53] ASoC: amd: yc: Add DMI quirk for Lenovo V15 G6 ARP The Lenovo V15 G6 ARP (product name 83UU) uses the ACP6x PDM interface for its internal digital microphone array. Its firmware exposes neither the AcpDmicConnected _DSD property nor the _WOV method, so acp6x_probe() falls through to the DMI check and returns -ENODEV, leaving acp_yc_mach.0 unbound and no capture device registered. Add a DMI quirk entry for this model. With it, the machine driver binds and the DMIC capture device is created: acp_yc_mach acp_yc_mach.0: Enabling ACP DMIC support via DMI card 2: acp6x [acp6x], device 0: DMIC capture dmic-hifi-0 Capture was verified working with arecord and PipeWire. Signed-off-by: Daniel Ortiz Link: https://patch.msgid.link/20260920011920.2482786-1-letargopausado@gmail.com Signed-off-by: Mark Brown --- 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 ea173bdab1d5..e798bf6a9cd0 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -388,6 +388,13 @@ 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, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83UU"), + } + }, { .driver_data = &acp6x_card, .matches = { From e31ae4aeb049983fe3be0cf1e12c8074811a7685 Mon Sep 17 00:00:00 2001 From: HyeongJun An Date: Sun, 20 Sep 2026 22:53:27 +0900 Subject: [PATCH 08/53] ASoC: wm8903: Move the DRC QR threshold to the register that holds it The DRC_THRESH_QR field is bits 7 and 6 of DRC_1, but the control sits on DRC_0. Its shift and max are right for the field, only the register is wrong. Those bits of DRC_0 belong to DRC_STARTUP_GAIN, which "DRC Startup Volume" claims as bits 10 to 6. So the two controls share bits 7 and 6. Writing either one moves the other, and the quick release threshold is never reached at all. The six fields of DRC_1 tile it exactly, so there is nowhere else the threshold could live. Point the control at DRC_1. Found by a sweep for two controls claiming the same bits of one register. No board with this codec was to hand, the register map is the driver's own wm8903.h. Fixes: f1c0a02f32f8 ("ALSA: ASoC: Add WM8903 CODEC driver") Signed-off-by: HyeongJun An Assisted-by: Claude:claude-opus-5 Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260920135327.3628704-1-sammiee5311@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 156e1e24a388..fbf388cf966d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -676,7 +676,7 @@ SOC_ENUM("DRC Decay Rate", drc_decay), SOC_ENUM("DRC FF Delay", drc_ff_delay), SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0), SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0), -SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), +SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_1, 6, 3, 0, drc_tlv_max), SOC_ENUM("DRC QR Decay Rate", drc_qr_decay), SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0), From 75fc8a3ee5aee72f2fea3b6436170fae175f7f19 Mon Sep 17 00:00:00 2001 From: Sheetal Date: Mon, 21 Sep 2026 08:57:02 +0000 Subject: [PATCH 09/53] ASoC: tegra: Fix ASRC Stream6 input threshold control The Stream6 Input Threshold control points at stream index 4, which is already used by Stream5. The threshold get and put callbacks derive the lane ID from the register offset, so the duplicate index makes Stream6 operate on lane 4 and leaves lane 5 inaccessible from userspace. Use stream index 5 for Stream6. Fixes: a2df8c2d5b36 ("ASoC: tegra: Add Tegra186 based ASRC driver") Signed-off-by: Sheetal Reviewed-by: Thierry Reding Link: https://patch.msgid.link/20260921085704.1248920-2-sheetal@nvidia.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra186_asrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra186_asrc.c b/sound/soc/tegra/tegra186_asrc.c index 7f360dfaf8b1..d8ae5d997615 100644 --- a/sound/soc/tegra/tegra186_asrc.c +++ b/sound/soc/tegra/tegra186_asrc.c @@ -828,7 +828,7 @@ static const struct snd_kcontrol_new tegra186_asrc_controls[] = { tegra186_asrc_put_input_threshold), SOC_SINGLE_EXT("Stream6 Input Threshold", - ASRC_STREAM_REG(TEGRA186_ASRC_RX_THRESHOLD, 4), 0, 3, 0, + ASRC_STREAM_REG(TEGRA186_ASRC_RX_THRESHOLD, 5), 0, 3, 0, tegra186_asrc_get_input_threshold, tegra186_asrc_put_input_threshold), From d30fba93ebfc11e4604a7fcc73fdf28684191dde Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 18 Sep 2026 13:17:44 +0200 Subject: [PATCH 10/53] ASoC: codecs: rt286: Revert delay value for jack setup Fat fingered the change when reorganizing the jack-detect initialization. Equivalent changes for rt274 and rt298: Commit a43b4394bb35 ("ASoC: codecs: rt274: Always init jack_detect_work") Commit 1eb73102da28 ("ASoC: codecs: rt298: Reorganize jack detect handling") carry no update to the delay value. >From user perspective, no difference has been observed between 1250 and 50 values on Intel's SkyLake, KabyLake and AmberLake configurations. Fixes: 3082afe097cc ("ASoC: codecs: rt286: Reorganize jack detect handling") Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20260918111745.3589393-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 4217467904d6..bcd9f5abcda8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -953,7 +953,7 @@ static int rt286_probe(struct snd_soc_component *component) if (rt286->i2c->irq) schedule_delayed_work(&rt286->jack_detect_work, - msecs_to_jiffies(50)); + msecs_to_jiffies(1250)); return 0; } From 6eda8938a790f65bd24fcfaae7b40ad4613fb7a2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 18 Sep 2026 13:17:45 +0200 Subject: [PATCH 11/53] ASoC: codecs: rt274: Fix format setting for 44100 rate MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As per specification, Sample Base Rate (bit 14) shall be set to 1 for 44.1 kHz. Fixes: c7e79b2b2d2d ("ASoC: rt274: add rt274 codec driver") Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20260918111745.3589393-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt274.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index f8c370106504..3c47f1ae8db9 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -619,9 +619,11 @@ static int rt274_hw_params(struct snd_pcm_substream *substream, unsigned int val = 0; int d_len_code = 0, c_len_code = 0; + /* bit 14 Sample Rate Base, 0: 48kHz 1: 44.1kHz */ switch (params_rate(params)) { - /* bit 14 0:48K 1:44.1K */ case 44100: + val |= BIT(14); + break; case 48000: break; default: From 5d35f9be9b54a1b8f7df85ec5261557e43a999bb Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Tue, 22 Sep 2026 20:30:29 +0800 Subject: [PATCH 12/53] ASoC: amd: acp: Add DMI quirk for Lenovo Yoga Pro 7 14ASP9 The Lenovo Yoga Pro 7 14ASP9 (83HN) has no digital microphone on the ACP PDM interface; all internal mics are wired to the Realtek ALC287 codec. Add 83HN to acp70_acpi_flag_override_table to prevent the legacy ACP driver from binding and exposing a non-functional PDM capture device, consistent with the sibling 83W5/83V9 entries. Reported-by: Mikail Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221886 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260922123029.534152-1-zhangheng@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 2d50be8190c1..4d8f86aa3d16 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -51,6 +51,13 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "83V9"), }, }, + { + /* Lenovo Yoga Pro 7 14ASP9 (Strix Point, ACP 7.0) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83HN"), + }, + }, { .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC"), From b16610e3770ef22fc8fcce818db0cd955332abf4 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Thu, 24 Sep 2026 10:54:57 +0100 Subject: [PATCH 13/53] ASoC: SDCA: Set suspended flag after resuming within system suspend drv->suspended path was executed before system suspension instead of after, move it lower to correct it. Fixes: 7a5214f769c7 ("ASoC: SDCA: Add basic system suspend support") Signed-off-by: Maciej Strozek Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260924095458.2683755-2-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_class_function.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sdca/sdca_class_function.c b/sound/soc/sdca/sdca_class_function.c index cc7045dc26e6..5039230280ad 100644 --- a/sound/soc/sdca/sdca_class_function.c +++ b/sound/soc/sdca/sdca_class_function.c @@ -490,8 +490,6 @@ static int class_function_suspend(struct device *dev) struct class_function_drv *drv = auxiliary_get_drvdata(auxdev); int ret; - drv->suspended = true; - /* Ensure runtime resume runs on resume */ ret = pm_runtime_resume_and_get(dev); if (ret) { @@ -499,6 +497,8 @@ static int class_function_suspend(struct device *dev) return ret; } + drv->suspended = true; + sdca_irq_disable(drv->function, drv->core->irq_info); ret = pm_runtime_force_suspend(dev); From 91094f9c86216898ce92141c26c8fec7e97544c0 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Thu, 24 Sep 2026 10:54:58 +0100 Subject: [PATCH 14/53] ASoC: SDCA: Add better pm_runtime error handling in func probe Add a common error path in probe that balances the runtime PM reference. Fixes: 3af1815a2f9c ("ASoC: SDCA: Add basic SDCA function driver") Signed-off-by: Maciej Strozek Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260924095458.2683755-3-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_class_function.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/sdca/sdca_class_function.c b/sound/soc/sdca/sdca_class_function.c index 5039230280ad..610931ae5531 100644 --- a/sound/soc/sdca/sdca_class_function.c +++ b/sound/soc/sdca/sdca_class_function.c @@ -388,20 +388,27 @@ static int class_function_probe(struct auxiliary_device *auxdev, ret = devm_pm_runtime_enable(dev); if (ret) - return ret; + goto err_pm; ret = class_function_boot(drv); if (ret) - return ret; + goto err_pm; ret = devm_snd_soc_register_component(dev, cmp_drv, dais, num_dais); - if (ret) - return dev_err_probe(dev, ret, "failed to register component\n"); + if (ret) { + dev_err_probe(dev, ret, "failed to register component\n"); + goto err_pm; + } pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); return 0; + +err_pm: + pm_runtime_put_sync(dev); + + return ret; } static void class_function_remove(struct auxiliary_device *auxdev) From c5a8fc2abe05cf6db71f57ff0177fddc59861a1a Mon Sep 17 00:00:00 2001 From: Jasper Firth Date: Fri, 25 Sep 2026 23:06:25 +1000 Subject: [PATCH 15/53] ASoC: Intel: bytcr_rt5651: Add quirk for Chuwi Hi8 with generic DMI strings Some Chuwi Hi8 tablets with model number CWI513 ship with a BIOS (American Megatrends 5.11, dated 05/21/2016) whose DMI sys_vendor and product_name are both "Default string". The tested unit is labelled just "Hi8", but its model number is the same as the existing "Chuwi Hi8 Pro (CWI513)" entry, and it needs the same quirks. That entry matches on sys_vendor "Hampoo" and product_name "X1D3_C806N", so it never applies to these units. Without its quirks the internal speaker, whose amplifier has a differential input wired across LOUT left and right, cancels to a faint distorted residual for any centred (L == R) content, and the headphone channels are swapped. Add an entry matching board_vendor "Hampoo" and board_name "Cherry Trail CR", plus the BIOS date since those two are too generic on their own. drivers/platform/x86/touchscreen_dmi.c already matches the Chuwi HiBook the same way. The quirks are those of the existing Hi8 Pro entry, except for the over-current detection scale factor: 1.0 instead of the 0.75 from BYT_RT5651_DEFAULT_QUIRKS. With 0.75 and a CTIA headset (inline mic and one button), loud speech into the mic, and headphone playback coupling onto the mic line, were reported as spurious KEY_PLAYPAUSE presses, one of them held for a whole 8 second speech window. With 1.0 the same test gave no spurious events, and 3 real button presses were reported as exactly 3 press/release pairs. Tested on a unit with the generic DMI strings, with a module built from this patch and no quirk module parameter. The quirks were applied via the new entry, the card components read "cfg-spk:1 cfg-mic:in2 cfg-hp:lrswap", and the following all work: speaker, headphone left/right order, jack and headset-mic detection, headset button, and the internal and headset mics. The problem was diagnosed with the help of an AI coding assistant, which also drafted this changelog. Assisted-by: Claude Opus 5.5 Signed-off-by: Jasper Firth Link: https://patch.msgid.link/20260925-hi8-quirk-v1-1-ffd00b0447f9@hotmail.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 62cb4856c797..c9b1053205de 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -410,6 +410,23 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_HP_LR_SWAPPED | BYT_RT5651_MONO_SPEAKER), }, + { + /* Chuwi Hi8 (CWI513) with a BIOS using generic DMI strings */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + /* Above matches are too generic, add bios-date match */ + DMI_MATCH(DMI_BIOS_DATE, "05/21/2016"), + }, + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_JD1_1 | + BYT_RT5651_OVCD_TH_2000UA | + BYT_RT5651_OVCD_SF_1P0 | + BYT_RT5651_IN2_MAP | + BYT_RT5651_HP_LR_SWAPPED | + BYT_RT5651_MONO_SPEAKER), + }, { /* Chuwi Vi8 Plus (CWI519) */ .callback = byt_rt5651_quirk_cb, From e0cb0570d0ebce6f452ab492e8743f6f7eb2558e Mon Sep 17 00:00:00 2001 From: Jianyu Zhou <997536907@qq.com> Date: Thu, 17 Sep 2026 14:58:29 +0800 Subject: [PATCH 16/53] ALSA: hda/realtek: Add quirk for IPASON SmartBook S1 From 2adfedabfdd9f5ec800462b1f1c0859a198e63aa Mon Sep 17 00:00:00 2001 From: Jianyu Zhou <997536907@qq.com> Date: Thu, 17 Sep 2026 14:34:46 +0800 Subject: [PATCH v2] ALSA: hda/realtek: Add quirk for IPASON SmartBook S1 The internal speaker of the IPASON SmartBook S1 (an Emdoor ODM laptop) stays silent under Linux, while the Bluetooth audio and the headset output work fine. The same machine plays sound under Windows with the vendor driver, so the hardware and the amplifier are known good. The BIOS reports the pin default of the actual internal speaker pin (NID 0x1b) as 0x411111f0, i.e. the default association 0xf, which means "unconnected". The pin parser therefore drops that node already at the pin-configuration time, hence no mixer control is ever created for it and the pin is never driven, no matter how the output pin 0x14 is set up. Enabling 0x1b makes the speaker work immediately. The existing ALC256_FIXUP_HONOR_MRB_XXX_M1020_AUDIO quirk makes the speaker work, too, but it additionally rewrites the pins 0x14, 0x19 and 0x1a with Honor-specific values, which creates a bogus "Dock Mic" device that does not exist on this hardware. Add a minimal fixup that only enables 0x1b as the internal speaker. Touching 0x14 is not needed: with the pin configuration overridden at runtime, both keeping its BIOS value (0x90170110) and disabling it (0x411111f0) leave the speaker working, so 0x1b alone decides the output. Codec: Realtek ALC256 (0x10ec0256), SSID 0x2782:0x0206 BIOS pincfg: 0x14 0x90170110 0x18 0x411111f0 0x19 0x02a19040 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4067b945 0x1e 0x411111f0 0x21 0x02214020 Tested on 7.2.0-1-MANJARO and 6.18.45-1-MANJARO, with both the SOF and the legacy HDA driver paths. Signed-off-by: Jianyu Zhou <997536907@qq.com> Link: https://patch.msgid.link/tencent_3AB9214FB134949B31E9C471BC3669117F05@qq.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 5127a111c59c..fdca7e5db3a9 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4384,6 +4384,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN_HEADSET, ALC285_LENOVO_DAC_RENAME, ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1, + ALC256_FIXUP_IPASON_SMARTBOOK_S1, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -7136,6 +7137,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_TXNW2781_I2C, }, + [ALC256_FIXUP_IPASON_SMARTBOOK_S1] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170110 }, /* the real internal speaker */ + { } + }, + }, }; static const struct hda_quirk alc269_fixup_tbl[] = { @@ -8354,6 +8362,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x2014, 0x800a, "Positivo ARN50", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x2039, 0x0001, "Inspur S14-G1", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x2145, 0x0001, "Star Labs StarFighter", ALC233_FIXUP_STARLABS_STARFIGHTER), + SND_PCI_QUIRK(0x2782, 0x0206, "IPASON SmartBook S1", ALC256_FIXUP_IPASON_SMARTBOOK_S1), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), From ae0f12706433995336784a0966d42d2f7f7b5dac Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Fri, 18 Sep 2026 19:03:01 +0800 Subject: [PATCH 17/53] ALSA: hda/realtek: Add quirk for Acer Nitro ANV16-41 The Acer Nitro ANV16-41 requires the ALC2XX_FIXUP_HEADSET_MIC quirk to make the headset microphone work properly. Link: https://bugzilla.kernel.org/show_bug.cgi?id=221953 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260918110301.664172-1-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index fdca7e5db3a9..30972cbe5f03 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7204,6 +7204,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1679, "Acer Nitro 16 AN16-41", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x171e, "Acer Nitro ANV15-51", ALC245_FIXUP_ACER_MICMUTE_LED), + SND_PCI_QUIRK(0x1025, 0x172c, "Acer Nitro ANV16-41", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1731, "Acer Predator PHN16-72", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x173a, "Acer Swift SFG14-73", ALC245_FIXUP_ACER_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1758, "Acer Nitro ANV15-41", ALC245_FIXUP_ACER_MICMUTE_LED), From 86f86e6fdedc33c5b8d38b9208b350f8accf57ee Mon Sep 17 00:00:00 2001 From: Xiang Mei Date: Fri, 18 Sep 2026 15:57:53 -0700 Subject: [PATCH 18/53] ALSA: ua101: reject mismatched capture/playback packet sizes detect_usb_format() cross-checks bSubframeSize, bBitResolution and tSamFreq between the capture and playback interfaces, but never relates the two endpoints' wMaxPacketSize and bNrChannels. Each playback URB gets a buffer of ua->playback.max_packet_bytes, while the number of bytes written into it is derived from the capture stream: capture_urb_complete() computes frames from the received capture packet and capture.frame_bytes, and start_usb_playback() and playback_work() multiply that by playback.frame_bytes. A device declaring a large capture wMaxPacketSize with few capture channels and a small playback wMaxPacketSize with many playback channels therefore memset()s and memcpy()s past the end of the playback buffer, in open() of the PCM node the driver registers during probe. usb_submit_urb() rejects the over-long iso_frame_desc[0].length with -EMSGSIZE, but only after the write. Reject such descriptors at probe time. Genuine UA-101/UA-1000 hardware declares proportional packet sizes and is unaffected. BUG: KASAN: slab-out-of-bounds in start_usb_playback (sound/usb/misc/ua101.c:586) Write of size 2048 at addr ffff8881098f3c00 by task exploit/5021 Call Trace: __asan_memset (mm/kasan/shadow.c:84) start_usb_playback (sound/usb/misc/ua101.c:586) playback_pcm_open (sound/usb/misc/ua101.c:679) snd_pcm_open_substream (sound/core/pcm_native.c:2829) snd_pcm_open (sound/core/pcm_native.c:2865 sound/core/pcm_native.c:2932) snd_pcm_playback_open (sound/core/pcm_native.c:2891) snd_open (sound/core/sound.c:166) chrdev_open (fs/char_dev.c:411) do_dentry_open (fs/open.c:996) vfs_open (fs/open.c:1101) path_openat (fs/namei.c:4837 fs/namei.c:5000) do_file_open (fs/namei.c:5029) do_sys_openat2 (fs/open.c:1417) __x64_sys_openat (fs/open.c:1423 fs/open.c:1439 fs/open.c:1434) do_syscall_64 (arch/x86/entry/syscall_64.c:61 arch/x86/entry/syscall_64.c:84) entry_SYSCALL_64_after_hwframe (arch/x86/entry/entry_64.S:121) The buggy address belongs to the object at ffff8881098f3c00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 0 bytes inside of allocated 168-byte region [ffff8881098f3c00, ffff8881098f3ca8) Cc: stable@vger.kernel.org Fixes: 63978ab3e3e9 ("sound: add Edirol UA-101 support") Reported-by: Assisted-by: LLM Signed-off-by: Xiang Mei Link: https://patch.msgid.link/20260918225753.1278505-1-xmei5@asu.edu Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 860a62a3d74b..58d8a3cdc8b9 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1007,6 +1007,15 @@ static int detect_usb_format(struct ua101 *ua) } ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, usb_endpoint_num(epd)); ua->playback.max_packet_bytes = usb_endpoint_maxp(epd); + + if (ua->capture.max_packet_bytes / ua->capture.frame_bytes * + ua->playback.frame_bytes > ua->playback.max_packet_bytes) { + dev_err(&ua->dev->dev, + "playback packet size %u too small for %u capture frames\n", + ua->playback.max_packet_bytes, + ua->capture.max_packet_bytes / ua->capture.frame_bytes); + return -ENXIO; + } return 0; } From 61b0a5a114fa4552892da5c0ff2f42e63bf08c6b Mon Sep 17 00:00:00 2001 From: Xiang Mei Date: Fri, 18 Sep 2026 16:40:14 -0700 Subject: [PATCH 19/53] ALSA: line6: reject oversized playback packets Each playback URB is handed a slice of out.buffer that is exactly LINE6_ISO_PACKETS * max_packet_size_out bytes, sized from the OUT endpoint, but the number of bytes written into that slice is never compared against it. submit_audio_out_urb() takes the frame count from prev_fsize, which audio_in_callback() derived from the *IN* endpoint's received packet length, and rescales it with the playback frame size; when prev_fsize is still zero it synthesizes a length from the sample rate instead. On a device declaring a large IN wMaxPacketSize and a small OUT wMaxPacketSize, the resulting memcpy(), or the memset() when the playback stream is idle, runs past its slice and, as the reproducer below shows, beyond the allocation. usb_submit_urb() is not a backstop. max_packet_size_out comes from usb_maxpacket(), which returns only the low 11 bits of wMaxPacketSize, while USB core validates a high-speed isochronous length against that base scaled by usb_endpoint_maxp_mult(). A length of up to three times the allocated slice is therefore accepted, and even a rejected URB is only rejected after the write. Reject a packet that does not fit the slice the driver allocated for it. Clamping it instead would silently shorten the capture-derived rate feedback and desynchronize the two streams. BUG: KASAN: slab-out-of-bounds in submit_audio_out_urb (sound/usb/line6/playback.c:229) Write of size 1024 at addr ffff888100bbd400 by task kworker/1:2/5002 Workqueue: events line6_startup_work Call Trace: __asan_memcpy (mm/kasan/shadow.c:106) submit_audio_out_urb (sound/usb/line6/playback.c:229) line6_submit_audio_out_all_urbs (sound/usb/line6/playback.c:291) line6_stream_start (sound/usb/line6/pcm.c:194) line6_pcm_acquire (sound/usb/line6/pcm.c:337) line6_startup_work (sound/usb/line6/driver.c:728) process_one_work (kernel/workqueue.c:3396) worker_thread (kernel/workqueue.c:3479 kernel/workqueue.c:3560) kthread (kernel/kthread.c:436) ret_from_fork (arch/x86/kernel/process.c:158) ret_from_fork_asm (arch/x86/entry/entry_64.S:245) The buggy address belongs to the object at ffff888100bbd400 which belongs to the cache kmalloc-256 of size 256 The buggy address is located 0 bytes inside of allocated 256-byte region [ffff888100bbd400, ffff888100bbd500) Cc: stable@vger.kernel.org Fixes: 7a0f55aeeb8f ("ALSA: line6: Support assymetrical in/out configurations") Reported-by: Assisted-by: LLM Signed-off-by: Xiang Mei Link: https://patch.msgid.link/20260918234014.1318325-1-xmei5@asu.edu Signed-off-by: Takashi Iwai --- sound/usb/line6/playback.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 7ebaf125f969..aa6ddf8746a0 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -182,6 +182,13 @@ static int submit_audio_out_urb(struct snd_line6_pcm *line6pcm) fsize *= bytes_per_frame; + if (fsize > line6pcm->max_packet_size_out) { + dev_err(line6pcm->line6->ifcdev, + "playback packet too large: %d > %d\n", + fsize, line6pcm->max_packet_size_out); + return -EMSGSIZE; + } + fout->offset = urb_size; fout->length = fsize; urb_size += fsize; From f4b506f735e3c24c4fd4d3a60239e0b1b241600e Mon Sep 17 00:00:00 2001 From: Matt Barr Date: Mon, 21 Sep 2026 17:48:12 -0400 Subject: [PATCH 20/53] ALSA: hda/realtek: Drop Yoga Pro 9 16IAH10 PCI SSID quirk The PCI SSID quirk added for the Lenovo Yoga Pro 9 16IAH10 (PCI SSID 17aa:3846, codec SSID 17aa:3920) also matches every other machine that shares PCI SSID 17aa:3846. snd_hda_pick_fixup() tries all PCI SSID entries before it falls back to the codec SSID, so those machines now get ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1 and never reach their own codec-SSID quirk. The Lenovo Legion Pro 7 16IRX8H (product 82WQ) is one such machine: PCI SSID 17aa:3846, codec SSID 17aa:3884. Before this quirk it picked ALC287_FIXUP_TAS2781_I2C via the 17aa:3884 codec SSID fallback and bound its TAS2781 amplifier. With it, the codec never binds the amplifier and the internal speakers are silent: snd_hda_codec_alc269 hdaudioC0D0: ALC287: picked fixup for PCI SSID 17aa:3846 versus, on a kernel without the quirk: snd_hda_codec_alc269 hdaudioC0D0: ALC287: picked fixup for codec SSID 17aa:3884 snd_hda_codec_alc269 hdaudioC0D0: bound i2c-TIAS2781:00 (ops tas2781_hda_comp_ops ...) Drop the PCI SSID entry. The Yoga Pro 9 16IAH10 has the same amplifier as the Yoga S990-16 (codec SSID 17aa:3920), so it still gets a fixup through the codec SSID fallback, using the existing 17aa:3920 entry. Switch that entry from ALC287_FIXUP_TXNW2781_I2C to ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1, which applies the DAC routing fix and then chains to ALC287_FIXUP_TXNW2781_I2C, so the amplifier setup is unchanged. Fixes: 41d60cbfde10 ("ALSA: hda/realtek: Fix bass speaker DAC routing for Lenovo Yoga Pro 9 16IAH10") Link: https://bugzilla.kernel.org/show_bug.cgi?id=220540 Suggested-by: Zhang Heng Signed-off-by: Matt Barr Link: https://patch.msgid.link/20260921214812.22139-1-matthewjaybarr@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 30972cbe5f03..b4164db7ecbc 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -8173,7 +8173,6 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3843, "Lenovo Yoga 9i / Yoga Book 9i", ALC287_FIXUP_LENOVO_YOGA_BOOK_9I), - SND_PCI_QUIRK(0x17aa, 0x3846, "Lenovo Yoga Pro 9 16IAH10", ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1), /* Yoga Pro 7 14IMH9 shares PCI SSID 17aa:3847 with Legion 7 16ACHG6; * use codec SSID to distinguish them */ @@ -8267,7 +8266,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3912, "Lenovo Xiaoxin 14 GT", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TXNW2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_YOGA9_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x3929, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x392b, "Thinkbook 13x Gen 5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), HDA_CODEC_QUIRK(0x17aa, 0x3936, "Legion R9000P ADR10H", ALC287_FIXUP_LENOVO_LEGION_AW88399), From d24b03248ac6f02fb87feff6beec0b5b1da5dafd Mon Sep 17 00:00:00 2001 From: Achilles Zhang Date: Mon, 21 Sep 2026 10:43:43 +0800 Subject: [PATCH 21/53] ALSA: usb-audio: Add native DSD quirk for HiBy FC4 The HiBy FC4 with USB ID 32bb:0004 advertises its native DSD stream as UAC2 RAW_DATA with a 4-byte, 32-bit subslot. Without a quirk snd-usb-audio leaves this alternate setting as SNDRV_PCM_FORMAT_SPECIAL, so userspace cannot select a native DSD format. Mark the device with QUIRK_FLAG_DSD_RAW so the existing generic raw DSD handling exposes the stream as DSD_U32_BE. Tested on an FC4 at DSD256. With the quirk the playback alternate setting changes from SPECIAL to DSD_U32_BE and reports DOP=0, bitrev=0; native DSD256 playback works correctly. Signed-off-by: Achilles Zhang Link: https://patch.msgid.link/20260921024344.391912-1-bnqzzdf@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c5870901a8d4..924d930a9f0e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2542,6 +2542,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x3255, 0x0000, /* Luxman D-10X */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x32bb, 0x0004, /* HiBy FC4 */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x3302, 0x17c2, /* TTGK Technology USB-C Audio */ QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x339b, 0x3a07, /* Synaptics HONOR USB-C HEADSET */ From 786fb6b3f513937f712db5a9ba6a12cbc6982ba9 Mon Sep 17 00:00:00 2001 From: Hernan Ganzo Date: Mon, 21 Sep 2026 21:55:38 +0100 Subject: [PATCH 22/53] ALSA: hda/realtek: Limit internal mic boost on HP Pavilion 15 (103c:2164) The HP Pavilion 15 with the ALC3227 codec (SSID 103c:2164) exposes the full four-step internal mic boost on pin 0x12: nsteps=3 with stepsize=0x2f, that is 0, 12, 24 and 36 dB. Combined with the ADC capture gain of +30 dB the analogue cascade reaches 66 dB, and the internal microphone clips on the ambient noise of a quiet room; the upper two steps are unusable at any useful input level. This is the defect class ALC269_FIXUP_LIMIT_INT_MIC_BOOST exists for. The machine matches no SSID quirk and falls through to the pin table, which selects ALC269_FIXUP_HP_MUTE_LED_MIC1 through a pin signature shared with several other HP models. Add an SSID entry instead, using ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED: the existing fixup that applies alc269_fixup_limit_int_mic_boost() and chains to the same mute LED fixup, as already done for 103c:218b. The boost becomes 0 or 12 dB and the LED handling is unchanged. Tested on the affected machine by selecting the same fixup through the model string (snd-hda-intel.model), which bypasses the SSID and pin tables, with the driver confirming the selection: snd_hda_codec_alc269 hdaudioC1D0: ALC3227: picked fixup limit-mic-boost (model specified) With that, Internal Mic Boost reports Limits 0 - 1 (0 or 12 dB) instead of 0 - 3 (0 to 36 dB), so alc269_fixup_limit_int_mic_boost() clamps this codec's internal mic pin as intended. Note that /proc/asound/card1/codec#0 keeps printing the unclamped hardware caps (nsteps=0x03): that file reads codec parameters uncached by design, while the fixup overrides the driver's cached parameter that the control is built from. This does not by itself make the default capture volume sane -- the remaining 12 dB on top of the +30 dB ADC gain is still hotter than this machine's usable operating point. It removes the two unusable steps from the hardware range, which is what a quirk of this class can do. Signed-off-by: Hernan Ganzo Link: https://patch.msgid.link/20260921205538.41192-1-gp@sneemgp.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index b4164db7ecbc..b94b0e29bda5 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7312,6 +7312,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x2164, "HP Pavilion 15 Notebook PC", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), From 775aa96c310d59f3d15a6e5a9bd1796fd1ba95ec Mon Sep 17 00:00:00 2001 From: Luis Banha Date: Tue, 22 Sep 2026 00:01:05 +0100 Subject: [PATCH 23/53] ALSA: hda/realtek: Add mute LED quirk for HP Pavilion 15-ec1xxx The mute LED on the HP Pavilion Gaming Laptop 15-ec1xxx (103c:87b2) does not work automatically. Add the ALC285_FIXUP_HP_MUTE_LED quirk to fix this behavior. Signed-off-by: Luis Banha Link: https://patch.msgid.link/20260921230105.16844-1-luisbanha26@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index b94b0e29bda5..8c29688d8e41 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7433,6 +7433,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x87b2, "HP Pavilion Gaming Laptop 15-ec1xxx", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87cb, "HP Pavilion 15-eg0xxx", ALC287_FIXUP_HP_GPIO_LED), From 3867e38b2d4b844d89076f988c7b9741c87a9858 Mon Sep 17 00:00:00 2001 From: KIM MinWoo Date: Wed, 23 Sep 2026 10:37:56 +0900 Subject: [PATCH 24/53] ALSA: hda/realtek: Add quirk for Lenovo ThinkBook 16p Gen 3 ARH (21EK) The ThinkBook 16p Gen 3 ARH (SSID 17aa:3871) has a TI TAS2563 amplifier device on I2C, exposed by ACPI as INT8866, the same arrangement as the Yoga 7 14ARB7 (SSID 17aa:3870). Without a quirk, the codec falls back to the vendor-wide Lenovo quirk (ALC269_FIXUP_LENOVO_XPAD_ACPI), the amplifier is never bound, and the internal speakers stay silent while the headphone output works. Reuse the Yoga 7 14ARB7 fixup, which binds the INT8866 device as a side codec. ALC287_FIXUP_TAS2781_I2C does not work on this machine because it looks for a TIAS2781 device. Assisted-by: LLM Signed-off-by: KIM MinWoo Link: https://patch.msgid.link/20260923013756.106033-1-phrimm136@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 8c29688d8e41..de478d01430e 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -8199,6 +8199,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */ SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), + SND_PCI_QUIRK(0x17aa, 0x3871, "Lenovo ThinkBook 16p Gen 3 ARH", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x3874, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), From d49f8d9680ee3f7f4e0339467cf0a244158c1adb Mon Sep 17 00:00:00 2001 From: Roman Bolotov Date: Wed, 23 Sep 2026 09:45:27 +0300 Subject: [PATCH 25/53] ALSA: usb-audio: Add quirk flag for ASUS SupremeFX Hi-Fi The ASUS SupremeFX Hi-Fi (0b05:1826, 0b05:1827) does not survive USB autosuspend. After a suspend/resume cycle the firmware degrades: HID probes begin to fail and the descriptors it returns become corrupted, until the device stops responding entirely and needs a physical power cycle to recover. Add QUIRK_FLAG_DISABLE_AUTOSUSPEND for both product IDs. Verified on 7.2.6 through the quirk_flags module parameter, with no code change, and with the local udev rules that had been forcing power/control commented out, so the parameter was the only mechanism in play. Both IDs were passed as 0b05:1827:disable_autosuspend;0b05:1826:disable_autosuspend The device then stayed awake across 2.5 hours (runtime_suspended_time remained 0), survived a system suspend/resume cycle, and continued to work afterwards with no failed probes and no power cycle needed. Signed-off-by: Roman Bolotov Link: https://patch.msgid.link/20260923064527.434441-1-hadros@ikeepitoblique.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 924d930a9f0e..294c7026b93c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2366,6 +2366,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0a73, 0x003a, /* Mackie DLZ Creator XS */ QUIRK_FLAG_ALWAYS_SET_RATE), + DEVICE_FLG(0x0b05, 0x1826, /* ASUS SupremeFX Hi-Fi */ + QUIRK_FLAG_DISABLE_AUTOSUSPEND), + DEVICE_FLG(0x0b05, 0x1827, /* ASUS SupremeFX Hi-Fi */ + QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x0b05, 0x18a6, /* ASUSTek Computer, Inc. */ QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE), DEVICE_FLG(0x0b0e, 0x0349, /* Jabra 550a */ From a01a223bd685be42e908d970de32eeb3646097f3 Mon Sep 17 00:00:00 2001 From: Tarun Noonemunthala Date: Wed, 23 Sep 2026 15:10:14 +0530 Subject: [PATCH 26/53] ALSA: hda/realtek: Add quirk for HP ENVY 13-aq1xxx The HP ENVY Laptop 13-aq1xxx (PCI SSID 103c:86ad) uses an ALC285 codec with two speaker pairs: front (pin 0x14) and bottom (pin 0x17). The BIOS leaves pin 0x14 as [N/A] ("not connected"), so the generic parser never enables the front pair and only the bottom speakers are audible. The vendor driver (RTKVHD64.sys) programs that pin with the value 0x90170150 and runs the HP amplifier initialization, which is already implemented in the driver as ALC285_FIXUP_HP_GPIO_AMP_INIT. Add a board fixup that applies the pin configuration for 0x14 and chains to the existing amplifier init fixup. The pin value was recovered from the vendor driver's own configuration data. The patch and this changelog were prepared with an AI coding assistant (see the Assisted-by tag below); the change was reviewed and tested by the submitter on the affected machine. Tested on an HP ENVY 13-aq1xxx by rebuilding snd-hda-codec-alc269 from the Fedora 7.1.10 source and loading it: the codec now reports "line_outs=2 (0x14/0x17)", pin 0x14 comes up as an output (Pin-ctls 0x40) with its amplifier unmuted, and both speaker pairs play. Assisted-by: OpenCode:deepseek-v4.1-flash checkpatch Signed-off-by: Tarun Noonemunthala Link: https://patch.msgid.link/20260923094014.58867-1-ntarun2000@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index de478d01430e..004f6a3eadfe 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4124,6 +4124,7 @@ enum { ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC285_FIXUP_HP_SPECTRE_X360_DF1, ALC285_FIXUP_HP_ENVY_X360, + ALC285_FIXUP_HP_ENVY_13AQ, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -6401,6 +6402,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_AMP_INIT, }, + [ALC285_FIXUP_HP_ENVY_13AQ] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170150 }, /* front speakers */ + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_AMP_INIT, + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -7405,6 +7415,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x864f, "HP Laptop 15-dy0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x86ad, "HP ENVY 13-aq1xxx", ALC285_FIXUP_HP_ENVY_13AQ), SND_PCI_QUIRK(0x103c, 0x86c1, "HP Laptop 15-da3001TU", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x86c8, "HP Laptop 14s-dr1xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), From cfc0a117d773eb585ca7e87d20c50d1c415058e5 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Wed, 23 Sep 2026 20:16:40 +0800 Subject: [PATCH 27/53] ALSA: hda/realtek: Fix square wave output on ASUS Strix G615LR ASUS ROG Strix G16 G615LR (PCI SSID 1043:3f20) outputs a loud full-scale square wave from internal speakers since the TAS2781 amplifier driver started loading. The quirk was using ALC287_FIXUP_TXNW2781_I2C which chains to the ThinkPad headset jack fixup, but this is an ASUS device that should use ALC287_FIXUP_TXNW2781_I2C_ASUS which chains to ALC294_FIXUP_ASUS_SPK for proper EAPD and pin configuration. Fixes: f7cede182c96 ("ALSA: hda/realtek: Add Asus quirk for TAS amplifiers") Cc: stable@vger.kernel.org Cc: Baojun Xu Cc: Antheas Kapenekakis Reported-by: Aleksei Arsenev Link: https://bugzilla.kernel.org/show_bug.cgi?id=222045 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260923121640.272509-1-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 004f6a3eadfe..b5cb9fa6e6d4 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7938,7 +7938,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3ef0, "ASUS Strix G635LR_LW_LX", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x1043, 0x3f00, "ASUS Strix G815LH_LM_LP", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x1043, 0x3f10, "ASUS Strix G835LR_LW_LX", ALC287_FIXUP_TXNW2781_I2C), - SND_PCI_QUIRK(0x1043, 0x3f20, "ASUS Strix G615LR_LW", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f20, "ASUS Strix G615LR_LW", ALC287_FIXUP_TXNW2781_I2C_ASUS), SND_PCI_QUIRK(0x1043, 0x3f30, "ASUS Strix G815LR_LW", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x1043, 0x3fd0, "ASUS B3605CVA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3ff0, "ASUS B5405CVA", ALC245_FIXUP_CS35L41_SPI_2), From 2a8ee6897302ca12ff39e8eb180503956797c60c Mon Sep 17 00:00:00 2001 From: Yu-Hsiang Tseng Date: Wed, 23 Sep 2026 22:13:44 +0800 Subject: [PATCH 28/53] ALSA: hda/realtek: Fix silent speaker on Higole F9B The Higole F9B is an Alder Lake-N mini PC with a 7" touchscreen and an ALC269VC codec (subsystem ID 0x10ec111e). Its DMI strings are all "Default string", so the SSID is the only way to identify it. The internal speaker is silent out of the box. The generic parser assigns DAC 0x03 to the speaker pin 0x14 and DAC 0x02 to the headphone pin 0x15, but the speaker only reproduces what DAC 0x02 plays, whatever its connection selection says. Checked one path at a time, only the gain of DAC 0x02 and the mute of pin 0x14 change what comes out of the speaker. User space only knows the path it was told the speaker uses, so selecting the speaker turns the path that actually reaches it all the way down. Route both 0x14 and 0x15 through mixer 0x0c (DAC 0x02), which is what ALC290_FIXUP_MONO_SPEAKERS already does. Tested on the device: the speaker plays. Cc: Assisted-by: LLM Signed-off-by: Yu-Hsiang Tseng Link: https://patch.msgid.link/20260923141344.1543158-1-asas1asas200@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index b5cb9fa6e6d4..502844dc4e26 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7961,6 +7961,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x111e, "Higole F9B", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x119e, "Positivo SU C1400", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x11bc, "VAIO VJFE-IL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), From 7b457ae8cf711831f19d20963f5ed2fdf4842d3a Mon Sep 17 00:00:00 2001 From: Tomasz Bojanowski Date: Wed, 23 Sep 2026 20:49:57 +0200 Subject: [PATCH 29/53] ALSA: usb-audio: Add mixer mapping for MSI MAG B850M MORTAR WIFI The USB audio device 0db0:cc78 (Realtek ALC4080) on the MSI MAG B850M MORTAR WIFI motherboard exposes all playback controls as "PCM", which makes it hard to tell the outputs apart. Its mixer layout matches the one of the MSI MPG X570S Carbon Max Wifi (units 29, 30 and 32), so reuse msi_mpg_x570s_carbon_max_wifi_alc4080_map for this device too. With the mapping applied, the controls show up as "Speaker", "Front Headphone" and "IEC958". Tested with a patched snd-usb-audio module on a 7.2.7 kernel. Signed-off-by: Tomasz Bojanowski Link: https://patch.msgid.link/20260923184957.6687-1-tomasz@bojanowski.me Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 41454cb403d0..f8e1f78039e1 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -749,6 +749,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0db0, 0xa073), .map = msi_mpg_x570s_carbon_max_wifi_alc4080_map, }, + { /* MSI MAG B850M MORTAR WIFI */ + .id = USB_ID(0x0db0, 0xcc78), + .map = msi_mpg_x570s_carbon_max_wifi_alc4080_map, + }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), .map = trx40_mobo_map, From 0b7bd20f6379bc5e431fd74d866390f5b5b63f3c Mon Sep 17 00:00:00 2001 From: Nguyen Ngoc Thang Date: Sat, 26 Sep 2026 00:07:17 +0700 Subject: [PATCH 30/53] ALSA: caiaq: unregister the input device when probe fails setup_card() registers the input device, whose name and phys point into struct snd_usb_caiaqdev, and then goes on to snd_card_register() and snd_usb_caiaq_control_init(). If either fails, snd_probe() calls snd_card_free(), which frees the device state but leaves the input device registered: card_free() only clears the pointer, and the input device is unregistered from snd_disconnect() alone. Reading its "uevent" attribute afterwards dereferences freed memory: BUG: KASAN: slab-use-after-free in string+0x4a9/0x4f0 Read of size 1 at addr ffff8880208f5043 by task caiaq/4967 add_uevent_var+0x183/0x3a0 input_dev_uevent+0x162/0x900 dev_uevent+0x2f1/0x870 uevent_show+0x1ca/0x3a0 ... Unregister the input device on the probe error path, as snd_disconnect() does. snd_usb_caiaq_input_disconnect() is a no-op when no input device was registered. Reproduced with a raw-gadget Audio Kontrol 1 and a temporary hack that makes snd_usb_caiaq_control_init() fail. Fixes: 28abd224db4a ("ALSA: caiaq: Handle probe errors properly") Cc: stable@vger.kernel.org Reported-by: syzbot+2a123f6269da57ffefaa@syzkaller.appspotmail.com Closes: https://syzkaller.appspot.com/bug?extid=2a123f6269da57ffefaa Signed-off-by: Nguyen Ngoc Thang Link: https://patch.msgid.link/20260925170717.22462-1-ngocthang2710.1999@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 3d821fde4582..f604846ec6e6 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -562,6 +562,9 @@ static int snd_probe(struct usb_interface *intf, ret = init_card(caiaqdev(card)); if (ret < 0) { dev_err(&usb_dev->dev, "unable to init card! (ret=%d)\n", ret); +#ifdef CONFIG_SND_USB_CAIAQ_INPUT + snd_usb_caiaq_input_disconnect(caiaqdev(card)); +#endif snd_card_free(card); return ret; } From ca51771c8efde5df05476ac98dd1b2af4e2b149f Mon Sep 17 00:00:00 2001 From: Zimeng Li Date: Sun, 27 Sep 2026 09:09:11 +0800 Subject: [PATCH 31/53] ASoC: qcom: lpass-cpu: use DAI table index for playback constraints The DAI ID is a hardware port ID and is not necessarily the index of the corresponding entry in variant->dai_driver. When configuring a DAI with LPAIF_I2SCTL_MODE_QUAD01, the probe code currently uses dai_id to index variant->dai_driver. This is incorrect for platforms where DAI IDs are sparse. For example, the IPQ806x MI2S DAI has ID 4 while it is the only entry in the DAI driver table. If that DAI is configured for QUAD01 and probe reaches this branch, the code writes beyond that single entry via dai_driver[4] instead of updating the DAI being processed. Use the loop index i when updating the current DAI's playback channel constraints, while retaining dai_id for indexing the hardware-port specific playback SD-line mode array. This fixes the incorrect DAI table access for platforms where the DAI ID does not match its position in the driver table. Fixes: c223f41c1a52 ("ASoC: qcom: Add four speaker support on MI2S secondary") Cc: stable@vger.kernel.org Signed-off-by: Zimeng Li Assisted-by: LLM-assisted source analysis Reviewed-by: Srinivas Kandagatla Link: https://patch.msgid.link/20260927010911.51980-1-me@lizi.moe Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 242bc16da36d..079177a6d81e 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1234,8 +1234,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) } if (drvdata->mi2s_playback_sd_mode[dai_id] == LPAIF_I2SCTL_MODE_QUAD01) { - variant->dai_driver[dai_id].playback.channels_min = 4; - variant->dai_driver[dai_id].playback.channels_max = 4; + variant->dai_driver[i].playback.channels_min = 4; + variant->dai_driver[i].playback.channels_max = 4; } } From 89365897f4210979966aa808d7b013edf4080558 Mon Sep 17 00:00:00 2001 From: Aaron Kling Date: Sat, 26 Sep 2026 02:24:19 -0500 Subject: [PATCH 32/53] ALSA: hda/tegra: Re-enable SDO workaround for Tegra194 This workaround was originally added for Tegra194, but no longer gets applied for it. Commit 615d43540043 ("ALSA: hda/tegra: fix tegra-hda on tegra30 soc") changed the guard to tegra30-hda, which worked because all affected archs had the fallback compatible. However, commit 7f0ea5acfc19 ("arm64: tegra: Use correct compatible string for Tegra194 HDA") removed the fallback compatible for Tegra194, causing this to break. Fixes: 7f0ea5acfc19 ("arm64: tegra: Use correct compatible string for Tegra194 HDA") Cc: stable@vger.kernel.org Assisted-by: LLM Signed-off-by: Aaron Kling Link: https://patch.msgid.link/20260926-tegra194-hda-sdo-v1-1-f7a636a07a58@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/controllers/tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/controllers/tegra.c b/sound/hda/controllers/tegra.c index 31c14c4bbe68..674cc0f79c3c 100644 --- a/sound/hda/controllers/tegra.c +++ b/sound/hda/controllers/tegra.c @@ -374,7 +374,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) * in powers of 2, next available ratio is 16 which can be * used as a limiting factor here. */ - if (of_device_is_compatible(np, "nvidia,tegra30-hda")) + if (of_device_is_compatible(np, "nvidia,tegra30-hda") || + of_device_is_compatible(np, "nvidia,tegra194-hda")) chip->bus.core.sdo_limit = 16; /* codec detection */ From 608c0c8947f03069b651ace54538a115f5b23123 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 17 Sep 2026 14:53:26 +0800 Subject: [PATCH 33/53] ALSA: hda/realtek - Add headset mode for Dell Pro QC1255 It lost its headset microphone functionality, and this patch will restore it. Fixes: 97272a5704bf ("ALSA: hda/realtek - Fixed Headphone noise issue for Dell QCM1255") Signed-off-by: Kailang Yang Link: https://lore.kernel.org/3c251e40a2944ae4ae479e4c2e4e63a4@realtek.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 502844dc4e26..fb3a223bf9fa 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7098,6 +7098,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC236_FIXUP_DELL_HP_POP_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_invalidate_dacs, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC274_FIXUP_HP_89E9_GPIO] = { .type = HDA_FIXUP_FUNC, From 4322ac247f4ce7932edbd46aa37f3dae62cde424 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Mon, 21 Sep 2026 20:44:49 +0800 Subject: [PATCH 34/53] ALSA: hda/realtek: Add quirk for Lenovo ThinkBook G8+ Fix Bass Speaker volume control on Lenovo ThinkBook G8+ by adding quirk 17aa:38dc with ALC285_FIXUP_SPEAKER2_TO_DAC1. Link: https://bugzilla.kernel.org/show_bug.cgi?id=221799 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260921124450.639601-1-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index fb3a223bf9fa..2b1fa3d4585b 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -8264,6 +8264,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38d5, "Yoga S990-16 Pro IMH YC Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d6, "Yoga S990-16 Pro IMH VECO Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38dc, "Lenovo ThinkBook G8", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x38df, "Yoga Y990 Intel YC Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38e0, "Yoga Y990 Intel VECO Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f8, "Yoga Book 9i", ALC287_FIXUP_TAS2781_I2C), From 0d90a929836344f1d3b39f22f08ccecc62378a59 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Mon, 21 Sep 2026 20:44:50 +0800 Subject: [PATCH 35/53] ALSA: hda/realtek: Add quirk for Acer Nitro AN515-45 The Acer Nitro AN515-45 requires ALC2XX_FIXUP_HEADSET_MIC to make the headset microphone work properly. Link: https://bugzilla.kernel.org/show_bug.cgi?id=221925 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260921124450.639601-2-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 2b1fa3d4585b..4bd37a15206e 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7203,6 +7203,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), + SND_PCI_QUIRK(0x1025, 0x151e, "Acer Nitro AN515-45", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1539, "Acer Nitro 5 AN515-57", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC287_FIXUP_ACER_MICMUTE_LED), From c748c42abb9b549fc79c41ea55e1f220eadb60ec Mon Sep 17 00:00:00 2001 From: Krish Gulati Date: Mon, 21 Sep 2026 18:47:03 +0530 Subject: [PATCH 36/53] ALSA: hda/realtek: Add mute LED quirk for HP Victus 15-fb2xxx The HP Victus 15-fb2xxx (PCI SSID 103c:8c2f) with the ALC245 codec does not turn on the mute LED when audio is muted. Add a quirk entry to enable it. Reported-by: erph.briones27@gmail.com Link: https://bugzilla.kernel.org/show_bug.cgi?id=221946 Signed-off-by: Krish Gulati Tested-by: Raphael Briones Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20260921131728.8999-1-krishgulati7@gmail.com --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 4bd37a15206e..1db335b18c4e 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7628,6 +7628,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x8c2d, "HP Victus 15-fa1xxx (MB 8C2D)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x8c2f, "HP Victus 15-fb2xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8c3f, "HP Victus 15-fa1xxx (MB 8C3F)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), From c26d18964a9352e83cb5fdd7b327d9a2566abddf Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 28 Sep 2026 11:56:10 +0100 Subject: [PATCH 37/53] ASoC: Intel: Add quirk to block match table for Lenovo Yoga Slim 7 A variant of the Lenovo Yoga Slim 7 exists on Lunar Lake that doesn't have a physical jack socket. Cirrus codecs still use match tables on LNL but the match table makes no distinction between a system with or without a jack. This leads to the system failing probe as the jack in the topology file is not matched by a DAI link as the machine driver is constructed. Add a new quirk to block using match tables for specific devices, and rely on function topologies which can handle this difference seamlessly. This is slightly preferred over moving everything to function topologies due to the lower chance of causing regressions on other systems. Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260928105610.242687-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- .../soc/intel/common/soc-acpi-intel-lnl-match.c | 1 + .../intel/common/soc-acpi-intel-sdca-quirks.c | 16 ++++++++++++++++ .../intel/common/soc-acpi-intel-sdca-quirks.h | 1 + 3 files changed, 18 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 3b9758f73238..62b8811395cb 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -745,6 +745,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .links = lnl_cs42l43_l0, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-cs42l43-l0.tplg", + .machine_check = snd_soc_acpi_intel_force_function_topology, .get_function_tplg_files = sof_sdw_get_tplg_files, }, { diff --git a/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.c b/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.c index 9fdeb6ad96ac..e62ef81214ff 100644 --- a/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.c +++ b/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.c @@ -53,6 +53,22 @@ bool snd_soc_acpi_intel_no_function_topology(void *arg) } EXPORT_SYMBOL_NS(snd_soc_acpi_intel_no_function_topology, "SND_SOC_ACPI_INTEL_SDCA_QUIRKS"); +static const struct dmi_system_id force_function_topology_quirk_table[] = { + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83R0"), + }, + }, + {} +}; + +bool snd_soc_acpi_intel_force_function_topology(void *arg) +{ + return !dmi_check_system(force_function_topology_quirk_table); +} +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_force_function_topology, "SND_SOC_ACPI_INTEL_SDCA_QUIRKS"); + bool snd_soc_acpi_intel_rt712_vb_no_function_topology(void *arg) { return snd_soc_acpi_intel_sdca_is_device_rt712_vb(arg) && diff --git a/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.h b/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.h index 60b665ab5924..10179004c86a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.h +++ b/sound/soc/intel/common/soc-acpi-intel-sdca-quirks.h @@ -11,6 +11,7 @@ bool snd_soc_acpi_intel_sdca_is_device_rt712_vb(void *arg); bool snd_soc_acpi_intel_no_function_topology(void *arg); +bool snd_soc_acpi_intel_force_function_topology(void *arg); bool snd_soc_acpi_intel_rt712_vb_no_function_topology(void *arg); #endif From f090e7ff26fe08be4cd57e0950872ede4af5351a Mon Sep 17 00:00:00 2001 From: Bartosz Brom Date: Mon, 28 Sep 2026 15:43:35 +0200 Subject: [PATCH 38/53] ALSA: hda/realtek: Add mute LED quirks for HP Pavilion 15-eh3174nw and HP Laptop 15-dw1xxx The HP Pavilion 15-eh3174nw with an ALC287 codec requires the ALC287_FIXUP_HP_GPIO_LED fixup for its mute LED to work correctly. Add subsystem ID 0x103c:0x8bc7 to the quirk table to apply the existing fixup. Tested on an HP Pavilion 15-eh3174nw; the mute LED now follows the speaker mute state. The HP Laptop 15-dw1xxx with an ALC236 codec requires the ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO fixup for its mute LED to work correctly. Add subsystem ID 0x103c:0x85f1 to the quirk table to apply the existing fixup. Tested on an HP Laptop 15-dw1xxx; the mute LED now follows the speaker mute state. Signed-off-by: Bartosz Brom Link: https://patch.msgid.link/20260928134335.1875-1-bartosz.brom06@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 1db335b18c4e..193e0fa3606d 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7411,6 +7411,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11), SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), SND_PCI_QUIRK(0x103c, 0x85f0, "HP Laptop 15-dw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x85f1, "HP Laptop 15-dw1xxx", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x85f3, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8603, "HP Omen 17-cb0xxx", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x860c, "HP ZBook 17 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), @@ -7606,6 +7607,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bb6, "HP Laptop 15-fd0039nt", ALC236_FIXUP_HP_15_FD0XXX), SND_PCI_QUIRK(0x103c, 0x8bbe, "HP Victus 16-r0xxx (MB 8BBE)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x8bc7, "HP Pavilion 15-eh3174nw", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bc8, "HP Victus 15-fa1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8bcd, "HP Omen 16-xd0xxx", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8bd4, "HP Victus 16-s0xxx (MB 8BD4)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), From de0be324fcb97a49caad131c23ad33265dd4b0b8 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Mon, 28 Sep 2026 16:39:25 +0800 Subject: [PATCH 39/53] ASoC: amd: acp-config: Add ACP70 DMI quirk for ASUS UM5406GA Add ASUS Zenbook S14 UM5406GA (Strix Point, ACP 7.0) to acp70_acpi_flag_override_table so that the broken BIOS acp-audio-config-flag is ignored and the default SoundWire enumeration path is used. Without this, no ASoC machine driver matches and internal speakers/mic fail to probe. Reported-by: Jannick Tobler Closes: https://bugzilla.kernel.org/show_bug.cgi?id=222009 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260928083925.266471-1-zhangheng@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 4d8f86aa3d16..b3b73096ac74 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -70,6 +70,13 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Vivobook 18 M1807GA"), }, }, + { + /* ASUS Zenbook S14 UM5406GA (Strix Point, ACP 7.0) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "Zenbook S14 UM5406GA"), + }, + }, { /* HP OmniBook X Flip 14-kc0xxx (Strix Point, ACP 7.2) */ .matches = { From 27a868b40e7d18e4be57d2bbbeac7581fd1219db Mon Sep 17 00:00:00 2001 From: Preston Lam Date: Mon, 28 Sep 2026 19:16:21 -0400 Subject: [PATCH 40/53] ALSA: hda/realtek: Add mute LED quirk for HP Laptop 15-fd2xxx The HP Laptop 15-fd2xxx (PCI SSID 103c:8e43, Realtek ALC236) has a speaker mute LED on the F5 key and a microphone mute LED on the F8 key. Neither lights up, because the model has no quirk entry and only the generic HP vendor fallback (ALC269_FIXUP_HP_MUTE_LED) is applied. Probing the codec with hda-verb showed how the LEDs are wired: - mute LED: COEF index 0x07, bit 0, on NID 0x20 (set = LED on) - mic mute LED: codec GPIO0, active low (driven low = LED on) Driving GPIO1 and GPIO2 did not light the mute LED. This wiring is exactly what the existing ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO fixup implements (COEF 0x07 bit 0 for the mute LED, GPIO0 with inverted polarity for the micmute LED), so use it for this model. The sibling HP Laptop 15-fd0xxx (103c:8dd7) already uses the same fixup. Tested on this machine running Ubuntu 7.0.0-34-generic (SOF driver, the quirk built into snd-hda-codec-alc269 from Ubuntu's 7.0.0-34.34 source tree): the hda::mute and hda::micmute LEDs are created, follow the Master mute state and the Dmic0 capture switch respectively, and the F5 and F8 keys toggle them correctly. Muting the microphone also silences the capture stream. The LEDs also kept the correct state across an s2idle suspend/resume cycle (speaker muted in one run, microphone in another) and still responded to mute changes afterwards; long suspends were not tested. The desktop session was SwayFX (wlroots) with PipeWire, which only changes the mute state through the normal mixer controls; no other desktop was tried. Not tested on a mainline kernel tree directly; the change is only the quirk table entry. AI assistance: this patch, the hardware analysis behind it (probing the codec with hda-verb and choosing the existing fixup) and this commit message were written by Claude Code (Anthropic, model claude-sonnet-5-5). The submitter ran the privileged commands and reboots and reported the LED behavior; Claude Code ran the tests and read the kernel logs. Assisted-by: Claude:claude-sonnet-5-5 Signed-off-by: Preston Lam Link: https://patch.msgid.link/20260928231621.1672704-1-plamlam99@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 193e0fa3606d..dfb5b5ab899d 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7740,6 +7740,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e37, "HP 16 Piston OmniBook X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e3a, "HP Agusta", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e3b, "HP Agusta", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8e43, "HP Laptop 15-fd2xxx", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8e60, "HP OmniBook 7 Laptop 16-bh0xxx", ALC245_FIXUP_CS35L41_I2C_2_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8e61, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e62, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), From 763183f76d6bc093fa603377862d6739cb7ccad6 Mon Sep 17 00:00:00 2001 From: Zhang Heng Date: Tue, 29 Sep 2026 17:11:09 +0800 Subject: [PATCH 41/53] ALSA: hda/realtek: Add quirk for HP Pavilion AiO 24 speaker Add ALC225_FIXUP_HP_AIO_SPK to enable the internal speaker on HP Pavilion AiO 24 (103c:85ba). Link: https://bugzilla.kernel.org/show_bug.cgi?id=221691 Signed-off-by: Zhang Heng Link: https://patch.msgid.link/20260929091109.400313-1-zhangheng@kylinos.cn Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index dfb5b5ab899d..d13cd276ec61 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4200,6 +4200,7 @@ enum { ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE, ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, + ALC225_FIXUP_HP_AIO_SPK, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -5721,6 +5722,14 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC225_FIXUP_HP_AIO_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0220 }, + { } + }, + }, [ALC286_FIXUP_ACER_AIO_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7408,6 +7417,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x854a, "HP EliteBook 830 G6", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x854d, "HP EliteBook 840 G6", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x856a, "HP Pavilion 15-cs1xxx", ALC295_FIXUP_HP_PAVILION_MUTE_LED_1B), + HDA_CODEC_QUIRK(0x103c, 0x85ba, "HP Pavilion AiO 24", ALC225_FIXUP_HP_AIO_SPK), SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11), SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), SND_PCI_QUIRK(0x103c, 0x85f0, "HP Laptop 15-dw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), From 0eee030c2c09c7a44be778de5f451d9e7a2f09c1 Mon Sep 17 00:00:00 2001 From: Muhtasham Nawr al-Mahmud Date: Thu, 24 Sep 2026 20:45:12 +0600 Subject: [PATCH 42/53] ALSA: hda/realtek: Fix speakers on ASUS PM3406CHA The ASUS ExpertBook PM3406CHA uses an ALC256 codec with subsystem ID 1043:1664. On Linux boot, the internal speakers are silent even though playback is routed to the codec and the speaker pin is configured for output. The codec starts with coefficient 0x10 set to 0x1d20. Writing 0x7f20 to coefficient 0x10 immediately restores internal speaker output. This was reproduced on the affected hardware from a fresh boot using hda-verb. Reducing a larger initialization sequence showed that this coefficient write alone is sufficient; no GPIO, EAPD, pin-control, or other coefficient changes are required. After applying the coefficient in userspace, speaker output also remains functional across suspend and resume. Add a machine-specific HDA fixup that writes the required coefficient. The fixup has been compile-tested, but could not be boot-tested because the affected hardware is no longer available. Closes: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/2162718 Signed-off-by: Muhtasham Nawr al-Mahmud Link: https://patch.msgid.link/20260924144512.3989-1-muhtaseem2005@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index d13cd276ec61..333689f31775 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -4025,6 +4025,7 @@ enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, + ALC256_FIXUP_ASUS_PM3406CHA, ALC269_FIXUP_SKU_IGNORE, ALC269_FIXUP_ASUS_G73JW, ALC269_FIXUP_ASUS_N7601ZM_PINS, @@ -4484,6 +4485,14 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC256_FIXUP_ASUS_PM3406CHA] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7f20 }, + {} + } + }, [ALC269_FIXUP_SKU_IGNORE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_sku_ignore, @@ -7853,6 +7862,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1652, "ASUS ROG Zephyrus Do 15 SE", ALC289_FIXUP_ASUS_ZEPHYRUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1664, "ASUS ExpertBook PM3406CHA", ALC256_FIXUP_ASUS_PM3406CHA), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS UX3402VA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), From b2047b8cadadccb1c9269ce756c399cd9aef2c82 Mon Sep 17 00:00:00 2001 From: Liviu Nicoara Date: Tue, 29 Sep 2026 07:57:55 -0400 Subject: [PATCH 43/53] ASoC: codecs: lpass-wsa-macro: rewrite the interpolator volume after enabling clocks Commit 902f497a1ff5 ("ASoC: codecs: lpass-wsa-macro: remove useless gain read/write sequence") removed the read and write of the digital volume register in wsa_macro_enable_interpolator(), on the grounds that writing back the value just read does nothing. The comment above it, "apply gain after int clk is enabled", was left in place. On the Dell XPS 13 9345 (X1E80100, four WSA8845 amplifiers on two WSA macros) the write does something: a volume change made while the path is idle does not take effect when playback starts. Lowering the digital volume from 81 to 63 with nothing playing, then playing a test tone, gave about the same level as before the change. With the rewrite restored, lowering it from 81 to 69 while idle played audibly quieter, and restoring 81 while idle brought the level back. Changes made during playback take effect with or without the rewrite. The register already holds the new value when this happens. Without the rewrite, after an idle change from 69 to 81, playback stayed at the old level while the register, read from the hardware through /dev/mem, held the new one (0xfd). Writing that same value back through /dev/mem brought the level up at once. This is the behaviour described in commit 46188db080bd ("ASoC: codecs: lpass-wsa-macro: fix compander volume hack"): "the volume registers still need to be written after enabling clocks in order for any prior updates to take effect." The value read comes from the register cache, so the write pushes the last requested volume to the hardware once its clock runs. Restore the rewrite in the interpolator's POST_PMU event only. The mix path event removed later in the same series is not brought back. Fixes: 902f497a1ff5 ("ASoC: codecs: lpass-wsa-macro: remove useless gain read/write sequence") Cc: stable@vger.kernel.org Assisted-by: Claude:claude-opus-5-5 Signed-off-by: Liviu Nicoara Reviewed-by: Johan Hovold Link: https://patch.msgid.link/20260929115755.6096-1-lnicoara@thinkoid.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-wsa-macro.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index cfd2ac0a6cda..38e37908baf9 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1841,6 +1841,7 @@ static int wsa_macro_enable_interpolator(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + unsigned int gain; u16 gain_reg; u16 reg; struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); @@ -1882,6 +1883,8 @@ static int wsa_macro_enable_interpolator(struct snd_soc_dapm_widget *w, CDC_WSA_RX_PGA_HALF_DB_MASK, CDC_WSA_RX_PGA_HALF_DB_ENABLE); } + gain = snd_soc_component_read(component, gain_reg); + snd_soc_component_write(component, gain_reg, gain); wsa_macro_config_ear_spkr_gain(component, wsa, event, gain_reg); break; From b5c4823cf3e49b3419a3bc8c6715101e1ac6cc59 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Sep 2026 14:29:25 +0200 Subject: [PATCH 44/53] ALSA: usb-audio: Apply boot quirk for Behringer models generically Jan reported that a few Behringer devices became broken since the recent optimization to avoid usb_string() call at probe time at the commit b364a0d23cae ("ALSA: usb-audio: Use strings in struct usb_dev for manufacturer & co"). Interestingly, the devices seem requiring the explicit descriptor read at probing time, and the optimization above dropped it. There is already a boot quirk for another model, Behringer CM1A (1397:1234), that adds a device descriptor read, and this seems working for them, too. As the quirk is safe and cheap, just apply the same boot quirk to all Behringer devices for avoiding the pitfall again. Fixes: b364a0d23cae ("ALSA: usb-audio: Use strings in struct usb_dev for manufacturer & co") Reported-by: Jan Lentfer Closes: https://lore.kernel.org/e7087d42-5e74-4d85-b1c5-b11eff235d41@web.de Link: https://patch.msgid.link/20260929122938.1471867-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 294c7026b93c..f55b02d6dd39 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1702,10 +1702,12 @@ int snd_usb_apply_boot_quirk_once(struct usb_device *dev, switch (id) { case USB_ID(0x07fd, 0x0008): /* MOTU M Series, 1st hardware version */ return snd_usb_motu_m_series_boot_quirk(dev); - case USB_ID(0x1397, 0x1234): /* Behringer CM1A */ - return snd_usb_cm1a_boot_quirk(dev); } + /* Behringer devices may need explicit device descriptor read at boot */ + if (USB_ID_VENDOR(id) == 0x1397) + return snd_usb_cm1a_boot_quirk(dev); + return 0; } From a154f7b9f197b3d5e41fa3ad62083f5aa93b1fe8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Sep 2026 16:41:26 +0200 Subject: [PATCH 45/53] ALSA: usb-audio: Apply IGNORE_CTL_ERROR quirk to all Audient devices Faaris reported a problem with Audient EVO4 device and it turned out to be a regression by the recent fix commit 87a6f2fa6e6c ("ALSA: usb-audio: Propagate write errors in generic mixer put callbacks"). It exhibited that the hardware gives errors at accessing the mixer unit 10 on certain channels, and the change above made it a fatal error. We had already a quirk for Audient iD14 to work around such errors from the mixer controls, and we can simply apply the same for EVO4. OTOH, it's highly possible that other Audient devices suffer from the same issue; they must be using similar firmware, after all. So, in this patch, we apply the quirk generically to all devices with the vendor ID Audient (2708), instead. Ignoring the control error isn't usually less critical than overreaction to the firmware misbehavior. Fixes: 87a6f2fa6e6c ("ALSA: usb-audio: Propagate write errors in generic mixer put callbacks") Reported-and-tested-by: Faaris Ansari Closes: https://lore.kernel.org/CANBVYRCL=8QdLxGg4S6qrahrFtwJxhv-aSGpW7-1S=+iOe4ZGA@mail.gmail.com Cc: Link: https://patch.msgid.link/20260929144132.1521617-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f55b02d6dd39..3afc49565963 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2516,8 +2516,6 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x262a, 0x9302, /* ddHiFi TC44C */ QUIRK_FLAG_DSD_RAW), - DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ - QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2772, 0x0502, /* Musical Fidelity M6s DAC */ 0), /* for avoiding QUIRK_FLAG_DSD_RAW with vendor match */ DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ @@ -2616,6 +2614,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2622, /* IAG Limited devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x2708, /* Audient devices */ + QUIRK_FLAG_IGNORE_CTL_ERROR), VENDOR_FLG(0x2772, /* Musical Fidelity devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x278b, /* Rotel? */ From 4fa0c91c2c4c604e6080690a698d4c6405ba96ca Mon Sep 17 00:00:00 2001 From: Harin Lee Date: Wed, 30 Sep 2026 20:58:24 +0900 Subject: [PATCH 46/53] ALSA: ctxfi: Fix silent playback on Titanium HD (SB1270) Commit 4b490e0d103c ("ALSA: ctxfi: Add ADC helper functions for GPIO") moved the GPIO_DATA read into hw_adc_stop(), but the Titanium HD (SB1270) setup in hw_adc_init() uses the local data variable holding the GPIO_CTRL value. Writing this value to GPIO_DATA sets the output mute bits. Playback runs without errors, but the outputs stay muted. Re-read GPIO_DATA before setting the speed mode bits. Also, add the delay after writing the mode bits, as in the original sequence. On affected kernels, switching the output selection was reported to restore clean playback. Fixes: 4b490e0d103c ("ALSA: ctxfi: Add ADC helper functions for GPIO") Cc: stable@vger.kernel.org Reported-by: Dany Patoine Closes: https://lore.kernel.org/linux-sound/DM6PR02MB5433A42E81FEFAC400DD1905EA8B2@DM6PR02MB5433.namprd02.prod.outlook.com/ Signed-off-by: Harin Lee Link: https://patch.msgid.link/20260930115824.739412-1-me@harin.net Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 36066ffe8784..26a6fa69f5f5 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1895,6 +1895,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) if (hw->model == CTSB1270) { /* Set up the PCM4220 ADC on Titanium HD */ + data = hw_read_20kx(hw, GPIO_DATA); data &= ~0x0C; if (1 == info->msr) data |= 0x00; /* Single Speed Mode 32-50kHz */ @@ -1903,6 +1904,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) else data |= 0x04; /* Quad Speed Mode 108kHz-216kHz */ hw_write_20kx(hw, GPIO_DATA, data); + usleep_range(10000, 11000); } hw_adc_start(hw); From ece609abc95cc0f75e633d93f3d868b030d9a8fe Mon Sep 17 00:00:00 2001 From: renpanpan Date: Wed, 30 Sep 2026 14:55:04 +0800 Subject: [PATCH 47/53] ALSA: hda/realtek: Add mute LED quirk for HP Pavilion 14-dv1001TU The HP Pavilion 14-dv1001TU (subsystem ID 103c:8987) uses a Realtek ALC287 codec. The mute LED on this machine is controlled by GPIO bit 0x10, which was verified via hda-verb, but the subsystem ID is missing from the quirk table. As a result, the ALC287_FIXUP_HP_GPIO_LED fixup that drives GPIO 0x10 is never applied and the mute LED stays off even though the associated LED class device toggles correctly. Add the missing SND_PCI_QUIRK entry to map this subsystem ID to ALC287_FIXUP_HP_GPIO_LED so the mute LED works as expected. Link: https://bugzilla.kernel.org/show_bug.cgi?id=221810 Signed-off-by: renpanpan Link: https://patch.msgid.link/20260930065504.24151-1-rppxj178543@163.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 333689f31775..566699482fb6 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7528,6 +7528,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x897d, "HP mt440 Mobile Thin Client U74", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8981, "HP Elite Dragonfly G3", ALC245_FIXUP_CS35L41_SPI_4), + SND_PCI_QUIRK(0x103c, 0x8987, "HP Pavilion 14-dv1001TU", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x898a, "HP Pavilion 15-eg100", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x898e, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x898f, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), From 97384ef06c4fa79f7265ecd65c240bf40a8a9c41 Mon Sep 17 00:00:00 2001 From: Harrison Lucas Date: Tue, 29 Sep 2026 11:43:48 -0600 Subject: [PATCH 48/53] ALSA: hda/realtek: Fix mute and micmute LEDs on HP ENVY x360 15-ey0xxx The HP ENVY x360 2-in-1 Laptop 15-ey0xxx (SSID 103c:8a31, ALC245 with CS35L41 I2C amps) currently only gets ALC287_FIXUP_CS35L41_I2C_2, so the mute and micmute LEDs on the keyboard never light up. On this machine the mute LED is driven by COEF 0x0b bits 2-3 and the micmute LED by GPIO 0x04 (active low), which is exactly what ALC245_FIXUP_HP_X360_MUTE_LEDS sets up. Switch the quirk to ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS, which keeps the CS35L41 setup and chains to that LED fixup. Verified by driving the COEF and GPIO values by hand with hda-verb, following Master Playback Switch and Mic ACP LED Capture Switch. Assisted-by: Claude:claude-opus-5-5 Signed-off-by: Harrison Lucas Link: https://patch.msgid.link/20260929174348.167537-1-tiiowsalt@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 566699482fb6..fd76d8b503a9 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7566,7 +7566,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a31, "HP ENVY x360 2-in-1 Laptop 15-ey0xxx", ALC287_FIXUP_CS35L41_I2C_2_HP_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x8a34, "HP Pavilion x360 2-in-1 Laptop 14-ek0xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a36, "HP Pavilion Plus 14-eh0xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a3d, "HP Victus 15-fb0xxx (MB 8A3D)", ALC245_FIXUP_HP_MUTE_LED_V2_COEFBIT), From 819f6aacbb1d1621c6a62721e708835fe1fb75d8 Mon Sep 17 00:00:00 2001 From: Jakub Gawlik Date: Wed, 30 Sep 2026 16:58:39 +0200 Subject: [PATCH 49/53] ALSA: hda/realtek: Add micmute LED quirk for Acer Swift SFG16-72 The Acer Swift SFG16-72 (1025:173b) uses the ALC245 codec, but its mic-mute LED is not enabled by default. Apply the existing ALC245_FIXUP_ACER_MICMUTE_LED quirk to enable the mic-mute LED. Assisted-by: LLM Cc: stable@vger.kernel.org Signed-off-by: Jakub Gawlik Link: https://patch.msgid.link/20260930145839.374762-1-jakubgawlik13@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index fd76d8b503a9..be5482530aa2 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7238,6 +7238,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x172c, "Acer Nitro ANV16-41", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1731, "Acer Predator PHN16-72", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x173a, "Acer Swift SFG14-73", ALC245_FIXUP_ACER_MICMUTE_LED), + SND_PCI_QUIRK(0x1025, 0x173b, "Acer Swift SFG16-72", ALC245_FIXUP_ACER_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1758, "Acer Nitro ANV15-41", ALC245_FIXUP_ACER_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1826, "Acer Helios ZPC", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), SND_PCI_QUIRK(0x1025, 0x182c, "Acer Helios ZPD", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), From 9ee4266827cfb67b9fbb89136ea6a8f3d7809f5c Mon Sep 17 00:00:00 2001 From: Zeyu Li <13776528859@163.com> Date: Sat, 26 Sep 2026 22:03:36 +0800 Subject: [PATCH 50/53] ALSA: hda/realtek: Add ALC235 support for headset mode The ALC235 (0x10ec0235) is registered as the ALC255 codec variant in patch_alc269(), but the headset mode functions were missing a case for it, so the CTIA/OMTP jack type detection and routing never ran on this codec. As a result, a plugged-in headset is detected (jack presence works) but the microphone signal is never routed, leaving the capture stream permanently silent. Verified on an ASUS M5451GA laptop (HDA:10ec0235,10431814): with the ALC255 coef sequence applied, the headset microphone works correctly. Signed-off-by: Zeyu Li <13776528859@163.com> Link: https://patch.msgid.link/20260926140336.27896-1-13776528859@163.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/hda/codecs/realtek/realtek.c b/sound/hda/codecs/realtek/realtek.c index db365a746b1a..bc5226fad52b 100644 --- a/sound/hda/codecs/realtek/realtek.c +++ b/sound/hda/codecs/realtek/realtek.c @@ -1292,6 +1292,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) } switch (codec->core.vendor_id) { + case 0x10ec0235: case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -1407,6 +1408,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, }; switch (codec->core.vendor_id) { + case 0x10ec0235: case 0x10ec0255: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -1562,6 +1564,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) alc_process_coef_fw(codec, coef0225); alc_hp_enable_unmute(codec, 75); break; + case 0x10ec0235: case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -1663,6 +1666,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0235: case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -1780,6 +1784,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0235: case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; @@ -1881,6 +1886,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) } switch (codec->core.vendor_id) { + case 0x10ec0235: case 0x10ec0255: alc_process_coef_fw(codec, coef0255); msleep(300); From f1d565dc92955f05383794e0fa27e61ee0555027 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2026 17:10:35 +0200 Subject: [PATCH 51/53] ALSA: core: Define auto-cleanup for snd_card_free() For the errors at the probe time, we'd need to call rather snd_card_free() instead of the snd_card_unref() -- the former calls explicitly snd_card_disconnect() that cleans up the registered devices, etc, while the latter may leak in corner cases. For convenience, define a new auto-clean with snd_card_free. Link: https://patch.msgid.link/20261001151039.592033-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/core.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 2ca24ac7e37f..15be947a730e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -340,6 +340,15 @@ static inline void snd_card_unref(struct snd_card *card) DEFINE_FREE(snd_card_unref, struct snd_card *, if (_T) snd_card_unref(_T)) +/* + * For automatic error handling at probe time, assign the card like: + * struct snd_card *card __free(snd_card_free) = NULL; + * then create and process as usual. + * But, don't forget to clear to NULL at successful return for avoiding + * the unexpected cleanup! + */ +DEFINE_FREE(snd_card_free, struct snd_card *, if (_T) snd_card_free(_T)) + #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) /* device.c */ From e6229c0402ea4863ec74802d79d5b52b9398d66c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2026 17:10:36 +0200 Subject: [PATCH 52/53] ALSA: ice1712: Fix the error handling via auto-cleanup at probe We used the auto-cleanup via snd_card_unref() for errors at probe of ice1712 driver, but this may be problematic in a subtle way when an error happens at snd_card_register(); since snd_card_unref() skips the snd_card_disconnect() call, it may miss some resource clearance. For fixing the issue, use the new __free(snd_card_free) instead, which does call snd_card_free() explicitly at errors for avoiding such a pitfall. Fixes: d736eba9c453 ("ALSA: ice1712: Fix the card leak at probe error with the auto-cleanup") Link: https://patch.msgid.link/20261001151039.592033-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d6bfd8fb20e6..1fad8c418132 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2523,7 +2523,7 @@ static int snd_ice1712_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { static int dev; - struct snd_card *card __free(snd_card_unref) = NULL; + struct snd_card *card __free(snd_card_free) = NULL; struct snd_ice1712 *ice; int pcm_dev = 0, err; const struct snd_ice1712_card_info * const *tbl, *c; From cad16d850e3759dd82cd869f739b56e9844a1fee Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 1 Oct 2026 09:58:30 +0100 Subject: [PATCH 53/53] spi: cs42l43: Workaround for wrong speaker ID on Dell XPS 13 DX13260 On Dell XPS 13 DX13260 create an acpi_gpio_mapping with exactly two GPIO entries to point at the two pins in the GpioIo(). Use this to read the speaker ID GPIOs. This fixes problems on Dell XPS 13 DX13260: - No speaker audio - The wrong firmware was loaded so the speaker protection did not match the speaker characteristics. The Dell XPS 13 DX13260 has two speaker ID GPIOs, to form a 2-bit ID. The ACPI GpioIo() has both pins but the Linux-specific spk-id-gpios property only has a mapping to the first pin. This meant that the speaker ID was wrong in most cases, and that would lead to the codec driver loading the wrong amp firmware, or not finding a firmware (as 0 is not a valid ID on this laptop). Assisted-by: Codex:gpt-6-sol Reported-by: Wiza Jalakasi Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221956 Tested-by: Wiza Jalakasi Fixes: f3c605147741e ("spi: cs42l43: Add GPIO speaker id support to the bridge configuration") Signed-off-by: Richard Fitzgerald Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20261001085830.4014291-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/spi/spi-cs42l43.c | 97 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 97 insertions(+) diff --git a/drivers/spi/spi-cs42l43.c b/drivers/spi/spi-cs42l43.c index 7106a8a4f280..6772abb483dd 100644 --- a/drivers/spi/spi-cs42l43.c +++ b/drivers/spi/spi-cs42l43.c @@ -11,6 +11,7 @@ #include #include #include +#include #include #include #include @@ -215,12 +216,108 @@ static size_t cs42l43_spi_max_length(struct spi_device *spi) return CS42L43_SPI_MAX_LENGTH; } +/* + * Workaround needed for two speaker ID pins in one ACPI GpioIo() but + * the Linux-specific _DSD property only contains one pin. + * Create a temporary acpi_gpio_mapping pointing at both pins. + */ +static const struct acpi_gpio_params cs42l43_2bit_speaker_id_from_one_gpioio_params[] = { + [0] = { + .crs_entry_index = 0, + .line_index = 0, + }, + [1] = { + .crs_entry_index = 0, + .line_index = 1, + }, +}; + +static const struct acpi_gpio_mapping cs42l43_2bit_speaker_id_from_one_gpioio_mapping[] = { + { + .name = "spk-id-quirk-gpios", + .data = cs42l43_2bit_speaker_id_from_one_gpioio_params, + .size = ARRAY_SIZE(cs42l43_2bit_speaker_id_from_one_gpioio_params), + }, + { } +}; + +static int cs42l43_get_2bit_speaker_id_from_one_gpioio(struct cs42l43_spi *priv, int *result) +{ + struct fwnode_reference_args args; + struct acpi_device *adev; + struct gpio_desc *desc; + u32 spkid = 0; + int i, ret; + + /* Use the _DSD property to get the node containing the GpioIo() */ + ret = fwnode_property_get_reference_args(dev_fwnode(priv->dev), "spk-id-gpios", + NULL, 3, 0, &args); + if (ret) + return ret; + + struct fwnode_handle *fwnode __free(fwnode_handle) = args.fwnode; + + /* An acpi_gpio_mapping must be added to the node that contains the GpioIo() */ + adev = to_acpi_device_node(fwnode); + if (!adev) + return -EINVAL; + + ret = acpi_dev_add_driver_gpios(adev, cs42l43_2bit_speaker_id_from_one_gpioio_mapping); + if (ret) + return ret; + + /* gpiod_get_array() can't read from a mapping in a child node */ + for (i = 0; i < ARRAY_SIZE(cs42l43_2bit_speaker_id_from_one_gpioio_params); i++) { + desc = fwnode_gpiod_get_index(fwnode, "spk-id-quirk", i, GPIOD_IN, + dev_name(priv->dev)); + if (IS_ERR(desc)) { + ret = PTR_ERR(desc); + goto out; + } + + ret = gpiod_get_value_cansleep(desc); + gpiod_put(desc); + if (ret < 0) + goto out; + + spkid |= (u32)ret << i; + } + + dev_dbg(priv->dev, "spk-id-gpios = %u\n", spkid); + *result = spkid; + ret = 0; +out: + acpi_dev_remove_driver_gpios(adev); + + return ret; +} + +static const struct dmi_system_id cs42l43_spk_id_quirks[] = { + { + .ident = "Dell XPS 13 DX13260", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0E53"), + }, + .driver_data = cs42l43_get_2bit_speaker_id_from_one_gpioio, + }, + { } +}; + static int cs42l43_get_speaker_id_gpios(struct cs42l43_spi *priv, int *result) { + const struct dmi_system_id *dmi_id; struct gpio_descs *descs; u32 spkid; int i, ret; + dmi_id = dmi_first_match(cs42l43_spk_id_quirks); + if (dmi_id) { + int (*get_speaker_id)(struct cs42l43_spi *priv, int *result) = dmi_id->driver_data; + + return get_speaker_id(priv, result); + } + descs = gpiod_get_array_optional(priv->dev, "spk-id", GPIOD_IN); if (!descs) return 0;