From aa2f4addab44407c7aa742321de5dc1914ab5762 Mon Sep 17 00:00:00 2001 From: Rong Zhang Date: Sun, 31 May 2026 03:52:49 +0800 Subject: [PATCH 1/9] ALSA: usb-audio: Set the value of potential sticky mixers to maximum It makes no sense to restore the saved value for a sticky mixer, since setting any value is a no-op. However, in some rare cases, SET_CUR is effective despite GET_CUR always returns a constant value. These mixers are not sticky, but there's no way to distinguish them. Without any additional information, the best thing we can do is to set the mixer value to the maximum before bailing out, so that a soft mixer can still reach the maximum hardware volume if the mixer turns out to be non-sticky. Meanwhile, all channels must be synchronized to prevent imbalance volume. Fixes: 86aa1ea1f15c ("ALSA: usb-audio: Do not expose sticky mixers") Signed-off-by: Rong Zhang Link: https://patch.msgid.link/20260531-uac-sticky-error-path-v1-1-12c2329d17ef@rong.moe Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5fba456eb4a9..fb37bb8ad9a9 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1371,10 +1371,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, goto no_checks; ret = check_sticky_volume_control(cval, minchn, saved); - if (ret < 0) { - snd_usb_set_cur_mix_value(cval, minchn, 0, saved); - return ret; - } + if (ret < 0) + goto sticky; if (cval->min + cval->res < cval->max) check_volume_control_res(cval, minchn, saved); @@ -1431,6 +1429,33 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, } return 0; + +sticky: + /* + * It makes no sense to restore the saved value for a sticky mixer, + * since setting any value is a no-op. + * + * However, in some rare cases, SET_CUR is effective despite GET_CUR + * always returns a constant value. These mixers are not sticky, but + * there's no way to distinguish them. Without any additional + * information, the best thing we can do is to set the mixer value to + * the maximum before bailing out, so that a soft mixer can still reach + * the maximum hardware volume if the mixer turns out to be non-sticky. + * Meanwhile, all channels must be synchronized to prevent imbalance + * volume. + */ + if (!cval->cmask) { + snd_usb_set_cur_mix_value(cval, 0, 0, cval->max); + } else { + for (i = 0; i < MAX_CHANNELS; i++) { + idx = 0; + if (cval->cmask & BIT(i)) { + snd_usb_set_cur_mix_value(cval, i + 1, idx, cval->max); + idx++; + } + } + } + return ret; } #define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL) From fc12cf16df9af867e1455b011cdd9310474a612e Mon Sep 17 00:00:00 2001 From: Jackie Dong Date: Wed, 27 May 2026 18:20:05 +0800 Subject: [PATCH 2/9] ASoC: amd: acp: Add DMI quirk for Lenovo Yoga Pro 7 15ASH11 Lenovo Yoga Pro 7 15ASH11 with AMD RYZEN AI MAX+ 388 (Strix Halo, ACP 7.0) uses Realtek ALC287 series codec and no any DMIC connected by ACP. All DMICs directly connet with ALC codec. Without this quirk, Input Device of Gnome Sound settings shows Internal Stereo Microphone and Digital Microphone by default. In fact, Digital Microphone of ACP doesn't work due to no connecting with ALC287 codec, the Internal Stereo Microphone as analog device based on snd_hda_intel driver can work well. Add a DMI quirk to override the flag to 0, consistent with the existing entry for the HN7306EA. Signed-off-by: Jackie Dong Link: https://patch.msgid.link/20260527102005.58528-1-xy-jackie@139.com 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 309dc9ed6e0d..0d977f4f758d 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -37,6 +37,13 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Zenbook S16 UM5606GA"), }, }, + { + /* Lenovo Yoga Pro 7 15ASH11 (Strix Halo, ACP 7.0) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "83W5"), + }, + }, {} }; From c9c64820a48dcc739311fd7a9138511888539d02 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Jo=C3=A3o=20Miguel?= Date: Sat, 23 May 2026 22:35:48 +0100 Subject: [PATCH 3/9] ASoC: amd: yc: Enable internal mic on MSI Bravo 17 C7VF MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The MSI Bravo 17 C7VF routes its internal digital microphone through the ACP6x. The machine driver only enables the DMIC for boards present in the DMI quirk table, so on this model the internal mic is never detected and no capture device is created. Add a quirk entry matching the board's DMI identifiers so the DMIC is enabled and the internal microphone works. Signed-off-by: João Miguel Link: https://patch.msgid.link/20260523213548.5219-1-jmiguel.ghp@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 7a637d6b5576..a93a7fd4877e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -794,6 +794,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M7601RM"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_BOARD_NAME, "MS-17LN"), + } + }, {} }; From 4790af1cc2e8871fb31f28c66e42b9a949a23992 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 1 Jun 2026 17:33:27 +0900 Subject: [PATCH 4/9] ASoC: fsl_sai: Fix 32 slots TDM broken by integer shift UB in xMR write When configuring 32 slots TDM (channels == slots == 32), the xMR (Mask Register) write used: ~0UL - ((1 << min(channels, slots)) - 1) The literal "1" is a signed 32-bit int. Shifting it by 32 positions is undefined behaviour which may set this register to 0xFFFFFFFF, masking all 32 slots. Use GENMASK_U32() macro instead. For 32 slots this produces a zero mask: ~GENMASK_U32(31, 0) = ~0xFFFFFFFF = 0x00000000 Behaviour for fewer than 32 slots is unchanged. Fixes: 770f58d7d2c5 ("ASoC: fsl_sai: Support multiple data channel enable bits") Cc: stable@vger.kernel.org Signed-off-by: Chancel Liu Reviewed-by: Shengjiu Wang Link: https://patch.msgid.link/20260601083327.1535185-1-chancel.liu@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e364552c1f47..78e953cd3e9f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -793,7 +793,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR4_FSD_MSTR, FSL_SAI_CR4_FSD_MSTR); regmap_write(sai->regmap, FSL_SAI_xMR(tx), - ~0UL - ((1 << min(channels, slots)) - 1)); + ~GENMASK_U32(min(channels, slots) - 1, 0)); return 0; } From cad530a8371124da988608f11df30b7dd82391ef Mon Sep 17 00:00:00 2001 From: David Glushkov Date: Sun, 31 May 2026 23:45:12 +0200 Subject: [PATCH 5/9] ASoC: amd: yc: Add MSI Raider A18 HX A9WJG to quirk table The MSI Raider A18 HX A9WJG has an internal digital microphone connected through AMD ACP6x, but this machine does not expose the AcpDmicConnected ACPI property, so acp_yc_mach does not bind. Add a DMI quirk for this model. This was tested on an MSI Raider A18 HX A9WJG with board MS-182L, BIOS E182LAMS.31A, AMD ACP6x rev 0x62, and Realtek ALC274. After applying the quirk, the internal microphone appears as an acp6x DMIC capture device and records correctly. Signed-off-by: David Glushkov Link: https://patch.msgid.link/20260531214512.170716-1-david.glushkov@sntiq.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 a93a7fd4877e..b426cda529a9 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -521,6 +521,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Vector A16 HX A8WHG"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Raider A18 HX A9WJG"), + } + }, { .driver_data = &acp6x_card, .matches = { From d2dcd85f9e09fe3566d4cdcd357856a42ac73f93 Mon Sep 17 00:00:00 2001 From: Aleksander Pshenitsyn Date: Sun, 31 May 2026 13:11:59 +0300 Subject: [PATCH 6/9] ASoC: amd: acp70: add standalone RT721 SoundWire machine The ASUS Vivobook 18 M1807GA (AMD ACP7.x, PCI 1022:15e2, subsystem 1043:3531) exposes a single Realtek RT721 SDCA codec on SoundWire link 1. The BIOS reports the ACP audio config flag as 0 (SoundWire mode), so snd_pci_ps claims the device, brings up the SoundWire managers and enumerates the RT721 peripheral (sdw:0:1:025d:0721:01); the rt721-sdca codec driver binds successfully. No sound card is created, however: acp63_sdw_machine_select() walks snd_soc_acpi_amd_acp70_sdw_machines[] and finds no entry whose declared SoundWire peripherals are all present on the bus. The only existing RT721 entry, acp70_rt721_l1u0_tas2783x2_l1u8b, additionally requires two TAS2783 amplifiers and deliberately exposes the RT721 as jack + DMIC only. This M1807GA variant has no external amplifiers - the RT721's internal AIF2 amplifier path drives the speakers - so that entry never matches and no machine device is registered. Add a standalone RT721 machine entry for link 1 exposing all three RT721 endpoints (jack/AIF1, speaker amplifier/AIF2, DMIC/AIF3), mirroring the standalone RT722 configuration. Place it after the TAS2783 combo entry so platforms that do have the external amplifiers continue to match the more specific entry first. ACPI _ADR of the codec: 0x000130025D072101 (link_id=1 version=3 mfg_id=0x025d Realtek part_id=0x0721 class=0x01). Verified on the hardware: with the entry present the amd_sdw machine binds, an "amd-soundwire" card is registered exposing the rt721-sdca AIF1 (SimpleJack) and AIF2 (SmartAmp) PCM devices, and audio plays out of the built-in speakers. Link: https://bugzilla.kernel.org/show_bug.cgi?id=221282 Signed-off-by: Aleksander Pshenitsyn Link: https://patch.msgid.link/20260531101159.14241-1-brains.fatman@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/amd-acp70-acpi-match.c | 44 ++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 1ae43df5da6c..18f2918d4ada 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -619,6 +619,45 @@ static const struct snd_soc_acpi_link_adr acp70_rt721_l1u0_tas2783x2_l1u8b[] = { {} }; +static const struct snd_soc_acpi_endpoint rt721_endpoints[] = { + { /* Jack Playback/Capture Endpoint (AIF1) */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Speaker Amplifier Endpoint (AIF2, internal amp) */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint (AIF3) */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device rt721_1_single_adr[] = { + { + .adr = 0x000130025D072101ull, + .num_endpoints = ARRAY_SIZE(rt721_endpoints), + .endpoints = rt721_endpoints, + .name_prefix = "rt721" + } +}; + +static const struct snd_soc_acpi_link_adr acp70_rt721_only[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt721_1_single_adr), + .adr_d = rt721_1_single_adr, + }, + {} +}; + struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { { .link_mask = BIT(0) | BIT(1), @@ -711,6 +750,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { .links = acp70_rt721_l1u0_tas2783x2_l1u8b, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(1), + .links = acp70_rt721_only, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sdw_machines); From 88fe2e3658726cb21ff2dcf9770bf672f9b9d31b Mon Sep 17 00:00:00 2001 From: Ji'an Zhou Date: Thu, 4 Jun 2026 14:25:59 +0000 Subject: [PATCH 7/9] ALSA: PCM: Fix wait queue list corruption in snd_pcm_drain() on linked streams snd_pcm_drain() uses init_waitqueue_entry which does not clear entry.prev/next, and add_wait_queue with a conditional remove_wait_queue that is skipped when to_check is no longer in the group after concurrent UNLINK. The orphaned wait entry remains on the unlinked substream sleep queue. On the next drain iteration, add_wait_queue adds the entry to a new queue while still linked on the old one, corrupting both lists. A subsequent wake_up dereferences NULL at the func pointer (mapped from the spinlock at offset 0 of the misinterpreted wait_queue_head_t), causing a kernel panic. Replace init_waitqueue_entry/add_wait_queue/conditional remove_wait_queue with init_wait_entry/prepare_to_wait/ finish_wait. init_wait_entry clears prev/next via INIT_LIST_HEAD on each iteration and sets autoremove_wake_function which auto-removes the entry on wake-up. finish_wait safely handles both the already-removed and still-queued cases. Fixes: 9b1dbd69ba6f ("ALSA: pcm: fix use-after-free on linked stream runtime in snd_pcm_drain") Signed-off-by: Ji'an Zhou Link: https://patch.msgid.link/20260604142559.3840881-1-eilaimemedsnaimel@gmail.com Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a541bb235cfa..302643c1c192 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2199,9 +2199,8 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, drain_no_period_wakeup = to_check->no_period_wakeup; drain_rate = to_check->rate; drain_bufsz = to_check->buffer_size; - init_waitqueue_entry(&wait, current); - set_current_state(TASK_INTERRUPTIBLE); - add_wait_queue(&to_check->sleep, &wait); + init_wait_entry(&wait, 0); + prepare_to_wait(&to_check->sleep, &wait, TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); if (drain_no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; @@ -2219,7 +2218,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, group = snd_pcm_stream_group_ref(substream); snd_pcm_group_for_each_entry(s, substream) { if (s->runtime == to_check) { - remove_wait_queue(&to_check->sleep, &wait); + finish_wait(&to_check->sleep, &wait); break; } } From d76b56b06b2734c6e0ff519599126919fa4f154b Mon Sep 17 00:00:00 2001 From: Lianqin Hu Date: Thu, 4 Jun 2026 12:40:28 +0000 Subject: [PATCH 8/9] ALSA: usb-audio: Add iface reset and delay quirk for AB13X USB Audio Setting up the interface when suspended/resumeing fail on this card. Adding a reset and delay quirk will eliminate this problem. usb 1-1: new full-speed USB device number 2 using xhci-hcd usb 1-1: New USB device found, idVendor=3c20, idProduct=3d21 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: AB13X USB Audio usb 1-1: Manufacturer: Generic usb 1-1: SerialNumber: 20210726905926 Signed-off-by: Lianqin Hu Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/TYUPR06MB62174610061C213260E1A992D2102@TYUPR06MB6217.apcprd06.prod.outlook.com --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3d1b3523b020..f4e885f96e83 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2487,6 +2487,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x3443, 0x930d, /* NexiGo N930W 60fps Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), + DEVICE_FLG(0x3c20, 0x3d21, /* AB13X USB Audio */ + QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x534d, 0x0021, /* MacroSilicon MS2100/MS2106 */ From 2b5ff4db5d7aa5b981d966df02e687f79ad7b311 Mon Sep 17 00:00:00 2001 From: Kyle Zeng Date: Fri, 5 Jun 2026 01:02:04 -0700 Subject: [PATCH 9/9] ALSA: seq: dummy: fix UMP event stack overread The dummy sequencer port forwards events by copying an incoming struct snd_seq_event into a stack temporary, rewriting source and destination, and dispatching the temporary to subscribers. That legacy event storage is smaller than struct snd_seq_ump_event. When a UMP event reaches the dummy client, the copy leaves the UMP flag set but only provides legacy-sized stack storage. The subscriber delivery path then uses snd_seq_event_packet_size() and copies a UMP-sized packet from that stack object, reading past the end of the temporary. Use the existing union __snd_seq_event storage and copy the packet size reported for the incoming event before rewriting the common routing fields. This preserves the full UMP packet for UMP events while keeping legacy event handling unchanged. Fixes: 32cb23a0f911 ("ALSA: seq: dummy: Allow UMP conversion") Signed-off-by: Kyle Zeng Link: https://patch.msgid.link/20260605080204.32045-1-kylebot@openai.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_dummy.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index af45f328ae99..8abe80985dad 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -9,6 +9,7 @@ #include #include #include "seq_clientmgr.h" +#include "seq_memory.h" #include #include @@ -81,19 +82,21 @@ dummy_input(struct snd_seq_event *ev, int direct, void *private_data, int atomic, int hop) { struct snd_seq_dummy_port *p; - struct snd_seq_event tmpev; + union __snd_seq_event tmpev; + size_t size; p = private_data; if (ev->source.client == SNDRV_SEQ_CLIENT_SYSTEM || ev->type == SNDRV_SEQ_EVENT_KERNEL_ERROR) return 0; /* ignore system messages */ - tmpev = *ev; + size = snd_seq_event_packet_size(ev); + memcpy(&tmpev, ev, size); if (p->duplex) - tmpev.source.port = p->connect; + tmpev.legacy.source.port = p->connect; else - tmpev.source.port = p->port; - tmpev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; - return snd_seq_kernel_client_dispatch(p->client, &tmpev, atomic, hop); + tmpev.legacy.source.port = p->port; + tmpev.legacy.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS; + return snd_seq_kernel_client_dispatch(p->client, &tmpev.legacy, atomic, hop); } /*