From 4a60127debb9e370d6c0e22a307326b624a141f3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?C=C3=A1ssio=20Gabriel?= Date: Mon, 15 Jun 2026 10:37:26 -0300 Subject: [PATCH 01/46] ALSA: compress: Fix task creation error unwind MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_compr_task_new() allocates the driver task before validating the returned DMA buffers and reserving file descriptors. When either of those later steps fails, the core frees its task wrapper and DMA-buffer references without calling the driver's task_free() callback. Any driver resources allocated by task_create() are therefore leaked. The dual-fd allocation path also jumps to cleanup without storing the negative get_unused_fd_flags() result in retval. Since retval still contains the successful task_create() return value, TASK_CREATE can incorrectly report success although the task was discarded. Preserve the fd allocation errors and call task_free() when failure occurs after a successful task_create() callback. Fixes: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Fixes: 3d3f43fab4cf ("ALSA: compress_offload: improve file descriptors installation for dma-buf") Cc: stable@vger.kernel.org Signed-off-by: Cássio Gabriel Link: https://patch.msgid.link/20260615-alsa-compress-task-unwind-v1-1-39e8ad3ddb27@gmail.com Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index fd63d219bf86..ea699491f0c3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1083,15 +1083,18 @@ static int snd_compr_task_new(struct snd_compr_stream *stream, struct snd_compr_ file descriptors are allocated before fd_install() */ if (!task->input || !task->input->file || !task->output || !task->output->file) { retval = -EINVAL; - goto cleanup; + goto free_driver_task; } fd_i = get_unused_fd_flags(O_WRONLY|O_CLOEXEC); - if (fd_i < 0) - goto cleanup; + if (fd_i < 0) { + retval = fd_i; + goto free_driver_task; + } fd_o = get_unused_fd_flags(O_RDONLY|O_CLOEXEC); if (fd_o < 0) { + retval = fd_o; put_unused_fd(fd_i); - goto cleanup; + goto free_driver_task; } /* keep dmabuf reference until freed with task free ioctl */ get_dma_buf(task->input); @@ -1103,6 +1106,8 @@ static int snd_compr_task_new(struct snd_compr_stream *stream, struct snd_compr_ list_add_tail(&task->list, &stream->runtime->tasks); stream->runtime->total_tasks++; return 0; +free_driver_task: + stream->ops->task_free(stream, task); cleanup: snd_compr_task_free(task); return retval; From f7c4968ae3af3e819428da5416c2dfd361473f5c Mon Sep 17 00:00:00 2001 From: Galen Hassen Date: Tue, 16 Jun 2026 10:32:57 -0700 Subject: [PATCH 02/46] ALSA: hda/conexant: Add pin config quirk for Lenovo IdeaPad Slim 5 16AKP10 The Lenovo IdeaPad Slim 5 16AKP10 (PCI SSID 17aa:38b6) uses the Conexant SN6140 codec. The internal microphone is on pin 0x1a but the BIOS configures it with pin default 0x95a60120, which includes a jack detection bit that causes the kernel to treat it as an unplugged external mic rather than a fixed internal mic. Add a pin config quirk that overrides pin 0x1a to 0x95a60130, setting the connectivity bits to indicate a fixed/always-connected device. This allows the internal microphone to be correctly identified and used. Signed-off-by: Galen Hassen Link: https://patch.msgid.link/20260616173257.37373-1-rwekyes@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/conexant.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index e3b6aaabe3a9..3d92262763f6 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -291,6 +291,7 @@ enum { CXT_FIXUP_HEADSET_MIC, CXT_FIXUP_HP_MIC_NO_PRESENCE, CXT_PINCFG_SWS_JS201D, + CXT_PINCFG_LENOVO_IDEAPAD_SLIM5_16AKP10, CXT_PINCFG_TOP_SPEAKER, CXT_FIXUP_HP_A_U, CXT_FIXUP_ACER_SWIFT_HP, @@ -826,6 +827,12 @@ static const struct hda_pintbl cxt_pincfg_lemote[] = { {} }; +/* Lenovo IdeaPad Slim 5 16AKP10 with SN6140 */ +static const struct hda_pintbl cxt_pincfg_lenovo_ideapad_slim5_16akp10[] = { + { 0x1a, 0x95a60130 }, /* Internal mic, fixed/always-connected */ + {} +}; + /* SuoWoSi/South-holding JS201D with sn6140 */ static const struct hda_pintbl cxt_pincfg_sws_js201d[] = { { 0x16, 0x03211040 }, /* hp out */ @@ -1006,6 +1013,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_sws_js201d, }, + [CXT_PINCFG_LENOVO_IDEAPAD_SLIM5_16AKP10] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lenovo_ideapad_slim5_16akp10, + }, [CXT_PINCFG_TOP_SPEAKER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1114,6 +1125,7 @@ static const struct hda_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), + SND_PCI_QUIRK(0x17aa, 0x38b6, "Lenovo IdeaPad Slim 5 16AKP10", CXT_PINCFG_LENOVO_IDEAPAD_SLIM5_16AKP10), SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), From 7d69804a35103a50852eae41bfe6a2e0061c68fd Mon Sep 17 00:00:00 2001 From: Xu Rao Date: Tue, 16 Jun 2026 19:59:16 +0800 Subject: [PATCH 03/46] ALSA: usb-audio: qcom: Free sideband sg_table objects The Qualcomm USB audio offload driver obtains an endpoint transfer-ring table by calling xhci_sideband_get_endpoint_buffer(). This getter passes the endpoint ring to xhci_ring_to_sgtable(), which allocates the outer struct sg_table with kzalloc_obj(*sgt). The event-ring path is equivalent: xhci_sideband_get_event_buffer() also returns the result of xhci_ring_to_sgtable(). Inside xhci_ring_to_sgtable(), sg_alloc_table_from_pages() separately allocates the scatterlist storage referenced by sgt->sgl. The returned object therefore has two allocation layers: the outer struct sg_table and its internal scatterlist storage. The Qualcomm caller only invokes sg_free_table(sgt). sg_free_table() releases the scatterlist storage owned by the table, but it does not free the separately allocated outer struct sg_table. The local sgt pointer is then discarded, so every successful endpoint or event-ring query leaks the outer object. Call kfree(sgt) after sg_free_table(sgt) in both setup paths, after the required page and DMA addresses have been copied out. Fixes: 326bbc348298 ("ALSA: usb-audio: qcom: Introduce QC USB SND offloading support") Signed-off-by: Xu Rao Link: https://patch.msgid.link/90B353283AA150C4+20260616115916.1222915-1-raoxu@uniontech.com Signed-off-by: Takashi Iwai --- sound/usb/qcom/qc_audio_offload.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index a3f90cc7c6ca..436d6821c5c9 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -1160,6 +1160,7 @@ uaudio_endpoint_setup(struct snd_usb_substream *subs, tr_pa = page_to_phys(pg); mem_info->dma = sg_dma_address(sgt->sgl); sg_free_table(sgt); + kfree(sgt); /* data transfer ring */ iova = uaudio_iommu_map_pa(MEM_XFER_RING, dma_coherent, tr_pa, @@ -1229,6 +1230,7 @@ static int uaudio_event_ring_setup(struct snd_usb_substream *subs, er_pa = page_to_phys(pg); mem_info->dma = sg_dma_address(sgt->sgl); sg_free_table(sgt); + kfree(sgt); iova = uaudio_iommu_map_pa(MEM_EVENT_RING, dma_coherent, er_pa, PAGE_SIZE); From 8956950dab22fbaefe92ca1980728165c5da793d Mon Sep 17 00:00:00 2001 From: Ai Chao Date: Wed, 17 Jun 2026 10:52:34 +0800 Subject: [PATCH 04/46] ALSA: usb-audio: Add quirk flags for SC13A The SC13A ( VID 0x1ff7, PID 0x0f81) not support reading the current sample rate and results in an error message printed to kmsg. Set QUIRK_FLAG_GET_SAMPLE_RATE to skip the sample rate check. Quirky device sample: usb 3-5.2.4.1: new high-speed USB device number 11 using xhci_hcd usb 3-5.2.4.1: New USB device found, idVendor=1ff7, idProduct=0f81 usb 3-5.2.4.1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 3-5.2.4.1: Product: SC13A usb 3-5.2.4.1: Manufacturer: Linux Foundation usb 3-5.2.4.1: SerialNumber: 000002 usb 3-5.2.4.1: Found UVC 1.50 device SC13A (1ff7:0f81) usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86 usb 3-5.2.4.1: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. usb 3-5.2.4.1: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 usbcore: registered new interface driver snd-usb-audio usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86 usb 3-5.2.4.1: 3:1: cannot get freq at ep 0x86 Signed-off-by: Ai Chao Link: https://patch.msgid.link/20260617025234.3344935-1-aichao@kylinos.cn 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 30f6d572e4d1..1cb588691e16 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2413,6 +2413,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), + DEVICE_FLG(0x1ff7, 0x0f81, /* SC13A Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x2040, 0x7200, /* Hauppauge HVR-950Q */ QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x2040, 0x7201, /* Hauppauge HVR-950Q-MXL */ From 9e6febe7891316182bcd80cb46b745a08ad0cacf Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Wed, 17 Jun 2026 08:42:18 +0200 Subject: [PATCH 05/46] ALSA: sh: Use more common error handling code in snd_aica_probe() Use an additional label so that a bit of exception handling can be better reused at the end of this function implementation. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Link: https://patch.msgid.link/47413de1-015b-4543-8e8c-25e41dfa9e39@web.de Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 9438c3a68ee9..8196e1bf0416 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -564,10 +564,9 @@ static int snd_aica_probe(struct platform_device *devptr) return -ENOMEM; err = snd_card_new(&devptr->dev, index, SND_AICA_DRIVER, THIS_MODULE, 0, &dreamcastcard->card); - if (unlikely(err < 0)) { - kfree(dreamcastcard); - return err; - } + if (unlikely(err < 0)) + goto free_card; + strscpy(dreamcastcard->card->driver, "snd_aica"); strscpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strscpy(dreamcastcard->card->longname, @@ -593,6 +592,7 @@ static int snd_aica_probe(struct platform_device *devptr) return 0; freedreamcast: snd_card_free(dreamcastcard->card); +free_card: kfree(dreamcastcard); return err; } From 2e9261761b35f0b67b7487688cd1365f535be0b3 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Jun 2026 18:02:57 +0100 Subject: [PATCH 06/46] ASoC: qcom: q6apm: fix NULL pointer dereference in graph_callback When q6apm_free_fragments() is called it frees rx_data.buf/tx_data.buf and sets them to NULL under graph->lock. A late DSP buffer-done response can race with this: graph_callback() passes the !graph->ar_graph guard (not yet NULL), acquires the lock, but then dereferences a now-NULL buf pointer to read buf[token].phys, crashing at virtual address 0x10. Add a NULL check for buf inside the mutex-protected section in both the write-done (DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2) and read-done (DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2) handlers and bail out cleanly if buffers have already been freed. This problem is only shown up recently while apr bus was updated to process the commands per service rather from single global queue. Fixes: 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support") Cc: Stable@vger.kernel.org Assisted-by: Claude:claude-4-6-sonnet Reported-by: Val Packett Closes: https://lore.kernel.org/all/133ced18-1aa9-475d-80d8-6120678bdde4@packett.cool/ Signed-off-by: Srinivas Kandagatla Link: https://patch.msgid.link/20260616170257.9381-1-srinivas.kandagatla@oss.qualcomm.com Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 2e5b25b8d00f..641d6d243229 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -587,6 +587,10 @@ static int graph_callback(const struct gpr_resp_pkt *data, void *priv, int op) token = hdr->token & APM_WRITE_TOKEN_MASK; done = data->payload; + if (!graph->rx_data.buf) { + mutex_unlock(&graph->lock); + break; + } phys = graph->rx_data.buf[token].phys; mutex_unlock(&graph->lock); /* token numbering starts at 0 */ @@ -609,6 +613,10 @@ static int graph_callback(const struct gpr_resp_pkt *data, void *priv, int op) client_event = APM_CLIENT_EVENT_DATA_READ_DONE; mutex_lock(&graph->lock); rd_done = data->payload; + if (!graph->tx_data.buf) { + mutex_unlock(&graph->lock); + break; + } phys = graph->tx_data.buf[hdr->token].phys; mutex_unlock(&graph->lock); /* token numbering starts at 0 */ From 1a3c8e28959790ae3e06029681418278b7821a3c Mon Sep 17 00:00:00 2001 From: Nicolas Frattaroli Date: Wed, 17 Jun 2026 13:46:04 +0200 Subject: [PATCH 07/46] ASoC: rockchip: Drop problematic guard() changes This reverts commit f7fe9f707360 ("ASoC: rockchip: rockchip_sai: Use guard() for spin locks"). This is very noisy pointless churn that was not tested by the submitter, nor was it addressed to the driver's maintainer. It mixes unrelated whitespace changes (eliminating the blank line between the includes - why?) with hard to review diffs that add a whole indentation level to the function for no benefit, while also not following kernel code style by doing stuff like "ret == 0". The driver is better off without these changes, and they're not worth the time to validate whether they really do make no functional changes. Signed-off-by: Nicolas Frattaroli Link: https://patch.msgid.link/20260617-sai-revert-v1-1-e46adda2213b@collabora.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_sai.c | 274 +++++++++++++++--------------- 1 file changed, 139 insertions(+), 135 deletions(-) diff --git a/sound/soc/rockchip/rockchip_sai.c b/sound/soc/rockchip/rockchip_sai.c index a195e96fed0a..ed393e5034a4 100644 --- a/sound/soc/rockchip/rockchip_sai.c +++ b/sound/soc/rockchip/rockchip_sai.c @@ -18,6 +18,7 @@ #include #include #include + #include "rockchip_sai.h" #define DRV_NAME "rockchip-sai" @@ -215,12 +216,14 @@ static void rockchip_sai_xfer_clk_stop_and_wait(struct rk_sai_dev *sai, unsigned static int rockchip_sai_runtime_suspend(struct device *dev) { struct rk_sai_dev *sai = dev_get_drvdata(dev); + unsigned long flags; rockchip_sai_fsync_lost_detect(sai, 0); rockchip_sai_fsync_err_detect(sai, 0); - scoped_guard(spinlock_irqsave, &sai->xfer_lock) - rockchip_sai_xfer_clk_stop_and_wait(sai, NULL); + spin_lock_irqsave(&sai->xfer_lock, flags); + rockchip_sai_xfer_clk_stop_and_wait(sai, NULL); + spin_unlock_irqrestore(&sai->xfer_lock, flags); regcache_cache_only(sai->regmap, true); /* @@ -480,6 +483,7 @@ static int rockchip_sai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai); unsigned int mask = 0, val = 0; unsigned int clk_gates; + unsigned long flags; int ret = 0; pm_runtime_get_sync(dai->dev); @@ -495,56 +499,56 @@ static int rockchip_sai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) sai->is_master_mode = false; break; default: - pm_runtime_put(dai->dev); - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } - scoped_guard(spinlock_irqsave, &sai->xfer_lock) { - rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates); - if (sai->initialized) { - if (sai->has_capture && sai->has_playback) - rockchip_sai_xfer_stop(sai, -1); - else if (sai->has_capture) - rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_CAPTURE); - else - rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_PLAYBACK); - } else { - rockchip_sai_clear(sai, 0); - sai->initialized = true; - } - - regmap_update_bits(sai->regmap, SAI_CKR, mask, val); - - mask = SAI_CKR_CKP_MASK | SAI_CKR_FSP_MASK; - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_NORMAL; - break; - case SND_SOC_DAIFMT_NB_IF: - val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_INVERTED; - break; - case SND_SOC_DAIFMT_IB_NF: - val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_NORMAL; - break; - case SND_SOC_DAIFMT_IB_IF: - val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_INVERTED; - break; - default: - ret = -EINVAL; - break; - } - - if (ret == 0) { - regmap_update_bits(sai->regmap, SAI_CKR, mask, val); - rockchip_sai_fmt_create(sai, fmt); - } - - if (clk_gates) - regmap_update_bits(sai->regmap, SAI_XFER, - SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK, - clk_gates); + spin_lock_irqsave(&sai->xfer_lock, flags); + rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates); + if (sai->initialized) { + if (sai->has_capture && sai->has_playback) + rockchip_sai_xfer_stop(sai, -1); + else if (sai->has_capture) + rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_CAPTURE); + else + rockchip_sai_xfer_stop(sai, SNDRV_PCM_STREAM_PLAYBACK); + } else { + rockchip_sai_clear(sai, 0); + sai->initialized = true; } + regmap_update_bits(sai->regmap, SAI_CKR, mask, val); + + mask = SAI_CKR_CKP_MASK | SAI_CKR_FSP_MASK; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_NORMAL; + break; + case SND_SOC_DAIFMT_NB_IF: + val = SAI_CKR_CKP_NORMAL | SAI_CKR_FSP_INVERTED; + break; + case SND_SOC_DAIFMT_IB_NF: + val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_NORMAL; + break; + case SND_SOC_DAIFMT_IB_IF: + val = SAI_CKR_CKP_INVERTED | SAI_CKR_FSP_INVERTED; + break; + default: + ret = -EINVAL; + goto err_xfer_unlock; + } + + regmap_update_bits(sai->regmap, SAI_CKR, mask, val); + + rockchip_sai_fmt_create(sai, fmt); + +err_xfer_unlock: + if (clk_gates) + regmap_update_bits(sai->regmap, SAI_XFER, + SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK, + clk_gates); + spin_unlock_irqrestore(&sai->xfer_lock, flags); +err_pm_put: pm_runtime_put(dai->dev); return ret; @@ -560,6 +564,7 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream, unsigned int ch_per_lane, slot_width; unsigned int val, fscr, reg; unsigned int lanes, req_lanes; + unsigned long flags; int ret = 0; if (!rockchip_sai_stream_valid(substream, dai)) @@ -586,8 +591,8 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream, dev_err(sai->dev, "not enough lanes (%d) for requested number of %s channels (%d)\n", lanes, reg == SAI_TXCR ? "playback" : "capture", params_channels(params)); - pm_runtime_put(sai->dev); - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } else { lanes = req_lanes; } @@ -613,88 +618,84 @@ static int rockchip_sai_hw_params(struct snd_pcm_substream *substream, val = SAI_XCR_VDW(32); break; default: - pm_runtime_put(sai->dev); - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } val |= SAI_XCR_CSR(lanes); - scoped_guard(spinlock_irqsave, &sai->xfer_lock) { + spin_lock_irqsave(&sai->xfer_lock, flags); - regmap_update_bits(sai->regmap, reg, SAI_XCR_VDW_MASK | SAI_XCR_CSR_MASK, val); + regmap_update_bits(sai->regmap, reg, SAI_XCR_VDW_MASK | SAI_XCR_CSR_MASK, val); - if (!sai->is_tdm) - regmap_update_bits(sai->regmap, reg, SAI_XCR_SBW_MASK, - SAI_XCR_SBW(params_physical_width(params))); + if (!sai->is_tdm) + regmap_update_bits(sai->regmap, reg, SAI_XCR_SBW_MASK, + SAI_XCR_SBW(params_physical_width(params))); - regmap_read(sai->regmap, reg, &val); + regmap_read(sai->regmap, reg, &val); - slot_width = SAI_XCR_SBW_V(val); - ch_per_lane = params_channels(params) / lanes; + slot_width = SAI_XCR_SBW_V(val); + ch_per_lane = params_channels(params) / lanes; - regmap_update_bits(sai->regmap, reg, SAI_XCR_SNB_MASK, - SAI_XCR_SNB(ch_per_lane)); + regmap_update_bits(sai->regmap, reg, SAI_XCR_SNB_MASK, + SAI_XCR_SNB(ch_per_lane)); - fscr = SAI_FSCR_FW(sai->fw_ratio * slot_width * ch_per_lane); + fscr = SAI_FSCR_FW(sai->fw_ratio * slot_width * ch_per_lane); - switch (sai->fpw) { - case FPW_ONE_BCLK_WIDTH: - fscr |= SAI_FSCR_FPW(1); - break; - case FPW_ONE_SLOT_WIDTH: - fscr |= SAI_FSCR_FPW(slot_width); - break; - case FPW_HALF_FRAME_WIDTH: - fscr |= SAI_FSCR_FPW(sai->fw_ratio * slot_width * ch_per_lane / 2); - break; - default: - dev_err(sai->dev, "Invalid Frame Pulse Width %d\n", sai->fpw); - ret = -EINVAL; - break; - } - - if (ret == 0) { - regmap_update_bits(sai->regmap, SAI_FSCR, - SAI_FSCR_FW_MASK | SAI_FSCR_FPW_MASK, fscr); - - if (sai->is_master_mode) { - bclk_rate = sai->fw_ratio * slot_width * - ch_per_lane * params_rate(params); - ret = clk_set_rate(sai->mclk, sai->mclk_rate); - if (ret) - dev_err(sai->dev, "Failed to set mclk to %u: %pe\n", - sai->mclk_rate, ERR_PTR(ret)); - else { - mclk_rate = clk_get_rate(sai->mclk); - if (mclk_rate < bclk_rate) { - dev_err(sai->dev, "Mismatch mclk: %u, at least %u\n", - mclk_rate, bclk_rate); - ret = -EINVAL; - } else { - - div_bclk = DIV_ROUND_CLOSEST(mclk_rate, bclk_rate); - mclk_req_rate = bclk_rate * div_bclk; - - if (mclk_rate < - mclk_req_rate - CLK_SHIFT_RATE_HZ_MAX || - mclk_rate > - mclk_req_rate + CLK_SHIFT_RATE_HZ_MAX) { - dev_err(sai->dev, - "Mismatch mclk: %u, expected %u (+/- %dHz)\n", - mclk_rate, mclk_req_rate, - CLK_SHIFT_RATE_HZ_MAX); - ret = -EINVAL; - } else - regmap_update_bits(sai->regmap, - SAI_CKR, - SAI_CKR_MDIV_MASK, - SAI_CKR_MDIV(div_bclk)); - } - } - } - } + switch (sai->fpw) { + case FPW_ONE_BCLK_WIDTH: + fscr |= SAI_FSCR_FPW(1); + break; + case FPW_ONE_SLOT_WIDTH: + fscr |= SAI_FSCR_FPW(slot_width); + break; + case FPW_HALF_FRAME_WIDTH: + fscr |= SAI_FSCR_FPW(sai->fw_ratio * slot_width * ch_per_lane / 2); + break; + default: + dev_err(sai->dev, "Invalid Frame Pulse Width %d\n", sai->fpw); + ret = -EINVAL; + goto err_xfer_unlock; } + regmap_update_bits(sai->regmap, SAI_FSCR, + SAI_FSCR_FW_MASK | SAI_FSCR_FPW_MASK, fscr); + + if (sai->is_master_mode) { + bclk_rate = sai->fw_ratio * slot_width * ch_per_lane * params_rate(params); + ret = clk_set_rate(sai->mclk, sai->mclk_rate); + if (ret) { + dev_err(sai->dev, "Failed to set mclk to %u: %pe\n", + sai->mclk_rate, ERR_PTR(ret)); + goto err_xfer_unlock; + } + + mclk_rate = clk_get_rate(sai->mclk); + if (mclk_rate < bclk_rate) { + dev_err(sai->dev, "Mismatch mclk: %u, at least %u\n", + mclk_rate, bclk_rate); + ret = -EINVAL; + goto err_xfer_unlock; + } + + div_bclk = DIV_ROUND_CLOSEST(mclk_rate, bclk_rate); + mclk_req_rate = bclk_rate * div_bclk; + + if (mclk_rate < mclk_req_rate - CLK_SHIFT_RATE_HZ_MAX || + mclk_rate > mclk_req_rate + CLK_SHIFT_RATE_HZ_MAX) { + dev_err(sai->dev, "Mismatch mclk: %u, expected %u (+/- %dHz)\n", + mclk_rate, mclk_req_rate, CLK_SHIFT_RATE_HZ_MAX); + ret = -EINVAL; + goto err_xfer_unlock; + } + + regmap_update_bits(sai->regmap, SAI_CKR, SAI_CKR_MDIV_MASK, + SAI_CKR_MDIV(div_bclk)); + } + +err_xfer_unlock: + spin_unlock_irqrestore(&sai->xfer_lock, flags); +err_pm_put: pm_runtime_put(sai->dev); return ret; @@ -704,6 +705,7 @@ static int rockchip_sai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai); + unsigned long flags; if (!rockchip_sai_stream_valid(substream, dai)) return 0; @@ -724,12 +726,13 @@ static int rockchip_sai_prepare(struct snd_pcm_substream *substream, * udelay falls short. */ udelay(20); - scoped_guard(spinlock_irqsave, &sai->xfer_lock) - regmap_update_bits(sai->regmap, SAI_XFER, - SAI_XFER_CLK_MASK | - SAI_XFER_FSS_MASK, - SAI_XFER_CLK_EN | - SAI_XFER_FSS_EN); + spin_lock_irqsave(&sai->xfer_lock, flags); + regmap_update_bits(sai->regmap, SAI_XFER, + SAI_XFER_CLK_MASK | + SAI_XFER_FSS_MASK, + SAI_XFER_CLK_EN | + SAI_XFER_FSS_EN); + spin_unlock_irqrestore(&sai->xfer_lock, flags); } rockchip_sai_fsync_lost_detect(sai, 1); @@ -912,6 +915,7 @@ static int rockchip_sai_set_tdm_slot(struct snd_soc_dai *dai, int slots, int slot_width) { struct rk_sai_dev *sai = snd_soc_dai_get_drvdata(dai); + unsigned long flags; unsigned int clk_gates; int sw = slot_width; @@ -927,16 +931,16 @@ static int rockchip_sai_set_tdm_slot(struct snd_soc_dai *dai, return -EINVAL; pm_runtime_get_sync(dai->dev); - scoped_guard(spinlock_irqsave, &sai->xfer_lock) { - rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates); - regmap_update_bits(sai->regmap, SAI_TXCR, SAI_XCR_SBW_MASK, - SAI_XCR_SBW(sw)); - regmap_update_bits(sai->regmap, SAI_RXCR, SAI_XCR_SBW_MASK, - SAI_XCR_SBW(sw)); - regmap_update_bits(sai->regmap, SAI_XFER, - SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK, - clk_gates); - } + spin_lock_irqsave(&sai->xfer_lock, flags); + rockchip_sai_xfer_clk_stop_and_wait(sai, &clk_gates); + regmap_update_bits(sai->regmap, SAI_TXCR, SAI_XCR_SBW_MASK, + SAI_XCR_SBW(sw)); + regmap_update_bits(sai->regmap, SAI_RXCR, SAI_XCR_SBW_MASK, + SAI_XCR_SBW(sw)); + regmap_update_bits(sai->regmap, SAI_XFER, + SAI_XFER_CLK_MASK | SAI_XFER_FSS_MASK, + clk_gates); + spin_unlock_irqrestore(&sai->xfer_lock, flags); pm_runtime_put(dai->dev); return 0; From fdf043f5f3bae150b678feae3d7bb1beed87ec14 Mon Sep 17 00:00:00 2001 From: Sen Wang Date: Tue, 16 Jun 2026 18:33:22 -0500 Subject: [PATCH 08/46] ASoC: tlv320aic3x: restrict CLKDIV bypass Q values in dual-rate mode The datasheet documents that when the PLL is disabled and dual-rate mode is enabled, only Q values {4, 8, 9, 12, 16} are valid for the CLKDIV bypass path; all other Q values produce invalid bitclock output. The existing loop iterates Q from 2 to 17 without this restriction, causing silent audio failure when an out-of-spec Q is picked. Restrict the Q search to the allowed set in dual-rate mode. Fixes: 4f9c16ccfa26 ("[ALSA] soc - tlv320aic3x - revisit clock setup") Suggested-by: Mir Jeffres Signed-off-by: Sen Wang Link: https://patch.msgid.link/20260616233322.873081-1-sen@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b12c1952823b..b38393a8130f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1049,11 +1049,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + static const u8 dual_rate_q[] = {4, 8, 9, 12, 16}; struct snd_soc_component *component = dai->component; struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 d, pll_d = 1; + bool dual_rate; int clk; int width = aic3x->slot_width; @@ -1079,14 +1081,25 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, /* Fsref can be 44100 or 48000 */ fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; + dual_rate = params_rate(params) >= 64000; /* Try to find a value for Q which allows us to bypass the PLL and * generate CODEC_CLK directly. */ - for (pll_q = 2; pll_q < 18; pll_q++) - if (aic3x->sysclk / (128 * pll_q) == fsref) { - bypass_pll = 1; - break; + if (dual_rate) { + for (int i = 0; i < ARRAY_SIZE(dual_rate_q); i++) { + pll_q = dual_rate_q[i]; + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } } + } else { + for (pll_q = 2; pll_q < 18; pll_q++) + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } + } if (bypass_pll) { pll_q &= 0xf; @@ -1106,13 +1119,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, * right DAC to right channel input */ data = (LDAC2LCH | RDAC2RCH); data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; - if (params_rate(params) >= 64000) + if (dual_rate) data |= DUAL_RATE_MODE; snd_soc_component_write(component, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ data = (fsref * 20) / params_rate(params); - if (params_rate(params) < 64000) + if (!dual_rate) data /= 2; data /= 5; data -= 2; From 5096a1634def431cd7f89e2b0af4456de91dd26d Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 15 Jun 2026 17:10:12 +0800 Subject: [PATCH 09/46] ASoC: rt5650: enhance spk protection function This patch adjusts several default settings to ensure the speaker protection function can be enabled safely. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20260615091012.718168-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 6 ++---- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 8a9af260e5f7..e9819653b30d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -83,6 +83,8 @@ static const struct reg_sequence rt5650_init_list[] = { {RT5645_PWR_ANLG1, 0x02}, {RT5645_IL_CMD3, 0x6728}, {RT5645_PR_BASE + 0x3a, 0x0000}, + {RT5645_CLSD_OUT_CTRL1, 0x4059}, + {RT5645_GEN_CTRL3, 0x0200}, }; static const struct reg_default rt5645_reg[] = { @@ -1855,13 +1857,9 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, RT5645_PWR_CLS_D_L, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L); - snd_soc_component_update_bits(component, RT5645_GEN_CTRL3, - RT5645_DET_CLK_MASK, RT5645_DET_CLK_MODE1); break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_component_update_bits(component, RT5645_GEN_CTRL3, - RT5645_DET_CLK_MASK, RT5645_DET_CLK_DIS); snd_soc_component_write(component, RT5645_EQ_CTRL2, 0); snd_soc_component_update_bits(component, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index bef74b29fd54..a5bfe9861b8b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -118,6 +118,7 @@ #define RT5645_A_JD_CTRL1 0x94 #define RT5645_VAD_CTRL4 0x9d #define RT5645_CLSD_OUT_CTRL 0xa0 +#define RT5645_CLSD_OUT_CTRL1 0xa1 /* Function - Digital */ #define RT5645_ADC_EQ_CTRL1 0xae #define RT5645_ADC_EQ_CTRL2 0xaf From 53d1ae7c20d97b08741d667ba54bb09d330eba3b Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 17 Jun 2026 11:29:43 +0100 Subject: [PATCH 10/46] ALSA: pcm: fix __le32 cast warning in snd_pcm_set_sync_per_card In snd_pcm_set_sync_per_card() the le32 value is written to an u32 instead of an __le32 pointer. Fix the following warning by fixing the type: sound/soc/soc-pcm.c:2166:9: warning: incorrect type in argument 7 (different base types) sound/soc/soc-pcm.c:2166:9: expected int sound/soc/soc-pcm.c:2166:9: got restricted snd_pcm_format_t Signed-off-by: Ben Dooks Link: https://patch.msgid.link/20260617102943.893950-1-ben.dooks@codethink.co.uk Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index fe597f7d522d..4d665b4148d7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -545,7 +545,7 @@ void snd_pcm_set_sync_per_card(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, const unsigned char *id, unsigned int len) { - *(__u32 *)params->sync = cpu_to_le32(substream->pcm->card->number); + *(__le32 *)params->sync = cpu_to_le32(substream->pcm->card->number); len = min(12, len); memcpy(params->sync + 4, id, len); memset(params->sync + 4 + len, 0, 12 - len); From f7f3f9fd81e7adbaa12c2e62ee07f0e094a543fd Mon Sep 17 00:00:00 2001 From: Maoyi Xie Date: Thu, 18 Jun 2026 14:03:15 +0800 Subject: [PATCH 11/46] ALSA: caiaq: fix out-of-bounds read in the Traktor Kontrol S4 input parser snd_usb_caiaq_tks4_dispatch() decodes the Traktor Kontrol S4 input stream in fixed 16-byte (TKS4_MSGBLOCK_SIZE) message blocks. On every iteration it advances buf and subtracts the block size while looping on "while (len)". len is urb->actual_length. That value is supplied by the device and is not guaranteed to be a multiple of 16. When a final short block leaves len between 1 and 15, the loop runs once more, reads up to buf[15], and then does "len -= TKS4_MSGBLOCK_SIZE". As len is unsigned this underflows to a huge value. The loop then keeps iterating and walking buf far past the end of the 512-byte ep4_in_buf, reading out of bounds until a bogus block id happens to be hit. Iterate only while a full message block is available. This stops the unsigned underflow and silently drops any trailing partial block, which carries no complete control value anyway. The sibling endpoint-4 parsers are not affected. The Traktor Kontrol X1 and Maschine arms in snd_usb_caiaq_ep4_reply_dispatch() floor urb->actual_length before dispatching. Fixes: 15c5ab607045 ("ALSA: snd-usb-caiaq: Add support for Traktor Kontrol S4") Cc: stable@vger.kernel.org Signed-off-by: Maoyi Xie Link: https://patch.msgid.link/178176259547.3343534.2724779296835237429@maoyixie.com Signed-off-by: Takashi Iwai --- sound/usb/caiaq/input.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 5c70fdf61cc1..2db4d1332df1 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -330,7 +330,7 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, { struct device *dev = caiaqdev_to_dev(cdev); - while (len) { + while (len >= TKS4_MSGBLOCK_SIZE) { unsigned int i, block_id = (buf[0] << 8) | buf[1]; switch (block_id) { From 58fc1275b3f288500ee79a02dbe89ed4197fdc3e Mon Sep 17 00:00:00 2001 From: Maoyi Xie Date: Thu, 18 Jun 2026 14:03:15 +0800 Subject: [PATCH 12/46] ALSA: caiaq: bound the length in the EP1 input parsers snd_caiaq_input_read_erp() and snd_caiaq_input_read_io() can be reached from snd_usb_caiaq_input_dispatch(). They read fixed byte offsets from the reply buffer without checking the reported length. On a short reply they decode stale bytes left from a previous, longer report and feed them to the input layer. This is not an out-of-bounds access. Every offset is a compile-time driver constant. The largest is buf[21] in the Maschine ERP case. The EP1 transfer buffer ep1_in_buf is EP1_BUFSIZE (64) bytes, and the USB core caps actual_length at 64, so a short reply only reads in-bounds stale data. Acting on data the device did not send is still wrong, so bail out per usb_id case when the reply is shorter than the bytes that case consumes. read_erp: AK1 needs 2 bytes, Kore needs 16, Maschine needs 22. read_io: the Kore case needs 5 bytes (buf[4]) and the Traktor Kontrol X1 case needs 7 (buf[5]/buf[6]). The preceding key bit loop is already bounded by "i < len * 8" and is left untouched. snd_caiaq_input_read_analog() and snd_usb_caiaq_maschine_dispatch() are not changed. Their callers already floor the reply length. Suggested-by: Takashi Iwai Signed-off-by: Maoyi Xie Link: https://patch.msgid.link/178176259547.3343534.6659489917322808916@maoyixie.com Signed-off-by: Takashi Iwai --- sound/usb/caiaq/input.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index 2db4d1332df1..eabbf41fdfb2 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -237,12 +237,16 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *cdev, switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): + if (len < 2) + return; i = decode_erp(buf[0], buf[1]); input_report_abs(input_dev, ABS_X, i); input_sync(input_dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + if (len < 16) + return; i = decode_erp(buf[7], buf[5]); input_report_abs(input_dev, ABS_HAT0X, i); i = decode_erp(buf[12], buf[14]); @@ -263,6 +267,8 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *cdev, break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + if (len < 22) + return; /* 4 under the left screen */ input_report_abs(input_dev, ABS_HAT0X, decode_erp(buf[21], buf[20])); input_report_abs(input_dev, ABS_HAT0Y, decode_erp(buf[15], buf[14])); @@ -308,9 +314,13 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *cdev, switch (cdev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + if (len < 5) + return; input_report_abs(cdev->input_dev, ABS_MISC, 255 - buf[4]); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (len < 7) + return; /* rotary encoders */ input_report_abs(cdev->input_dev, ABS_X, buf[5] & 0xf); input_report_abs(cdev->input_dev, ABS_Y, buf[5] >> 4); From 4520fbbaedbeda8861f0f0bfbe50b3bca4deae08 Mon Sep 17 00:00:00 2001 From: Haowen Tu Date: Thu, 18 Jun 2026 15:54:31 +0800 Subject: [PATCH 13/46] ALSA: hda/realtek: Add headset mic quirk for Acer S40-54 Acer S40-54 with ALC256 does not restore headset mic detection properly after S4 resume. After resume, headset plug events may no longer update the headset mic state, leaving the headset microphone unavailable. The system uses subsystem ID 1025:161f. Applying the existing ALC256_FIXUP_ACER_MIC_NO_PRESENCE fixup restores headset mic detection on this machine. Add a machine-specific quirk for this system. Signed-off-by: Haowen Tu Link: https://patch.msgid.link/20260618075431.1116988-1-tuhaowen@uniontech.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 5d09baa6d32e..d209efdb8272 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6807,6 +6807,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x159c, "Acer Nitro 5 AN515-58", ALC287_FIXUP_ACER_MICMUTE_LED), SND_PCI_QUIRK(0x1025, 0x1597, "Acer Nitro 5 AN517-55", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x160e, "Acer PT316-51S", ALC2XX_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x161f, "Acer S40-54", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1640, "Acer Aspire A315-44P", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), 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), From bdb640be82e645e2828731648f485224d0c2587b Mon Sep 17 00:00:00 2001 From: Michael Bommarito Date: Wed, 17 Jun 2026 22:51:25 -0400 Subject: [PATCH 14/46] ALSA: usb-audio: qcom: reject stream disable with no active interface handle_uaudio_stream_req() resolves an interface index with info_idx_from_ifnum(), which returns -EINVAL when no interface matches. The enable branch and the response: cleanup label both guard against a negative index, but the disable branch does not: it forms info = &uadev[pcm_card_num].info[info_idx] and dereferences it. uadev[].info is a pointer allocated only when a stream is first enabled, so a negative info_idx on the disable path is unsafe in two ways: - If the card was never enabled, .info is NULL and &info[-EINVAL] is a wild pointer; reading info->data_ep_pipe faults (kernel oops). - If the card was enabled at least once (.info allocated) and the disable names an interface that does not match, &info[-EINVAL] points before the allocation; info->data_ep_pipe / info->sync_ep_pipe are an out-of-bounds slab read and, when non-zero, an out-of-bounds 4-byte write (both pipe fields are cleared to 0). That is memory corruption, not just a NULL dereference. The request is reachable from unprivileged local userspace over AF_QIPCRTR. Reject a disable request with no resolved interface, matching the guard the enable path already has. Fixes: 326bbc348298a ("ALSA: usb-audio: qcom: Introduce QC USB SND offloading support") Assisted-by: Claude:claude-opus-4-8 Signed-off-by: Michael Bommarito Link: https://patch.msgid.link/20260618025126.1862954-2-michael.bommarito@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/qcom/qc_audio_offload.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index 436d6821c5c9..3b05dadbbeae 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -1642,6 +1642,11 @@ static void handle_uaudio_stream_req(struct qmi_handle *handle, subs->opened = 0; } } else { + if (info_idx < 0) { + ret = -EINVAL; + goto response; + } + info = &uadev[pcm_card_num].info[info_idx]; if (info->data_ep_pipe) { ep = usb_pipe_endpoint(uadev[pcm_card_num].udev, From 3c7af07943b2718087ae791cad450af5cf646d90 Mon Sep 17 00:00:00 2001 From: Michael Bommarito Date: Wed, 17 Jun 2026 22:51:26 -0400 Subject: [PATCH 15/46] ALSA: usb-audio: qcom: clear opened when stream enable fails On enable, subs->opened is set before the service_interval is validated; an invalid interval jumps to the response label without clearing it, so the substream is wedged at -EBUSY until a disable or disconnect. Clear subs->opened on the enable error path. Fixes: 326bbc348298a ("ALSA: usb-audio: qcom: Introduce QC USB SND offloading support") Assisted-by: Claude:claude-opus-4-8 Signed-off-by: Michael Bommarito Link: https://patch.msgid.link/20260618025126.1862954-3-michael.bommarito@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/qcom/qc_audio_offload.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index 3b05dadbbeae..3a586fd16e72 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -1620,8 +1620,13 @@ static void handle_uaudio_stream_req(struct qmi_handle *handle, if (req_msg->service_interval_valid) { ret = get_data_interval_from_si(subs, req_msg->service_interval); - if (ret == -EINVAL) + if (ret == -EINVAL) { + if (req_msg->enable) { + guard(mutex)(&chip->mutex); + subs->opened = 0; + } goto response; + } datainterval = ret; } From e6fa716c9d7248e03cb93f566874bd5709901bcf Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Wed, 17 Jun 2026 16:55:08 +0200 Subject: [PATCH 16/46] ASoC: audio-graph-card2: Drop warning for manually selected DAI formats MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Sometimes the DAI format must be specified in the audio-graph-card2 device tree, so emitting a warning can be misleading. Revert back to emitting no warning. A few examples where automatic format selection might not be applicable: - For DPCM, where the other side of the DAI link is not apparent, no proper selection can actually be made. This can lead to disagreeing formats. - Due to hardware peculiarities, some ostensibly supported formats might not work in practice. In either case, the only correct solution is for the sound card to set the format Link: https://lore.kernel.org/all/87ik7s36k2.wl-kuninori.morimoto.gx@renesas.com/ Signed-off-by: Alvin Šipraga Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20260617145508.327213-1-alvin@pqrs.dk Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 6894bb936cfd..0202ed0ee78e 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -778,18 +778,6 @@ static void graph_link_init(struct simple_util_priv *priv, graph_parse_daifmt(ports_cpu, &daifmt); graph_parse_daifmt(ports_codec, &daifmt); graph_parse_daifmt(lnk, &daifmt); - if (daifmt) { - struct device *dev = simple_priv_to_dev(priv); - - /* - * Recommend to use Auto Select by using .auto_selectable_formats. - * linux/sound/soc/renesas/rcar/core.c can be good sample for it. - * - * One note is that Audio Graph Card2 still keeps compatible to set - * DAI format via DT. - */ - dev_warn_once(dev, "use .auto_selectable_formats on each corresponding CPU/Codec driver"); - } graph_util_parse_link_direction(lnk, &playback_only, &capture_only); graph_util_parse_link_direction(ports_cpu, &playback_only, &capture_only); From 348f69320e4db6ebec6940c81154bec4b9eb275a Mon Sep 17 00:00:00 2001 From: Jean-Louis Colaco Date: Thu, 18 Jun 2026 13:32:02 +0200 Subject: [PATCH 17/46] ALSA: usb-audio: Add quirk for YAMAHA CDS3000 This quirk is identical to the one for the Yamaha Steinberg UR22, here applied to a CD player that also uses the Steinberg USB interface. This quirk is necessary to avoid sporadic "clic" noise when using the DAC of the player. Signed-off-by: Jean-Louis Colaco Link: https://patch.msgid.link/20260618113202.8363-1-jean-louis.colaco@orange.fr Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 97f28c53d201..71444c2898b4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -390,6 +390,20 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150d), + QUIRK_DRIVER_INFO { + /* .vendor_name = "Yamaha", */ + /* .product_name = "CDS3000", */ + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + { QUIRK_DATA_IGNORE(4) }, + QUIRK_COMPOSITE_END + } + } +}, { USB_DEVICE(0x0499, 0x1718), QUIRK_DRIVER_INFO { From b59aff62767bf59ca0c787015c0ddc14f60ab10d Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Thu, 18 Jun 2026 12:24:36 +0200 Subject: [PATCH 18/46] ALSA: emu10k1: Use common error handling code in snd_emu10k1_playback_open() Use an additional label so that a bit of exception handling can be better reused at the end of this function implementation. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Link: https://patch.msgid.link/d709474d-62b0-4f7e-9011-a0f716b35383@web.de Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 9023f3444d20..b8749e0131ad 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1181,19 +1181,17 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_emu10k1_pcm_free_substream; runtime->hw = snd_emu10k1_playback; err = snd_emu10k1_playback_set_constraints(runtime); - if (err < 0) { - kfree(epcm); - return err; - } + if (err < 0) + goto free_epcm; + if (emu->card_capabilities->emu_model) sample_rate = emu->emu1010.word_clock; else sample_rate = 48000; err = snd_pcm_hw_rule_noresample(runtime, sample_rate); - if (err < 0) { - kfree(epcm); - return err; - } + if (err < 0) + goto free_epcm; + mix = &emu->pcm_mixer[substream->number]; for (i = 0; i < 8; i++) mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i; @@ -1204,6 +1202,10 @@ static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream) mix->epcm = epcm; snd_emu10k1_pcm_mixer_notify(emu, substream->number, 1); return 0; + +free_epcm: + kfree(epcm); + return err; } static int snd_emu10k1_playback_close(struct snd_pcm_substream *substream) From f199c8a8bdd54296d3458777e70fe82a78bd9817 Mon Sep 17 00:00:00 2001 From: Cen Zhang Date: Fri, 19 Jun 2026 01:00:10 +0800 Subject: [PATCH 19/46] ALSA: usb-audio: Kill MIDI 2.0 URBs before freeing endpoints MIDI 2.0 input URBs are started during snd_usb_midi_v2_create(). A later setup failure can still jump to snd_usb_midi_v2_free(), which currently frees each endpoint and its coherent URB buffers without first stopping the submitted URBs. A completion can then dereference the embedded URB context and endpoint state after they have been freed, or try to resubmit from the stale endpoint. This was observed as a KASAN slab-use-after-free in input_urb_complete(). The buggy scenario involves two paths, with each column showing the order within that path: probe error path: USB completion path: 1. start_input_streams() submits 1. The HCD still owns a input URBs. submitted input URB. 2. A later setup helper returns 2. input_urb_complete() runs an error. with urb->context in ep. 3. snd_usb_midi_v2_free() frees 3. The completion reads ep endpoint storage and URB buffers. state and can requeue URBs. Make the endpoint destructor follow the same teardown ordering used for disconnect when the endpoint has not already been disconnected: publish ep->disconnected, kill the URBs synchronously, and drain the endpoint before freeing URB buffers and endpoint storage. The guard avoids repeating the stop sequence after the normal snd_usb_midi_v2_disconnect_all() path, while still synchronizing the direct MIDI 2.0 create-error free path. Validation reproduced this kernel report: BUG: KASAN: slab-use-after-free in input_urb_complete+0x37/0x1b0 Workqueue: usb_hub_wq hub_event RIP: 0010:_raw_spin_unlock_irq+0x2e/0x50 Read of size 8 Call trace: dump_stack_lvl+0x77/0xb0 print_report+0xce/0x5f0 input_urb_complete+0x37/0x1b0 (sound/usb/midi2.c:186) srso_alias_return_thunk+0x5/0xfbef5 __virt_addr_valid+0x19f/0x330 kasan_report+0xe0/0x110 __usb_hcd_giveback_urb+0x112/0x1d0 dummy_timer+0xaaa/0x19a0 lock_is_held_type+0x9a/0x110 __lock_acquire+0x467/0x28b0 mark_held_locks+0x40/0x70 _raw_spin_unlock_irqrestore+0x44/0x60 lockdep_hardirqs_on_prepare+0xbb/0x1a0 __hrtimer_run_queues+0x101/0x520 hrtimer_run_softirq+0xd0/0x130 handle_softirqs+0x15b/0x670 __irq_exit_rcu+0xd0/0x170 irq_exit_rcu+0xe/0x20 sysvec_apic_timer_interrupt+0x6c/0x80 asm_sysvec_apic_timer_interrupt+0x1a/0x20 Fixes: d9c99876868c ("ALSA: usb-audio: Create UMP blocks from USB MIDI GTBs") Assisted-by: Codex:gpt-5.5 Signed-off-by: Cen Zhang Link: https://patch.msgid.link/20260618170010.191433-1-zzzccc427@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/midi2.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index 04aeb9052f13..3ec633291772 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -470,6 +470,11 @@ static int create_midi2_endpoint(struct snd_usb_midi2_interface *umidi, static void free_midi2_endpoint(struct snd_usb_midi2_endpoint *ep) { list_del(&ep->list); + if (!ep->disconnected) { + ep->disconnected = 1; + kill_midi_urbs(ep, false); + drain_urb_queue(ep); + } free_midi_urbs(ep); kfree(ep); } From f56521ab6f76faeaa5a524320898835454e3bc31 Mon Sep 17 00:00:00 2001 From: Ramcharan Rajpurohit Date: Fri, 19 Jun 2026 11:54:35 +0530 Subject: [PATCH 20/46] ALSA: hda/realtek: Enable mute LED on HP EliteBook 840 G6 The HP EliteBook 840 G6 (PCI SSID 103c:854d) has an ALC215 codec whose mic-mute and audio-mute LEDs are wired to the same GPIOs as the already supported EliteBook 830 G6 and 840 G7. Without a matching quirk the LEDs are never registered, so the front-panel mic-mute LED stays permanently lit and does not track the mute state. Apply ALC285_FIXUP_HP_GPIO_LED, mirroring the sibling EliteBook entries. With this fixup the codec registers an "hda::micmute" LED class device and the LED correctly follows the capture-mute state. This was verified on the affected machine by forcing the same fixup at runtime via snd_sof_intel_hda_generic.hda_model=103c:8548, which made the LED work as expected. Signed-off-by: Ramcharan Rajpurohit Link: https://patch.msgid.link/20260619062435.26256-1-b23ci1032@iitj.ac.in 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 d209efdb8272..2df491335a76 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6987,6 +6987,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8548, "HP EliteBook x360 830 G6", ALC285_FIXUP_HP_GPIO_LED), 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), 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), From b70f007a9fc665ee988683fd5085ab34e2c10ad3 Mon Sep 17 00:00:00 2001 From: Chris Aherin Date: Sun, 21 Jun 2026 19:12:10 -0500 Subject: [PATCH 21/46] ALSA: hda/realtek: Add quirk for Lenovo Yoga 7 16IAP7 The Yoga 7 16IAP7 (board LNVNB161216, codec SSID 17aa:386a) has pin complex 0x17 (bass speakers) wrongly reported as unconnected, causing only one of four speaker pins (0x14) to be configured and resulting in mono/tinny audio. SOF corrupts the PCI subsystem ID to 17aa:0000, preventing SND_PCI_QUIRK from matching. HDA_CODEC_QUIRK is used instead, which matches against codec->core.subsystem_id read directly from the HDA codec register and unaffected by the SOF bug. Applies ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, the same fixup used for the Yoga 7 14IAL7, which corrects pin 0x17's default configuration and enables both speaker pairs. Signed-off-by: Chris Aherin Link: https://patch.msgid.link/20260622001210.20553-1-chrisaherin@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 2df491335a76..c83c1654ed45 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7755,6 +7755,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + HDA_CODEC_QUIRK(0x17aa, 0x386a, "Lenovo Yoga 7 16IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_SPEAKER2_TO_DAC1), HDA_CODEC_QUIRK(0x17aa, 0x38a8, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), /* this must match before PCI SSID 17aa:386f below */ From 6efb1897209ab50940c58384d15a68fd4212821c Mon Sep 17 00:00:00 2001 From: Dirk Su Date: Mon, 22 Jun 2026 15:20:04 +0800 Subject: [PATCH 22/46] ALSA: hda/realtek: Add LED fixup for HP EliteBook 6 G2i Laptops The HP EliteBook 6 G2i laptops requires specific LED control method ALC236_FIXUP_HP_GPIO_LED to work Signed-off-by: Dirk Su Link: https://patch.msgid.link/20260622072019.56351-1-dirk.su@canonical.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index c83c1654ed45..e88ca720f4e5 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7328,6 +7328,10 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8f0e, "HP ZBook X G2i 16W", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8f2d, "HP Auster 14", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8f2e, "HP Auster 14", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8f37, "HP EliteBook 6 G2i", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8f38, "HP EliteBook 6 G2i", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8f39, "HP EliteBook 6 G2i", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8f3a, "HP EliteBook 6 G2i", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8f3c, "HP EliteBook 6 G2a", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8f3d, "HP EliteBook 6 G2a", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8f40, "HP ZBook 8 G2a 14", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), From 493765b8e922a506e09e22e80b6cc9ff05e8295b Mon Sep 17 00:00:00 2001 From: Aaron Erhardt Date: Tue, 19 May 2026 17:49:48 +0200 Subject: [PATCH 23/46] ALSA: hda/realtek: Fix noisy mic for Clevo V6xxAW Add a PCI quirk to reduce the volume of the internal microphone to prevent extremely noisy signal. Signed-off-by: Aaron Erhardt Signed-off-by: Werner Sembach Cc: stable@vger.kernel.org Link: https://patch.msgid.link/20260519155047.106096-1-wse@tuxedocomputers.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 e88ca720f4e5..0d932f6ae3e8 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7603,6 +7603,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5700, "Clevo X560WN[RST]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x6480, "Clevo V6xxAW", ALC245_FIXUP_CLEVO_NOISY_MIC), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), From cf81b260916c92cf610c907d41627a175e86e37d Mon Sep 17 00:00:00 2001 From: Ahmad Fatoum Date: Wed, 17 Jun 2026 16:47:53 +0200 Subject: [PATCH 24/46] ASoC: cs530x: Fix expected MCLK rates for CS5302/4/8 When this driver was first added, it accepted rates of 24.56 MHz and 22.572 MHz for the MCLK when PLL bypass is enabled. These rates seem to have no basis in the datasheets and were thus replaced with 45.1584 MHz and 49.152 MHz, respectively, in commit e7ab858390f2 ("ASoC: cs530x: Correct MCLK reference frequency values"). While the new rates are indeed correct for the CS4xxx ICs[0][1][2][3], they are incorrect for the CS530x ICs the driver was originally written to support as the MCLK frequencies are halved there[4][5][6]. Fix this by checking against the correct type-appropriate rates. While at it, drop the CS530X_SYSCLK_REF_* macros. They arguably confuse more than they help, especially as they are not applicable to the cs5302/4/8. [0]: https://statics.cirrus.com/pubs/proDatasheet/CS4282P_DS1318F1.pdf [1]: https://statics.cirrus.com/pubs/proDatasheet/CS4302P_DS1315F1.pdf [2]: https://statics.cirrus.com/pubs/proDatasheet/CS4304P_DS1316F1.pdf [3]: https://statics.cirrus.com/pubs/proDatasheet/CS4308P_DS1317F1.pdf [4]: https://statics.cirrus.com/pubs/proDatasheet/CS5302P_DS1312F1.pdf [5]: https://statics.cirrus.com/pubs/proDatasheet/CS5304P_DS1313F1.pdf [6]: https://statics.cirrus.com/pubs/proDatasheet/CS5308P_DS1314F1.pdf Fixes: 2884c29152c0 ("ASoC: cs530x: Support for cs530x ADCs") Signed-off-by: Ahmad Fatoum Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260617-cs530x-mclk-v1-1-0215b5f1a0a4@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/cs530x.c | 29 ++++++++++++++++++++++++----- sound/soc/codecs/cs530x.h | 6 ------ 2 files changed, 24 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/cs530x.c b/sound/soc/codecs/cs530x.c index 18b5ff75feec..2c7e33135911 100644 --- a/sound/soc/codecs/cs530x.c +++ b/sound/soc/codecs/cs530x.c @@ -1093,6 +1093,29 @@ static int cs530x_component_probe(struct snd_soc_component *component) return 0; } +static bool cs530x_mclk_freq_is_valid(struct cs530x_priv *cs530x, + unsigned int freq) +{ + /* + * All these chips support 48 kHz- and 44.1 kHz-related sample rates, + * but they differ in what MCLK frequency is required for achieving + * the sample rate. + */ + switch (cs530x->devtype) { + case CS4282: + case CS4302: + case CS4304: + case CS4308: + return freq == 49152000 || freq == 45158400; + case CS5302: + case CS5304: + case CS5308: + return freq == 24576000 || freq == 22579200; + } + + return false; +} + static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir) { @@ -1101,11 +1124,7 @@ static int cs530x_set_sysclk(struct snd_soc_component *component, int clk_id, switch (source) { case CS530X_SYSCLK_SRC_MCLK: - switch (freq) { - case CS530X_SYSCLK_REF_45_1MHZ: - case CS530X_SYSCLK_REF_49_1MHZ: - break; - default: + if (!cs530x_mclk_freq_is_valid(cs530x, freq)) { dev_err(component->dev, "Invalid MCLK source rate %d\n", freq); return -EINVAL; } diff --git a/sound/soc/codecs/cs530x.h b/sound/soc/codecs/cs530x.h index 1e2f6a7a589c..18aa4dfd0c86 100644 --- a/sound/soc/codecs/cs530x.h +++ b/sound/soc/codecs/cs530x.h @@ -200,12 +200,6 @@ /* IN_VOL_CTL5 and OUT_VOL_CTL5 */ #define CS530X_INOUT_VU BIT(0) -/* MCLK Reference Source Frequency */ -/* 41KHz related */ -#define CS530X_SYSCLK_REF_45_1MHZ 45158400 -/* 48KHz related */ -#define CS530X_SYSCLK_REF_49_1MHZ 49152000 - /* System Clock Source */ #define CS530X_SYSCLK_SRC_MCLK 0 #define CS530X_SYSCLK_SRC_PLL 1 From e0ecb324246be9cf3a0689346a658e48a38546b2 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Tue, 23 Jun 2026 02:59:04 +0930 Subject: [PATCH 25/46] ALSA: FCP: Add Focusrite ISA C8X support Add USB PID 0x821e to the list of devices handled by the Focusrite Control Protocol (FCP) driver. Cc: stable@vger.kernel.org Signed-off-by: Geoffrey D. Bennett Link: https://patch.msgid.link/ajlw4HK+2RSW3nUl@m.b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 628f841b04aa..f6f2bc8e97a7 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -4526,6 +4526,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x821b): /* Focusrite Scarlett 16i16 4th Gen */ case USB_ID(0x1235, 0x821c): /* Focusrite Scarlett 18i16 4th Gen */ case USB_ID(0x1235, 0x821d): /* Focusrite Scarlett 18i20 4th Gen */ + case USB_ID(0x1235, 0x821e): /* Focusrite ISA C8X */ err = snd_fcp_init(mixer); break; From 5714c8359f4fc171ab8e0bd0dfc4c61fb36e1db6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 23 Jun 2026 11:18:10 +0100 Subject: [PATCH 26/46] soundwire: Move wait for initialisation helper to header As SoundWire devices tend to enumerate on the bus after probe, drivers frequently need to wait for the device to initialise from common driver code. The common system is to split drivers into a core module and then a module for each communication bus. These two facts tend to cause Kconfig issues, the issue tends to be when SOUNDWIRE=m and DRIVER_I2C=y, this usually selects DRIVER=y. The driver code then wants to call sdw_slave_wait_for_init(), but this results in calling a module function from built in code. A depends on SOUNDWIRE | !SOUNDWIRE could be added to the end driver but this seems slightly off as it adds a lot of counter intuitive depends. A simpler solution is to make sdw_slave_wait_for_init() a static inline function. As part of doing this add a check for the slave device being NULL acknowledging that this is likely called from code that is shared between control buses. It does require dropping the call to sdw_show_ping_status() but this can be added back in end drivers that used it originally. Currently this is causing rand config issues on RT5682 and will soon also cause similar problems on cs42l43. Acked-by: Vinod Koul Acked-by: Arnd Bergmann Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260623101814.24044-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/soundwire/bus.c | 28 -------------------------- include/linux/soundwire/sdw.h | 38 +++++++++++++++++++++++++++-------- 2 files changed, 30 insertions(+), 36 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index b7bdf19ebb42..fe5316d93fef 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1372,34 +1372,6 @@ int sdw_slave_get_current_bank(struct sdw_slave *slave) } EXPORT_SYMBOL_GPL(sdw_slave_get_current_bank); -/** - * sdw_slave_wait_for_init - Wait for device initialisation - * @slave: Pointer to the SoundWire peripheral. - * @timeout_ms: Timeout in milliseconds. - * - * Wait for a peripheral device to enumerate and be initialised by the - * SoundWire core. - * - * Return: Zero on success, and a negative error code on failure. - */ -int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms) -{ - unsigned long time; - - time = wait_for_completion_timeout(&slave->initialization_complete, - msecs_to_jiffies(timeout_ms)); - if (!time) { - dev_err(&slave->dev, "Initialization not complete\n"); - sdw_show_ping_status(slave->bus, true); - return -ETIMEDOUT; - } - - slave->unattach_request = 0; - - return 0; -} -EXPORT_SYMBOL_GPL(sdw_slave_wait_for_init); - static int sdw_slave_set_frequency(struct sdw_slave *slave) { int scale_index; diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index a46cbaec5949..b484784e2690 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -11,6 +11,7 @@ #include #include #include +#include #include #include #include @@ -1093,8 +1094,6 @@ int sdw_slave_get_current_bank(struct sdw_slave *sdev); int sdw_slave_get_scale_index(struct sdw_slave *slave, u8 *base); -int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms); - /* messaging and data APIs */ int sdw_read(struct sdw_slave *slave, u32 addr); int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); @@ -1138,12 +1137,6 @@ static inline int sdw_slave_get_current_bank(struct sdw_slave *sdev) return -EINVAL; } -static inline int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms) -{ - WARN_ONCE(1, "SoundWire API is disabled"); - return -EINVAL; -} - /* messaging and data APIs */ static inline int sdw_read(struct sdw_slave *slave, u32 addr) { @@ -1207,4 +1200,33 @@ static inline int sdw_update_no_pm(struct sdw_slave *slave, u32 addr, u8 mask, u #endif /* CONFIG_SOUNDWIRE */ +/** + * sdw_slave_wait_for_init - Wait for device initialisation + * @slave: Pointer to the SoundWire peripheral. + * @timeout_ms: Timeout in milliseconds. + * + * Wait for a peripheral device to enumerate and be initialised by the + * SoundWire core. + * + * Return: Zero on success, and a negative error code on failure. + */ +static inline int sdw_slave_wait_for_init(struct sdw_slave *slave, int timeout_ms) +{ + unsigned long time; + + if (!slave) + return 0; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(timeout_ms)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete\n"); + return -ETIMEDOUT; + } + + slave->unattach_request = 0; + + return 0; +} + #endif /* __SOUNDWIRE_H */ From ce52450319fb0ee122e5be5ec8dfb888ee1e0237 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 23 Jun 2026 11:18:11 +0100 Subject: [PATCH 27/46] ASoC: es9356: Add back local call to sdw_show_ping_status() As the core no longer calls this debug helper add it back to the drivers that originally called it. Acked-by: Arnd Bergmann Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260623101814.24044-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/es9356.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es9356.c b/sound/soc/codecs/es9356.c index 670e918b56a4..8db81d574624 100644 --- a/sound/soc/codecs/es9356.c +++ b/sound/soc/codecs/es9356.c @@ -1111,8 +1111,10 @@ static int es9356_sdca_dev_resume(struct device *dev) es9356->disable_irq = false; ret = sdw_slave_wait_for_init(slave, es9356_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(es9356->regmap, false); regcache_sync(es9356->regmap); From 1921303a1d2f26ae70446aa18fb218767bddd913 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 23 Jun 2026 11:18:12 +0100 Subject: [PATCH 28/46] ASoC: max98373: Add back local call to sdw_show_ping_status() As the core no longer calls this debug helper add it back to the drivers that originally called it. Acked-by: Arnd Bergmann Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260623101814.24044-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-sdw.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 6829fa07c9ec..7a42052dc051 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -272,8 +272,10 @@ static int max98373_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, MAX98373_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); From ea9ff3b7bcfbcd1e61d34590c7c632005ef3d9aa Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 23 Jun 2026 11:18:13 +0100 Subject: [PATCH 29/46] ASoC: ti: Add back local call to sdw_show_ping_status() As the core no longer calls this debug helper add it back to the drivers that originally called it. Acked-by: Arnd Bergmann Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260623101814.24044-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/tac5xx2-sdw.c | 4 +++- sound/soc/codecs/tas2783-sdw.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tac5xx2-sdw.c b/sound/soc/codecs/tac5xx2-sdw.c index bb12cfb6da12..ace06f5ab58c 100644 --- a/sound/soc/codecs/tac5xx2-sdw.c +++ b/sound/soc/codecs/tac5xx2-sdw.c @@ -1445,8 +1445,10 @@ static s32 tac5xx2_sdca_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, TAC5XX2_PROBE_TIMEOUT_MS); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(tac_dev->regmap, false); regcache_mark_dirty(tac_dev->regmap); diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index 7d70e7e3f24f..1127ea59b5e4 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -1083,8 +1083,10 @@ static s32 tas2783_sdca_dev_resume(struct device *dev) int ret; ret = sdw_slave_wait_for_init(slave, TAS2783_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(tas_dev->regmap, false); regcache_sync(tas_dev->regmap); From 6540b9d9ccc32ad1546dcc7b4d4bcbb68c667714 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 23 Jun 2026 11:18:14 +0100 Subject: [PATCH 30/46] ASoC: realtek: Add back local call to sdw_show_ping_status() As the core no longer calls this debug helper add it back to the drivers that originally called it. Acked-by: Arnd Bergmann Signed-off-by: Charles Keepax Link: https://patch.msgid.link/20260623101814.24044-6-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1017-sdca-sdw.c | 4 +++- sound/soc/codecs/rt1308-sdw.c | 4 +++- sound/soc/codecs/rt1316-sdw.c | 4 +++- sound/soc/codecs/rt5682-sdw.c | 4 +++- sound/soc/codecs/rt700-sdw.c | 4 +++- sound/soc/codecs/rt711-sdca-sdw.c | 4 +++- sound/soc/codecs/rt712-sdca-dmic.c | 4 +++- sound/soc/codecs/rt712-sdca-sdw.c | 4 +++- sound/soc/codecs/rt715-sdca-sdw.c | 4 +++- sound/soc/codecs/rt715-sdw.c | 4 +++- sound/soc/codecs/rt721-sdca-sdw.c | 4 +++- sound/soc/codecs/rt722-sdca-sdw.c | 4 +++- 12 files changed, 36 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index d62e8a253676..91d3d43cd998 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -779,8 +779,10 @@ static int rt1017_sdca_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT1017_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt1017->regmap, false); regcache_sync(rt1017->regmap); diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 39e06a3a7560..60e5040b6dd9 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -774,8 +774,10 @@ static int rt1308_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT1308_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt1308->regmap, false); regcache_sync_region(rt1308->regmap, 0xc000, 0xcfff); diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 1828fd9d5af6..5e8eda6a5f7f 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -751,8 +751,10 @@ static int rt1316_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT1316_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt1316->regmap, false); regcache_sync(rt1316->regmap); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index ec2a35a0cacd..dec8c2147d68 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -769,8 +769,10 @@ static int rt5682_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, RT5682_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt5682->sdw_regmap, false); regcache_cache_only(rt5682->regmap, false); diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 30fcca210f05..6bc636c86f42 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -528,8 +528,10 @@ static int rt700_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT700_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt700->regmap, false); regcache_sync_region(rt700->regmap, 0x3000, 0x8fff); diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index a8164fc3979a..461315844ba9 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -454,8 +454,10 @@ static int rt711_sdca_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, RT711_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt711->regmap, false); regcache_sync(rt711->regmap); diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 4c5c2f5ba5ed..8b7d50a80ff9 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -911,8 +911,10 @@ static int rt712_sdca_dmic_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT712_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt712->regmap, false); regcache_sync(rt712->regmap); diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 581732180473..2787524c796e 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -467,8 +467,10 @@ static int rt712_sdca_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, RT712_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt712->regmap, false); regcache_sync(rt712->regmap); diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 4b9815b5628d..fabd21bbbe5b 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -230,8 +230,10 @@ static int rt715_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT715_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt715->regmap, false); regcache_sync_region(rt715->regmap, diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 7f83a8f1a06e..a4a3945522e8 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -507,8 +507,10 @@ static int rt715_dev_resume(struct device *dev) return 0; ret = sdw_slave_wait_for_init(slave, RT715_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt715->regmap, false); regcache_sync_region(rt715->regmap, 0x3000, 0x8fff); diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c index 58606209316a..02df04a0ddad 100644 --- a/sound/soc/codecs/rt721-sdca-sdw.c +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -505,8 +505,10 @@ static int rt721_sdca_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, RT721_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt721->regmap, false); regcache_sync(rt721->regmap); diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 0f76492ff915..284900933ebf 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -552,8 +552,10 @@ static int rt722_sdca_dev_resume(struct device *dev) } ret = sdw_slave_wait_for_init(slave, RT722_PROBE_TIMEOUT); - if (ret) + if (ret) { + sdw_show_ping_status(slave->bus, true); return ret; + } regcache_cache_only(rt722->regmap, false); regcache_sync(rt722->regmap); From 1ce42a11bed134903e352010a01fa53073a6b395 Mon Sep 17 00:00:00 2001 From: HyeongJun An Date: Tue, 23 Jun 2026 20:05:26 +0900 Subject: [PATCH 31/46] ASoC: SDCA: Validate written enum value in ge_put_enum_double() ge_put_enum_double() passes the user-supplied enumeration index item[0] to snd_soc_enum_item_to_val() without checking it against the number of items in the enum: ret = snd_soc_enum_item_to_val(e, item[0]); snd_soc_enum_item_to_val() indexes the heap-allocated e->values[] array with that index (e->values is set from a devm_kcalloc() of e->items entries), so a control write with an out-of-range item[0] reads past the end of the values buffer. The bounds check in snd_soc_dapm_put_enum_double() only runs afterwards, so it does not prevent the read here. Reject an out-of-range item before using it, matching the other enum put handlers. This issue was pointed out by the Sashiko AI review bot while reviewing a related enum-validation series: https://lore.kernel.org/all/20260609125735.CEB651F00893@smtp.kernel.org/ Fixes: 812ff1baa764 ("ASoC: SDCA: Limit values user can write to Selected Mode") Signed-off-by: HyeongJun An Reviewed-by: Charles Keepax Link: https://patch.msgid.link/20260623110526.813217-1-sammiee5311@gmail.com Signed-off-by: Mark Brown --- sound/soc/sdca/sdca_asoc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sdca/sdca_asoc.c b/sound/soc/sdca/sdca_asoc.c index e76afa396b0a..b4dedba719dc 100644 --- a/sound/soc/sdca/sdca_asoc.c +++ b/sound/soc/sdca/sdca_asoc.c @@ -160,6 +160,9 @@ static int ge_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int reg = e->reg; int ret; + if (item[0] >= e->items) + return -EINVAL; + reg &= ~SDW_SDCA_CTL_CSEL(0x3F); reg |= SDW_SDCA_CTL_CSEL(SDCA_CTL_GE_DETECTED_MODE); From e26bb459d0f3dad83c6a31d5e4480e60760c262b Mon Sep 17 00:00:00 2001 From: Bartosz Juraszewski Date: Mon, 22 Jun 2026 20:27:33 +0200 Subject: [PATCH 32/46] ASoC: tas2783: Update loaded firmware names to linux-firmware 20260519 In linux-firmware commit from 2026-05-19 `2f90f4fe5c67f51a8410907a...` `ASoC: tas2783: Add Firmware files for tas2783A projects` by Baojun Xu 138 new firmware files for tas2783 were added, none of which are loaded by the kernel. Kernel expects files to be named with the following convention: "%04X-%1X-%1X.bin". However the added firmware files follow "-0x%1X.bin" naming instead with `0x` hex prefix, which fails to load resulting in following dmesg log: slave-tas2783 sdw:0:1:0102:0000:01:8: Direct firmware load for 1714-1-8.bin failed with error -2 slave-tas2783 sdw:0:1:0102:0000:01:8: Failed to read fw binary 1714-1-8.bin slave-tas2783 sdw:0:1:0102:0000:01:b: Direct firmware load for 1714-1-B.bin failed with error -2 slave-tas2783 sdw:0:1:0102:0000:01:b: Failed to read fw binary 1714-1-B.bin slave-tas2783 sdw:0:1:0102:0000:01:8: error playback without fw download slave-tas2783 sdw:0:1:0102:0000:01:8: ASoC error (-22): at snd_soc_dai_hw_params() on tas2783-codec This same commit removes all 22 symlinks from WHENCE, that used naming without the '0x' prefix to only 6 prevoiusly existing .bin files. This patch adds `0x` prefix explicitly to the generated firmware name allowing file to successfully load. In case prefixed firmware is missing due to out of date linux-firmware, we set the fallback flag and attempt to load firmware again based on the old file names. This prefix change results in functioning firmware loading on ASUS ProArt PX13 HN7306EAC, which uses 1714-1-0x8.bin and 1714-1-0xB.bin firmware files. Tested on top of 7.1 and next-20260619 with SND_SOC_AMD_ACP7X set to no. Signed-off-by: Bartosz Juraszewski Link: https://patch.msgid.link/20260622182733.23947-1-bjuraszewski@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2783-sdw.c | 73 +++++++++++++++++++++++----------- 1 file changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c index 1127ea59b5e4..3d0b116544cc 100644 --- a/sound/soc/codecs/tas2783-sdw.c +++ b/sound/soc/codecs/tas2783-sdw.c @@ -100,6 +100,8 @@ struct tas2783_prv { wait_queue_head_t fw_wait; bool fw_dl_task_done; bool fw_dl_success; + /* use fallback fw name */ + bool fw_use_fallback; }; static const struct reg_default tas2783_reg_default[] = { @@ -740,11 +742,19 @@ static void tas2783_fw_ready(const struct firmware *fmw, void *context) goto out; } + /* firmware binary not found*/ if (!fmw || !fmw->data) { - /* firmware binary not found*/ - dev_err(tas_dev->dev, - "Failed to read fw binary %s\n", - tas_dev->rca_binaryname); + if (!tas_dev->fw_use_fallback) { + tas_dev->fw_use_fallback = true; + dev_info(tas_dev->dev, + "Failed to read preferred fw binary: %s, attempting fallback binary load\n", + tas_dev->rca_binaryname); + } else { + dev_err(tas_dev->dev, + "Failed to read fallback fw binary %s\n", + tas_dev->rca_binaryname); + } + ret = -EINVAL; goto out; } @@ -1105,13 +1115,16 @@ static void tas_generate_fw_name(struct sdw_slave *slave, char *name, size_t siz bool pci_found = false; #if IS_ENABLED(CONFIG_PCI) struct device *dev = &slave->dev; + struct tas2783_prv *tas_dev = dev_get_drvdata(&slave->dev); struct pci_dev *pci = NULL; + const char *fw_uid_prefix = tas_dev->fw_use_fallback ? "" : "0x"; for (; dev; dev = dev->parent) { if (dev->bus == &pci_bus_type) { pci = to_pci_dev(dev); - scnprintf(name, size, "%04X-%1X-%1X.bin", - pci->subsystem_device, bus->link_id, unique_id); + scnprintf(name, size, "%04X-%1X-%s%1X.bin", + pci->subsystem_device, bus->link_id, + fw_uid_prefix, unique_id); pci_found = true; break; } @@ -1123,28 +1136,15 @@ static void tas_generate_fw_name(struct sdw_slave *slave, char *name, size_t siz bus->link_id, unique_id); } -static s32 tas_io_init(struct device *dev, struct sdw_slave *slave) +static s32 tas_fw_load(struct tas2783_prv *tas_dev, struct sdw_slave *slave) { - struct tas2783_prv *tas_dev = dev_get_drvdata(dev); s32 ret; u8 unique_id = tas_dev->sdw_peripheral->id.unique_id; - if (tas_dev->hw_init) - return 0; - - tas_dev->fw_dl_task_done = false; - tas_dev->fw_dl_success = false; - - ret = regmap_write(tas_dev->regmap, TAS2783_SW_RESET, 0x1); - if (ret) { - dev_err(dev, "sw reset failed, err=%d", ret); - return ret; - } - usleep_range(2000, 2200); - tas_generate_fw_name(slave, tas_dev->rca_binaryname, sizeof(tas_dev->rca_binaryname)); + tas_dev->fw_dl_task_done = false; ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, tas_dev, tas2783_fw_ready); @@ -1159,8 +1159,35 @@ static s32 tas_io_init(struct device *dev, struct sdw_slave *slave) msecs_to_jiffies(TIMEOUT_FW_DL_MS)); if (!ret) { dev_err(tas_dev->dev, "fw request, wait_event timeout\n"); - ret = -EAGAIN; - } else { + return -EAGAIN; + } + + return 0; +} + +static s32 tas_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tas2783_prv *tas_dev = dev_get_drvdata(dev); + s32 ret; + + if (tas_dev->hw_init) + return 0; + + tas_dev->fw_dl_success = false; + + ret = regmap_write(tas_dev->regmap, TAS2783_SW_RESET, 0x1); + if (ret) { + dev_err(dev, "sw reset failed, err=%d", ret); + return ret; + } + usleep_range(2000, 2200); + + tas_dev->fw_use_fallback = false; + ret = tas_fw_load(tas_dev, slave); + if (!ret && tas_dev->fw_use_fallback) + ret = tas_fw_load(tas_dev, slave); + + if (!ret) { if (tas_dev->sa_func_data) ret = sdca_regmap_write_init(dev, tas_dev->regmap, tas_dev->sa_func_data); From d0c415f0076b71b956f62ff6f31de885f0fa2489 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 23 Jun 2026 18:25:14 +0800 Subject: [PATCH 33/46] ASoC: rt5575: Use __le32 for SPI burst write address The addr field in the SPI burst write buffer represents on-wire little-endian data. Define it as __le32 so that the assignment of cpu_to_le32() is type correct and avoids sparse endian warnings. Closes: https://lore.kernel.org/oe-kbuild-all/202606230139.rFZUVpCa-lkp@intel.com/ Fixes: 420739112e95 ("ASoC: rt5575: Add the codec driver for the ALC5575") Reported-by: kernel test robot Signed-off-by: Oder Chiou Link: https://patch.msgid.link/20260623102514.2422990-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5575-spi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c index 9a349965435b..d5b3a57c8866 100644 --- a/sound/soc/codecs/rt5575-spi.c +++ b/sound/soc/codecs/rt5575-spi.c @@ -17,7 +17,7 @@ struct rt5575_spi_burst_write { u8 cmd; - u32 addr; + __le32 addr; u8 data[RT5575_SPI_BUF_LEN]; u8 dummy; } __packed; From 435990e25bf1f4af3e6df12a6fbfd1f7ba4a97d4 Mon Sep 17 00:00:00 2001 From: HyeongJun An Date: Wed, 24 Jun 2026 08:38:40 +0900 Subject: [PATCH 34/46] ALSA: seq: Fix uninitialised heap leak in snd_seq_event_dup() snd_seq_event_dup() copies an incoming event into a pool cell and, in the UMP-enabled build, clears the trailing cell->ump.raw.extra word that the memcpy() did not cover. The guard deciding whether to clear it compares the copied size against sizeof(cell->event): memcpy(&cell->ump, event, size); if (size < sizeof(cell->event)) cell->ump.raw.extra = 0; For a legacy (non-UMP) event, size == sizeof(struct snd_seq_event) == sizeof(cell->event), so the condition is false and the extra word keeps stale data. The cell pool is allocated with kvmalloc() (not zeroed) and cells are reused via a free list, so that word holds uninitialised heap or leftover event data. When such a cell is delivered to a UMP client (client->midi_version > 0) that set SNDRV_SEQ_FILTER_NO_CONVERT -- so the legacy event reaches it unconverted -- snd_seq_read() reads it out as the larger struct snd_seq_ump_event and copies the stale word to user space, a 4-byte kernel heap infoleak to an unprivileged /dev/snd/seq client. Compare against sizeof(cell->ump) instead, so the trailing word is zeroed for every event shorter than the UMP cell. Fixes: 46397622a3fa ("ALSA: seq: Add UMP support") Cc: stable@vger.kernel.org Assisted-by: Claude:claude-opus-4-8 Signed-off-by: HyeongJun An Link: https://patch.msgid.link/20260623233841.853326-1-sammiee5311@gmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ca9f6db0022c..209b08c2a940 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -364,7 +364,7 @@ int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event, size = snd_seq_event_packet_size(event); memcpy(&cell->ump, event, size); #if IS_ENABLED(CONFIG_SND_SEQ_UMP) - if (size < sizeof(cell->event)) + if (size < sizeof(cell->ump)) cell->ump.raw.extra = 0; #endif From 83d53eca7e5512b53ecef6cf67ea8cbfb595516e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig=20=28The=20Capable=20Hub=29?= Date: Wed, 24 Jun 2026 10:37:07 +0200 Subject: [PATCH 35/46] ASoC: rockchip: rockchip_sai: #include explicitly MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently that header is only included via: -> -> which doesn't look reliable, still more in the presence of the comment: /* For the current users of sound/soc.h to avoid build issues */ in . Signed-off-by: Uwe Kleine-König (The Capable Hub) Reviewed-by: Nicolas Frattaroli Link: https://patch.msgid.link/20260624083708.254517-2-u.kleine-koenig@baylibre.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_sai.c b/sound/soc/rockchip/rockchip_sai.c index ed393e5034a4..585e89f61f0d 100644 --- a/sound/soc/rockchip/rockchip_sai.c +++ b/sound/soc/rockchip/rockchip_sai.c @@ -11,6 +11,7 @@ #include #include #include +#include #include #include #include From a7ea04d1ad39d60da397de75b503062ad5fa562b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 23 Jun 2026 14:58:21 +0100 Subject: [PATCH 36/46] ASoC: soc-core: Don't fail if device_link could not be created If device_link_add() fails in snd_soc_bind_card() just skip that driver pair and carry on. This means that ASoC must now keep track of which components it was able to device_link to the card->dev. The new card_device_link member of struct snd_soc_component is non-NULL if a device_link exists. The intent of the device link is to ensure that the machine driver system-suspends before the component drivers, to prevent ASoC suspend attempting to reconfigure a driver that has already suspended. It isn't possible to create this device link if the machine driver is a parent of the component driver or already has a device_link in the opposite direction. In this case skip the link. A warn is placed in kernel log since this might indicate a genuine design problem with those two drivers (this can be downgraded to dbg in future when people are happy that all these special drivers correctly handle their reversed shutdown order). Fixes: 0f54ce994b23 ("ASoC: soc-core: Create device_link to ensure correct suspend order") Reported-by: Marek Szyprowski Closes: https://lore.kernel.org/linux-sound/61bd38e7-5eb9-4448-a93f-afa2ccbd1c9d@opensource.cirrus.com/T/#m496fe5a11b0a3649afd2e85da5e1cea82bb16d8a Tested-by: Marek Szyprowski Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260623135821.4125543-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 4 ++++ sound/soc/soc-core.c | 35 +++++++++++++++-------------------- 2 files changed, 19 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 11bc9527653f..aa423865dbe7 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -10,6 +10,8 @@ #include +struct device_link; + /* * Component probe and remove ordering levels for components with runtime * dependencies. @@ -216,6 +218,8 @@ struct snd_soc_component { struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; + struct device_link *card_device_link; + const struct snd_soc_component_driver *driver; struct list_head dai_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 86b6c752a56b..7817beea5b3b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1980,19 +1980,15 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) } } -static void snd_soc_remove_device_links(struct snd_soc_card *card, - struct snd_soc_component *stop_at) +static void snd_soc_remove_device_links(struct snd_soc_card *card) { struct snd_soc_component *component; for_each_card_components(card, component) { - if (card->dev == component->dev) - continue; - - device_link_remove(card->dev, component->dev); - - if (component == stop_at) - return; + if (component->card_device_link) { + device_link_del(component->card_device_link); + component->card_device_link = NULL; + } } } @@ -2001,7 +1997,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card) if (snd_soc_card_is_instantiated(card)) { card->instantiated = false; - snd_soc_remove_device_links(card, NULL); + snd_soc_remove_device_links(card); soc_cleanup_card_resources(card); } @@ -2012,7 +2008,6 @@ static int snd_soc_bind_card(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card); - struct snd_soc_component *last_devlinked_component = NULL; int ret; snd_soc_card_mutex_lock_root(card); @@ -2141,19 +2136,21 @@ static int snd_soc_bind_card(struct snd_soc_card *card) * Add device_link from card to component so that system_suspend * will be done in the correct order. The card must suspend first * to stop audio activity before the components suspend. + * + * If a driver pair already have a link in the opposite direction + * they must manage their own suspend order. */ for_each_card_components(card, component) { if (card->dev == component->dev) continue; - if (!device_link_add(card->dev, component->dev, DL_FLAG_STATELESS)) { - dev_warn(card->dev, "Failed to create device link to %s\n", + component->card_device_link = device_link_add(card->dev, + component->dev, + DL_FLAG_STATELESS); + if (!component->card_device_link) { + dev_warn(card->dev, "Could not create device link to %s\n", dev_name(component->dev)); - ret = -EINVAL; - goto probe_end; } - - last_devlinked_component = component; } ret = snd_soc_card_late_probe(card); @@ -2185,9 +2182,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) probe_end: if (ret < 0) { - if (last_devlinked_component) - snd_soc_remove_device_links(card, last_devlinked_component); - + snd_soc_remove_device_links(card); soc_cleanup_card_resources(card); } From 00e44a68efef50f65b12854b41f098b4d50f10be Mon Sep 17 00:00:00 2001 From: Oleg Kucheryavenko Date: Wed, 24 Jun 2026 22:13:01 +0300 Subject: [PATCH 37/46] ALSA: hda/realtek: Add quirk for Acer Nitro ANV15-41 The Acer Nitro ANV15-41 laptop with ALC245 codec does not detect the headset microphone in the combo jack by default. Apply the ALC2XX_FIXUP_HEADSET_MIC quirk to fix it. Signed-off-by: Oleg Kucheryavenko Link: https://patch.msgid.link/20260624191301.10162-1-oleg.kucheryavenko2018@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 0d932f6ae3e8..2a7a693ecedf 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6813,6 +6813,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { 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, 0x173a, "Acer Swift SFG14-73", ALC245_FIXUP_ACER_MICMUTE_LED), + SND_PCI_QUIRK(0x1025, 0x1758, "Acer Nitro ANV15-41", ALC2XX_FIXUP_HEADSET_MIC), 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), SND_PCI_QUIRK(0x1025, 0x1844, "Acer Helios ZPS", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), From 7693c0cc415f3a16a7a3355f245474a5e661be4e Mon Sep 17 00:00:00 2001 From: Darvell Long Date: Wed, 24 Jun 2026 07:37:23 -0700 Subject: [PATCH 38/46] ALSA: usb-audio: avoid kobject path lookup in DualSense match The DualSense jack-detection input handler verifies that a matching input device belongs to the same physical controller by building kobject path strings for both the input device and the USB audio device, then comparing the path prefix. This was observed when a weak physical connection caused the controller to rapidly disconnect and reconnect. During that repeated hotplug, snd_dualsense_ih_match() can run while the controller's USB device is being disconnected. kobject_get_path() walks ancestor kobjects and dereferences their names; if the USB device kobject name is no longer valid, this can fault in strlen(): RIP: 0010:strlen+0x10/0x30 Call Trace: kobject_get_path+0x34/0x150 snd_dualsense_ih_match+0x49/0xd0 [snd_usb_audio] input_register_device+0x566/0x6a0 ps_probe+0xb89/0x1590 [hid_playstation] The same ownership check can be done without building kobject path strings. The input device is parented below the HID device, USB interface and USB device, so walking the input device parent chain and comparing against the mixer USB device preserves the check without dereferencing kobject names during disconnect. Fixes: 79d561c4ec04 ("ALSA: usb-audio: Add mixer quirk for Sony DualSense PS5") Cc: Assisted-by: Cute:gpt-5.5 Signed-off-by: Darvell Long Link: https://patch.msgid.link/20260624143723.2986353-1-contact@darvell.me Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 40 ++++++++++++---------------------------- 1 file changed, 12 insertions(+), 28 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f6f2bc8e97a7..10792d26fa94 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -577,46 +577,30 @@ static bool snd_dualsense_ih_match(struct input_handler *handler, { struct dualsense_mixer_elem_info *mei; struct usb_device *snd_dev; - char *input_dev_path, *usb_dev_path; - size_t usb_dev_path_len; - bool match = false; + struct device *parent; mei = container_of(handler, struct dualsense_mixer_elem_info, ih); snd_dev = mei->info.head.mixer->chip->dev; - input_dev_path = kobject_get_path(&dev->dev.kobj, GFP_KERNEL); - if (!input_dev_path) { - dev_warn(&snd_dev->dev, "Failed to get input dev path\n"); - return false; - } - - usb_dev_path = kobject_get_path(&snd_dev->dev.kobj, GFP_KERNEL); - if (!usb_dev_path) { - dev_warn(&snd_dev->dev, "Failed to get USB dev path\n"); - goto free_paths; - } - /* * Ensure the VID:PID matched input device supposedly owned by the * hid-playstation driver belongs to the actual hardware handled by - * the current USB audio device, which implies input_dev_path being - * a subpath of usb_dev_path. + * the current USB audio device. * * This verification is necessary when there is more than one identical * controller attached to the host system. + * + * The input device is registered below the HID device, USB interface and + * USB device, so compare the parent chain directly instead of building + * kobject path strings. This avoids dereferencing kobject names while the + * USB device hierarchy is being torn down during disconnect. */ - usb_dev_path_len = strlen(usb_dev_path); - if (usb_dev_path_len >= strlen(input_dev_path)) - goto free_paths; + for (parent = dev->dev.parent; parent; parent = parent->parent) { + if (parent == &snd_dev->dev) + return true; + } - usb_dev_path[usb_dev_path_len] = '/'; - match = !memcmp(input_dev_path, usb_dev_path, usb_dev_path_len + 1); - -free_paths: - kfree(input_dev_path); - kfree(usb_dev_path); - - return match; + return false; } static int snd_dualsense_ih_connect(struct input_handler *handler, From 6485da8579e2b0132630faa1ee2ac72ccaf01501 Mon Sep 17 00:00:00 2001 From: Kamlesh Chhetty Date: Mon, 22 Jun 2026 13:22:47 -0400 Subject: [PATCH 39/46] ALSA: hda: Add Lenovo Legion 7i 16IAX7 17AA3874 quirk Lenovo Legion 7i 16IAX7 systems with Realtek ALC287 codec SSID 17aa:3874 and CSC3551/CS35L41 speaker amps do not provide the required CS35L41 _DSD properties in ACPI. Without a quirk, cs35l41-hda fails probing the amps with missing cirrus,dev-index / Platform not supported errors, leaving the built-in speakers silent. This model is similar to the already-supported 17AA386F Legion 7i 16IAX7 variant. Add the Realtek ALC287 quirk to select ALC287_FIXUP_CS35L41_I2C_2 and add 17AA3874 to the CS35L41 property table using the same two-amp external-boost configuration. Tested on a Lenovo Legion 7 16IAX7 with Ubuntu 7.0.0-22-generic. Both CSC3551 CS35L41 amps probe and bind, firmware loads, calibration applies, built-in speaker playback works, and the cirrus,dev-index failure is gone. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=221663 Link: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/2157060 Signed-off-by: Kamlesh Chhetty Link: https://patch.msgid.link/20260622172247.19301-1-kamleshkc2002@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/realtek/alc269.c | 1 + sound/hda/codecs/side-codecs/cs35l41_hda_property.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 2a7a693ecedf..71625394aa99 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7767,6 +7767,7 @@ 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, 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), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), diff --git a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c index 732ae534db36..416d7bf3e289 100644 --- a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c +++ b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c @@ -128,6 +128,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA386E", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3874", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, @@ -554,6 +555,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "17AA3866", generic_dsd_config }, { "CSC3551", "17AA386E", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, + { "CSC3551", "17AA3874", generic_dsd_config }, { "CSC3551", "17AA3877", generic_dsd_config }, { "CSC3551", "17AA3878", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, From b8ff1f3578f83dedf5feb796d97a3f67c1bea58f Mon Sep 17 00:00:00 2001 From: Xu Rao Date: Tue, 23 Jun 2026 15:13:08 +0800 Subject: [PATCH 40/46] ALSA: usb-audio: qcom: Free QMI handle qc_usb_audio_probe() allocates svc->uaudio_svc_hdl separately from the uaudio_qmi_svc object. qmi_handle_release() releases the resources owned by an initialized QMI handle, but does not free the memory containing the struct qmi_handle itself. The probe error path and the remove path currently release the handle and then free svc, losing the last pointer to the separately allocated handle. This leaks one struct qmi_handle on each affected probe unwind and on each successful probe/remove cycle. Free the handle after qmi_handle_release() in both paths. Signed-off-by: Xu Rao Link: https://patch.msgid.link/9108EC860F3F87DF+20260623071308.2549182-1-raoxu@uniontech.com Signed-off-by: Takashi Iwai --- sound/usb/qcom/qc_audio_offload.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index 3a586fd16e72..e4bfd43a2488 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -1988,6 +1988,7 @@ static int qc_usb_audio_probe(struct auxiliary_device *auxdev, release_qmi: qc_usb_audio_cleanup_qmi_dev(); qmi_handle_release(svc->uaudio_svc_hdl); + kfree(svc->uaudio_svc_hdl); free_svc: kfree(svc); @@ -2012,6 +2013,7 @@ static void qc_usb_audio_remove(struct auxiliary_device *auxdev) qc_usb_audio_cleanup_qmi_dev(); qmi_handle_release(svc->uaudio_svc_hdl); + kfree(svc->uaudio_svc_hdl); kfree(svc); uaudio_svc = NULL; } From 29b9667982e4df2ed7744f86b1144f8bb58eb698 Mon Sep 17 00:00:00 2001 From: Maoyi Xie Date: Sun, 21 Jun 2026 23:09:07 +0800 Subject: [PATCH 41/46] ALSA: firewire: isight: bound the sample count to the packet payload isight_packet() takes the frame count from the device iso packet and checks it only against the device claimed iso length. count = be32_to_cpu(payload->sample_count); if (likely(count <= (length - 16) / 4)) isight_samples(isight, payload->samples, count); length is the iso header data_length. It can be up to 0xffff. So the gate allows a count up to about 16379. isight_samples() then copies count frames out of payload->samples into the PCM DMA buffer. payload->samples holds only 2 * MAX_FRAMES_PER_PACKET values. The device multiplexes two samples per frame. A count past MAX_FRAMES_PER_PACKET reads past the payload. A count past the buffer size writes past runtime->dma_area. The smallest PCM buffer is larger than MAX_FRAMES_PER_PACKET. Bounding the count to MAX_FRAMES_PER_PACKET keeps both the read and the write in range. A malicious or faulty Apple iSight on the FireWire bus reaches this during a normal capture. Add the MAX_FRAMES_PER_PACKET bound to the gate. Fixes: 3a691b28a0ca ("ALSA: add Apple iSight microphone driver") Suggested-by: Takashi Sakamoto Cc: stable@vger.kernel.org Signed-off-by: Maoyi Xie Link: https://patch.msgid.link/178205454729.1900991.7807310178296762772@maoyixie.com Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 2b7f071d593b..33c9dd48b3b0 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -179,7 +179,8 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle, if (likely(length >= 16 && payload->signature == cpu_to_be32(0x73676874/*"sght"*/))) { count = be32_to_cpu(payload->sample_count); - if (likely(count <= (length - 16) / 4)) { + if (likely(count <= (length - 16) / 4 && + count <= MAX_FRAMES_PER_PACKET)) { total = be32_to_cpu(payload->sample_total); if (unlikely(total != isight->total_samples)) { if (!isight->first_packet) From 64184f07e7516eb3ac1839387eb7cbe2dbeaa96f Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Thu, 25 Jun 2026 18:28:15 +0800 Subject: [PATCH 42/46] ASoC: tas2781: Update default register address to TAS2563 The previous TAS2781 address is overwritten during calibration, so switching to the TAS2563 address is required. Signed-off-by: Baojun Xu Link: https://patch.msgid.link/20260625102815.20043-1-baojun.xu@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index d45f805b5d18..b763da613769 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -59,9 +59,9 @@ #define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) /* XM_340 */ -#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) /* XM_341 */ -#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0c) From c5315c8dcbcb9271f59662104428cd9229c9a7cb Mon Sep 17 00:00:00 2001 From: Christian Hewitt Date: Thu, 25 Jun 2026 12:28:11 +0000 Subject: [PATCH 43/46] ASoC: codecs: pcm512x: only print info once on no sclk If sclk is not provided the driver falls back to using bclk and prints an info message in the system log. Under normal operations the message is repeated many times: [ 17.929576] pcm512x 0-004c: No SCLK, using BCLK: -2 [ 17.949172] pcm512x 0-004c: No SCLK, using BCLK: -2 [ 17.953029] pcm512x 0-004c: No SCLK, using BCLK: -2 [ 17.965059] pcm512x 0-004c: No SCLK, using BCLK: -2 [ 82.592980] pcm512x 0-004c: No SCLK, using BCLK: -2 [ 82.866293] pcm512x 0-004c: No SCLK, using BCLK: -2 Switch from dev_info to dev_info_once to reduce log noise. Signed-off-by: Christian Hewitt Link: https://patch.msgid.link/20260625122811.4056274-1-christianshewitt@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index fdef98ce52f1..fe3b5011fa16 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -633,7 +633,7 @@ static int pcm512x_dai_startup_slave(struct snd_pcm_substream *substream, struct regmap *regmap = pcm512x->regmap; if (IS_ERR(pcm512x->sclk)) { - dev_info(dev, "No SCLK, using BCLK: %ld\n", + dev_info_once(dev, "No SCLK, using BCLK: %ld\n", PTR_ERR(pcm512x->sclk)); /* Disable reporting of missing SCLK as an error */ From cf6f56990ea21172e085f0588e5bbf2089ce8f58 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Jun 2026 18:24:16 +0800 Subject: [PATCH 44/46] ASoC: fsl_asrc_dma: fix eDMA maxburst misalignment with channel count The back-end consumes data in units of the number of channels. When the maxburst value is not evenly divisible by the channel count, the DMA transfer length does not align with the FIFO frame boundary, causing wrong data to be copied and audible noise at the end of the stream. This is specific to eDMA: eDMA only responds to DMA requests from the back-end, whereas SDMA handles requests from both the front-end and the back-end and is not affected. For eDMA, when the back-end maxburst is not evenly divisible by the channel count, align it to the nearest valid boundary: - If maxburst >= channel count, override to the channel count so each transfer corresponds to exactly one audio frame. - If maxburst < channel count, override to 1 to avoid partial-frame transfers. Retain the original maxburst for SDMA or when it already aligns with the channel count. Fixes: c05f10f28ef6 ("ASoC: fsl_asrc: Add support for imx8qm & imx8qxp") Signed-off-by: Shengjiu Wang Link: https://patch.msgid.link/20260625102416.424911-1-shengjiu.wang@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc_dma.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5aa96af994c4..38f2b7c63133 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -288,6 +288,26 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, config_be.dst_addr_width = buswidth; config_be.dst_maxburst = dma_params_be->maxburst; + /* + * For eDMA, the back-end may report a maxburst size that is not evenly + * divisible by the channel count. This causes the DMA transfer length + * to misalign with the FIFO boundary, resulting in wrong data and + * audible noise. Align maxburst to the nearest valid boundary: + * - If maxburst >= channel count, override to the channel count so + * each transfer equals exactly one audio frame. + * - If maxburst < channel count, override to 1 to avoid partial-frame + * transfers. + */ + if (asrc->use_edma && (dma_params_be->maxburst % params_channels(params))) { + if (dma_params_be->maxburst >= params_channels(params)) { + config_be.src_maxburst = params_channels(params); + config_be.dst_maxburst = params_channels(params); + } else { + config_be.src_maxburst = 1; + config_be.dst_maxburst = 1; + } + } + memset(&audio_config, 0, sizeof(audio_config)); config_be.peripheral_config = &audio_config; config_be.peripheral_size = sizeof(audio_config); From 9dbbe81962b973fe71592ad8615d1e6cd28451bf Mon Sep 17 00:00:00 2001 From: Oleg Kucheryavenko Date: Thu, 25 Jun 2026 16:49:55 +0300 Subject: [PATCH 45/46] ALSA: hda/realtek: Update Acer Nitro ANV15-41 quirk to enable mute LED The laptop has a microphone mute LED on the F4 key, but it was not taken in mind when the previous quirk was added in commit 00e44a68efef50f65b12854b41f098b4d50f10be ("ALSA: hda/realtek: Add quirk for Acer Nitro ANV15-41"). Replace ALC2XX_FIXUP_HEADSET_MIC with ALC245_FIXUP_ACER_MICMUTE_LED, which enables the LED and chains the previous quirk for the headset microphone. Fixes: 00e44a68efef ("ALSA: hda/realtek: Add quirk for Acer Nitro ANV15-41") Signed-off-by: Oleg Kucheryavenko Link: https://patch.msgid.link/20260625134955.27465-1-oleg.kucheryavenko2018@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 71625394aa99..d9e2384fc0ba 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6813,7 +6813,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { 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, 0x173a, "Acer Swift SFG14-73", ALC245_FIXUP_ACER_MICMUTE_LED), - SND_PCI_QUIRK(0x1025, 0x1758, "Acer Nitro ANV15-41", ALC2XX_FIXUP_HEADSET_MIC), + 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), SND_PCI_QUIRK(0x1025, 0x1844, "Acer Helios ZPS", ALC287_FIXUP_PREDATOR_SPK_CS35L41_I2C_2), From e1e31e0ec8a609e17fd2e86b77bc00d9cbb24d7c Mon Sep 17 00:00:00 2001 From: Jiaming Zhang Date: Thu, 25 Jun 2026 21:49:33 +0800 Subject: [PATCH 46/46] ALSA: FCP: Fix NULL pointer dereference in interface lookup A malformed USB device can provide a vendor-specific interface without any endpoint descriptors. fcp_find_fc_interface() currently selects the first vendor-specific interface and reads endpoint 0 from it, without checking whether the interface actually has any endpoints. When bNumEndpoints is zero, no endpoint array is allocated for the parsed alternate setting, so get_endpoint(..., 0) yields an invalid endpoint descriptor pointer. Dereferencing it through usb_endpoint_num() then triggers a NULL pointer dereference. Skip vendor-specific interfaces that do not have any endpoints. Fixes: 46757a3e7d50 ("ALSA: FCP: Add Focusrite Control Protocol driver") Reported-by: Jiaming Zhang Closes: https://lore.kernel.org/lkml/CANypQFb1EHj0xX8bA1WxSOSK-5xca6ZNKzOQcp12=s=puY7VFw@mail.gmail.com/ Signed-off-by: Jiaming Zhang Link: https://patch.msgid.link/20260625134933.425785-1-r772577952@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/fcp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/fcp.c b/sound/usb/fcp.c index ea746bdb36ff..6f5dcd35e1d4 100644 --- a/sound/usb/fcp.c +++ b/sound/usb/fcp.c @@ -1083,6 +1083,8 @@ static int fcp_find_fc_interface(struct usb_mixer_interface *mixer) if (desc->bInterfaceClass != 255) continue; + if (desc->bNumEndpoints < 1) + continue; epd = get_endpoint(intf->altsetting, 0); private->bInterfaceNumber = desc->bInterfaceNumber;