From 5cff1529a2f9b3461a7f5a6e36a86682fc290534 Mon Sep 17 00:00:00 2001 From: WenTao Liang Date: Sat, 27 Jun 2026 12:29:49 +0800 Subject: ALSA: us144mkii: capture_urb_complete: redundant usb_anchor_urb corrupts anchor list on each resubmission In capture_urb_complete(), usb_anchor_urb() is called on every completion callback, but the URB is already anchored from the initial submission in tascam_trigger_start(). Each redundant call corrupts the anchor's doubly-linked list and inflates the URB refcount. When usb_kill_anchored_urbs() traverses the list during stream stop / suspend / disconnect, the corrupted list leads to use-after-free. Remove the redundant usb_anchor_urb() from the resubmit path. Cc: stable@vger.kernel.org Fixes: c1bb0c13e430 ("ALSA: usb-audio: us144mkii: Implement audio capture and decoding") Signed-off-by: WenTao Liang Link: https://patch.msgid.link/20260627042949.61767-1-vulab@iscas.ac.cn Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us144mkii_capture.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/us144mkii_capture.c b/sound/usb/usx2y/us144mkii_capture.c index af120bf62173..fa01da98151a 100644 --- a/sound/usb/usx2y/us144mkii_capture.c +++ b/sound/usb/usx2y/us144mkii_capture.c @@ -302,7 +302,6 @@ void capture_urb_complete(struct urb *urb) } usb_get_urb(urb); - usb_anchor_urb(urb, &tascam->capture_anchor); ret = usb_submit_urb(urb, GFP_ATOMIC); if (ret < 0) { dev_err_ratelimited(tascam->card->dev, @@ -312,6 +311,7 @@ void capture_urb_complete(struct urb *urb) usb_put_urb(urb); atomic_dec( &tascam->active_urbs); /* Decrement on failed resubmission */ + return; } out: usb_put_urb(urb); -- cgit v1.2.3 From 0773610eef71c30df3cb4c113c8215625d2a7c23 Mon Sep 17 00:00:00 2001 From: Luca Weiss Date: Tue, 26 May 2026 17:03:05 +0200 Subject: ASoC: codecs: lpass-va-macro: Fix LPASS Codec Version for SC7280 According to both the static definition in downstream... yupik-audio-overlay.dtsi: qcom,bolero-version = <4>; #define BOLERO_VERSION_2_0 0x0004) and the runtime detection: CDC_VA_TOP_CSR_CORE_ID_0=0x1 CDC_VA_TOP_CSR_CORE_ID_1=0xf SC7280 has LPASS Codec Version 2.0 and not, as declared with sm8250_va_data LPASS_CODEC_VERSION_1_0. Create new va_macro_data with .version not set to use the runtime detection and correctly get .version = LPASS_CODEC_VERSION_2_0. Fixes: 77212f300bfd ("ASoC: codecs: lpass-va-macro: set the default codec version for sm8250") Signed-off-by: Luca Weiss Reviewed-by: Srinivas Kandagatla Link: https://patch.msgid.link/20260526-sc7280-va-macro-2-0-v1-1-2c1b572fa388@fairphone.com Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-va-macro.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 528d5b167ecf..58a5798823d7 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -244,6 +244,11 @@ static const struct va_macro_data sm8250_va_data = { .version = LPASS_CODEC_VERSION_1_0, }; +static const struct va_macro_data sc7280_va_data = { + .has_swr_master = false, + .has_npl_clk = false, +}; + static const struct va_macro_data sm8450_va_data = { .has_swr_master = true, .has_npl_clk = true, @@ -1755,7 +1760,7 @@ static const struct dev_pm_ops va_macro_pm_ops = { }; static const struct of_device_id va_macro_dt_match[] = { - { .compatible = "qcom,sc7280-lpass-va-macro", .data = &sm8250_va_data }, + { .compatible = "qcom,sc7280-lpass-va-macro", .data = &sc7280_va_data }, { .compatible = "qcom,sm6115-lpass-va-macro", .data = &sm8450_va_data }, { .compatible = "qcom,sm8250-lpass-va-macro", .data = &sm8250_va_data }, { .compatible = "qcom,sm8450-lpass-va-macro", .data = &sm8450_va_data }, -- cgit v1.2.3 From 91a6dba5183d269b8215d6170c079b654ef4cab7 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 29 Jun 2026 10:57:22 +0800 Subject: MAINTAINERS: ASoC: SOF: add AMD reviewer for Sound Open Firmware SOF spans multiple vendors and hardware paths. AMD ships ACP-based platforms and contributes to the shared SOF tree, so list an AMD reviewer (R:) on the SOF MAINTAINERS entry for balanced review and correct get_maintainer coverage when shared SOF code changes. Add: R: Vijendar Mukunda Signed-off-by: Vijendar Mukunda Reviewed-by: Liam Girdwood Signed-off-by: Bard Liao Link: https://patch.msgid.link/20260629025722.1982120-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 15011f5752a9..558bc04f2e45 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -25377,6 +25377,7 @@ M: Bard Liao M: Daniel Baluta R: Kai Vehmanen R: Pierre-Louis Bossart +R: Vijendar Mukunda L: sound-open-firmware@alsa-project.org (moderated for non-subscribers) S: Supported W: https://github.com/thesofproject/linux/ -- cgit v1.2.3 From 6c18817c01f6f76d9e2739903abde4d69397f2c6 Mon Sep 17 00:00:00 2001 From: Shubham Nayak Date: Mon, 29 Jun 2026 19:04:55 +0530 Subject: ALSA: hda/realtek: Add quirk for HP Victus 16-e0xxx (88EE) to enable mute LED The mute LED on the HP Victus 16-e0xxx (board ID 88EE, ALC245 codec) does not function by default. Add the ALC245_FIXUP_HP_MUTE_LED_COEFBIT quirk to enable it. Tested on my HP Victus 16-e0xxx with kernel 7.1.2. Signed-off-by: Shubham Nayak Link: https://patch.msgid.link/20260629-hp-victus-16-mute-led-v1-1-ab0f4a8a533b@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 d9e2384fc0ba..f7700713dc62 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7068,6 +7068,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x88d1, "HP Pavilion 15-eh1xxx (mainboard 88D1)", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT), SND_PCI_QUIRK(0x103c, 0x88dd, "HP Pavilion 15z-ec200", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x88eb, "HP Victus 16-e0xxx", ALC245_FIXUP_HP_MUTE_LED_V2_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x88ee, "HP Victus 16-e0xxx (MB 88EE)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x890e, "HP 255 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8919, "HP Pavilion Aero Laptop 13-be0xxx", ALC287_FIXUP_HP_GPIO_LED), -- cgit v1.2.3 From e782d687d2f5bf8b8113dc48ba22cca4b472c252 Mon Sep 17 00:00:00 2001 From: Jetha Chan Date: Tue, 30 Jun 2026 09:33:28 +0900 Subject: ASoC: amd: yc: Add Alienware m15 R7 AMD to DMIC quirk table The Alienware m15 R7 AMD exposes an ACP6x DMIC path, but its DMI product name is not present in the Yellow Carp ACP quirk table. As a result, the ACP machine driver does not enable the DMIC card on this system. Add the DMI product name for this machine. With this quirk applied, the kernel reports: acp_yc_mach acp_yc_mach.0: Enabling ACP DMIC support via DMI and ALSA exposes the ACP DMIC capture device: card 3: acp6x device 0: DMIC capture dmic-hifi-0 Tested on an Alienware m15 R7 AMD with product SKU 0B59. Link: https://jethachan.net/dev/2026/03/21/fixing-internal-microphone-alienware-linux.html Assisted-by: OpenAI-Codex:gpt-5.5 Signed-off-by: Jetha Chan Link: https://patch.msgid.link/20260630003328.15675-1-jethachan@gmail.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index a405cc2c11a3..d6df7de70b27 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -528,6 +528,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Raider A18 HX A9WJG"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m15 R7 AMD"), + } + }, { .driver_data = &acp6x_card, .matches = { -- cgit v1.2.3 From 147996e7e7c9e8339c0e04f6fa7ccb3e4d448ff7 Mon Sep 17 00:00:00 2001 From: HyeongJun An Date: Wed, 1 Jul 2026 18:52:31 +0900 Subject: ALSA: usx2y: us144mkii: fix work UAF on disconnect tascam_disconnect() cancels capture_work and midi_in_work before usb_kill_anchored_urbs() kills the capture/MIDI-in URBs. Those URBs self-resubmit, and their completion handlers reschedule the work. A URB that completes in the small window between cancel_work_sync() and usb_kill_anchored_urbs() therefore re-arms the work after its only cancel. Nothing cancels it again before snd_card_free() frees the card-private tascam structure, so the work handler then runs on freed memory. Kill the anchored URBs before cancelling the work; once the work is cancelled no remaining URB can complete to re-arm it. Fixes: c1bb0c13e430 ("ALSA: usb-audio: us144mkii: Implement audio capture and decoding") Cc: stable@vger.kernel.org Assisted-by: Claude:claude-opus-4-8 Signed-off-by: HyeongJun An Link: https://patch.msgid.link/20260701095231.1020811-1-sammiee5311@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us144mkii.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/usb/usx2y/us144mkii.c b/sound/usb/usx2y/us144mkii.c index 94553b61013c..58ef23146f20 100644 --- a/sound/usb/usx2y/us144mkii.c +++ b/sound/usb/usx2y/us144mkii.c @@ -585,19 +585,24 @@ static void tascam_disconnect(struct usb_interface *intf) return; if (intf->cur_altsetting->desc.bInterfaceNumber == 0) { - /* Ensure all deferred work is complete before freeing resources */ snd_card_disconnect(tascam->card); - cancel_work_sync(&tascam->stop_work); - cancel_work_sync(&tascam->capture_work); - cancel_work_sync(&tascam->midi_in_work); - cancel_work_sync(&tascam->midi_out_work); - cancel_work_sync(&tascam->stop_pcm_work); + /* + * Kill the URBs before cancelling the work, so a late URB + * completion cannot re-arm a work that then runs after + * snd_card_free(). + */ usb_kill_anchored_urbs(&tascam->playback_anchor); usb_kill_anchored_urbs(&tascam->capture_anchor); usb_kill_anchored_urbs(&tascam->feedback_anchor); usb_kill_anchored_urbs(&tascam->midi_in_anchor); usb_kill_anchored_urbs(&tascam->midi_out_anchor); + + cancel_work_sync(&tascam->stop_work); + cancel_work_sync(&tascam->capture_work); + cancel_work_sync(&tascam->midi_in_work); + cancel_work_sync(&tascam->midi_out_work); + cancel_work_sync(&tascam->stop_pcm_work); timer_delete_sync(&tascam->error_timer); tascam_free_urbs(tascam); snd_card_free(tascam->card); -- cgit v1.2.3 From 95edf2dbb492f3ea2420111e9c0044c7dec9113c Mon Sep 17 00:00:00 2001 From: Yousef Alhouseen Date: Sun, 28 Jun 2026 02:03:29 +0200 Subject: ASoC: SOF: validate probe info element counts Probe information replies contain a firmware-provided element count. IPC3 uses that count to copy an array, then returns the unchecked count to its caller. A short reply can therefore make the caller walk beyond the copied array. IPC4 similarly uses the count both to allocate the destination array and to walk the reply. On 32-bit systems the allocation size can wrap, while on all systems an excessive count reads beyond the reply payload. Validate each count against the actual reply size before copying or allocating the array, and use kcalloc() for the IPC4 allocation. Signed-off-by: Yousef Alhouseen Link: https://patch.msgid.link/20260628000329.18606-1-alhouseenyousef@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-probes-ipc3.c | 23 +++++++++++++++++++---- sound/soc/sof/sof-client-probes-ipc4.c | 11 ++++++++++- 2 files changed, 29 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/sof-client-probes-ipc3.c b/sound/soc/sof/sof-client-probes-ipc3.c index a78ec0954a61..a3e382d6161f 100644 --- a/sound/soc/sof/sof-client-probes-ipc3.c +++ b/sound/soc/sof/sof-client-probes-ipc3.c @@ -107,7 +107,7 @@ static int ipc3_probes_info(struct sof_client_dev *cdev, unsigned int cmd, struct device *dev = &cdev->auxdev.dev; struct sof_ipc_probe_info_params msg = {{{0}}}; struct sof_ipc_probe_info_params *reply; - size_t bytes; + size_t bytes, elem_size, payload_size; int ret; *params = NULL; @@ -128,14 +128,29 @@ static int ipc3_probes_info(struct sof_client_dev *cdev, unsigned int cmd, if (ret < 0 || reply->rhdr.error < 0) goto exit; + payload_size = reply->rhdr.hdr.size; + if (payload_size < offsetof(struct sof_ipc_probe_info_params, dma)) { + ret = -EINVAL; + goto exit; + } + if (!reply->num_elems) goto exit; if (cmd == SOF_IPC_PROBE_DMA_INFO) - bytes = sizeof(reply->dma[0]); + elem_size = sizeof(reply->dma[0]); else - bytes = sizeof(reply->desc[0]); - bytes *= reply->num_elems; + elem_size = sizeof(reply->desc[0]); + + payload_size -= offsetof(struct sof_ipc_probe_info_params, dma); + if (reply->num_elems > payload_size / elem_size) { + dev_err(dev, "%s: invalid probe info element count %u\n", + __func__, reply->num_elems); + ret = -EINVAL; + goto exit; + } + + bytes = reply->num_elems * elem_size; *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); if (!*params) { ret = -ENOMEM; diff --git a/sound/soc/sof/sof-client-probes-ipc4.c b/sound/soc/sof/sof-client-probes-ipc4.c index 88397c7dc4c3..2eef32b55395 100644 --- a/sound/soc/sof/sof-client-probes-ipc4.c +++ b/sound/soc/sof/sof-client-probes-ipc4.c @@ -248,10 +248,19 @@ static int ipc4_probes_points_info(struct sof_client_dev *cdev, return ret; } info = msg.data_ptr; + if (msg.data_size < sizeof(*info) || + info->num_elems > (msg.data_size - sizeof(*info)) / + sizeof(info->points[0])) { + dev_err(dev, "%s: invalid probe info element count %u\n", + __func__, info->num_elems); + kfree(msg.data_ptr); + return -EINVAL; + } + *num_desc = info->num_elems; dev_dbg(dev, "%s: got %zu probe points", __func__, *num_desc); - *desc = kzalloc(*num_desc * sizeof(**desc), GFP_KERNEL); + *desc = kcalloc(*num_desc, sizeof(**desc), GFP_KERNEL); if (!*desc) { kfree(msg.data_ptr); return -ENOMEM; -- cgit v1.2.3 From ed0abc8be27e23aa65716bcaab8976ada2503cab Mon Sep 17 00:00:00 2001 From: John Madieu Date: Wed, 10 Jun 2026 16:47:04 +0000 Subject: ASoC: rsnd: adg: make rsnd_adg_clk_control() idempotent rsnd_adg_clk_control() is asymmetric on the disable path: the clkin clocks are guarded by clkin_rate[], but the "adg" clock is disabled unconditionally. If an enable attempt fails (for example a clkin failing to turn on during resume), the error path correctly rolls everything back, but rsnd_resume() ignores the return value, so the following system suspend calls rsnd_adg_clk_disable() again and underflows the "adg" clock enable count: adg_0_clks1 already disabled WARNING: drivers/clk/clk.c:1188 clk_core_disable+0xa4/0xac Call trace: clk_core_disable+0xa4/0xac (P) clk_disable+0x30/0x4c rsnd_adg_clk_control+0x9c/0x2cc rsnd_suspend+0x20/0x74 device_suspend+0x140/0x3ec dpm_suspend+0x168/0x270 Track the enable state explicitly and bail out of redundant enable/disable calls, mirroring what is already done for the per-SSI clock prepare state. A failed enable leaves the state as disabled, so the next suspend becomes a no-op and the next resume retries cleanly. Fixes: 47899d53f86f ("ASoC: rsnd: adg: Add per-SSI ADG and SSIF supply clock management") Signed-off-by: John Madieu Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20260610164704.2211321-1-john.madieu.xa@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/adg.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/renesas/rcar/adg.c b/sound/soc/renesas/rcar/adg.c index 5479cefb6dbe..53efd1be5139 100644 --- a/sound/soc/renesas/rcar/adg.c +++ b/sound/soc/renesas/rcar/adg.c @@ -45,6 +45,7 @@ struct rsnd_adg { struct rsnd_mod mod; int clkin_rate[CLKINMAX]; bool ssi_clk_prepared; + bool clk_enabled; int clkin_size; int clkout_size; u32 ckr; @@ -463,6 +464,22 @@ int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) struct clk *clk; int ret = 0, i; + /* + * rsnd_adg_clk_enable() and rsnd_adg_clk_disable() can be called + * redundantly, for example when system suspend follows a resume + * whose enable failed. Make this function idempotent so that the + * "adg" clock, which has no clkin_rate[] style guard, is never + * disabled twice. + */ + if (enable) { + if (adg->clk_enabled) + return 0; + } else { + if (!adg->clk_enabled) + return 0; + adg->clk_enabled = false; + } + if (enable) { ret = clk_prepare_enable(adg->adg); if (ret < 0) @@ -520,12 +537,22 @@ int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) * rsnd_adg_clk_enable() might return error (_disable() will not). * We need to rollback in such case */ - if (ret < 0) + if (ret < 0) { + /* + * Mark as enabled so that the rollback below is not + * short-circuited by the idempotency guard. It clears + * the flag again on its way through. + */ + adg->clk_enabled = true; rsnd_adg_clk_disable(priv); + return ret; + } /* disable adg */ if (!enable) clk_disable_unprepare(adg->adg); + else + adg->clk_enabled = true; return ret; } -- cgit v1.2.3 From 12272cb1b23e3032e5c627fb52f183a61913a88b Mon Sep 17 00:00:00 2001 From: Sen Wang Date: Tue, 30 Jun 2026 13:31:20 -0500 Subject: ASoC: codecs: tas675x: use READ_ONCE for params to be used concurrently active_playback_dais and active_capture_dais are written atomically via set_bit()/clear_bit() and can be read concurrently from the fault_check_work delayed work handler. fault_check_work already uses READ_ONCE; extend the same guard to all other reads in tas675x_hw_params() and tas675x_mute_stream(). Fixes: 133c81f84471 ("ASoC: codecs: Add TAS67524 quad-channel audio amplifier driver") Signed-off-by: Sen Wang Link: https://patch.msgid.link/20260630183126.2588322-2-sen@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas675x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas675x.c b/sound/soc/codecs/tas675x.c index 6f89a422f3c6..f1d2bbc85008 100644 --- a/sound/soc/codecs/tas675x.c +++ b/sound/soc/codecs/tas675x.c @@ -1133,7 +1133,7 @@ static int tas675x_hw_params(struct snd_pcm_substream *substream, * Single clock domain: SDIN and SDOUT share one SCLK/FSYNC pair, * so all active DAIs must use the same sample rate. */ - if ((tas->active_playback_dais || tas->active_capture_dais) && + if ((READ_ONCE(tas->active_playback_dais) || READ_ONCE(tas->active_capture_dais)) && tas->rate && tas->rate != rate) { dev_err(component->dev, "Rate %u conflicts with active rate %u\n", @@ -1397,14 +1397,14 @@ static int tas675x_mute_stream(struct snd_soc_dai *dai, int mute, int direction) set_bit(dai->id, &tas->active_playback_dais); /* Last playback stream */ - if (mute && !tas->active_playback_dais) { + if (mute && !READ_ONCE(tas->active_playback_dais)) { ret = tas675x_set_state_all(tas, TAS675X_STATE_SLEEP_BOTH); regmap_read(tas->regmap, TAS675X_CLK_FAULT_LATCHED_REG, &discard); return ret; } return tas675x_set_state_all(tas, - tas->active_playback_dais ? + READ_ONCE(tas->active_playback_dais) ? TAS675X_STATE_PLAY_BOTH : TAS675X_STATE_SLEEP_BOTH); } -- cgit v1.2.3 From a044f99d000dca7e1d3e8fc847d9ad60467b6793 Mon Sep 17 00:00:00 2001 From: Sen Wang Date: Tue, 30 Jun 2026 13:31:21 -0500 Subject: ASoC: codecs: tas675x: Fix CHx temperature range register bit fields The initial merged patch mixed up the bits for temp reg with LDG report, now fixing to the right bits according to TRM (SLOU589A). Fixes: 133c81f84471 ("ASoC: codecs: Add TAS67524 quad-channel audio amplifier driver") Signed-off-by: Sen Wang Link: https://patch.msgid.link/20260630183126.2588322-3-sen@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas675x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas675x.c b/sound/soc/codecs/tas675x.c index f1d2bbc85008..82526362de7b 100644 --- a/sound/soc/codecs/tas675x.c +++ b/sound/soc/codecs/tas675x.c @@ -954,10 +954,10 @@ static const struct snd_kcontrol_new tas675x_snd_controls[] = { /* Temperature and Voltage Monitoring */ SOC_SINGLE_RO("PVDD Sense", TAS675X_PVDD_SENSE_REG, 0, 0xFF), SOC_SINGLE_RO("Global Temperature", TAS675X_TEMP_GLOBAL_REG, 0, 0xFF), - SOC_SINGLE_RO("CH1 Temperature Range", TAS675X_TEMP_CH1_CH2_REG, 6, 3), - SOC_SINGLE_RO("CH2 Temperature Range", TAS675X_TEMP_CH1_CH2_REG, 4, 3), - SOC_SINGLE_RO("CH3 Temperature Range", TAS675X_TEMP_CH3_CH4_REG, 2, 3), - SOC_SINGLE_RO("CH4 Temperature Range", TAS675X_TEMP_CH3_CH4_REG, 0, 3), + SOC_SINGLE_RO("CH1 Temperature Range", TAS675X_TEMP_CH1_CH2_REG, 0, 7), + SOC_SINGLE_RO("CH2 Temperature Range", TAS675X_TEMP_CH1_CH2_REG, 3, 7), + SOC_SINGLE_RO("CH3 Temperature Range", TAS675X_TEMP_CH3_CH4_REG, 0, 7), + SOC_SINGLE_RO("CH4 Temperature Range", TAS675X_TEMP_CH3_CH4_REG, 3, 7), /* Speaker Protection & Detection */ SOC_SINGLE("Tweeter Detection Switch", TAS675X_TWEETER_DETECT_CTRL_REG, 0, 1, 1), -- cgit v1.2.3 From c34a4be8b846c7a220fe56442ecca27f6ab91943 Mon Sep 17 00:00:00 2001 From: Sen Wang Date: Tue, 30 Jun 2026 13:31:22 -0500 Subject: Documentation: sound: tas675x: Fix temperature range and impedance documentation Two corrections against the TRM (SLOU589A): - Corrected channel temperature range - Corrected conversion formula for global temperature Fixes: ba46edca354e ("Documentation: sound: Add TAS675x codec mixer controls documentation") Signed-off-by: Sen Wang Link: https://patch.msgid.link/20260630183126.2588322-4-sen@ti.com Signed-off-by: Mark Brown --- Documentation/sound/codecs/tas675x.rst | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/codecs/tas675x.rst b/Documentation/sound/codecs/tas675x.rst index c08b0e392306..2d9e009b46b1 100644 --- a/Documentation/sound/codecs/tas675x.rst +++ b/Documentation/sound/codecs/tas675x.rst @@ -281,7 +281,7 @@ Global Temperature :Description: Global die temperature sense register. :Type: Integer (read-only) :Range: 0 to 255 -:Conversion: (value × 0.5 °C) − 50 °C +:Conversion: value × 2.19 K; subtract 273.15 for °C :Register: 0x75 CHx Temperature Range @@ -289,10 +289,11 @@ CHx Temperature Range :Description: Per-channel coarse temperature range indicator (x = 1, 2, 3, 4). :Type: Integer (read-only) -:Range: 0 to 3 -:Mapping: 0 = <80 °C, 1 = 80–100 °C, 2 = 100–120 °C, 3 = >120 °C -:Register: 0xBB bits [7:6] (CH1), bits [5:4] (CH2), - 0xBC bits [3:2] (CH3), bits [1:0] (CH4) +:Range: 0 to 7 +:Mapping: 0 = <95 °C, 1 = 95–110 °C, 2 = 110–125 °C, 3 = 125–135 °C, + 4 = 135–145 °C, 5 = 145–155 °C, 6 = 155–165 °C, 7 = >165 °C +:Register: 0xBB bits [2:0] (CH1), bits [5:3] (CH2), + 0xBC bits [2:0] (CH3), bits [5:3] (CH4) Load Diagnostics ================ -- cgit v1.2.3 From 83245e7a436c04e511378af14dd81fd188b41541 Mon Sep 17 00:00:00 2001 From: John Madieu Date: Tue, 30 Jun 2026 17:53:29 +0000 Subject: ASoC: rsnd: src: Add missing scu_supply clock to suspend/resume scu_supply is enabled alongside scu and scu_x2 during normal SRC operation, but rsnd_src_suspend() and rsnd_src_resume() only disable and re-enable scu and scu_x2. The supply clock is left enabled across a system suspend and its prepare/enable refcount becomes unbalanced after a suspend/resume cycle. Disable scu_supply in rsnd_src_suspend() and re-enable it in rsnd_src_resume() so the SRC clocks are managed consistently across system PM transitions. Fixes: ef19ecf042b4 ("ASoC: rsnd: Add system suspend/resume support") Signed-off-by: John Madieu Acked-by: Kuninori Morimoto Link: https://patch.msgid.link/20260630175329.4145703-1-john.madieu.xa@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/renesas/rcar/src.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/renesas/rcar/src.c b/sound/soc/renesas/rcar/src.c index ac806bdc96d9..2cdb39e898af 100644 --- a/sound/soc/renesas/rcar/src.c +++ b/sound/soc/renesas/rcar/src.c @@ -850,6 +850,7 @@ void rsnd_src_suspend(struct rsnd_priv *priv) clk_disable_unprepare(src_ctrl->scu_x2); clk_disable_unprepare(src_ctrl->scu); + clk_disable_unprepare(src_ctrl->scu_supply); } void rsnd_src_resume(struct rsnd_priv *priv) @@ -861,6 +862,7 @@ void rsnd_src_resume(struct rsnd_priv *priv) if (!src_ctrl) return; + clk_prepare_enable(src_ctrl->scu_supply); clk_prepare_enable(src_ctrl->scu); clk_prepare_enable(src_ctrl->scu_x2); -- cgit v1.2.3