From 4e1462e7cd6a9b920f441842726cc2b9d232673e Mon Sep 17 00:00:00 2001 From: Matus Gajdos Date: Fri, 2 Oct 2026 14:52:37 +0200 Subject: [PATCH 01/13] ALSA: usb-audio: Add native DSD support for Cambridge Audio streamers The Cambridge Audio streamer advertises the UAC2 RAW_DATA format on altsetting 2 of its playback interface, but without the DSD_RAW quirk flag the driver exposes it only as a SPECIAL format that cannot be used. Add QUIRK_FLAG_DSD_RAW for the device (USB ID 22e8:ca10), so the altsetting is exposed as DSD_U32_BE, enabling native DSD64/128/256 playback. Tested with: gst-launch-1.0 filesrc location=file.dsf ! \ avdemux_dsf ! dsdconvert ! \ alsasink device=hw:2,0 Signed-off-by: Matus Gajdos Link: https://patch.msgid.link/20261002125237.82188-1-matuszpd@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3afc49565963..a54fabce7920 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2510,6 +2510,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x21b4, 0x0232, /* Ayre QX-5 Twenty */ QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x22e8, 0xca10, /* Cambridge Audio streamers */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x25aa, 0x600b, /* TAE1159 */ From 9a44b828dd7c0be1d0d6d6cf178d9727ceb6da6e Mon Sep 17 00:00:00 2001 From: Eckhart Mohr Date: Thu, 1 Oct 2026 20:34:00 +0200 Subject: [PATCH 02/13] ALSA: hda/realtek: Add quirk for ASRock NUC BOX-358H Fix headphone jack detection for some devices Signed-off-by: Eckhart Mohr Signed-off-by: Werner Sembach Link: https://patch.msgid.link/20261001183403.103990-1-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 be5482530aa2..0b5e0b45b6b8 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -8346,6 +8346,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x0269, "Positivo Master C6400", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), + SND_PCI_QUIRK(0x1849, 0x6256, "ASRock NUC Box 358H", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), From 593d46fc6af9a0178a46e77d6d24b20d2314165f Mon Sep 17 00:00:00 2001 From: Harin Lee Date: Thu, 1 Oct 2026 23:45:01 +0900 Subject: [PATCH 03/13] ALSA: ctxfi: Fix memory leakage when clearing DAO input mappers Commit e576e7843c0d ("ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions") centralized the DAO input mapper clearing logic in dao_clear_input(), but omitted the kfree() calls. It removes the mapper entries from the list and sets their pointers to NULL without freeing the allocated array. This leaks memory each time dao_clear_input() is called. Store the base pointer of the allocation before clearing the mapper entries and free the array after removing entries from the list. Fixes: e576e7843c0d ("ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions") Cc: stable@vger.kernel.org Signed-off-by: Harin Lee Link: https://patch.msgid.link/20261001144501.3050667-2-me@harin.net Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctdaio.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 9be70c6862ab..27e502fa493e 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -221,14 +221,18 @@ static int dao_set_right_input(struct dao *dao, struct rsc *input) static int dao_clear_input(struct dao *dao, unsigned int start, unsigned int end) { + struct imapper *entry; unsigned int i; - if (!dao->imappers[start]) + entry = dao->imappers[start]; + if (!entry) return 0; + for (i = start; i < end; i++) { dao->mgr->imap_delete(dao->mgr, dao->imappers[i]); dao->imappers[i] = NULL; } + kfree(entry); return 0; } From 408249089c48df1a4d23f60b831be3f3c311e6d1 Mon Sep 17 00:00:00 2001 From: Mohamed Yassine Jebabli Date: Thu, 1 Oct 2026 12:24:42 +0200 Subject: [PATCH 04/13] ALSA: hda/tas2781: Accept V1 calibration data without CRC Some Lenovo BIOSes (e.g. Legion Pro 7 16IRX9H, SSID 17aa:38cd) store valid V1 calibration data in the CALI_DATA EFI variable but leave the TimeStamp and CRC fields zeroed. The driver rejects such data, so the amplifiers always run with default parameters on these machines. The rejection was silent until commit 4fe238513407 ("ALSA: hda/tas2781: Move and unified the calibrated-data getting function for SPI and I2C into the tas2781_hda lib") turned it into "V1 CRC error". The stored values are consistent (R0 = 3.82/3.83 ohm, 1/R0 matching within 1e-4), so accept V1 data when both TimeStamp and CRC are zero and the first device entry is non-zero, and emit a warning. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Cc: stable@vger.kernel.org Signed-off-by: Mohamed Yassine Jebabli Link: https://patch.msgid.link/20261001-tas2781-v1-nocrc-v1-1-ff0c1032ce76@gmail.com Signed-off-by: Takashi Iwai --- sound/hda/codecs/side-codecs/tas2781_hda.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index b22f93424c62..e0da299df780 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -139,9 +139,21 @@ static void tas2781_apply_calib(struct tasdevice_priv *p) */ crc = crc32(~0, data, 84) ^ ~0; if (crc != tmp_val[21]) { - cali_data->total_sz = 0; - dev_err(p->dev, "%s: V1 CRC error\n", __func__); - return; + /* + * Some Lenovo BIOSes (e.g. Legion Pro 7 16IRX9H, + * SSID 17aa:38cd) store valid V1 calibration data + * but leave TimeStamp and CRC zeroed. + */ + if (!tmp_val[20] && !tmp_val[21] && + tmp_val[0] && tmp_val[1]) { + dev_warn(p->dev, + "%s: V1 data without CRC, accepting\n", + __func__); + } else { + cali_data->total_sz = 0; + dev_err(p->dev, "%s: V1 CRC error\n", __func__); + return; + } } for (j = p->ndev - 1; j >= 0; j--) { From 13dffca9043500ed6033d770b491a043fb952bc5 Mon Sep 17 00:00:00 2001 From: Jaidheer Sirigineedi Date: Fri, 2 Oct 2026 13:59:51 -0500 Subject: [PATCH 05/13] ALSA: hda/realtek: Add mute LED quirk for HP Pavilion 15-eg3xxx (103c:8bfd) The mute LED on the HP Pavilion Laptop 15-eg3xxx (PCI SSID 103c:8bfd, Realtek ALC287) does not work because the machine has no entry in the quirk table. No fixup is applied, so the mute LED is never driven, even though the mute function itself works. The LED is controlled via GPIO pin 4 (mask 0x10), which is exactly what ALC287_FIXUP_HP_GPIO_LED configures. This was verified on the hardware with hda-verb: setting the GPIO data bit lights the mute LED, and clearing it turns the LED off. hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x10 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x10 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x10 # LED on hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x00 # LED off Add a quirk entry for 103c:8bfd using ALC287_FIXUP_HP_GPIO_LED. Signed-off-by: Jaidheer Sirigineedi Link: https://patch.msgid.link/20261002185951.33855-1-jaidheer7@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 0b5e0b45b6b8..1623cf926c71 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -7646,6 +7646,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy x360 2-in-1 Laptop 15-fh0xxx", ALC245_FIXUP_HP_ENVY_X360_15_FH0XXX), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8bfd, "HP Pavilion 15-eg3xxx", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre x360 2-in-1 Laptop 16-aa0xxx", ALC245_FIXUP_HP_SPECTRE_X360_16_AA0XXX), From d0cb7a7b2a490c381ad97f53155d02a748b60dea Mon Sep 17 00:00:00 2001 From: Michael Garland Date: Mon, 5 Oct 2026 01:10:32 -0600 Subject: [PATCH 06/13] ASoC: amd: yc: Add DMI quirk for Acer Aspire Go 15 (AG15-21P) The Acer Aspire Go 15 (AG15-21P, board MDC Herbag2_MDU) has two PDM DMICs connected to the ACP. The BIOS does not set the AcpDmicConnected _DSD property, so the acp6x machine driver never registers the DMIC card. The existing Herbag_MDU entry does not match this board name, so add a DMI quirk for it. Tested on BIOS V1.11: the acp6x DMIC capture device is registered and records audio on both channels. Cc: stable@vger.kernel.org Signed-off-by: Michael Garland Link: https://patch.msgid.link/20261005-acp6x-herbag2-quirk-v1-1-a1f2683d758f@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 e798bf6a9cd0..1be8504571ad 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -787,6 +787,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "Herbag_MDU"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "MDC"), + DMI_MATCH(DMI_BOARD_NAME, "Herbag2_MDU"), + } + }, { .driver_data = &acp6x_card, .matches = { From 64dd172bbffe6135d79a06ddad53dd2da4d8bd62 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Tue, 6 Oct 2026 14:55:56 +0200 Subject: [PATCH 07/13] ASoC: adau1372: allow sleepy powerdown GPIOs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit gpiod_set_value() emits a WARN when the GPIO provider indicates that it can sleep, as is typical for I2C controlled GPIO expanders. Since the powerdown GPIO is only manipulated in process context, use the _cansleep variant instead. This lets the driver use sleepy GPIOs to power off the chip. Fixes: 6cd4c6459e47 ("ASoC: Add ADAU1372 audio CODEC support") Signed-off-by: Alvin Šipraga Acked-by: Nuno Sá Link: https://patch.msgid.link/20261006-asoc-adau1372-fixes-v1-1-599cc17fb5f6@analog.com Signed-off-by: Mark Brown --- sound/soc/codecs/adau1372.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c index cc174ec3a1f8..b84495371b0b 100644 --- a/sound/soc/codecs/adau1372.c +++ b/sound/soc/codecs/adau1372.c @@ -798,7 +798,7 @@ static int adau1372_set_power(struct adau1372 *adau1372, bool enable) if (ret) return ret; if (adau1372->pd_gpio) - gpiod_set_value(adau1372->pd_gpio, 0); + gpiod_set_value_cansleep(adau1372->pd_gpio, 0); if (adau1372->switch_mode) adau1372->switch_mode(adau1372->dev); @@ -818,8 +818,7 @@ static int adau1372_set_power(struct adau1372 *adau1372, bool enable) ADAU1372_CLK_CTRL_PLL_EN, 0); regcache_cache_only(adau1372->regmap, true); - if (adau1372->pd_gpio) - gpiod_set_value(adau1372->pd_gpio, 1); + gpiod_set_value_cansleep(adau1372->pd_gpio, 1); clk_disable_unprepare(adau1372->mclk); return ret; } @@ -836,7 +835,7 @@ static int adau1372_set_power(struct adau1372 *adau1372, bool enable) * map. No need to do any register writes to manually * turn things off. */ - gpiod_set_value(adau1372->pd_gpio, 1); + gpiod_set_value_cansleep(adau1372->pd_gpio, 1); regcache_mark_dirty(adau1372->regmap); } else { regmap_update_bits(adau1372->regmap, ADAU1372_REG_CLK_CTRL, From 36d0e05815ce84266c8b138cd4f84c3feeaf15f4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Tue, 6 Oct 2026 14:55:57 +0200 Subject: [PATCH 08/13] ASoC: adau1372: fix invalid default state of output ASRC and DAC muxes MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The chip has strange power-on reset values for the registers that control the output ASRC and DAC source muxes - the values are actually marked "Reserved" in the datasheet [1]. This means that until the underlying kcontrols are modified, the codec will not function as expected. The registers in question are: register / field reset value valid values ASRCO_SOURCE_0_1 0x10 ASRC_OUT_SOURCE0 0 4 .. 16 ASRC_OUT_SOURCE1 1 4 .. 16 ASRCO_SOURCE_2_3 0x10 ASRC_OUT_SOURCE2 2 4 .. 16 ASRC_OUT_SOURCE3 3 4 .. 16 DAC_SOURCE_0_1 0x10 DAC_SOURCE_0 0 12 .. 13 DAC_SOURCE_1 1 12 .. 13 Set some sane defaults during the driver probe: ADC Decimator0 -> Output ASRC0 ADC Decimator1 -> Output ASRC1 ADC Decimator2 -> Output ASRC2 ADC Decimator3 -> Output ASRC3 Input ASRC0 -> DAC0 Input ASRC1 -> DAC1 With this, the chip will behave as advertised by the default kcontrol state exposed by the driver. [1] https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1372.pdf Fixes: 6cd4c6459e47 ("ASoC: Add ADAU1372 audio CODEC support") Signed-off-by: Alvin Šipraga Acked-by: Nuno Sá Link: https://patch.msgid.link/20261006-asoc-adau1372-fixes-v1-2-599cc17fb5f6@analog.com Signed-off-by: Mark Brown --- sound/soc/codecs/adau1372.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c index b84495371b0b..c3d3be898ba6 100644 --- a/sound/soc/codecs/adau1372.c +++ b/sound/soc/codecs/adau1372.c @@ -1002,6 +1002,17 @@ int adau1372_probe(struct device *dev, struct regmap *regmap, regmap_write(regmap, ADAU1372_REG_OP_STAGE_MUTE, 0x0); + /* + * Set sane default values for the Output ASRC and DAC input muxes, + * since the power-on reset defaults are invalid "Reserved" states. + */ + regmap_write(regmap, ADAU1372_REG_ASRCO_SOURCE_0_1, + 0x54); /* Decimator0, Decimator1 */ + regmap_write(regmap, ADAU1372_REG_ASRCO_SOURCE_2_3, + 0x76); /* Decimator2, Decimator3 */ + regmap_write(regmap, ADAU1372_REG_DAC_SOURCE, + 0xdc); /* Input ASRC0, Input ASRC1 */ + regmap_write(regmap, 0x7, 0x01); /* CLOCK OUT */ return devm_snd_soc_register_component(dev, &adau1372_driver, &adau1372_dai_driver, 1); From 7529254014764ae0b8c33d5c61a58f36c65b1a1c Mon Sep 17 00:00:00 2001 From: Marek Szyprowski Date: Fri, 2 Oct 2026 11:21:12 +0200 Subject: [PATCH 09/13] ASoC: samsung: i2s: enable op_clk in probe to balance runtime PM The probe function puts the device into the runtime PM active state with only the "iis" bus clock enabled, but it also sets priv->op_clk to the parent of the RCLK_SRC mux without enabling it. The runtime PM suspend callback disables both clocks, so the first runtime suspend after probe disables op_clk once more than it was enabled. On Exynos5250 Arndale both clocks resolve to the same EXYNOS_I2S_BUS gate, so this triggers the following warning during boot: WARNING: drivers/clk/clk.c:1270 at clk_core_disable+0xe4/0x23c, CPU#0: kworker/u8:3/45 i2s_bus already disabled ... Workqueue: pm pm_runtime_work Call trace: ... clk_core_disable from clk_disable+0x28/0x34 clk_disable from i2s_runtime_suspend+0x58/0x68 [snd_soc_i2s] i2s_runtime_suspend [snd_soc_i2s] from genpd_runtime_suspend+0xc4/0x298 genpd_runtime_suspend from __rpm_callback+0x48/0x18c __rpm_callback from rpm_callback+0x4c/0x50 rpm_callback from rpm_suspend+0xf4/0x718 rpm_suspend from pm_runtime_work+0x9c/0xa0 ... Fix this by enabling op_clk in probe right after getting it, so the clock state matches the runtime PM active state, and by disabling it in remove. Fixes: 48279c53fd1d ("ASoC: samsung: i2s: Prevent external abort on exynos5433 I2S1 access") Assisted-by: Claude Opus 5.5 Signed-off-by: Marek Szyprowski Link: https://patch.msgid.link/20261002092116.585466-1-m.szyprowski@samsung.com Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 3e76cb81462f..442e70b107da 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1534,9 +1534,16 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err_disable_pm; priv->op_clk = clk_get_parent(priv->clk_table[CLK_I2S_RCLK_SRC]); + ret = clk_prepare_enable(priv->op_clk); + if (ret < 0) { + priv->op_clk = NULL; + goto err_unregister_clk; + } return 0; +err_unregister_clk: + i2s_unregister_clock_provider(priv); err_disable_pm: pm_runtime_disable(&pdev->dev); err_del_sec: @@ -1559,6 +1566,7 @@ static void samsung_i2s_remove(struct platform_device *pdev) i2s_unregister_clock_provider(priv); i2s_delete_secondary_device(priv); + clk_disable_unprepare(priv->op_clk); clk_disable_unprepare(priv->clk); pm_runtime_put_noidle(&pdev->dev); From 86c8360c97f66fa7830fd930e2dc69374007284b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Oct 2026 16:52:34 +0100 Subject: [PATCH 10/13] ASoC: cs35l56: Add DAI for SDCA OT25 stream Add a DAI for the SDCA OT25 stream on SoundWire DP4. The SDCA-defined OT25 stream is an amp reference feedback, typically used for AEC. The firmware outputs this on DP4. The firmware owns all control registers for this stream so there are no mixer controls or any other configuration options. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20261005155235.1386525-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56-sdw.c | 9 ++++-- sound/soc/codecs/cs35l56.c | 54 ++++++++++++++++++++++++++++++++++ 3 files changed, 62 insertions(+), 2 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 45a5df574aa6..747e10ec1a27 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -312,6 +312,7 @@ struct snd_ctl_elem_value; #define CS35L56_SDW1_PLAYBACK_PORT 1 #define CS35L56_SDW1_CAPTURE_PORT 3 +#define CS35L56_OT25_CAPTURE_PORT 4 #define CS35L56_NUM_BULK_SUPPLIES 3 #define CS35L56_NUM_DSP_REGIONS 5 diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index 98bb4542b914..e10f04d677f4 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -265,11 +265,11 @@ static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) if (ret == 0) prop->clk_stop_mode1 = !!clock_stop_1; - ports = devm_kcalloc(cs35l56->base.dev, 2, sizeof(*ports), GFP_KERNEL); + ports = devm_kcalloc(cs35l56->base.dev, 3, sizeof(*ports), GFP_KERNEL); if (!ports) return -ENOMEM; - prop->source_ports = BIT(CS35L56_SDW1_CAPTURE_PORT); + prop->source_ports = BIT(CS35L56_SDW1_CAPTURE_PORT) | BIT(CS35L56_OT25_CAPTURE_PORT); prop->sink_ports = BIT(CS35L56_SDW1_PLAYBACK_PORT); prop->paging_support = true; prop->use_domain_irq = true; @@ -286,6 +286,11 @@ static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) ports[1].num = CS35L56_SDW1_CAPTURE_PORT; ports[1].type = SDW_DPN_FULL; ports[1].ch_prep_timeout = 10; + + /* DP4 - capture */ + ports[2].num = CS35L56_OT25_CAPTURE_PORT; + ports[2].type = SDW_DPN_FULL; + ports[2].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[1]; dev_dbg(&peripheral->dev, "clock stop mode 1 supported: %s\n", diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 35d210626627..279db3f13f1e 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include @@ -324,6 +325,7 @@ static const struct snd_soc_dapm_widget cs35l56_dapm_widgets[] = { SND_SOC_DAPM_SIGGEN("VDDBMON ADC"), SND_SOC_DAPM_SIGGEN("VBSTMON ADC"), SND_SOC_DAPM_SIGGEN("TEMPMON ADC"), + SND_SOC_DAPM_SIGGEN("OT25 Reference"), SND_SOC_DAPM_INPUT("Calibrate"), }; @@ -388,6 +390,8 @@ static const struct snd_soc_dapm_route cs35l56_audio_map[] = { { "SDW1 Capture", NULL, "SDW1 TX2 Source" }, { "SDW1 Capture", NULL, "SDW1 TX3 Source" }, { "SDW1 Capture", NULL, "SDW1 TX4 Source" }, + + { "OT25", NULL, "OT25 Reference" }, }; static int cs35l56_dsp_event(struct snd_soc_dapm_widget *w, @@ -712,6 +716,37 @@ static int cs35l56_sdw_dai_set_stream(struct snd_soc_dai *dai, return 0; } +static int cs35l56_ot25_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l56_private *cs35l56 = snd_soc_component_get_drvdata(dai->component); + struct sdw_stream_runtime *sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + struct sdw_stream_config sconfig = { }; + struct sdw_port_config pconfig = { }; + int ret; + + dev_dbg(cs35l56->base.dev, "%s: rate %d\n", __func__, params_rate(params)); + + if (!cs35l56->base.init_done) + return -ENODEV; + + if (!sdw_stream) + return -EINVAL; + + snd_sdw_params_to_config(substream, params, &sconfig, &pconfig); + pconfig.num = CS35L56_OT25_CAPTURE_PORT; + + ret = sdw_stream_add_slave(cs35l56->sdw_peripheral, &sconfig, &pconfig, + 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "Failed to add OT25 stream: %d\n", ret); + return ret; + } + + return 0; +} + static const struct snd_soc_dai_ops cs35l56_sdw_dai_ops = { .set_tdm_slot = cs35l56_sdw_dai_set_tdm_slot, .shutdown = cs35l56_sdw_dai_shutdown, @@ -720,6 +755,13 @@ static const struct snd_soc_dai_ops cs35l56_sdw_dai_ops = { .set_stream = cs35l56_sdw_dai_set_stream, }; +static const struct snd_soc_dai_ops cs35l56_ot25_dai_ops = { + .shutdown = cs35l56_sdw_dai_shutdown, + .hw_params = cs35l56_ot25_dai_hw_params, + .hw_free = cs35l56_sdw_dai_hw_free, + .set_stream = cs35l56_sdw_dai_set_stream, +}; + static struct snd_soc_dai_driver cs35l56_dai[] = { { .name = "cs35l56-asp1", @@ -768,6 +810,18 @@ static struct snd_soc_dai_driver cs35l56_dai[] = { .symmetric_rate = 1, .ops = &cs35l56_sdw_dai_ops, }, + { + .name = "cs35l56-ot25", + .id = 3, + .capture = { + .stream_name = "OT25", + .channels_min = 1, + .channels_max = 2, + .rates = CS35L56_RATES, + .formats = CS35L56_TX_FORMATS, + }, + .ops = &cs35l56_ot25_dai_ops, + }, }; static int cs35l56_write_cal(struct cs35l56_private *cs35l56) From 4b7f393a01f88e8d03cba94cf4948652de14789b Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 5 Oct 2026 16:52:35 +0100 Subject: [PATCH 11/13] ASoC: sdw_utils: Switch CS35L56/57/62/63 to use OT25 DAI Change the CS35L56/57/62/63 entries in codec_info_list[] to use the OT25 DAI for amp feedback. The AMP feedback was incorrectly using a DAI connected to DP3, but that is reserved for SDCA companion amp. The correct amp output is DP4, which is the SDCA OT25 feedback output. Fixes: 898cd43bde307 ("ASoC: intel: sof_sdw: Add support for CS35L63 into machine driver") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20261005155235.1386525-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index d2eeef4931c6..eda7c567f71e 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -808,7 +808,7 @@ struct asoc_sdw_codec_info codec_info_list[] = { }, { .direction = {false, true}, - .dai_name = "cs35l56-sdw1c", + .dai_name = "cs35l56-ot25", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, }, @@ -836,7 +836,7 @@ struct asoc_sdw_codec_info codec_info_list[] = { }, { .direction = {false, true}, - .dai_name = "cs35l56-sdw1c", + .dai_name = "cs35l56-ot25", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, }, @@ -864,7 +864,7 @@ struct asoc_sdw_codec_info codec_info_list[] = { }, { .direction = {false, true}, - .dai_name = "cs35l56-sdw1c", + .dai_name = "cs35l56-ot25", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, }, @@ -892,7 +892,7 @@ struct asoc_sdw_codec_info codec_info_list[] = { }, { .direction = {false, true}, - .dai_name = "cs35l56-sdw1c", + .dai_name = "cs35l56-ot25", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, }, From 7587fc10cba6c4a8d4c30ba480200370bc68dc74 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 7 Oct 2026 13:08:35 +0000 Subject: [PATCH 12/13] ASoC: amd: acp: Add DMI override for ASUS EXPERTBOOK AM7406CKA A BIOS for this SKU has been observed with an incorrect acp-audio-config-flag, add the override so that it is forced to the correct value. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20261007130835.102325-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index b3b73096ac74..3d48bb0f100f 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -30,6 +30,13 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "HN7306EA"), }, }, + { + /* ASUS EXPERTBOOK AM7406CKA (Kraken Point, ACP 7.0) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "ASUS EXPERTBOOK AM7406CKA"), + }, + }, { /* ASUS Zenbook S16 UM5606GA (Strix Point, ACP 7.0) */ .matches = { From 9ca0d0d0249230135dbf6fa1b9168e260151babc Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 7 Oct 2026 13:08:57 +0000 Subject: [PATCH 13/13] ASoC: amd: acp: Add more ACP7.0 match entries for Cirrus Logic parts This adds the following SoundWire configuration: cs42l43b link 0 UID 1 cs35l57 link 0 UID 0 cs35l57 link 0 UID 1 Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20261007130857.102432-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/amd-acp70-acpi-match.c | 35 ++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 2a9da6de7bbc..c2f0479301eb 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -335,6 +335,27 @@ static const struct snd_soc_acpi_adr_device cs42l43b_l0u1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs42l43b_l0u1_cs35l57x2_l0u01_adr[] = { + { + .adr = 0x00003101FA2A3B01ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + }, + { + .adr = 0x00003001FA355701ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00003101FA355701ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, +}; + static const struct snd_soc_acpi_adr_device cs42l43_l1u0_cs35l56x4_l1u0123_adr[] = { { .adr = 0x00013001FA424301ull, @@ -438,6 +459,15 @@ static const struct snd_soc_acpi_link_adr acp70_cs42l43_l0u0_cs35l56x4_l1u3210[] {} }; +static const struct snd_soc_acpi_link_adr acp70_cs42l43b_l0u1_cs35l57x2_l0u01[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43b_l0u1_cs35l57x2_l0u01_adr), + .adr_d = cs42l43b_l0u1_cs35l57x2_l0u01_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr acp70_cs42l43_l1u0_cs35l56x4_l1u0123[] = { { .mask = BIT(1), @@ -820,6 +850,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { .links = acp70_cs42l45_l1u0_cs35l63x2_l0u01, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(0), + .links = acp70_cs42l43b_l0u1_cs35l57x2_l0u01, + .drv_name = "amd_sdw", + }, { .link_mask = BIT(1), .links = acp70_cs42l43_l1u0_cs35l56x4_l1u0123,