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[PATCH v1 1/1] ASoC: atmel: tse850-pcm5142: Remove unused of_gpio.h
by Andy Shevchenko 08 May '24

08 May '24

of_gpio.h is deprecated and subject to remove. The driver doesn't use it, simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/atmel/tse850-pcm5142.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 611da23325d3..5d208e0b4b90 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -35,10 +35,9 @@ // of the (filtered) output from the PCM5142 codec. #include <linux/clk.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/regulator/consumer.h> #include <sound/soc.h> -- 2.43.0.rc1.1336.g36b5255a03ac

2 1

[PATCH v2 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 08 May '24

08 May '24

Hi, First patch is a build dependency. Changes in v2: - Re-work most of the idea according to Srini comments: set channel mapping for backend DAIs, not frontend. - Patch #1: no changes - Patch #2 is entirely replaced - now channel mapping is implemented in q6apm-lpass-dais. - Patch #3: rework to new approach, but most of the code stays. - Patch #4: rework significantly, because only backend DAIs is now affected. - Link to v1: https://lore.kernel.org/r/20240507-asoc-x1e80100-4-channel-mapping-v1-0-b12… Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6apm-lpass-dais: Implement proper channel mapping ASoC: qcom: qdsp6: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 30 +++++++----------------------- sound/soc/qcom/qdsp6/audioreach.h | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 2 ++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 21 +++++++++++++-------- sound/soc/qcom/x1e80100.c | 18 ++++++++++++++++++ sound/soc/soc-dai.c | 4 ++-- 15 files changed, 79 insertions(+), 57 deletions(-) --- base-commit: 2b84edefcad14934796fad37b16512b6a2ca467e change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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[PATCH v1] ALSA: hda/realtek: Add support for ASUS Zenbook 2024 HN7306W
by Stefan Binding 08 May '24

08 May '24

This laptop uses CS35L41 HDA with 2 amps using I2C and Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8502b30d5d5d..8c157158f1cb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10276,6 +10276,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

2 1

[PATCH 0/4] ASoC: Constify static snd_pcm_hardware
by Krzysztof Kozlowski 08 May '24

08 May '24

No dependencies. Static 'struct snd_pcm_hardware' is not modified by few drivers and its copy is passed to the core, so it can be made const for increased code safety. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: qcom: Constify static snd_pcm_hardware ASoC: fsl: Constify static snd_pcm_hardware ASoC: meson: Constify static snd_pcm_hardware ASoC: uniphier: Constify static snd_pcm_hardware sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/meson/aiu-fifo-i2s.c | 2 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.h | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/uniphier/aio-dma.c | 2 +- 8 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: ebbaba914053d8bcadd7d64eb8d2555299509a27 change-id: 20240429-n-asoc-const-snd-pcm-hardware-f0a5ade885ca Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 5

[PATCH AUTOSEL 6.1 01/25] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 094445036c20f..d6ef8e850412b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.6 01/43] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e609249cc38d5..651408c6f399d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH AUTOSEL 6.8 01/52] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Sasha Levin 08 May '24

08 May '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit e50729d742ec364895f1c389c32315984a987aa5 ] The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05f38d1f7d824..b41a1147f1c34 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,28 +636,30 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | BYT_RT5640_OVCD_TH_2000UA | BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MONO_SPEAKER | - BYT_RT5640_DIFF_MIC | - BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, { -- 2.43.0

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[PATCH 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, First patch is a build dependency. Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6dsp: Implement proper channel mapping in Audioreach ASoC: qcom: q6dsp: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 +++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 14 +++++-------- sound/soc/qcom/qdsp6/audioreach.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 12 +++++++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 ++++-- sound/soc/qcom/qdsp6/q6apm.c | 28 ++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6apm.h | 8 +++++++ sound/soc/qcom/qdsp6/topology.c | 12 +++++++++++ sound/soc/qcom/x1e80100.c | 37 +++++++++++++++++++++++++++++++-- sound/soc/soc-dai.c | 4 ++-- 18 files changed, 142 insertions(+), 39 deletions(-) --- base-commit: c5e512ffe106f751c61e5a036560f522e58eadcd change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 9

[PATCH 0/3] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 07 May '24

07 May '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Artur Weber (3): ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection .../bindings/sound/samsung,midas-audio.yaml | 30 +++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 23 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 265 ++++++++++++++++++++- 4 files changed, 305 insertions(+), 15 deletions(-) --- base-commit: e67572cd2204894179d89bd7b984072f19313b03 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

2 8

[PATCH v2] ASoC: Intel: avs: boards: Properly name input device
by Amadeusz Sławiński 07 May '24

07 May '24

Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- v2: - Rebase on top of correct tree. --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- 5 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index ff3bd15132694..f4919a6abc278 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..eef718c6b8ad9 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..51ebc790c6b4d 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..67fd362962c40 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..1003e8689eacc 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; -- 2.34.1

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[PATCH 0/4] ASoC: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 07 May '24

07 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (4): ASoC: codecs: wm8962: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8993: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8994: use 'time_left' variable with wait_for_completion_timeout() ASoC: codecs: wm8996: use 'time_left' variable with wait_for_completion_timeout() sound/soc/codecs/wm8962.c | 12 ++++++------ sound/soc/codecs/wm8993.c | 12 ++++++------ sound/soc/codecs/wm8994.c | 8 ++++---- sound/soc/codecs/wm8996.c | 14 +++++++------- 4 files changed, 23 insertions(+), 23 deletions(-) -- 2.43.0

3 6

[PATCH 00/13] ASoC: Use snd_soc_substream_to_rtd() for accessing private_data
by Krzysztof Kozlowski 07 May '24

07 May '24

Hi, Do not open-code snd_soc_substream_to_rtd() when accessing snd_pcm_substream->private_data. This makes code more consistent with rest of ASoC and allows in the future to move the field to any other place or add additional checks in snd_soc_substream_to_rtd(). Best regards, Krzysztof --- Krzysztof Kozlowski (13): ASoC: qcom: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: tegra: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: ti: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: arm: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: amd: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: fsl: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: img: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: kirkwood: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: loongson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: mediatek: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: meson: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: samsung: Use snd_soc_substream_to_rtd() for accessing private_data ASoC: sunxi: Use snd_soc_substream_to_rtd() for accessing private_data sound/arm/pxa2xx-pcm-lib.c | 4 ++-- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-card.c | 6 +++--- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/loongson/loongson_card.c | 2 +- sound/soc/loongson/loongson_dma.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 14 +++++++------- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++---- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 6 +++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +++++----- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ++-- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 4 ++-- sound/soc/qcom/qdsp6/q6apm-dai.c | 2 +- sound/soc/qcom/sc7180.c | 10 +++++----- sound/soc/qcom/sc7280.c | 12 ++++++------ sound/soc/qcom/sc8280xp.c | 8 ++++---- sound/soc/qcom/sdw.c | 8 ++++---- sound/soc/qcom/sm8250.c | 10 +++++----- sound/soc/qcom/x1e80100.c | 8 ++++---- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra_asoc_machine.c | 2 +- sound/soc/tegra/tegra_pcm.c | 6 +++--- sound/soc/ti/omap-hdmi.c | 2 +- 35 files changed, 82 insertions(+), 82 deletions(-) --- base-commit: 82415cf72c7e224be7a6496f3a53c0b365c2fe9d change-id: 20240430-asoc-snd-substream-clean-924b717d8f54 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH] ASoC: qcom: audioreach: Correct mapping of back speakers
by Krzysztof Kozlowski 07 May '24

07 May '24

Qualcomm DSP, according to downstream sources, expects back speakers to be mapped as "back", not "surround". The surround is used only for 8+ speakers configuration. Reported-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 3c5fcb20e07e ("ASoC: qcom: audioreach: Add 4 channel support") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/audioreach.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5291deac0a0b..c655f0213723 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -277,8 +277,8 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) } else if (num_channels == 4) { ch_map[0] = PCM_CHANNEL_FL; ch_map[1] = PCM_CHANNEL_FR; - ch_map[2] = PCM_CHANNEL_LS; - ch_map[3] = PCM_CHANNEL_RS; + ch_map[2] = PCM_CHANNEL_LB; + ch_map[3] = PCM_CHANNEL_RB; } } @@ -851,8 +851,8 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, } else if (num_channels == 4) { media_format->channel_mapping[0] = PCM_CHANNEL_FL; media_format->channel_mapping[1] = PCM_CHANNEL_FR; - media_format->channel_mapping[2] = PCM_CHANNEL_LS; - media_format->channel_mapping[3] = PCM_CHANNEL_RS; + media_format->channel_mapping[2] = PCM_CHANNEL_LB; + media_format->channel_mapping[3] = PCM_CHANNEL_RB; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); -- 2.43.0

1 1

[PATCH v3 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 07 May '24

07 May '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 78 + .../bindings/sound/qcom,wcd937x.yaml | 58 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1147 +++++++ sound/soc/codecs/wcd937x.c | 3036 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 4999 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 5e4f84f18c4ee9b0ccdc19e39b7de41df21699dd -- 2.25.1

4 13

[PATCH v1] slimbus: qcom-ngd-ctrl: Add stream disable support
by Viken Dadhaniya 07 May '24

07 May '24

Currently slimbus driver doesn't support stream disable callback, it only supports stream enable callback. In slimbus usecase, client is switching to new frequency with same channel and calling enable stream callback for new frequency but DSP subsystem is crashing as we are switching to new frequency with same channel without disabling stream for older frequency. Ideally, before switching to another frequency, client should call disable stream callback and then enable stream for newer frequency. Hence add support to disable stream via qcom_slim_ngd_disable_stream(). Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 70 +++++++++++++++++++++++++++++++++ drivers/slimbus/slimbus.h | 13 ++++++ 2 files changed, 83 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index e0b21f0f79c1..d952827d2e12 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1,6 +1,7 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved. // Copyright (c) 2018, Linaro Limited +// Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. #include <linux/irq.h> #include <linux/kernel.h> @@ -1084,6 +1085,74 @@ static int qcom_slim_ngd_enable_stream(struct slim_stream_runtime *rt) return ret; } +static int qcom_slim_ngd_disable_stream(struct slim_stream_runtime *rt) +{ + struct slim_device *sdev = rt->dev; + struct slim_controller *ctrl = sdev->ctrl; + struct slim_val_inf msg = {0}; + u8 wbuf[SLIM_MSGQ_BUF_LEN]; + u8 rbuf[SLIM_MSGQ_BUF_LEN]; + struct slim_msg_txn txn = {0,}; + int i, ret; + + txn.mt = SLIM_MSG_MT_DEST_REFERRED_USER; + txn.dt = SLIM_MSG_DEST_LOGICALADDR; + txn.la = SLIM_LA_MGR; + txn.ec = 0; + txn.msg = &msg; + txn.msg->num_bytes = 0; + txn.msg->wbuf = wbuf; + txn.msg->rbuf = rbuf; + + for (i = 0; i < rt->num_ports; i++) { + struct slim_port *port = &rt->ports[i]; + + if (txn.msg->num_bytes == 0) { + wbuf[txn.msg->num_bytes++] = (u8)(SLIM_CH_REMOVE << 6) + | (sdev->laddr & 0x1f); + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(&sdev->dev, "Fail to allocate TID\n"); + return -ENXIO; + } + wbuf[txn.msg->num_bytes++] = txn.tid; + } + wbuf[txn.msg->num_bytes++] = port->ch.id; + } + + txn.mc = SLIM_USR_MC_CHAN_CTRL; + txn.rl = txn.msg->num_bytes + 4; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + return ret; + } + + txn.mc = SLIM_USR_MC_RECONFIG_NOW; + txn.msg->num_bytes = 2; + wbuf[1] = sdev->laddr; + txn.rl = txn.msg->num_bytes + 4; + + ret = slim_alloc_txn_tid(ctrl, &txn); + if (ret) { + dev_err(ctrl->dev, "Fail to allocate TID ret:%d\n", ret); + return ret; + } + + wbuf[0] = txn.tid; + ret = qcom_slim_ngd_xfer_msg_sync(ctrl, &txn); + if (ret) { + slim_free_txn_tid(ctrl, &txn); + dev_err(&sdev->dev, "TX timed out:MC:0x%x,mt:0x%x ret:%d\n", + txn.mc, txn.mt, ret); + } + + return ret; +} + static int qcom_slim_ngd_get_laddr(struct slim_controller *ctrl, struct slim_eaddr *ea, u8 *laddr) { @@ -1642,6 +1711,7 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) ctrl->ctrl.clkgear = SLIM_MAX_CLK_GEAR; ctrl->ctrl.get_laddr = qcom_slim_ngd_get_laddr; ctrl->ctrl.enable_stream = qcom_slim_ngd_enable_stream; + ctrl->ctrl.disable_stream = qcom_slim_ngd_disable_stream; ctrl->ctrl.xfer_msg = qcom_slim_ngd_xfer_msg; ctrl->ctrl.wakeup = NULL; ctrl->state = QCOM_SLIM_NGD_CTRL_DOWN; diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..21543d125614 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -1,6 +1,7 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2024 Qualcomm Innovation Center, Inc. All rights reserved. */ #ifndef _DRIVERS_SLIMBUS_H @@ -316,6 +317,18 @@ enum slim_transport_protocol { SLIM_PROTO_EXT_HALF_DUP, }; +/* + * enum slim_ch_control: Channel control. + * Activate will schedule channel and/or group of channels in the TDM frame. + * Suspend will keep the schedule but data-transfer won't happen. + * Remove will remove the channel/group from the TDM frame. + */ +enum slim_ch_control { + SLIM_CH_ACTIVATE, + SLIM_CH_SUSPEND, + SLIM_CH_REMOVE, +}; + /** * struct slim_stream_runtime - SLIMbus stream runtime instance * -- 2.17.1

2 1

[PATCH] [RFC PATCH] ALSA: usb-audio: endpoint: Prevent NULL pointer deference in snd_usb_endpoint_close
by Rajashekar kuruva 07 May '24

07 May '24

When multiple plug-in and plug-out events occur, there is a risk of encountering a NULL pointer dereference leading to a kernel panic during a headset use-case. this issue arises in the snd_usb_endpoint_close function To avoid check if ep->iface_ref is not null before decrementing its opened count. If ep->iface_ref is null, we skip the decrement and the subsequent logic. Signed-off-by: Rajashekar kuruva <quic_kuruva(a)quicinc.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8f65349a06d3..0e3101b7e392 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -950,7 +950,7 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened && + if (ep->iface_ref && !--ep->iface_ref->opened && !(chip->quirk_flags & QUIRK_FLAG_IFACE_SKIP_CLOSE)) endpoint_set_interface(chip, ep, false); -- 2.25.1

3 9

[PATCH] ASoC: qcom: q6apm-dai: drop unused 'q6apm_dai_rtd' fields
by Krzysztof Kozlowski 07 May '24

07 May '24

Remove few unused fields from 'struct q6apm_dai_rtd'. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5c..cee23d4bda6f 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -70,14 +70,10 @@ struct q6apm_dai_rtd { unsigned int bytes_received; unsigned int copied_total; uint16_t bits_per_sample; - uint16_t source; /* Encoding source bit mask */ - uint16_t session_id; bool next_track; enum stream_state state; struct q6apm_graph *graph; spinlock_t lock; - uint32_t initial_samples_drop; - uint32_t trailing_samples_drop; bool notify_on_drain; }; @@ -720,14 +716,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component, switch (metadata->key) { case SNDRV_COMPRESS_ENCODER_PADDING: - prtd->trailing_samples_drop = metadata->value[0]; q6apm_remove_trailing_silence(component->dev, prtd->graph, - prtd->trailing_samples_drop); + metadata->value[0]); break; case SNDRV_COMPRESS_ENCODER_DELAY: - prtd->initial_samples_drop = metadata->value[0]; q6apm_remove_initial_silence(component->dev, prtd->graph, - prtd->initial_samples_drop); + metadata->value[0]); break; default: ret = -EINVAL; -- 2.43.0

2 1

[PATCH v2] ASoC: tas2781: Fix a warning reported by robot kernel test
by Shenghao Ding 07 May '24

07 May '24

Fix a warning reported by robot kernel test that 'fw_entry' in function 'tas2781_load_calibration' is used uninitialized with compiler sh4-linux-gcc (GCC) 13.2.0, an update of copyright and a correction of the comments. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Describe more what was actually an issue reported by robot kernel test. - Correct the subject suffix, ALSA: ASoc/tas2781 --> ASoC: tas2781. v1: - Update the copyright year to 2024 - tasdevice-fmw.c --> tas2781-fmwlib.c - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202405021200.YHInjV43-lkp@intel.com/ --- sound/soc/codecs/tas2781-fmwlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..a6be81adcb83 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tasdevice-fmw.c -- TASDEVICE firmware support +// tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; -- 2.34.1

3 2

[PATCH 0/7] ASoC: SOF: Intel: remove circular dependency for SoundWire BPT
by Pierre-Louis Bossart 07 May '24

07 May '24

The SoundWire BPT support will rely on the HDaudio DMA. This exposes a circular dependency module dependency which has to be resolved by splitting common parts used by HDaudio and SoundWire parts, and 'generic' parts used by HDaudio only. This patchset does not change any functionality, it just moves code around, exposes symbols that are used in the new module. The code has been in use for more than one kernel cycle already so it really shouldn't break any existing platforms. The main issue with such code moves is that it makes backports or fixes more complicated. That's the main reason why we held back these patches until we were reasonably confident on the maturity of MTL and LNL drivers. Pierre-Louis Bossart (7): ASoC: SOF: Intel: hda-stream: export stream_get_position() helper ASoC: SOF: Intel: regroup all SoundWire/Intel functions in hda.c ASoC: SOF: Intel: start splitting top-level from common parts ASoC: SOF: Intel: move common code from hda.c ASoC: SOF: Intel: move tracepoint creation ASoC: SOF: Intel: remove circular dependency on hda_sdw_process_wakeen() ASoC: SOF: Intel: move hda.c to different module sound/soc/sof/intel/Kconfig | 25 +- sound/soc/sof/intel/Makefile | 23 +- sound/soc/sof/intel/apl.c | 3 - sound/soc/sof/intel/cnl.c | 15 +- sound/soc/sof/intel/hda-bus.c | 2 + sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-ctrl.c | 10 + sound/soc/sof/intel/hda-dai.c | 6 + sound/soc/sof/intel/hda-dsp.c | 502 ++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 115 +++++ sound/soc/sof/intel/hda-loader.c | 45 +- sound/soc/sof/intel/hda-pcm.c | 6 + sound/soc/sof/intel/hda-probes.c | 2 + sound/soc/sof/intel/hda-stream.c | 13 + sound/soc/sof/intel/hda-trace.c | 3 + sound/soc/sof/intel/hda.c | 605 +++------------------------ sound/soc/sof/intel/hda.h | 15 + sound/soc/sof/intel/icl.c | 4 +- sound/soc/sof/intel/lnl.c | 4 +- sound/soc/sof/intel/mtl.c | 21 +- sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 2 + sound/soc/sof/intel/pci-lnl.c | 3 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 2 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/telemetry.c | 1 + sound/soc/sof/intel/tgl.c | 10 +- sound/soc/sof/intel/tracepoints.c | 5 + 31 files changed, 811 insertions(+), 637 deletions(-) create mode 100644 sound/soc/sof/intel/tracepoints.c -- 2.40.1

2 8

[PATCH 0/3] ASoC: SOF: Intel: HDA/DMIC updates
by Pierre-Louis Bossart 07 May '24

07 May '24

The first patch handles a problematic configuration where the wrong machine driver/topology is used: when the hardware reports an external HDaudio codec the direction is to ignore/discard ACPI SoundWire devices. The last two patch deal with DMIC format configurations and allow users to select S16_LE even if the DMIC and internal copiers only support 24 or 32-bits. The code changes are located in sound/soc/sof/ but in the scope of Intel DAIs. Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Allow selective update in sof_ipc4_update_hw_params ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request Pierre-Louis Bossart (1): ASoC: SOF: Intel: discard SoundWire configuration if HDaudio codec is reported sound/soc/sof/intel/hda.c | 9 +- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/ipc4-topology.c | 245 ++++++++++++++++++++++++++-------- 3 files changed, 196 insertions(+), 60 deletions(-) -- 2.40.1

2 4

[PATCH v3] ALSA: usb-audio: Add sampling rates support for Mbox3
by Manuel Barrio Linares 07 May '24

07 May '24

This adds support for all sample rates supported by the hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: Manuel Barrio Linares <mbarriolinares(a)gmail.com> --- v2: Cleaned up code. Explicit __le32 usage. v3: Get rid of sparse warnings sound/usb/quirks-table.h | 38 +++++++++++++-------- sound/usb/quirks.c | 74 +++++++++++++++++++++++++++++++--------- 2 files changed, 81 insertions(+), 31 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8..73abc38 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .implicit_fb = 0, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e6..2f961f0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,46 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + __le32 buff4 = 0; + u8 buff1 = 0x01; + u32 new_rate = subs->data_endpoint->cur_rate; + u32 current_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + current_rate = le32_to_cpu(buff4); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", current_rate); + if (current_rate == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed (current rate:%d == new rate:%d)", + current_rate, new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + buff4 = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &buff4, 4); + + // Set clock source to Internal + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buff1, 1); + + // Check whether the change was successful + buff4 = 0; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + if (new_rate != le32_to_cpu(buff4)) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1792,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } -- 2.45.0

3 4

[PATCH] ASoC: Intel: avs: boards: Properly name input device
by Amadeusz Sławiński 06 May '24

06 May '24

Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- sound/soc/intel/avs/boards/rt5640.c | 3 ++- 6 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 3d77bc3344702..6440a46e795ec 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..eef718c6b8ad9 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..51ebc790c6b4d 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..67fd362962c40 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..1003e8689eacc 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt5640.c b/sound/soc/intel/avs/boards/rt5640.c index 07cd8db4e6c2b..7c94680c4b3ad 100644 --- a/sound/soc/intel/avs/boards/rt5640.c +++ b/sound/soc/intel/avs/boards/rt5640.c @@ -62,7 +62,8 @@ static int avs_rt5640_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; -- 2.34.1

1 0

[PATCH v3 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Modification for add tas2781 driver for SPI ALSA: hda/tas2781: Main code of tas2781 driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 driver for SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 10 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 165 ++ sound/pci/hda/tas2781_hda_spi.c | 1305 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2272 +++++++++++++++++ 8 files changed, 3783 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

4 10

[PATCH v4 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Add tas2781 hda driver based on SPI ALSA: hda/tas2781: Tas2781 hda driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 hda driver based on SPI drivers/acpi/scan.c | 2 + .../platform/x86/serial-multi-instantiate.c | 13 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 149 ++ sound/pci/hda/tas2781_hda_spi.c | 1240 +++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2252 +++++++++++++++++ 8 files changed, 3685 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

5 12

[PATCH] ALSA: hda: Add Intel BMG PCI ID and HDMI codec vid
by Chaitanya Kumar Borah 06 May '24

06 May '24

Add HD Audio PCI ID and HDMI codec vendor ID for Intel Battlemage. Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah(a)intel.com> --- include/linux/pci_ids.h | 1 + include/sound/hdaudio.h | 1 + sound/pci/hda/hda_intel.c | 2 ++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 5 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index a0c75e467df3..ae8e66242927 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -3106,6 +3106,7 @@ #define PCI_DEVICE_ID_INTEL_HDA_CML_S 0xa3f0 #define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828 #define PCI_DEVICE_ID_INTEL_S21152BB 0xb152 +#define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7 #define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8 #define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index a73d7f34f4e5..07638111b28f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -731,6 +731,7 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_0) }, \ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_1) }, \ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_2) }, \ + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_BMG) }, \ { } \ }, pci) || HDA_CONTROLLER_IS_HSW(pci)) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9ea5925401f9..1d8171b11720 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2499,6 +2499,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_RPL_M, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, { PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, { PCI_DEVICE_DATA(INTEL, HDA_MTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, + /* Battlemage */ + { PCI_DEVICE_DATA(INTEL, HDA_BMG, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Lunarlake-P */ { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Arrow Lake-S */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 495d63101186..0a00c5ea42e0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4642,6 +4642,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281d, "Meteor Lake HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281e, "Battlemage HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptor Lake P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862820, "Lunar Lake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.25.1

2 1

[PATCH 1/2] ASoC: amd: acp: fix for acp platform device creation failure
by Vijendar Mukunda 05 May '24

05 May '24

ACP pin configuration varies based on acp version. ACP PCI driver should read the ACP PIN config value and based on config value, it has to create a platform device in below two conditions. 1) If ACP PDM configuration is selected from BIOS and ACP PDM controller exists. 2) If ACP I2S configuration is selected from BIOS. Other than above scenarios, ACP PCI driver should skip the platform device creation logic, i.e. ACP PCI driver probe sequence should never fail if other acp pin configuration is selected. It should skip platform device creation logic. check_acp_pdm() function was implemented for ACP6.x platforms to check ACP PDM configuration. Previously, this code was safe guarded by FLAG_AMD_LEGACY_ONLY_DMIC flag check. This implementation breaks audio use cases for Huawei Matebooks which are based on ACP3.x varaint uses I2S configuration. In current scenario, check_acp_pdm() function returns -ENODEV value which results in ACP PCI driver probe failure without creating a platform device even in case of valid ACP pin configuration. Implement check_acp_config() as a common function which invokes platform specific acp pin configuration check functions for ACP3.x, ACP6.0 & ACP6.3 & ACP7.0 variants and checks for ACP PDM controller. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218780 Fixes: 4af565de9f8c ("ASoC: amd: acp: fix for acp pdm configuration check") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 96 ++++++++++++++++++++++----- sound/soc/amd/acp/acp-pci.c | 9 ++- sound/soc/amd/acp/amd.h | 10 ++- sound/soc/amd/acp/chip_offset_byte.h | 1 + 4 files changed, 95 insertions(+), 21 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index b5aff3f230be..3be7c6d55a6f 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -358,11 +358,25 @@ int smn_read(struct pci_dev *dev, u32 smn_addr) } EXPORT_SYMBOL_NS_GPL(smn_read, SND_SOC_ACP_COMMON); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) +static void check_acp3x_config(struct acp_chip_info *chip) { - struct acpi_device *pdm_dev; - const union acpi_object *obj; - u32 pdm_addr, val; + u32 val; + + val = readl(chip->base + ACP3X_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + chip->is_pdm_config = true; + break; + } +} + +static void check_acp6x_config(struct acp_chip_info *chip) +{ + u32 val; val = readl(chip->base + ACP_PIN_CONFIG); switch (val) { @@ -371,42 +385,94 @@ int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) case ACP_CONFIG_6: case ACP_CONFIG_7: case ACP_CONFIG_8: - case ACP_CONFIG_10: case ACP_CONFIG_11: + case ACP_CONFIG_14: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: case ACP_CONFIG_12: case ACP_CONFIG_13: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + break; + } +} + +static void check_acp70_config(struct acp_chip_info *chip) +{ + u32 val; + + val = readl(chip->base + ACP_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + case ACP_CONFIG_5: + case ACP_CONFIG_6: + case ACP_CONFIG_7: + case ACP_CONFIG_8: + case ACP_CONFIG_11: case ACP_CONFIG_14: + case ACP_CONFIG_17: + case ACP_CONFIG_18: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: + case ACP_CONFIG_12: + case ACP_CONFIG_13: + case ACP_CONFIG_19: + case ACP_CONFIG_20: + chip->is_i2s_config = true; + chip->is_pdm_config = true; break; default: - return -EINVAL; + break; } +} + +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) +{ + struct acpi_device *pdm_dev; + const union acpi_object *obj; + u32 pdm_addr; switch (chip->acp_rev) { case ACP3X_DEV: pdm_addr = ACP_RENOIR_PDM_ADDR; + check_acp3x_config(chip); break; case ACP6X_DEV: pdm_addr = ACP_REMBRANDT_PDM_ADDR; + check_acp6x_config(chip); break; case ACP63_DEV: pdm_addr = ACP63_PDM_ADDR; + check_acp6x_config(chip); break; case ACP70_DEV: pdm_addr = ACP70_PDM_ADDR; + check_acp70_config(chip); break; default: - return -EINVAL; + break; } - pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); - if (pdm_dev) { - if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == pdm_addr) - return 0; + if (chip->is_pdm_config) { + pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); + if (pdm_dev) { + if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == pdm_addr) + chip->is_pdm_dev = true; + } } - return -ENODEV; } -EXPORT_SYMBOL_NS_GPL(check_acp_pdm, SND_SOC_ACP_COMMON); +EXPORT_SYMBOL_NS_GPL(check_acp_config, SND_SOC_ACP_COMMON); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 5f35b90eab8d..ad320b29e87d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -100,7 +100,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id ret = -EINVAL; goto release_regions; } - dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0); if (IS_ERR(dmic_dev)) { dev_err(dev, "failed to create DMIC device\n"); @@ -119,6 +118,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id if (ret) goto unregister_dmic_dev; + check_acp_config(pci, chip); + if (!chip->is_pdm_dev && !chip->is_i2s_config) + goto skip_pdev_creation; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; @@ -136,10 +139,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 5017e868f39b..d75b4eb34de8 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -138,6 +138,9 @@ struct acp_chip_info { void __iomem *base; /* ACP memory PCI base */ struct platform_device *chip_pdev; unsigned int flag; /* Distinguish b/w Legacy or Only PDM */ + bool is_pdm_dev; /* flag set to true when ACP PDM controller exists */ + bool is_pdm_config; /* flag set to true when PDM configuration is selected from BIOS */ + bool is_i2s_config; /* flag set to true when I2S configuration is selected from BIOS */ }; struct acp_stream { @@ -212,6 +215,11 @@ enum acp_config { ACP_CONFIG_13, ACP_CONFIG_14, ACP_CONFIG_15, + ACP_CONFIG_16, + ACP_CONFIG_17, + ACP_CONFIG_18, + ACP_CONFIG_19, + ACP_CONFIG_20, }; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; @@ -240,7 +248,7 @@ void restore_acp_pdm_params(struct snd_pcm_substream *substream, int restore_acp_i2s_params(struct snd_pcm_substream *substream, struct acp_dev_data *adata, struct acp_stream *stream); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip); +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip); static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction) { diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index cfd6c4d07594..18da734c0e9e 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -20,6 +20,7 @@ #define ACP_SOFT_RESET 0x1000 #define ACP_CONTROL 0x1004 #define ACP_PIN_CONFIG 0x1440 +#define ACP3X_PIN_CONFIG 0x1400 #define ACP_EXTERNAL_INTR_REG_ADDR(adata, offset, ctrl) \ (adata->acp_base + adata->rsrc->irq_reg_offset + offset + (ctrl * 0x04)) -- 2.34.1

3 3

[PATCH] ASoC: codecs: Drop explicit initialization of struct i2c_device_id::driver_data to 0
by Uwe Kleine-König 05 May '24

05 May '24

These drivers don't use the driver_data member of struct i2c_device_id, so don't explicitly initialize this member. This prepares putting driver_data in an anonymous union which requires either no initialization or named designators. But it's also a nice cleanup on its own. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/adau1372-i2c.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 8 ++++---- sound/soc/codecs/adau7118-i2c.c | 2 +- sound/soc/codecs/adav803.c | 2 +- sound/soc/codecs/ak4118.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-i2c.c | 8 ++++---- sound/soc/codecs/cs35l45-i2c.c | 2 +- sound/soc/codecs/cs35l56-i2c.c | 2 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271-i2c.c | 2 +- sound/soc/codecs/cs42l42-i2c.c | 2 +- sound/soc/codecs/cs42l51-i2c.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 8 ++++---- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 4 ++-- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8326.c | 2 +- sound/soc/codecs/es8328-i2c.c | 4 ++-- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98371.c | 2 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/mt6660.c | 2 +- sound/soc/codecs/nau8325.c | 2 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8810.c | 6 +++--- sound/soc/codecs/nau8821.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm1789-i2c.c | 2 +- sound/soc/codecs/pcm179x-i2c.c | 2 +- sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 4 ++-- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 4 ++-- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 4 ++-- sound/soc/codecs/rt5640.c | 6 +++--- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 4 ++-- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 6 +++--- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sma1303.c | 2 +- sound/soc/codecs/src4xxx-i2c.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm4567.c | 2 +- sound/soc/codecs/sta32x.c | 6 +++--- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tas2780.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tda7419.c | 2 +- sound/soc/codecs/tfa9879.c | 2 +- sound/soc/codecs/tlv320aic23-i2c.c | 2 +- sound/soc/codecs/ts3a227e.c | 2 +- sound/soc/codecs/tscs42xx.c | 4 ++-- sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731-i2c.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 4 ++-- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8804-i2c.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 4 ++-- 141 files changed, 168 insertions(+), 168 deletions(-) diff --git a/sound/soc/codecs/adau1372-i2c.c b/sound/soc/codecs/adau1372-i2c.c index 132b9e2cca59..2869325f9ace 100644 --- a/sound/soc/codecs/adau1372-i2c.c +++ b/sound/soc/codecs/adau1372-i2c.c @@ -21,7 +21,7 @@ static int adau1372_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1372_i2c_ids[] = { - { "adau1372", 0 }, + { "adau1372" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1372_i2c_ids); diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 3582c4b968a0..a910e252aa12 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1496,7 +1496,7 @@ static int adau1373_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1373_i2c_id[] = { - { "adau1373", 0 }, + { "adau1373" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d1392d9abccd..8bd6067df7f7 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -862,10 +862,10 @@ static int adau1701_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1701_i2c_id[] = { - { "adau1401", 0 }, - { "adau1401a", 0 }, - { "adau1701", 0 }, - { "adau1702", 0 }, + { "adau1401" }, + { "adau1401a" }, + { "adau1701" }, + { "adau1702" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c index b302b28eca7c..f9dc8f4ef9a4 100644 --- a/sound/soc/codecs/adau7118-i2c.c +++ b/sound/soc/codecs/adau7118-i2c.c @@ -68,7 +68,7 @@ static const struct of_device_id adau7118_of_match[] = { MODULE_DEVICE_TABLE(of, adau7118_of_match); static const struct i2c_device_id adau7118_id[] = { - {"adau7118", 0}, + {"adau7118"}, {} }; MODULE_DEVICE_TABLE(i2c, adau7118_id); diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c index 78a317947df9..8b96c41f0354 100644 --- a/sound/soc/codecs/adav803.c +++ b/sound/soc/codecs/adav803.c @@ -14,7 +14,7 @@ #include "adav80x.h" static const struct i2c_device_id adav803_id[] = { - { "adav803", 0 }, + { "adav803" }, { } }; MODULE_DEVICE_TABLE(i2c, adav803_id); diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 74a10108c1d4..9a43235e6a11 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -396,7 +396,7 @@ MODULE_DEVICE_TABLE(of, ak4118_of_match); #endif static const struct i2c_device_id ak4118_id_table[] = { - { "ak4118", 0 }, + { "ak4118" }, {} }; MODULE_DEVICE_TABLE(i2c, ak4118_id_table); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 904bf91090aa..aadc46a47280 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -430,7 +430,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ak4535_i2c_id[] = { - { "ak4535", 0 }, + { "ak4535" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 5b7df2f0dd6a..ec33e7d73c6c 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -619,7 +619,7 @@ static void ak4641_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id ak4641_i2c_id[] = { - { "ak4641", 0 }, + { "ak4641" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 5b849b390c2a..d545aa2e0a39 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -646,7 +646,7 @@ static int ak4671_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id ak4671_i2c_id[] = { - { "ak4671", 0 }, + { "ak4671" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index d1350ffbf3bd..96555263e10b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -558,7 +558,7 @@ MODULE_DEVICE_TABLE(of, cs35l32_of_match); static const struct i2c_device_id cs35l32_id[] = { - {"cs35l32", 0}, + {"cs35l32"}, {} }; diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index a19a2bafb37c..b03aab147530 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1264,7 +1264,7 @@ static const struct of_device_id cs35l33_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l33_of_match); static const struct i2c_device_id cs35l33_id[] = { - {"cs35l33", 0}, + {"cs35l33"}, {} }; diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index cca59de66b73..4c517231d765 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1198,7 +1198,7 @@ static const struct of_device_id cs35l34_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l34_of_match); static const struct i2c_device_id cs35l34_id[] = { - {"cs35l34", 0}, + {"cs35l34"}, {} }; MODULE_DEVICE_TABLE(i2c, cs35l34_id); diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index ddb7d63213a3..c39b3cfe9574 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1639,7 +1639,7 @@ static const struct of_device_id cs35l35_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l35_of_match); static const struct i2c_device_id cs35l35_id[] = { - {"cs35l35", 0}, + {"cs35l35"}, {} }; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index f5bd32e434a0..bc79990615e8 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1930,7 +1930,7 @@ static const struct of_device_id cs35l36_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l36_of_match); static const struct i2c_device_id cs35l36_id[] = { - {"cs35l36", 0}, + {"cs35l36"}, {} }; diff --git a/sound/soc/codecs/cs35l41-i2c.c b/sound/soc/codecs/cs35l41-i2c.c index a0c457c0d04b..34097996b784 100644 --- a/sound/soc/codecs/cs35l41-i2c.c +++ b/sound/soc/codecs/cs35l41-i2c.c @@ -20,10 +20,10 @@ #include "cs35l41.h" static const struct i2c_device_id cs35l41_id_i2c[] = { - { "cs35l40", 0 }, - { "cs35l41", 0 }, - { "cs35l51", 0 }, - { "cs35l53", 0 }, + { "cs35l40" }, + { "cs35l41" }, + { "cs35l51" }, + { "cs35l53" }, {} }; diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index bc2af1ed0fe9..f5fc42dcc8c7 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -53,7 +53,7 @@ static const struct of_device_id cs35l45_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l45_of_match); static const struct i2c_device_id cs35l45_id_i2c[] = { - { "cs35l45", 0 }, + { "cs35l45" }, {} }; MODULE_DEVICE_TABLE(i2c, cs35l45_id_i2c); diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 7063c400e896..2bd2ff75cd50 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -57,7 +57,7 @@ static void cs35l56_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id cs35l56_id_i2c[] = { - { "cs35l56", 0 }, + { "cs35l56" }, {} }; MODULE_DEVICE_TABLE(i2c, cs35l56_id_i2c); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 1ed1e60d8e53..78ffb7fa7fc5 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -638,7 +638,7 @@ static const struct of_device_id cs4265_of_match[] = { MODULE_DEVICE_TABLE(of, cs4265_of_match); static const struct i2c_device_id cs4265_id[] = { - { "cs4265", 0 }, + { "cs4265" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4265_id); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3bbb90c827f2..67e92bfecb56 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -734,7 +734,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client) * cs4270_id - I2C device IDs supported by this driver */ static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, + {"cs4270"}, {} }; MODULE_DEVICE_TABLE(i2c, cs4270_id); diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c index 89fe7d1665df..1d210b969173 100644 --- a/sound/soc/codecs/cs4271-i2c.c +++ b/sound/soc/codecs/cs4271-i2c.c @@ -23,7 +23,7 @@ static int cs4271_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id cs4271_i2c_id[] = { - { "cs4271", 0 }, + { "cs4271" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); diff --git a/sound/soc/codecs/cs42l42-i2c.c b/sound/soc/codecs/cs42l42-i2c.c index 2552a1e6b82f..8d10f9328e02 100644 --- a/sound/soc/codecs/cs42l42-i2c.c +++ b/sound/soc/codecs/cs42l42-i2c.c @@ -78,7 +78,7 @@ static const struct acpi_device_id __maybe_unused cs42l42_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, cs42l42_acpi_match); static const struct i2c_device_id cs42l42_id[] = { - {"cs42l42", 0}, + {"cs42l42"}, {} }; diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 5ed2ef83dcdb..e7cc50096297 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -14,7 +14,7 @@ #include "cs42l51.h" static struct i2c_device_id cs42l51_i2c_id[] = { - {"cs42l51", 0}, + {"cs42l51"}, {} }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4fc8a6ae8d92..7128d4c62f50 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1215,7 +1215,7 @@ MODULE_DEVICE_TABLE(of, cs42l52_of_match); static const struct i2c_device_id cs42l52_id[] = { - { "cs42l52", 0 }, + { "cs42l52" }, { } }; MODULE_DEVICE_TABLE(i2c, cs42l52_id); diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3e3a86dab4fc..aaa10c459b52 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1330,7 +1330,7 @@ MODULE_DEVICE_TABLE(of, cs42l56_of_match); static const struct i2c_device_id cs42l56_id[] = { - { "cs42l56", 0 }, + { "cs42l56" }, { } }; MODULE_DEVICE_TABLE(i2c, cs42l56_id); diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6ab67d196d10..21ba796a5cd9 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1372,7 +1372,7 @@ static const struct of_device_id cs42l73_of_match[] = { MODULE_DEVICE_TABLE(of, cs42l73_of_match); static const struct i2c_device_id cs42l73_id[] = { - {"cs42l73", 0}, + {"cs42l73"}, {} }; diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index b6d829bbe3cc..be4037890fdb 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2763,10 +2763,10 @@ MODULE_DEVICE_TABLE(acpi, cs43130_acpi_match); static const struct i2c_device_id cs43130_i2c_id[] = { - {"cs43130", 0}, - {"cs4399", 0}, - {"cs43131", 0}, - {"cs43198", 0}, + {"cs43130"}, + {"cs4399"}, + {"cs43131"}, + {"cs43198"}, {} }; diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 2ceca5d0e5bf..d87aae31c516 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -248,7 +248,7 @@ static int cs4341_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id cs4341_i2c_id[] = { - { "cs4341", 0 }, + { "cs4341" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id); diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index ca8f21aa4837..a134ca722892 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -361,7 +361,7 @@ static const struct of_device_id cs4349_of_match[] = { MODULE_DEVICE_TABLE(of, cs4349_of_match); static const struct i2c_device_id cs4349_i2c_id[] = { - {"cs4349", 0}, + {"cs4349"}, {} }; diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f4065555c36e..c0893146423b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -1104,7 +1104,7 @@ static const struct of_device_id cs53l30_of_match[] = { MODULE_DEVICE_TABLE(of, cs53l30_of_match); static const struct i2c_device_id cs53l30_id[] = { - { "cs53l30", 0 }, + { "cs53l30" }, {} }; diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index f8b128084015..e8e22b1a1963 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1686,8 +1686,8 @@ static void cx2072x_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id cx2072x_i2c_id[] = { - { "cx20721", 0 }, - { "cx20723", 0 }, + { "cx20721" }, + { "cx20723" }, {} }; MODULE_DEVICE_TABLE(i2c, cx2072x_i2c_id); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1e232d01809c..da2d0242019e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1238,7 +1238,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id da7210_i2c_id[] = { - { "da7210", 0 }, + { "da7210" }, { } }; MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 369c62078780..a2b328f3b39f 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2238,7 +2238,7 @@ static const struct dev_pm_ops da7213_pm = { }; static const struct i2c_device_id da7213_i2c_id[] = { - { "da7213", 0 }, + { "da7213" }, { } }; MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f8ca1afa8af5..b747f6fa12e4 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1546,7 +1546,7 @@ static int da732x_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id da732x_i2c_id[] = { - { "da7320", 0}, + { "da7320"}, { } }; MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index c8a34572965d..8bb8fef2a1d1 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1511,7 +1511,7 @@ static int da9055_i2c_probe(struct i2c_client *i2c) * and PMIC, which must be different to operate together. */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055-codec", 0 }, + { "da9055-codec" }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index e53b2856d625..61729e5b50a8 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -887,7 +887,7 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client) } static const struct i2c_device_id es8316_i2c_id[] = { - {"es8316", 0 }, + {"es8316" }, {} }; MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 93385f181d2c..833ea52638ab 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1263,7 +1263,7 @@ static void es8326_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id es8326_i2c_id[] = { - {"es8326", 0 }, + {"es8326" }, {} }; MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c index 3c4aaa0032a0..56bfbe9261ce 100644 --- a/sound/soc/codecs/es8328-i2c.c +++ b/sound/soc/codecs/es8328-i2c.c @@ -16,8 +16,8 @@ #include "es8328.h" static const struct i2c_device_id es8328_id[] = { - { "es8328", 0 }, - { "es8388", 0 }, + { "es8328" }, + { "es8388" }, { } }; MODULE_DEVICE_TABLE(i2c, es8328_id); diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index f9456133a89a..b7a94631d77d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1133,7 +1133,7 @@ static int isabelle_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id isabelle_i2c_id[] = { - { "isabelle", 0 }, + { "isabelle" }, { } }; MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index e7542f71323d..26cdb750cbca 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -128,7 +128,7 @@ static int lm4857_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "lm4857", 0 }, + { "lm4857" }, { } }; MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index a4094689b3dd..ab89af7965bf 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1442,7 +1442,7 @@ static int lm49453_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id lm49453_i2c_id[] = { - { "lm49453", 0 }, + { "lm49453" }, { } }; MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 8d0ca1be99c0..e4793a5d179e 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -206,7 +206,7 @@ static int max9768_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id max9768_i2c_id[] = { - { "max9768", 0 }, + { "max9768" }, { } }; MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index f0e49179c38f..852db211ba1e 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -400,7 +400,7 @@ static int max98371_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98371_i2c_id[] = { - { "max98371", 0 }, + { "max98371" }, { } }; diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index e7ec7875c4a9..1f7ff3dbcbbe 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -578,7 +578,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98373_i2c_id[] = { - { "max98373", 0}, + { "max98373"}, { }, }; diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 078adec29312..b847d7c59ec0 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -976,7 +976,7 @@ static int max98388_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98388_i2c_id[] = { - { "max98388", 0}, + { "max98388"}, { }, }; diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 5b8e78e51630..57fa2db1e148 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1104,7 +1104,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98390_i2c_id[] = { - { "max98390", 0}, + { "max98390"}, {}, }; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 8b012a85360a..c395132689b4 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -320,7 +320,7 @@ static int max9850_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max9850_i2c_id[] = { - { "max9850", 0 }, + { "max9850" }, { } }; MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c index edd05253d37c..479ded22672e 100644 --- a/sound/soc/codecs/max98520.c +++ b/sound/soc/codecs/max98520.c @@ -734,7 +734,7 @@ static int max98520_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98520_i2c_id[] = { - { "max98520", 0}, + { "max98520"}, { }, }; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 3b9dd158c34b..50db88fce904 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -684,7 +684,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max9867_i2c_id[] = { - { "max9867", 0 }, + { "max9867" }, { } }; MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 2ae64fcf29c7..1bd0d4761ca6 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -151,7 +151,7 @@ static int max9877_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id max9877_i2c_id[] = { - { "max9877", 0 }, + { "max9877" }, { } }; MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index a9c1d85cd0d5..66c78051bd09 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -617,7 +617,7 @@ static int max98925_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98925_i2c_id[] = { - { "max98925", 0 }, + { "max98925" }, { } }; MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 922ce0dc4e60..ae962bda163e 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -565,7 +565,7 @@ static int max98926_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98926_i2c_id[] = { - { "max98926", 0 }, + { "max98926" }, { } }; MODULE_DEVICE_TABLE(i2c, max98926_i2c_id); diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 70db9d3ff5a5..747aa6f1d54f 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -875,7 +875,7 @@ static void max98927_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id max98927_i2c_id[] = { - { "max98927", 0}, + { "max98927"}, { }, }; diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index a45ef9d65703..c6585e8143a5 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -572,7 +572,7 @@ static int ml26124_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ml26124_i2c_id[] = { - { "ml26124", 0 }, + { "ml26124" }, { } }; MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 5c50c7de26cd..39a57f643d81 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -559,7 +559,7 @@ static const struct of_device_id __maybe_unused mt6660_of_id[] = { MODULE_DEVICE_TABLE(of, mt6660_of_id); static const struct i2c_device_id mt6660_i2c_id[] = { - {"mt6660", 0 }, + {"mt6660" }, {}, }; MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c index d65f73144597..2266f320a8f2 100644 --- a/sound/soc/codecs/nau8325.c +++ b/sound/soc/codecs/nau8325.c @@ -871,7 +871,7 @@ static int nau8325_i2c_probe(struct i2c_client *i2c, } static const struct i2c_device_id nau8325_i2c_ids[] = { - { "nau8325", 0 }, + { "nau8325" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8325_i2c_ids); diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 22251fb2fa1f..7e59448e7ac6 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -965,7 +965,7 @@ static int nau8540_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8540_i2c_ids[] = { - { "nau8540", 0 }, + { "nau8540" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8540_i2c_ids); diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 97a54059474c..dc3aaca89919 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -895,9 +895,9 @@ static int nau8810_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8810_i2c_id[] = { - { "nau8810", 0 }, - { "nau8812", 0 }, - { "nau8814", 0 }, + { "nau8810" }, + { "nau8812" }, + { "nau8814" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 6818bbd1d3c7..de5c4db05c8f 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1922,7 +1922,7 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8821_i2c_ids[] = { - { "nau8821", 0 }, + { "nau8821" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids); diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 7199d734c79f..e6909e64dfa3 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -1151,7 +1151,7 @@ static int nau8822_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8822_i2c_id[] = { - { "nau8822", 0 }, + { "nau8822" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 704af1cf8cbf..f92b95b21cae 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -2003,7 +2003,7 @@ static int nau8824_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8824_i2c_ids[] = { - { "nau8824", 0 }, + { "nau8824" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8824_i2c_ids); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index cd30ad649bae..bde25bc6909d 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2934,7 +2934,7 @@ static void nau8825_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id nau8825_i2c_ids[] = { - { "nau8825", 0 }, + { "nau8825" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 316ad53bc66a..fc152496d5dc 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -291,7 +291,7 @@ static const struct snd_soc_component_driver soc_component_dev_pcm1681 = { }; static const struct i2c_device_id pcm1681_i2c_id[] = { - {"pcm1681", 0}, + {"pcm1681"}, {} }; MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); diff --git a/sound/soc/codecs/pcm1789-i2c.c b/sound/soc/codecs/pcm1789-i2c.c index f2d0b4d21e41..abadf4f8ed5e 100644 --- a/sound/soc/codecs/pcm1789-i2c.c +++ b/sound/soc/codecs/pcm1789-i2c.c @@ -41,7 +41,7 @@ MODULE_DEVICE_TABLE(of, pcm1789_of_match); #endif static const struct i2c_device_id pcm1789_i2c_ids[] = { - { "pcm1789", 0 }, + { "pcm1789" }, { } }; MODULE_DEVICE_TABLE(i2c, pcm1789_i2c_ids); diff --git a/sound/soc/codecs/pcm179x-i2c.c b/sound/soc/codecs/pcm179x-i2c.c index 10579681f44b..effc1dd6df22 100644 --- a/sound/soc/codecs/pcm179x-i2c.c +++ b/sound/soc/codecs/pcm179x-i2c.c @@ -38,7 +38,7 @@ MODULE_DEVICE_TABLE(of, pcm179x_of_match); #endif static const struct i2c_device_id pcm179x_i2c_ids[] = { - { "pcm179x", 0 }, + { "pcm179x" }, { } }; MODULE_DEVICE_TABLE(i2c, pcm179x_i2c_ids); diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index d5285baad53a..a0e75b03e9dc 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2206,7 +2206,7 @@ MODULE_DEVICE_TABLE(acpi, rt1011_acpi_match); #endif static const struct i2c_device_id rt1011_i2c_id[] = { - { "rt1011", 0 }, + { "rt1011" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1011_i2c_id); diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 1250cfaf2adc..0f806dde9c39 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1097,7 +1097,7 @@ static const struct regmap_config rt1015_regmap = { }; static const struct i2c_device_id rt1015_i2c_id[] = { - { "rt1015", 0 }, + { "rt1015" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1015_i2c_id); diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c index 919a1f25e584..fed4da23cba2 100644 --- a/sound/soc/codecs/rt1016.c +++ b/sound/soc/codecs/rt1016.c @@ -608,7 +608,7 @@ static const struct regmap_config rt1016_regmap = { }; static const struct i2c_device_id rt1016_i2c_id[] = { - { "rt1016", 0 }, + { "rt1016" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1016_i2c_id); diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index ceb8baa6a20d..d989d06a2614 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -540,7 +540,7 @@ static const struct regmap_config rt1019_regmap = { }; static const struct i2c_device_id rt1019_i2c_id[] = { - { "rt1019", 0 }, + { "rt1019" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1019_i2c_id); diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 80888cbcf49c..c2b55be8d165 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -981,8 +981,8 @@ MODULE_DEVICE_TABLE(acpi, rt1305_acpi_match); #endif static const struct i2c_device_id rt1305_i2c_id[] = { - { "rt1305", 0 }, - { "rt1306", 0 }, + { "rt1305" }, + { "rt1306" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1305_i2c_id); diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 86afb429d423..b366338cea71 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -795,7 +795,7 @@ MODULE_DEVICE_TABLE(acpi, rt1308_acpi_match); #endif static const struct i2c_device_id rt1308_i2c_id[] = { - { "rt1308", 0 }, + { "rt1308" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1308_i2c_id); diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 6e7843484250..bd61a257d7b5 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1097,7 +1097,7 @@ MODULE_DEVICE_TABLE(of, rt274_of_match); #endif static const struct i2c_device_id rt274_i2c_id[] = { - {"rt274", 0}, + {"rt274"}, {} }; MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f8994f4968c5..d0f533120c33 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1075,8 +1075,8 @@ static const struct regmap_config rt286_regmap = { }; static const struct i2c_device_id rt286_i2c_id[] = { - {"rt286", 0}, - {"rt288", 0}, + {"rt286"}, + {"rt288"}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 03d9839a5de3..13aef6c5e91c 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1137,7 +1137,7 @@ static const struct regmap_config rt298_regmap = { }; static const struct i2c_device_id rt298_i2c_id[] = { - {"rt298", 0}, + {"rt298"}, {} }; MODULE_DEVICE_TABLE(i2c, rt298_i2c_id); diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a8cdc3d6994d..2b3c0f9e178c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1199,7 +1199,7 @@ static const struct regmap_config rt5514_regmap = { }; static const struct i2c_device_id rt5514_i2c_id[] = { - { "rt5514", 0 }, + { "rt5514" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5514_i2c_id); diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index e7aa60e73961..34461c462009 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1320,7 +1320,7 @@ static const struct regmap_config rt5616_regmap = { }; static const struct i2c_device_id rt5616_i2c_id[] = { - { "rt5616", 0 }, + { "rt5616" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index a64e66c2d3c4..12df0c4f2097 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1669,8 +1669,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5631 = { }; static const struct i2c_device_id rt5631_i2c_id[] = { - { "rt5631", 0 }, - { "alc5631", 0 }, + { "rt5631" }, + { "alc5631" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 174872ef35d2..16f3425a3e35 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2949,9 +2949,9 @@ static const struct regmap_config rt5640_regmap = { }; static const struct i2c_device_id rt5640_i2c_id[] = { - { "rt5640", 0 }, - { "rt5639", 0 }, - { "rt5642", 0 }, + { "rt5640" }, + { "rt5639" }, + { "rt5642" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d0d24a53df74..05f574bf8b8f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3630,8 +3630,8 @@ static const struct regmap_config temp_regmap = { }; static const struct i2c_device_id rt5645_i2c_id[] = { - { "rt5645", 0 }, - { "rt5650", 0 }, + { "rt5645" }, + { "rt5650" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 33a34bd0b405..00421a1f54bf 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2199,7 +2199,7 @@ MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match); #endif static const struct i2c_device_id rt5651_i2c_id[] = { - { "rt5651", 0 }, + { "rt5651" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index fb094c0fe740..a2652fa6e1d7 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3815,8 +3815,8 @@ static const struct regmap_config rt5659_regmap = { }; static const struct i2c_device_id rt5659_i2c_id[] = { - { "rt5658", 0 }, - { "rt5659", 0 }, + { "rt5658" }, + { "rt5659" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5659_i2c_id); diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index d5c2f0f2df98..3ac41d2c279b 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1224,7 +1224,7 @@ static const struct regmap_config rt5660_regmap = { }; static const struct i2c_device_id rt5660_i2c_id[] = { - { "rt5660", 0 }, + { "rt5660" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 161dcb3915f9..9d32debd3689 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3307,7 +3307,7 @@ static const struct regmap_config temp_regmap = { }; static const struct i2c_device_id rt5663_i2c_id[] = { - { "rt5663", 0 }, + { "rt5663" }, {} }; MODULE_DEVICE_TABLE(i2c, rt5663_i2c_id); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6f778c8f0832..47df14ba5278 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4635,7 +4635,7 @@ static const struct regmap_config rt5665_regmap = { }; static const struct i2c_device_id rt5665_i2c_id[] = { - {"rt5665", 0}, + {"rt5665"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5665_i2c_id); diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6d8e228ccb57..494ca3ce9b96 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2378,7 +2378,7 @@ static const struct regmap_config rt5668_regmap = { }; static const struct i2c_device_id rt5668_i2c_id[] = { - {"rt5668b", 0}, + {"rt5668b"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0e34293f3395..30bf96c35b58 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2871,9 +2871,9 @@ static const struct regmap_config rt5670_regmap = { }; static const struct i2c_device_id rt5670_i2c_id[] = { - { "rt5670", 0 }, - { "rt5671", 0 }, - { "rt5672", 0 }, + { "rt5670" }, + { "rt5671" }, + { "rt5672" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 62f26ce9d476..ff9e14fad0cd 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -318,7 +318,7 @@ static const struct acpi_device_id rt5682_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); static const struct i2c_device_id rt5682_i2c_id[] = { - {"rt5682", 0}, + {"rt5682"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 12741668fdb3..f50f196d700d 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -3319,7 +3319,7 @@ static const struct acpi_device_id rt5682s_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match); static const struct i2c_device_id rt5682s_i2c_id[] = { - {"rt5682s", 0}, + {"rt5682s"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2f468f41b94d..7aa89e34657e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1809,7 +1809,7 @@ static void sgtl5000_i2c_shutdown(struct i2c_client *client) } static const struct i2c_device_id sgtl5000_id[] = { - {"sgtl5000", 0}, + {"sgtl5000"}, {}, }; diff --git a/sound/soc/codecs/sma1303.c b/sound/soc/codecs/sma1303.c index 61072e7574a0..980c48cbc348 100644 --- a/sound/soc/codecs/sma1303.c +++ b/sound/soc/codecs/sma1303.c @@ -1791,7 +1791,7 @@ static void sma1303_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id sma1303_i2c_id[] = { - {"sma1303", 0}, + {"sma1303"}, {} }; MODULE_DEVICE_TABLE(i2c, sma1303_i2c_id); diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c index 93af8e209b05..55f00ce7c718 100644 --- a/sound/soc/codecs/src4xxx-i2c.c +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -19,7 +19,7 @@ static int src4xxx_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id src4xxx_i2c_ids[] = { - { "src4392", 0 }, + { "src4392" }, { } }; MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index d20d897407eb..06016e88dd27 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -793,7 +793,7 @@ MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); #endif static const struct i2c_device_id ssm2518_i2c_ids[] = { - { "ssm2518", 0 }, + { "ssm2518" }, { } }; MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 0a6f04d8f636..3e09c85abedb 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -471,7 +471,7 @@ static int ssm4567_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ssm4567_i2c_ids[] = { - { "ssm4567", 0 }, + { "ssm4567" }, { } }; MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fcf0dbfbbbca..bd8848ea1ec2 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1154,9 +1154,9 @@ static int sta32x_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id sta32x_i2c_id[] = { - { "sta326", 0 }, - { "sta328", 0 }, - { "sta329", 0 }, + { "sta326" }, + { "sta328" }, + { "sta329" }, { } }; MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index 612cc1d7eafe..d1450de92652 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1238,7 +1238,7 @@ static void sta350_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id sta350_i2c_id[] = { - { "sta350", 0 }, + { "sta350" }, { } }; MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index eedafef775e5..f7718491c899 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -363,7 +363,7 @@ static int sta529_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id sta529_i2c_id[] = { - { "sta529", 0 }, + { "sta529" }, { } }; MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 8c9dc318b0e8..40f5f27e74c0 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -742,7 +742,7 @@ static void tas2552_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas2552_id[] = { - { "tas2552", 0 }, + { "tas2552" }, { } }; MODULE_DEVICE_TABLE(i2c, tas2552_id); diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index a9838e0738cc..1dc719d726ab 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -738,7 +738,7 @@ static int tas2764_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2764_i2c_id[] = { - { "tas2764", 0}, + { "tas2764"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2764_i2c_id); diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 99bf402eb566..67bc1c8b0131 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -702,7 +702,7 @@ static int tas2770_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2770_i2c_id[] = { - { "tas2770", 0}, + { "tas2770"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 972e8ea5ebde..a18ccf5fb7ad 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -626,7 +626,7 @@ static int tas2780_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2780_i2c_id[] = { - { "tas2780", 0}, + { "tas2780"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2780_i2c_id); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index f52c14b43f28..6d45df3b9ba4 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -891,7 +891,7 @@ static const struct snd_soc_component_driver soc_component_dev_tas5086 = { }; static const struct i2c_device_id tas5086_i2c_id[] = { - { "tas5086", 0 }, + { "tas5086" }, { } }; MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index da89e8c681dd..bb0500e9d3ea 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -792,7 +792,7 @@ static void tas6424_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas6424_i2c_ids[] = { - { "tas6424", 0 }, + { "tas6424" }, { } }; MODULE_DEVICE_TABLE(i2c, tas6424_i2c_ids); diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index e187d74a1737..386b99c8023b 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -614,7 +614,7 @@ static int tda7419_probe(struct i2c_client *i2c) } static const struct i2c_device_id tda7419_i2c_id[] = { - { "tda7419", 0 }, + { "tda7419" }, { } }; MODULE_DEVICE_TABLE(i2c, tda7419_i2c_id); diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 8cca2ceadd9e..ac0c5c337677 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -296,7 +296,7 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tfa9879_i2c_id[] = { - { "tfa9879", 0 }, + { "tfa9879" }, { } }; MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index 9692ae007c91..a31fb95048b8 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -28,7 +28,7 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tlv320aic23_id[] = { - {"tlv320aic23", 0}, + {"tlv320aic23"}, {} }; diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 6d9166d9116a..dbf448dd8864 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -427,7 +427,7 @@ static const struct dev_pm_ops ts3a227e_pm = { }; static const struct i2c_device_id ts3a227e_i2c_ids[] = { - { "ts3a227e", 0 }, + { "ts3a227e" }, { } }; MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 1eefc1fe6ea8..f8a3d1b40990 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1485,8 +1485,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tscs42xx_i2c_id[] = { - { "tscs42A1", 0 }, - { "tscs42A2", 0 }, + { "tscs42A1" }, + { "tscs42A2" }, { } }; MODULE_DEVICE_TABLE(i2c, tscs42xx_i2c_id); diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 744aef32a21f..850e5de9271e 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3457,7 +3457,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tscs454_i2c_id[] = { - { "tscs454", 0 }, + { "tscs454" }, { } }; MODULE_DEVICE_TABLE(i2c, tscs454_i2c_id); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5c5751dc14e5..4f8fdd574585 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -782,7 +782,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id uda1380_i2c_id[] = { - { "uda1380", 0 }, + { "uda1380" }, { } }; MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 9fa6df48799b..1f59309d8c69 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -204,7 +204,7 @@ static int wm1250_ev1_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm1250_ev1_i2c_id[] = { - { "wm1250-ev1", 0 }, + { "wm1250-ev1" }, { } }; MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id); diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 9571ea53cb3f..a07a443ba196 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -929,7 +929,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm2000_i2c_id[] = { - { "wm2000", 0 }, + { "wm2000" }, { } }; MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 69c9c2bd7e7b..841247173d98 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2474,7 +2474,7 @@ static const struct dev_pm_ops wm2200_pm = { }; static const struct i2c_device_id wm2200_i2c_id[] = { - { "wm2200", 0 }, + { "wm2200" }, { } }; MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7ee4b45c0834..11bbc94a282c 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2670,7 +2670,7 @@ static const struct dev_pm_ops wm5100_pm = { }; static const struct i2c_device_id wm5100_i2c_id[] = { - { "wm5100", 0 }, + { "wm5100" }, { } }; MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 0e671cce8447..4a31d6f89502 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -668,7 +668,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8510_i2c_id[] = { - { "wm8510", 0 }, + { "wm8510" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 41b14538b03c..138eba7e577a 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -517,7 +517,7 @@ static int wm8523_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8523_i2c_id[] = { - { "wm8523", 0 }, + { "wm8523" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 7d339cc65208..a1c99bbf5aa1 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -454,7 +454,7 @@ static int wm8711_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id wm8711_i2c_id[] = { - { "wm8711", 0 }, + { "wm8711" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index d9cc78fbf1ea..2cbd6b189416 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -295,7 +295,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8728_i2c_id[] = { - { "wm8728", 0 }, + { "wm8728" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); diff --git a/sound/soc/codecs/wm8731-i2c.c b/sound/soc/codecs/wm8731-i2c.c index 7f68ad0380e0..1254e583af51 100644 --- a/sound/soc/codecs/wm8731-i2c.c +++ b/sound/soc/codecs/wm8731-i2c.c @@ -47,7 +47,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8731_i2c_id[] = { - { "wm8731", 0 }, + { "wm8731" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index a0ba1e7dee98..efdc242c2ede 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -639,7 +639,7 @@ static int wm8737_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8737_i2c_id[] = { - { "wm8737", 0 }, + { "wm8737" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8737_i2c_id); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a0848774427b..4863d6ac461b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -606,7 +606,7 @@ static int wm8741_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8741_i2c_id[] = { - { "wm8741", 0 }, + { "wm8741" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b8d76cd001da..cae97fa3bcb0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -802,8 +802,8 @@ static int wm8750_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8750_i2c_id[] = { - { "wm8750", 0 }, - { "wm8987", 0 }, + { "wm8750" }, + { "wm8987" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index f42ed24314f3..38b76b7275e5 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1580,7 +1580,7 @@ static int wm8753_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8753_i2c_id[] = { - { "wm8753", 0 }, + { "wm8753" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index 7062a8b2f8b5..e80dad87219b 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -31,7 +31,7 @@ static void wm8804_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, + { "wm8804" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 84d06c190411..e44fdf97796f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1286,7 +1286,7 @@ static void wm8900_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id wm8900_i2c_id[] = { - { "wm8900", 0 }, + { "wm8900" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 84ae1102ac88..c643b5377d3a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2199,7 +2199,7 @@ static const struct of_device_id wm8903_of_match[] = { MODULE_DEVICE_TABLE(of, wm8903_of_match); static const struct i2c_device_id wm8903_i2c_id[] = { - { "wm8903", 0 }, + { "wm8903" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b9432f8b64e5..8a532f7d750c 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -844,7 +844,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8940_i2c_id[] = { - { "wm8940", 0 }, + { "wm8940" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 4f4338326438..bae52a8a2e11 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -994,7 +994,7 @@ static int wm8955_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8955_i2c_id[] = { - { "wm8955", 0 }, + { "wm8955" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7689fe3cc86d..00858b9c9568 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1549,7 +1549,7 @@ static void wm8960_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8960_i2c_id[] = { - { "wm8960", 0 }, + { "wm8960" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 8f8330efb341..d1c731e25777 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -966,7 +966,7 @@ static int wm8961_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8961_i2c_id[] = { - { "wm8961", 0 }, + { "wm8961" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7c6ed2983128..3ef95b7dcbdb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3938,7 +3938,7 @@ static const struct dev_pm_ops wm8962_pm = { }; static const struct i2c_device_id wm8962_i2c_id[] = { - { "wm8962", 0 }, + { "wm8962" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index e88f323d28b2..b97c7d5bd4e7 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -691,7 +691,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8971_i2c_id[] = { - { "wm8971", 0 }, + { "wm8971" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 260bac695b20..0ee3655cad01 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -707,7 +707,7 @@ static int wm8974_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8974_i2c_id[] = { - { "wm8974", 0 }, + { "wm8974" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 718bfef302cc..40d22b36b7a9 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1056,7 +1056,7 @@ static int wm8978_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8978_i2c_id[] = { - { "wm8978", 0 }, + { "wm8978" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index b26d6a68e8d2..252b4a6cac04 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1059,7 +1059,7 @@ static int wm8983_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8983_i2c_id[] = { - { "wm8983", 0 }, + { "wm8983" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id); diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 76f214f12ce0..f0e9d6e38dc0 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -896,7 +896,7 @@ static int wm8988_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8988_i2c_id[] = { - { "wm8988", 0 }, + { "wm8988" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5a8e765090af..573bd3d487ba 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1238,7 +1238,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8990_i2c_id[] = { - { "wm8990", 0 }, + { "wm8990" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 590318aafaea..3bd9b362051b 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1314,7 +1314,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8991_i2c_id[] = { - { "wm8991", 0 }, + { "wm8991" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5b788f35e5e4..98a83f51ab26 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1732,7 +1732,7 @@ static void wm8993_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8993_i2c_id[] = { - { "wm8993", 0 }, + { "wm8993" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 59ef2ef8ce00..1f9a9b636935 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2258,7 +2258,7 @@ static int wm8995_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8995_i2c_id[] = { - {"wm8995", 0}, + {"wm8995"}, {} }; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e738326e33ed..f6c5adce164e 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3069,7 +3069,7 @@ static void wm8996_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8996_i2c_id[] = { - { "wm8996", 0 }, + { "wm8996" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8996_i2c_id); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index e7ec799573d3..cb9d040b34d6 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1360,7 +1360,7 @@ static void wm9081_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id wm9081_i2c_id[] = { - { "wm9081", 0 }, + { "wm9081" }, { } }; MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 50c1cbccfdb9..26191bcc161d 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -606,8 +606,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm9090_id[] = { - { "wm9090", 0 }, - { "wm9093", 0 }, + { "wm9090" }, + { "wm9093" }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); -- 2.43.0

3 4

Fix seg. fault when accessing `ConstantObject` extra
by GitHub pull_request - opened 04 May '24

04 May '24

alsa-project/alsa-python pull request #10 was opened from FrancescoCeruti: This fixes (#9) a segmentation fault that started happening with python 3.12, changes in the PyLongObject implementation have broken some pointer arithmetic, which, from what I understand, is not necessary. Probably someone with a better understanding of C should check this :) I've tested with 3.12, 3.11 and 3.7. Request URL : https://github.com/alsa-project/alsa-python/pull/10 Patch URL : https://github.com/alsa-project/alsa-python/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH v1] ALSA: ASoc/tas2781: Fix an issue reported by robot kernel test
by Shenghao Ding 04 May '24

04 May '24

Fix an issue reported by robot kernel test and two harmless changes. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 - tasdevice-fmw.c --> tas2781-fmwlib.c - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202405021200.YHInjV43-lkp@intel.com/ --- sound/soc/codecs/tas2781-fmwlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..a6be81adcb83 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tasdevice-fmw.c -- TASDEVICE firmware support +// tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; -- 2.34.1

2 1

[PATCH] ALSA: hda/cs_dsp_ctl: Actually remove ALSA controls
by Richard Fitzgerald 03 May '24

03 May '24

hda_cs_dsp_control_remove() must remove the ALSA control when deleting all the infrastructure for handling the control. Without this it is possible for ALSA controls to be left in the Soundcard after the amp driver module has been unloaded. So the get/set callbacks point to code that no longer exists. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 3233b978af23 ("ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls") --- Note: it would be better to use the control private_free to do the cleanup, and that is my plan long-term. But that is a larger change to the code. I like to keep bugfix patches as simple as possible so they are low-risk and easy to cherry-pick into older kernels. So this patch fixes the bug. Sometime I will send a patch for future kernel versions that reworks the cleanup to use private_free. --- sound/pci/hda/hda_cs_dsp_ctl.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 463ca06036bf..a42653d3473d 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -203,6 +203,10 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv; + /* Only public firmware controls will have an associated kcontrol */ + if (ctl && ctl->kctl) + snd_ctl_remove(ctl->card, ctl->kctl); + kfree(ctl); } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); -- 2.39.2

2 3

[PATCH v3] ALSA: ASoc/tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 03 May '24

03 May '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. 'Fixes: 0a0877812628 ("ASoc: tas2781: Fix spelling mistake "calibraiton" -> "calibration"")' Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Remove redundant return in tasdev_load_calibrated_data - Put the second function parameter into the previous line for tasdev_load_calibrated_data - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202405021200.YHInjV43-lkp@intel.com/ v2: - In the Subject, fixed --> Fix - In tas2781-fmwlib.c, tasdevice-fmw.c ---> tas2781-fmwlib.c - dsp --> DSP - Remove unneeded parentheses for & (dereference) operator - Add Fixes tag - Changed the copyright year to 2024 in the related files - In tas2781-dsp.h, __TASDEVICE_DSP_H__ --> __TAS2781_DSP_H__ v1: - Download calibrated data after loading the new DSP config params - call tasdevice_prmg_load instead of tasdevice_prmg_calibdata_load, it is unnecessary to load calibrated data after loading DSP program. Load it after loading DSP config params each time. - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading DSP program. --- include/sound/tas2781-dsp.h | 7 +-- sound/soc/codecs/tas2781-fmwlib.c | 99 +++++++------------------------ sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 27 insertions(+), 83 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..2ab4c7a55b90 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tasdevice-fmw.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; @@ -2151,6 +2151,18 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, return ret; } +static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) +{ + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(priv, &cal->dev_data); + } +} + int tasdevice_select_tuningprm_cfg(void *context, int prm_no, int cfg_no, int rca_conf_no) { @@ -2210,21 +2222,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2247,9 +2247,13 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, if (tas_priv->tasdevice[i].is_loaderr == true) { status |= 1 << (i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + tasdev_load_calibrated_data(tas_priv, i); tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2312,6 @@ int tasdevice_prmg_load(void *context, int prm_no) } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { -- 2.34.1

3 2

[PATCH V2 1/2] ASoC: amd: acp: fix for acp platform device creation failure
by Vijendar Mukunda 03 May '24

03 May '24

ACP pin configuration varies based on acp version. ACP PCI driver should read the ACP PIN config value and based on config value, it has to create a platform device in below two conditions. 1) If ACP PDM configuration is selected from BIOS and ACP PDM controller exists. 2) If ACP I2S configuration is selected from BIOS. Other than above scenarios, ACP PCI driver should skip the platform device creation logic, i.e. ACP PCI driver probe sequence should never fail if other acp pin configuration is selected. It should skip platform device creation logic. check_acp_pdm() function was implemented for ACP6.x platforms to check ACP PDM configuration. Previously, this code was safe guarded by FLAG_AMD_LEGACY_ONLY_DMIC flag check. This implementation breaks audio use cases for Huawei Matebooks which are based on ACP3.x varaint uses I2S configuration. In current scenario, check_acp_pdm() function returns -ENODEV value which results in ACP PCI driver probe failure without creating a platform device even in case of valid ACP pin configuration. Implement check_acp_config() as a common function which invokes platform specific acp pin configuration check functions for ACP3.x, ACP6.0 & ACP6.3 & ACP7.0 variants and checks for ACP PDM controller. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218780 Fixes: 4af565de9f8c ("ASoC: amd: acp: fix for acp pdm configuration check") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes since v1: - drop white line change which is not relevant to this patch. sound/soc/amd/acp/acp-legacy-common.c | 96 ++++++++++++++++++++++----- sound/soc/amd/acp/acp-pci.c | 8 +-- sound/soc/amd/acp/amd.h | 10 ++- sound/soc/amd/acp/chip_offset_byte.h | 1 + 4 files changed, 95 insertions(+), 20 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index b5aff3f230be..3be7c6d55a6f 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -358,11 +358,25 @@ int smn_read(struct pci_dev *dev, u32 smn_addr) } EXPORT_SYMBOL_NS_GPL(smn_read, SND_SOC_ACP_COMMON); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) +static void check_acp3x_config(struct acp_chip_info *chip) { - struct acpi_device *pdm_dev; - const union acpi_object *obj; - u32 pdm_addr, val; + u32 val; + + val = readl(chip->base + ACP3X_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + chip->is_pdm_config = true; + break; + } +} + +static void check_acp6x_config(struct acp_chip_info *chip) +{ + u32 val; val = readl(chip->base + ACP_PIN_CONFIG); switch (val) { @@ -371,42 +385,94 @@ int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) case ACP_CONFIG_6: case ACP_CONFIG_7: case ACP_CONFIG_8: - case ACP_CONFIG_10: case ACP_CONFIG_11: + case ACP_CONFIG_14: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: case ACP_CONFIG_12: case ACP_CONFIG_13: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + break; + } +} + +static void check_acp70_config(struct acp_chip_info *chip) +{ + u32 val; + + val = readl(chip->base + ACP_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + case ACP_CONFIG_5: + case ACP_CONFIG_6: + case ACP_CONFIG_7: + case ACP_CONFIG_8: + case ACP_CONFIG_11: case ACP_CONFIG_14: + case ACP_CONFIG_17: + case ACP_CONFIG_18: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: + case ACP_CONFIG_12: + case ACP_CONFIG_13: + case ACP_CONFIG_19: + case ACP_CONFIG_20: + chip->is_i2s_config = true; + chip->is_pdm_config = true; break; default: - return -EINVAL; + break; } +} + +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) +{ + struct acpi_device *pdm_dev; + const union acpi_object *obj; + u32 pdm_addr; switch (chip->acp_rev) { case ACP3X_DEV: pdm_addr = ACP_RENOIR_PDM_ADDR; + check_acp3x_config(chip); break; case ACP6X_DEV: pdm_addr = ACP_REMBRANDT_PDM_ADDR; + check_acp6x_config(chip); break; case ACP63_DEV: pdm_addr = ACP63_PDM_ADDR; + check_acp6x_config(chip); break; case ACP70_DEV: pdm_addr = ACP70_PDM_ADDR; + check_acp70_config(chip); break; default: - return -EINVAL; + break; } - pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); - if (pdm_dev) { - if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == pdm_addr) - return 0; + if (chip->is_pdm_config) { + pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); + if (pdm_dev) { + if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == pdm_addr) + chip->is_pdm_dev = true; + } } - return -ENODEV; } -EXPORT_SYMBOL_NS_GPL(check_acp_pdm, SND_SOC_ACP_COMMON); +EXPORT_SYMBOL_NS_GPL(check_acp_config, SND_SOC_ACP_COMMON); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 5f35b90eab8d..d8314d2b331b 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -119,6 +119,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id if (ret) goto unregister_dmic_dev; + check_acp_config(pci, chip); + if (!chip->is_pdm_dev && !chip->is_i2s_config) + goto skip_pdev_creation; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; @@ -136,10 +140,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 5017e868f39b..d75b4eb34de8 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -138,6 +138,9 @@ struct acp_chip_info { void __iomem *base; /* ACP memory PCI base */ struct platform_device *chip_pdev; unsigned int flag; /* Distinguish b/w Legacy or Only PDM */ + bool is_pdm_dev; /* flag set to true when ACP PDM controller exists */ + bool is_pdm_config; /* flag set to true when PDM configuration is selected from BIOS */ + bool is_i2s_config; /* flag set to true when I2S configuration is selected from BIOS */ }; struct acp_stream { @@ -212,6 +215,11 @@ enum acp_config { ACP_CONFIG_13, ACP_CONFIG_14, ACP_CONFIG_15, + ACP_CONFIG_16, + ACP_CONFIG_17, + ACP_CONFIG_18, + ACP_CONFIG_19, + ACP_CONFIG_20, }; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; @@ -240,7 +248,7 @@ void restore_acp_pdm_params(struct snd_pcm_substream *substream, int restore_acp_i2s_params(struct snd_pcm_substream *substream, struct acp_dev_data *adata, struct acp_stream *stream); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip); +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip); static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction) { diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index cfd6c4d07594..18da734c0e9e 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -20,6 +20,7 @@ #define ACP_SOFT_RESET 0x1000 #define ACP_CONTROL 0x1004 #define ACP_PIN_CONFIG 0x1440 +#define ACP3X_PIN_CONFIG 0x1400 #define ACP_EXTERNAL_INTR_REG_ADDR(adata, offset, ctrl) \ (adata->acp_base + adata->rsrc->irq_reg_offset + offset + (ctrl * 0x04)) -- 2.34.1

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[PATCH v20 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 03 May '24

03 May '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Requesting to see if we can get some Acked-By tags, and merge on usb-next. Changelog -------------------------------------------- Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (39): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 444 ++++ drivers/usb/host/xhci.c | 48 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 182 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6311 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

4 54

usb:gadget:f_uac2: RFC: allowing multiple altsetttings for channel/samplesize combinations
by Pavel Hofman 02 May '24

02 May '24

Hi, I am considering implementation of multiple altsettings to f_uac2, so that multiple combinations of channels and samplesizes can be offered to the host. Configuration: -------------- * each altsetting for each direction should define * channel mask * samplesize * hs_bint bInterval * c_sync type (for capture only) Perhaps the easiest config would be allowing lists for the existing parameters (like the multiple samplerates were implemented). All the list params would have to have the same number of items - initial check. First values in the list would apply to altsetting 1, second to altsetting 2 etc. Or the altsetting could be some structured configfs param - please is there any recommended standard for structured configfs params? Should the config also adjust the list of allowed samplerates for each altsetting? Technically it makes sense as higher number of channels can decrease the max samplerate, e.g. for via a TDM interface. If so, it would need either the structured configuration or some "list of lists" format. Implementation: --------------- Parameters could be turned to arrays of fixed predefined sizes, like the p/s_srates. E.g. 5 max. altsettings in each direction would consume only 4 * (5-1) + 3* (5-1) = 28 extra ints (excluding the samplerates config). Currently all descriptor structs are statically pre-allocated as there are only two hard-coded altsettings. IMO the descriptors specific for each altsetting could be allocated dynamically in a loop over all none-zero alsettings. Please may I ask UAC2 gadget "stakeholders" for comments, suggestions, recommendations, so that my eventual initial version was in some generally acceptable direction? Thanks a lot, Pavel.

4 8

ASRock N100DC-ITX snd_usb_audio problems
by Florian Paul Schmidt 02 May '24

02 May '24

Hi! I recently acquired a ASRock N100DC-ITX mainboard with the goal to run some audio processing on it. After some tuning I got the board to behave nicely with e.g. cyclictest giving worst case latencies around 10 us with a recent -rt kernel. So the system shoudl be well equipped to run small period sizes. I have two USB audio class 2.0 audio interfaces here. 1. Focusrite 2i2 2nd gen 2. Tascam iXR Both these interfaces work fine on other systems, so I guess they are not the problem. And I used at least the Tascam with period sizes of e.g. 64 and 32 frames and 2 periods without problems (the 0.125 ms microframe resolution is good enough when it comes to the resulting jitter). But on this new system I cannot get either of the two interfaces to run with even 64 frames, 2 periods reliably. I tried jackd1, jack2, and the alsa_loopback program from Fons' libzita-alsa-pcmi. The system is well tuned with xhci IRQs at prio 90, jackd/alsa_loopback at prio 80 and all other kernel irq threads at prio 50. For 64 frames 2 periods 48000 samplerate $ LD_LIBRARY_PATH=../source/ ./alsa_loopback hw:USB hw:USB 48000 64 2 I get: May 02 12:37:35 ogfx100 kernel: retire_capture_urb: 1090 callbacks suppressed in the kernel log (every 5 seconds with the number of callbacks in the same ballpark) and alsa_loopback complaining: Error: pcm_wait returned 0. For 48 frames 2 periods 48000 samplerate (which should actually work better since the period boundaries should fall onto usb frame boundaries.) LD_LIBRARY_PATH=../source/ ./alsa_loopback hw:USB hw:USB 48000 48 2 I get: May 02 12:39:58 ogfx100 kernel: xhci_hcd 0000:00:14.0: WARN Event TRB for slot 4 ep 1 with no TDs queued? over and over in the kernel log and alsa_loopback again complaining as above. On the other hand 48 frames, 3 periods works great! I measure jitter with jack_wakeup in an otherwise empty jack-graph: [fps@ogfx100:~]$ ./src/jack_wakeup/jack_wakeup min: 0.92522 ms; mean: 0.999884ms; max: 1.07727 ms 64 frames, 3 periods works ok-ish: [fps@ogfx100:~]$ ./src/jack_wakeup/jack_wakeup min: 0.938189 ms; mean: 1.33265ms; max: 1.59372 ms with 3 to 4 times the jitter I would expect giving the 0.125 ms microframe length of USB 2. Here I would expect output more like min: 1.25... ms; mean: 1.33265ms; max: 1.5... ms or min: 1.25... ms; mean: 1.33265ms; max: 1.375... ms I suspect that this is not a problem of the snd_usb_audio driver from ALSA but rather a problem with the XHCI in this board, since the combination of cards and snd_usb_audio driver work well on other systems. But maybe you have some insight or suggestions for me to try before reporting to LKML.. Looking forward to your replies and kind regards, FPS P.S.: [fps@ogfx100:~]$ uname -a Linux ogfx100 6.6.25-rt29 #1-NixOS SMP PREEMPT_RT Thu Apr 4 18:23:07 UTC 2024 x86_64 GNU/Linux I have tried with many other kernels though -rt and non-rt. That did not change much IIRC. I am P.P.S.: Find below the output of lspci -v, lspci -t, lsusb -v, cat /proc/cpuinfo [fps@ogfx100:~]$ lspci -v 00:00.0 Host bridge: Intel Corporation Device 461c Subsystem: ASRock Incorporation Device 461c Flags: bus master, fast devsel, latency 0 Capabilities: <access denied> 00:02.0 VGA compatible controller: Intel Corporation Alder Lake-N [UHD Graphics] (prog-if 00 [VGA controller]) Subsystem: ASRock Incorporation Device 46d1 Flags: bus master, fast devsel, latency 0, IRQ 134 Memory at 6000000000 (64-bit, non-prefetchable) [size=16M] Memory at 4000000000 (64-bit, prefetchable) [size=256M] I/O ports at 4000 [size=64] Expansion ROM at 000c0000 [virtual] [disabled] [size=128K] Capabilities: <access denied> Kernel driver in use: i915 Kernel modules: i915 00:0d.0 USB controller: Intel Corporation Device 464e (prog-if 30 [XHCI]) Flags: medium devsel, IRQ 129 Memory at 6001110000 (64-bit, non-prefetchable) [size=64K] Capabilities: <access denied> Kernel driver in use: xhci_hcd Kernel modules: xhci_pci 00:14.0 USB controller: Intel Corporation Device 54ed (prog-if 30 [XHCI]) Subsystem: ASRock Incorporation Device 54ed Flags: bus master, medium devsel, latency 0, IRQ 131 Memory at 6001100000 (64-bit, non-prefetchable) [size=64K] Capabilities: <access denied> Kernel driver in use: xhci_hcd Kernel modules: xhci_pci 00:14.2 RAM memory: Intel Corporation Device 54ef Flags: fast devsel Memory at 6001124000 (64-bit, non-prefetchable) [disabled] [size=16K] Memory at 600112a000 (64-bit, non-prefetchable) [disabled] [size=4K] Capabilities: <access denied> 00:16.0 Communication controller: Intel Corporation Device 54e0 Subsystem: ASRock Incorporation Device 54e0 Flags: bus master, fast devsel, latency 0, IRQ 132 Memory at 6001129000 (64-bit, non-prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: mei_me Kernel modules: mei_me 00:17.0 SATA controller: Intel Corporation Device 54d3 (prog-if 01 [AHCI 1.0]) Subsystem: ASRock Incorporation Device 54d3 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 130 Memory at 80a00000 (32-bit, non-prefetchable) [size=8K] Memory at 80a03000 (32-bit, non-prefetchable) [size=256] I/O ports at 4090 [size=8] I/O ports at 4080 [size=4] I/O ports at 4060 [size=32] Memory at 80a02000 (32-bit, non-prefetchable) [size=2K] Capabilities: <access denied> Kernel driver in use: ahci Kernel modules: ahci 00:1c.0 PCI bridge: Intel Corporation Device 54be (prog-if 00 [Normal decode]) Subsystem: ASRock Incorporation Device 54be Flags: bus master, fast devsel, latency 0, IRQ 122 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 3000-3fff [size=4K] [16-bit] Memory behind bridge: 80900000-809fffff [size=1M] [32-bit] Prefetchable memory behind bridge: [disabled] [64-bit] Capabilities: <access denied> Kernel driver in use: pcieport 00:1d.0 PCI bridge: Intel Corporation Device 54b0 (prog-if 00 [Normal decode]) Subsystem: ASRock Incorporation Device 54b0 Flags: bus master, fast devsel, latency 0, IRQ 123 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: [disabled] [16-bit] Memory behind bridge: 80800000-808fffff [size=1M] [32-bit] Prefetchable memory behind bridge: [disabled] [64-bit] Capabilities: <access denied> Kernel driver in use: pcieport 00:1f.0 ISA bridge: Intel Corporation Device 5481 Subsystem: ASRock Incorporation Device 5481 Flags: bus master, fast devsel, latency 0 00:1f.3 Audio device: Intel Corporation Device 54c8 Subsystem: ASRock Incorporation Device 3897 Flags: bus master, fast devsel, latency 32, IRQ 135 Memory at 6001120000 (64-bit, non-prefetchable) [size=16K] Memory at 6001000000 (64-bit, non-prefetchable) [size=1M] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel, snd_sof_pci_intel_tgl 00:1f.4 SMBus: Intel Corporation Device 54a3 Subsystem: ASRock Incorporation Device 54a3 Flags: medium devsel, IRQ 16 Memory at 6001128000 (64-bit, non-prefetchable) [size=256] I/O ports at efa0 [size=32] Kernel driver in use: i801_smbus Kernel modules: i2c_i801 00:1f.5 Serial bus controller: Intel Corporation Device 54a4 Subsystem: ASRock Incorporation Device 54a4 Flags: fast devsel Memory at 80a04000 (32-bit, non-prefetchable) [size=4K] Kernel driver in use: intel-spi Kernel modules: spi_intel_pci 01:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168/8411 PCI Express Gigabit Ethernet Controller (rev 15) Subsystem: ASRock Incorporation Motherboard (one of many) Flags: bus master, fast devsel, latency 0, IRQ 18 I/O ports at 3000 [size=256] Memory at 80904000 (64-bit, non-prefetchable) [size=4K] Memory at 80900000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: r8169 Kernel modules: r8169 02:00.0 Non-Volatile memory controller: Kingston Technology Company, Inc. Device 5017 (rev 03) (prog-if 02 [NVM Express]) Subsystem: Kingston Technology Company, Inc. Device 5017 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 80800000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: nvme Kernel modules: nvme [fps@ogfx100:~]$ lspci -t -[0000:00]-+-00.0 +-02.0 +-0d.0 +-14.0 +-14.2 +-16.0 +-17.0 +-1c.0-[01]----00.0 +-1d.0-[02]----00.0 +-1f.0 +-1f.3 +-1f.4 \-1f.5 [fps@ogfx100:~]$ sudo lsusb -v Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 6.06 iManufacturer 3 Linux 6.6.25-rt29 xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:0d.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0019 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 1 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0100 power Device Status: 0x0001 Self Powered Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 3.10 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 3 bMaxPacketSize0 9 idVendor 0x1d6b Linux Foundation idProduct 0x0003 3.0 root hub bcdDevice 6.06 iManufacturer 3 Linux 6.6.25-rt29 xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:0d.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x001f bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 bMaxBurst 0 Binary Object Store Descriptor: bLength 5 bDescriptorType 15 wTotalLength 0x003b bNumDeviceCaps 2 SuperSpeed USB Device Capability: bLength 10 bDescriptorType 16 bDevCapabilityType 3 bmAttributes 0x02 Latency Tolerance Messages (LTM) Supported wSpeedsSupported 0x0008 Device can operate at SuperSpeed (5Gbps) bFunctionalitySupport 1 Lowest fully-functional device speed is Full Speed (12Mbps) bU1DevExitLat 0 micro seconds bU2DevExitLat 400 micro seconds SuperSpeedPlus USB Device Capability: bLength 44 bDescriptorType 16 bDevCapabilityType 10 bmAttributes 0x00000067 Sublink Speed Attribute count 8 Sublink Speed ID count 4 wFunctionalitySupport 0x1104 Min functional Speed Attribute ID: 4 Min functional RX lanes: 1 Min functional TX lanes: 1 bmSublinkSpeedAttr[0] 0x00050034 Speed Attribute ID: 4 5Gb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[1] 0x000500b4 Speed Attribute ID: 4 5Gb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[2] 0x000a4035 Speed Attribute ID: 5 10Gb/s Symmetric RX SuperSpeedPlus bmSublinkSpeedAttr[3] 0x000a40b5 Speed Attribute ID: 5 10Gb/s Symmetric TX SuperSpeedPlus bmSublinkSpeedAttr[4] 0x00054036 Speed Attribute ID: 6 5Gb/s Symmetric RX SuperSpeedPlus bmSublinkSpeedAttr[5] 0x000540b6 Speed Attribute ID: 6 5Gb/s Symmetric TX SuperSpeedPlus bmSublinkSpeedAttr[6] 0x000a4037 Speed Attribute ID: 7 10Gb/s Symmetric RX SuperSpeedPlus bmSublinkSpeedAttr[7] 0x000a40b7 Speed Attribute ID: 7 10Gb/s Symmetric TX SuperSpeedPlus Hub Descriptor: bLength 12 bDescriptorType 42 nNbrPorts 2 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 0 milli Ampere bHubDecLat 0.0 micro seconds wHubDelay 0 nano seconds DeviceRemovable 0x00 Hub Port Status: Port 1: 0000.02a0 5Gbps power Rx.Detect Port 2: 0000.02a0 5Gbps power Rx.Detect Device Status: 0x0001 Self Powered Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 6.06 iManufacturer 3 Linux 6.6.25-rt29 xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:14.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0019 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 11 bDescriptorType 41 nNbrPorts 12 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0xa0 0x00 PortPwrCtrlMask 0xff 0xff Hub Port Status: Port 1: 0000.0100 power Port 2: 0000.0303 lowspeed power enable connect Port 3: 0000.0100 power Port 4: 0000.0100 power Port 5: 0000.0503 highspeed power enable connect Port 6: 0000.0503 highspeed power enable connect Port 7: 0000.0507 highspeed power suspend enable connect Port 8: 0000.0100 power Port 9: 0000.0100 power Port 10: 0000.0100 power Port 11: 0000.0100 power Port 12: 0000.0100 power Device Status: 0x0001 Self Powered Bus 003 Device 002: ID 046d:c31c Logitech, Inc. Keyboard K120 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 [unknown] bDeviceSubClass 0 [unknown] bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x046d Logitech, Inc. idProduct 0xc31c Keyboard K120 bcdDevice 64.00 iManufacturer 1 Logitech iProduct 2 USB Keyboard iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x003b bNumInterfaces 2 bConfigurationValue 1 iConfiguration 3 U64.00_B0001 bmAttributes 0xa0 (Bus Powered) Remote Wakeup MaxPower 90mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 1 Boot Interface Subclass bInterfaceProtocol 1 Keyboard iInterface 2 USB Keyboard HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 65 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 iInterface 2 USB Keyboard HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 159 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 255 Device Status: 0x0000 (Bus Powered) Bus 003 Device 003: ID 174c:2074 ASMedia Technology Inc. ASM1074 High-Speed hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.10 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 2 TT per port bMaxPacketSize0 64 idVendor 0x174c ASMedia Technology Inc. idProduct 0x2074 ASM1074 High-Speed hub bcdDevice 2.00 iManufacturer 2 Asmedia iProduct 3 ASM107x iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0029 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 1 Single TT iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 12 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 2 TT per port iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 12 Binary Object Store Descriptor: bLength 5 bDescriptorType 15 wTotalLength 0x002a bNumDeviceCaps 3 USB 2.0 Extension Device Capability: bLength 7 bDescriptorType 16 bDevCapabilityType 2 bmAttributes 0x00000000 (Missing must-be-set LPM bit!) SuperSpeed USB Device Capability: bLength 10 bDescriptorType 16 bDevCapabilityType 3 bmAttributes 0x00 wSpeedsSupported 0x000e Device can operate at Full Speed (12Mbps) Device can operate at High Speed (480Mbps) Device can operate at SuperSpeed (5Gbps) bFunctionalitySupport 1 Lowest fully-functional device speed is Full Speed (12Mbps) bU1DevExitLat 10 micro seconds bU2DevExitLat 2047 micro seconds Container ID Device Capability: bLength 20 bDescriptorType 16 bDevCapabilityType 4 bReserved 0 ContainerID {cfe2b8a8-57df-4db7-9881-214005c7fbe5} Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 4 wHubCharacteristic 0x00a9 Per-port power switching Per-port overcurrent protection TT think time 16 FS bits Port indicators bPwrOn2PwrGood 1 * 2 milli seconds bHubContrCurrent 100 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0100 power Port 2: 0000.0503 highspeed power enable connect Port 3: 0000.0100 power Port 4: 0000.0100 power Device Status: 0x0001 Self Powered Bus 003 Device 004: ID 1235:8202 Focusrite-Novation Focusrite Scarlett 2i2 2nd Gen Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 [unknown] bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x1235 Focusrite-Novation idProduct 0x8202 Focusrite Scarlett 2i2 2nd Gen bcdDevice 4.1b iManufacturer 1 Focusrite iProduct 3 Scarlett 2i2 USB iSerial 0 bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0116 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 9 Internal bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 3 Scarlett 2i2 USB AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0077 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 3 Internal programmable clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 10 USB Internal AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 40 bNrInPins 1 baCSourceID(0) 41 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 8 Clock Source AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 12 Output 1 bmControls 0x0000 iTerminal 6 Scarlett 2i2 USB AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 14 Input 1 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 11 bCSourceID 40 bmControls 0x0000 iTerminal 7 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Scarlett 2i2 USB AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 12 Output 1 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0008 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Scarlett 2i2 USB AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 14 Input 1 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0008 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 [unknown] bInterfaceProtocol 16 iInterface 11 Focusrite Control Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 8 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0116 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 9 Internal bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 3 Scarlett 2i2 USB AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0077 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 41 bmAttributes 3 Internal programmable clock bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 10 USB Internal AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bClockID 40 bNrInPins 1 baCSourceID(0) 41 bmControls 0x03 Clock Selector Control (read/write) iClockSelector 8 Clock Source AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 12 Output 1 bmControls 0x0000 iTerminal 6 Scarlett 2i2 USB AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 10 bCSourceID 40 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 40 bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 14 Input 1 bmControls 0x0000 iTerminal 0 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 22 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 11 bCSourceID 40 bmControls 0x0000 iTerminal 7 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 4 Scarlett 2i2 USB AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 12 Output 1 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0008 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0004 1x 4 bytes bInterval 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Scarlett 2i2 USB Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 5 Scarlett 2i2 USB AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 22 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 2 bmChannelConfig 0x00000000 iChannelNames 14 Input 1 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0008 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 [unknown] bInterfaceProtocol 16 iInterface 11 Focusrite Control Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 8 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 [unknown] bDeviceSubClass 0 [unknown] bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) Bus 003 Device 005: ID 148f:2070 Ralink Technology, Corp. RT2070 Wireless Adapter Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 [unknown] bDeviceSubClass 0 [unknown] bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x148f Ralink Technology, Corp. idProduct 0x2070 RT2070 Wireless Adapter bcdDevice 1.01 iManufacturer 1 Ralink iProduct 2 802.11 g WLAN iSerial 3 1.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0043 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 450mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 7 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 255 Vendor Specific Protocol iInterface 5 1.0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 [unknown] bDeviceSubClass 0 [unknown] bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) Bus 003 Device 006: ID 05e3:0610 Genesys Logic, Inc. Hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 2 TT per port bMaxPacketSize0 64 idVendor 0x05e3 Genesys Logic, Inc. idProduct 0x0610 Hub bcdDevice 60.52 iManufacturer 0 iProduct 1 USB2.0 Hub iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0029 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 1 Single TT iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 12 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 2 TT per port iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 4 wHubCharacteristic 0x00e9 Per-port power switching Per-port overcurrent protection TT think time 32 FS bits Port indicators bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 100 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0100 power Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0100 power Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 3.10 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 3 bMaxPacketSize0 9 idVendor 0x1d6b Linux Foundation idProduct 0x0003 3.0 root hub bcdDevice 6.06 iManufacturer 3 Linux 6.6.25-rt29 xhci-hcd iProduct 2 xHCI Host Controller iSerial 1 0000:00:14.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x001f bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 bMaxBurst 0 Binary Object Store Descriptor: bLength 5 bDescriptorType 15 wTotalLength 0x005b bNumDeviceCaps 2 SuperSpeed USB Device Capability: bLength 10 bDescriptorType 16 bDevCapabilityType 3 bmAttributes 0x02 Latency Tolerance Messages (LTM) Supported wSpeedsSupported 0x0008 Device can operate at SuperSpeed (5Gbps) bFunctionalitySupport 1 Lowest fully-functional device speed is Full Speed (12Mbps) bU1DevExitLat 10 micro seconds bU2DevExitLat 160 micro seconds SuperSpeedPlus USB Device Capability: bLength 76 bDescriptorType 16 bDevCapabilityType 10 bmAttributes 0x000000ef Sublink Speed Attribute count 16 Sublink Speed ID count 8 wFunctionalitySupport 0x1106 Min functional Speed Attribute ID: 6 Min functional RX lanes: 1 Min functional TX lanes: 1 bmSublinkSpeedAttr[0] 0x00050034 Speed Attribute ID: 4 5Gb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[1] 0x000500b4 Speed Attribute ID: 4 5Gb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[2] 0x000a4035 Speed Attribute ID: 5 10Gb/s Symmetric RX SuperSpeedPlus bmSublinkSpeedAttr[3] 0x000a40b5 Speed Attribute ID: 5 10Gb/s Symmetric TX SuperSpeedPlus bmSublinkSpeedAttr[4] 0x00e00026 Speed Attribute ID: 6 224Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[5] 0x00e000a6 Speed Attribute ID: 6 224Mb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[6] 0x00c00027 Speed Attribute ID: 7 192Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[7] 0x00c000a7 Speed Attribute ID: 7 192Mb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[8] 0x00800028 Speed Attribute ID: 8 128Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[9] 0x008000a8 Speed Attribute ID: 8 128Mb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[10] 0x00b10029 Speed Attribute ID: 9 177Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[11] 0x00b100a9 Speed Attribute ID: 9 177Mb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[12] 0x0063002a Speed Attribute ID: 10 99Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[13] 0x006300aa Speed Attribute ID: 10 99Mb/s Symmetric TX SuperSpeed bmSublinkSpeedAttr[14] 0x00c6002b Speed Attribute ID: 11 198Mb/s Symmetric RX SuperSpeed bmSublinkSpeedAttr[15] 0x00c600ab Speed Attribute ID: 11 198Mb/s Symmetric TX SuperSpeed Hub Descriptor: bLength 12 bDescriptorType 42 nNbrPorts 4 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 0 milli Ampere bHubDecLat 0.0 micro seconds wHubDelay 0 nano seconds DeviceRemovable 0x02 Hub Port Status: Port 1: 0000.0263 5Gbps power suspend enable connect Ext Status: 0000.0044 RX Speed Attribute ID: 4 Lanes: 1 TX Speed Attribute ID: 4 Lanes: 1 Port 2: 0000.02a0 5Gbps power Rx.Detect Port 3: 0000.02a0 5Gbps power Rx.Detect Port 4: 0000.02a0 5Gbps power Rx.Detect Device Status: 0x0001 Self Powered Bus 004 Device 002: ID 174c:3074 ASMedia Technology Inc. ASM1074 SuperSpeed hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 3.00 bDeviceClass 9 Hub bDeviceSubClass 0 [unknown] bDeviceProtocol 3 bMaxPacketSize0 9 idVendor 0x174c ASMedia Technology Inc. idProduct 0x3074 ASM1074 SuperSpeed hub bcdDevice 2.00 iManufacturer 2 Asmedia iProduct 3 ASM107x iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x001f bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 8mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 [unknown] bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 19 Transfer Type Interrupt Synch Type None Usage Type Feedback wMaxPacketSize 0x0002 1x 2 bytes bInterval 8 bMaxBurst 0 Binary Object Store Descriptor: bLength 5 bDescriptorType 15 wTotalLength 0x002a bNumDeviceCaps 3 USB 2.0 Extension Device Capability: bLength 7 bDescriptorType 16 bDevCapabilityType 2 bmAttributes 0x00000000 (Missing must-be-set LPM bit!) SuperSpeed USB Device Capability: bLength 10 bDescriptorType 16 bDevCapabilityType 3 bmAttributes 0x00 wSpeedsSupported 0x000e Device can operate at Full Speed (12Mbps) Device can operate at High Speed (480Mbps) Device can operate at SuperSpeed (5Gbps) bFunctionalitySupport 1 Lowest fully-functional device speed is Full Speed (12Mbps) bU1DevExitLat 10 micro seconds bU2DevExitLat 2047 micro seconds Container ID Device Capability: bLength 20 bDescriptorType 16 bDevCapabilityType 4 bReserved 0 ContainerID {cfe2b8a8-57df-4db7-9881-214005c7fbe5} Hub Descriptor: bLength 12 bDescriptorType 42 nNbrPorts 4 wHubCharacteristic 0x0009 Per-port power switching Per-port overcurrent protection bPwrOn2PwrGood 60 * 2 milli seconds bHubContrCurrent 400 milli Ampere bHubDecLat 0.4 micro seconds wHubDelay 3492 nano seconds DeviceRemovable 0x00 Hub Port Status: Port 1: 0000.02a0 5Gbps power Rx.Detect Port 2: 0000.02a0 5Gbps power Rx.Detect Port 3: 0000.02a0 5Gbps power Rx.Detect Port 4: 0000.02a0 5Gbps power Rx.Detect Device Status: 0x000d Self Powered U1 Enabled U2 Enabled [fps@ogfx100:~]$ sudo lsusb -t /: Bus 001.Port 001: Dev 001, Class=root_hub, Driver=xhci_hcd/1p, 480M /: Bus 002.Port 001: Dev 001, Class=root_hub, Driver=xhci_hcd/2p, 20000M/x2 /: Bus 003.Port 001: Dev 001, Class=root_hub, Driver=xhci_hcd/12p, 480M |__ Port 002: Dev 002, If 0, Class=Human Interface Device, Driver=usbhid, 1.5M |__ Port 002: Dev 002, If 1, Class=Human Interface Device, Driver=usbhid, 1.5M |__ Port 005: Dev 003, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 002: Dev 005, If 0, Class=Vendor Specific Class, Driver=rt2800usb, 480M |__ Port 006: Dev 004, If 0, Class=Audio, Driver=snd-usb-audio, 480M |__ Port 006: Dev 004, If 1, Class=Audio, Driver=snd-usb-audio, 480M |__ Port 006: Dev 004, If 2, Class=Audio, Driver=snd-usb-audio, 480M |__ Port 006: Dev 004, If 3, Class=Vendor Specific Class, Driver=[none], 480M |__ Port 007: Dev 006, If 0, Class=Hub, Driver=hub/4p, 480M /: Bus 004.Port 001: Dev 001, Class=root_hub, Driver=xhci_hcd/4p, 10000M |__ Port 001: Dev 002, If 0, Class=Hub, Driver=hub/4p, 5000M [fps@ogfx100:~]$ cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 190 model name : Intel(R) N100 stepping : 0 microcode : 0x15 cpu MHz : 700.000 cache size : 6144 KB physical id : 0 siblings : 4 core id : 0 cpu cores : 4 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 32 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc art arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc cpuid aperfmperf tsc_known_freq pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 sdbg fma cx16 xtpr pdcm sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm 3dnowprefetch cpuid_fault epb cat_l2 cdp_l2 ssbd ibrs ibpb stibp ibrs_enhanced tpr_shadow flexpriority ept vpid ept_ad fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid rdt_a rdseed adx smap clflushopt clwb intel_pt sha_ni xsaveopt xsavec xgetbv1 xsaves split_lock_detect user_shstk avx_vnni dtherm ida arat pln pts hwp hwp_notify hwp_act_window hwp_epp hwp_pkg_req vnmi umip pku ospke waitpkg gfni vaes vpclmulqdq rdpid movdiri movdir64b fsrm md_clear serialize arch_lbr ibt flush_l1d arch_capabilities vmx flags : vnmi preemption_timer posted_intr invvpid ept_x_only ept_ad ept_1gb flexpriority apicv tsc_offset vtpr mtf vapic ept vpid unrestricted_guest vapic_reg vid ple shadow_vmcs ept_mode_based_exec tsc_scaling usr_wait_pause bugs : spectre_v1 spectre_v2 spec_store_bypass swapgs rfds bogomips : 1612.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 190 model name : Intel(R) N100 stepping : 0 microcode : 0x15 cpu MHz : 3052.591 cache size : 6144 KB physical id : 0 siblings : 4 core id : 1 cpu cores : 4 apicid : 2 initial apicid : 2 fpu : yes fpu_exception : yes cpuid level : 32 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc art arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc cpuid aperfmperf tsc_known_freq pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 sdbg fma cx16 xtpr pdcm sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm 3dnowprefetch cpuid_fault epb cat_l2 cdp_l2 ssbd ibrs ibpb stibp ibrs_enhanced tpr_shadow flexpriority ept vpid ept_ad fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid rdt_a rdseed adx smap clflushopt clwb intel_pt sha_ni xsaveopt xsavec xgetbv1 xsaves split_lock_detect user_shstk avx_vnni dtherm ida arat pln pts hwp hwp_notify hwp_act_window hwp_epp hwp_pkg_req vnmi umip pku ospke waitpkg gfni vaes vpclmulqdq rdpid movdiri movdir64b fsrm md_clear serialize arch_lbr ibt flush_l1d arch_capabilities vmx flags : vnmi preemption_timer posted_intr invvpid ept_x_only ept_ad ept_1gb flexpriority apicv tsc_offset vtpr mtf vapic ept vpid unrestricted_guest vapic_reg vid ple shadow_vmcs ept_mode_based_exec tsc_scaling usr_wait_pause bugs : spectre_v1 spectre_v2 spec_store_bypass swapgs rfds bogomips : 1612.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 2 vendor_id : GenuineIntel cpu family : 6 model : 190 model name : Intel(R) N100 stepping : 0 microcode : 0x15 cpu MHz : 3148.532 cache size : 6144 KB physical id : 0 siblings : 4 core id : 2 cpu cores : 4 apicid : 4 initial apicid : 4 fpu : yes fpu_exception : yes cpuid level : 32 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc art arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc cpuid aperfmperf tsc_known_freq pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 sdbg fma cx16 xtpr pdcm sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm 3dnowprefetch cpuid_fault epb cat_l2 cdp_l2 ssbd ibrs ibpb stibp ibrs_enhanced tpr_shadow flexpriority ept vpid ept_ad fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid rdt_a rdseed adx smap clflushopt clwb intel_pt sha_ni xsaveopt xsavec xgetbv1 xsaves split_lock_detect user_shstk avx_vnni dtherm ida arat pln pts hwp hwp_notify hwp_act_window hwp_epp hwp_pkg_req vnmi umip pku ospke waitpkg gfni vaes vpclmulqdq rdpid movdiri movdir64b fsrm md_clear serialize arch_lbr ibt flush_l1d arch_capabilities vmx flags : vnmi preemption_timer posted_intr invvpid ept_x_only ept_ad ept_1gb flexpriority apicv tsc_offset vtpr mtf vapic ept vpid unrestricted_guest vapic_reg vid ple shadow_vmcs ept_mode_based_exec tsc_scaling usr_wait_pause bugs : spectre_v1 spectre_v2 spec_store_bypass swapgs rfds bogomips : 1612.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: processor : 3 vendor_id : GenuineIntel cpu family : 6 model : 190 model name : Intel(R) N100 stepping : 0 microcode : 0x15 cpu MHz : 3125.526 cache size : 6144 KB physical id : 0 siblings : 4 core id : 3 cpu cores : 4 apicid : 6 initial apicid : 6 fpu : yes fpu_exception : yes cpuid level : 32 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx pdpe1gb rdtscp lm constant_tsc art arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc cpuid aperfmperf tsc_known_freq pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 sdbg fma cx16 xtpr pdcm sse4_1 sse4_2 x2apic movbe popcnt tsc_deadline_timer aes xsave avx f16c rdrand lahf_lm abm 3dnowprefetch cpuid_fault epb cat_l2 cdp_l2 ssbd ibrs ibpb stibp ibrs_enhanced tpr_shadow flexpriority ept vpid ept_ad fsgsbase tsc_adjust bmi1 avx2 smep bmi2 erms invpcid rdt_a rdseed adx smap clflushopt clwb intel_pt sha_ni xsaveopt xsavec xgetbv1 xsaves split_lock_detect user_shstk avx_vnni dtherm ida arat pln pts hwp hwp_notify hwp_act_window hwp_epp hwp_pkg_req vnmi umip pku ospke waitpkg gfni vaes vpclmulqdq rdpid movdiri movdir64b fsrm md_clear serialize arch_lbr ibt flush_l1d arch_capabilities vmx flags : vnmi preemption_timer posted_intr invvpid ept_x_only ept_ad ept_1gb flexpriority apicv tsc_offset vtpr mtf vapic ept vpid unrestricted_guest vapic_reg vid ple shadow_vmcs ept_mode_based_exec tsc_scaling usr_wait_pause bugs : spectre_v1 spectre_v2 spec_store_bypass swapgs rfds bogomips : 1612.80 clflush size : 64 cache_alignment : 64 address sizes : 39 bits physical, 48 bits virtual power management: -- https://dfdx.eu

1 0

[PATCH] ALSA: hda: cs35l56: Perform firmware download in the background
by Simon Trimmer 02 May '24

02 May '24

It is possible that during system boot when there multiple devices attempting simultaneous initialization on a slow control bus the download of firmware and tuning data may take a user perceivable amount of time. Adopt a pattern used in the ASoC driver and perform this activity in a background thread so that interactive performance is not impaired. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 89 ++++++++++++++++++++++++++++++------- sound/pci/hda/cs35l56_hda.h | 4 ++ 2 files changed, 76 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 1a3f84599cb5..042c2407c007 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -13,6 +13,7 @@ #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/workqueue.h> #include <sound/core.h> #include <sound/cs-amp-lib.h> #include <sound/hda_codec.h> @@ -50,11 +51,19 @@ static const struct reg_sequence cs35l56_hda_dai_config[] = { }; +static void cs35l56_hda_wait_dsp_ready(struct cs35l56_hda *cs35l56) +{ + /* Wait for patching to complete */ + flush_work(&cs35l56->dsp_work); +} + static void cs35l56_hda_play(struct cs35l56_hda *cs35l56) { unsigned int val; int ret; + cs35l56_hda_wait_dsp_ready(cs35l56); + pm_runtime_get_sync(cs35l56->base.dev); ret = cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_PLAY); if (ret == 0) { @@ -180,6 +189,8 @@ static int cs35l56_hda_mixer_get(struct snd_kcontrol *kcontrol, unsigned int reg_val; int i; + cs35l56_hda_wait_dsp_ready(cs35l56); + regmap_read(cs35l56->base.regmap, kcontrol->private_value, &reg_val); reg_val &= CS35L56_ASP_TXn_SRC_MASK; @@ -203,6 +214,8 @@ static int cs35l56_hda_mixer_put(struct snd_kcontrol *kcontrol, if (item >= CS35L56_NUM_INPUT_SRC) return -EINVAL; + cs35l56_hda_wait_dsp_ready(cs35l56); + regmap_update_bits_check(cs35l56->base.regmap, kcontrol->private_value, CS35L56_INPUT_MASK, cs35l56_tx_input_values[item], &changed); @@ -227,6 +240,8 @@ static int cs35l56_hda_posture_get(struct snd_kcontrol *kcontrol, unsigned int pos; int ret; + cs35l56_hda_wait_dsp_ready(cs35l56); + ret = regmap_read(cs35l56->base.regmap, CS35L56_MAIN_POSTURE_NUMBER, &pos); if (ret) return ret; @@ -248,6 +263,8 @@ static int cs35l56_hda_posture_put(struct snd_kcontrol *kcontrol, (pos > CS35L56_MAIN_POSTURE_MAX)) return -EINVAL; + cs35l56_hda_wait_dsp_ready(cs35l56); + ret = regmap_update_bits_check(cs35l56->base.regmap, CS35L56_MAIN_POSTURE_NUMBER, CS35L56_MAIN_POSTURE_MASK, @@ -291,6 +308,8 @@ static int cs35l56_hda_vol_get(struct snd_kcontrol *kcontrol, int vol; int ret; + cs35l56_hda_wait_dsp_ready(cs35l56); + ret = regmap_read(cs35l56->base.regmap, CS35L56_MAIN_RENDER_USER_VOLUME, &raw_vol); if (ret) @@ -323,6 +342,8 @@ static int cs35l56_hda_vol_put(struct snd_kcontrol *kcontrol, raw_vol = (vol + CS35L56_MAIN_RENDER_USER_VOLUME_MIN) << CS35L56_MAIN_RENDER_USER_VOLUME_SHIFT; + cs35l56_hda_wait_dsp_ready(cs35l56); + ret = regmap_update_bits_check(cs35l56->base.regmap, CS35L56_MAIN_RENDER_USER_VOLUME, CS35L56_MAIN_RENDER_USER_VOLUME_MASK, @@ -539,8 +560,9 @@ static void cs35l56_hda_release_firmware_files(const struct firmware *wmfw_firmw kfree(coeff_filename); } -static void cs35l56_hda_add_dsp_controls(struct cs35l56_hda *cs35l56) +static void cs35l56_hda_create_dsp_controls_work(struct work_struct *work) { + struct cs35l56_hda *cs35l56 = container_of(work, struct cs35l56_hda, control_work); struct hda_cs_dsp_ctl_info info; info.device_name = cs35l56->amp_name; @@ -566,7 +588,7 @@ static void cs35l56_hda_apply_calibration(struct cs35l56_hda *cs35l56) dev_info(cs35l56->base.dev, "Calibration applied\n"); } -static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) +static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) { const struct firmware *coeff_firmware = NULL; const struct firmware *wmfw_firmware = NULL; @@ -574,15 +596,27 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) char *wmfw_filename = NULL; unsigned int preloaded_fw_ver; bool firmware_missing; - int ret = 0; + bool add_dsp_controls_required = false; + int ret; - /* Prepare for a new DSP power-up */ + /* + * Prepare for a new DSP power-up. If the DSP has had firmware + * downloaded previously then it needs to be powered down so that it + * can be updated and if hadn't been patched before then the controls + * will need to be added once firmware download succeeds. + */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); + else + add_dsp_controls_required = true; cs35l56->base.fw_patched = false; - pm_runtime_get_sync(cs35l56->base.dev); + ret = pm_runtime_resume_and_get(cs35l56->base.dev); + if (ret < 0) { + dev_err(cs35l56->base.dev, "Failed to resume and get %d\n", ret); + return; + } /* * The firmware can only be upgraded if it is currently running @@ -606,7 +640,6 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) */ if (!coeff_firmware && firmware_missing) { dev_err(cs35l56->base.dev, ".bin file required but not found\n"); - ret = -ENOENT; goto err_fw_release; } @@ -657,6 +690,15 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) CS35L56_FIRMWARE_MISSING); cs35l56->base.fw_patched = true; + /* + * Adding controls is deferred to prevent a lock inversion - ALSA takes + * the controls_rwsem when adding a control, the get() / put() + * functions of a control are called holding controls_rwsem and those + * that depend on running firmware wait for dsp_work() to complete. + */ + if (add_dsp_controls_required) + queue_work(cs35l56->dsp_wq, &cs35l56->control_work); + ret = cs_dsp_run(&cs35l56->cs_dsp); if (ret) dev_dbg(cs35l56->base.dev, "%s: cs_dsp_run ret %d\n", __func__, ret); @@ -676,15 +718,19 @@ static int cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) coeff_firmware, coeff_filename); err_pm_put: pm_runtime_put(cs35l56->base.dev); +} - return ret; +static void cs35l56_hda_dsp_work(struct work_struct *work) +{ + struct cs35l56_hda *cs35l56 = container_of(work, struct cs35l56_hda, dsp_work); + + cs35l56_hda_fw_load(cs35l56); } static int cs35l56_hda_bind(struct device *dev, struct device *master, void *master_data) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); struct hda_component *comps = master_data; - int ret; if (!comps || cs35l56->index < 0 || cs35l56->index >= HDA_MAX_COMPONENTS) return -EINVAL; @@ -698,12 +744,9 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas strscpy(comps->name, dev_name(dev), sizeof(comps->name)); comps->playback_hook = cs35l56_hda_playback_hook; - ret = cs35l56_hda_fw_load(cs35l56); - if (ret) - return ret; + queue_work(cs35l56->dsp_wq, &cs35l56->dsp_work); cs35l56_hda_create_controls(cs35l56); - cs35l56_hda_add_dsp_controls(cs35l56); #if IS_ENABLED(CONFIG_SND_DEBUG) cs35l56->debugfs_root = debugfs_create_dir(dev_name(cs35l56->base.dev), sound_debugfs_root); @@ -720,6 +763,9 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); struct hda_component *comps = master_data; + cancel_work_sync(&cs35l56->dsp_work); + cancel_work_sync(&cs35l56->control_work); + cs35l56_hda_remove_controls(cs35l56); #if IS_ENABLED(CONFIG_SND_DEBUG) @@ -747,6 +793,10 @@ static int cs35l56_hda_system_suspend(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); + cs35l56_hda_wait_dsp_ready(cs35l56); + + flush_work(&cs35l56->control_work); + if (cs35l56->playing) cs35l56_hda_pause(cs35l56); @@ -842,11 +892,8 @@ static int cs35l56_hda_system_resume(struct device *dev) ret = cs35l56_is_fw_reload_needed(&cs35l56->base); dev_dbg(cs35l56->base.dev, "fw_reload_needed: %d\n", ret); - if (ret > 0) { - ret = cs35l56_hda_fw_load(cs35l56); - if (ret) - return ret; - } + if (ret > 0) + queue_work(cs35l56->dsp_wq, &cs35l56->dsp_work); if (cs35l56->playing) cs35l56_hda_play(cs35l56); @@ -964,6 +1011,14 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) mutex_init(&cs35l56->base.irq_lock); dev_set_drvdata(cs35l56->base.dev, cs35l56); + cs35l56->dsp_wq = create_singlethread_workqueue("cs35l56-dsp"); + if (!cs35l56->dsp_wq) { + ret = -ENOMEM; + goto err; + } + INIT_WORK(&cs35l56->dsp_work, cs35l56_hda_dsp_work); + INIT_WORK(&cs35l56->control_work, cs35l56_hda_create_dsp_controls_work); + ret = cs35l56_hda_read_acpi(cs35l56, hid, id); if (ret) goto err; diff --git a/sound/pci/hda/cs35l56_hda.h b/sound/pci/hda/cs35l56_hda.h index 464e4aa63cd1..d207e4e0d167 100644 --- a/sound/pci/hda/cs35l56_hda.h +++ b/sound/pci/hda/cs35l56_hda.h @@ -14,6 +14,7 @@ #include <linux/firmware/cirrus/cs_dsp.h> #include <linux/firmware/cirrus/wmfw.h> #include <linux/regulator/consumer.h> +#include <linux/workqueue.h> #include <sound/cs35l56.h> struct dentry; @@ -21,6 +22,9 @@ struct dentry; struct cs35l56_hda { struct cs35l56_base base; struct hda_codec *codec; + struct work_struct dsp_work; + struct work_struct control_work; + struct workqueue_struct *dsp_wq; int index; const char *system_name; -- 2.34.1

3 5

[PATCH v5 00/18] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 02 May '24

02 May '24

Changes in v5: - Cleaned up MT8186 soundcard migration commit which erroneously had leftovers from development - Changed the mtk_pcm_constraints_data structure to hold pointers to snd_pcm_hw_constraint_list, as to really reuse the const data - Tested again on all of the listed MTK platforms. Changes in v4: - Rebased over next-20240409 - Dropped the first 4 patches from v3 as were already picked by Mark - Fixed W=1 build issue Changes in v3: - Added audio-routing names in enum in all yaml files - Added mention of disallowing old and new properties together in commit message of bindings patches - Fixed validation errors with sound-card-common.yaml inclusion due to missing model property in examples - Removed `else` enforcing headset-codec/speaker-codecs requirement if xxx-dai-link not present to avoid future commit noise as the deprecated statement will disallow deprecated properties as required Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (18): ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +- .../bindings/sound/mt8195-mt6359.yaml | 134 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 199 +++ .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8186/Makefile | 3 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 25 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 25 files changed, 1597 insertions(+), 2190 deletions(-) delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

3 20

[tiwai-sound:for-linus 1/4] sound/pci/hda/patch_realtek.c:7222:8: error: no member named 'power_hook' in 'struct alc_spec'
by kernel test robot 01 May '24

01 May '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 9b3c22ab3929672b117f5c46d718692abb65da0b commit: 1e707769df072757bdcafab158bb159ead73daa4 [1/4] ALSA: hda/realtek - Set GPIO3 to default at S4 state for Thinkpad with ALC1318 config: hexagon-allmodconfig (https://download.01.org/0day-ci/archive/20240502/202405020226.ddBTVUNI-lkp@…) compiler: clang version 19.0.0git (https://github.com/llvm/llvm-project 37ae4ad0eef338776c7e2cffb3896153d43dcd90) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240502/202405020226.ddBTVUNI-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202405020226.ddBTVUNI-lkp@intel.com/ Note: the tiwai-sound/for-linus HEAD 9b3c22ab3929672b117f5c46d718692abb65da0b builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): In file included from sound/pci/hda/patch_realtek.c:17: In file included from include/linux/pci.h:38: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:547:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 547 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:560:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 560 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/pci/hda/patch_realtek.c:17: In file included from include/linux/pci.h:38: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:573:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 573 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/pci/hda/patch_realtek.c:17: In file included from include/linux/pci.h:38: In file included from include/linux/interrupt.h:11: In file included from include/linux/hardirq.h:11: In file included from ./arch/hexagon/include/generated/asm/hardirq.h:1: In file included from include/asm-generic/hardirq.h:17: In file included from include/linux/irq.h:20: In file included from include/linux/io.h:13: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:584:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 584 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:594:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 594 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:604:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 604 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ In file included from sound/pci/hda/patch_realtek.c:17: In file included from include/linux/pci.h:2693: In file included from include/linux/dma-mapping.h:11: In file included from include/linux/scatterlist.h:8: In file included from include/linux/mm.h:2208: include/linux/vmstat.h:522:36: warning: arithmetic between different enumeration types ('enum node_stat_item' and 'enum lru_list') [-Wenum-enum-conversion] 522 | return node_stat_name(NR_LRU_BASE + lru) + 3; // skip "nr_" | ~~~~~~~~~~~ ^ ~~~ >> sound/pci/hda/patch_realtek.c:7222:8: error: no member named 'power_hook' in 'struct alc_spec' 7222 | spec->power_hook = alc287_s4_power_gpio3_default; | ~~~~ ^ 7 warnings and 1 error generated. vim +7222 sound/pci/hda/patch_realtek.c 7214 7215 static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec, 7216 const struct hda_fixup *fix, int action) 7217 { 7218 struct alc_spec *spec = codec->spec; 7219 7220 if (action != HDA_FIXUP_ACT_PRE_PROBE) 7221 return; > 7222 spec->power_hook = alc287_s4_power_gpio3_default; 7223 spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook; 7224 } 7225 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[GIT PULL] ASoC fixes for v6.9-rc6
by Mark Brown 01 May '24

01 May '24

The following changes since commit fec50db7033ea478773b159e0e2efb135270e3b7: Linux 6.9-rc3 (2024-04-07 13:22:46 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.9-rc6 for you to fetch changes up to c5782bb5468acf86d8ca8e161267e8d055fb4161: ASoC: meson: tdm fixes (2024-04-30 23:36:23 +0900) ---------------------------------------------------------------- ASoC: Fixes for v6.9 This is much larger than is ideal, partly due to your holiday but also due to several vendors having come in with relatively large fixes at similar times. It's all driver specific stuff. The meson fixes from Jerome fix some rare timing issues with blocking operations happening in triggers, plus the continuous clock support which fixes clocking for some platforms. The SOF series from Peter builds to the fix to avoid spurious resets of ChainDMA which triggered errors in cleanup paths with both PulseAudio and PipeWire, and there's also some simple new debugfs files from Pierre which make support a lot eaiser. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: avs: Set name of control as in topology Cezary Rojewski (1): ASoC: Intel: avs: Fix debug window description Derek Fang (2): ASoC: rt5645: Fix the electric noise due to the CBJ contacts floating ASoC: dt-bindings: rt5645: add cbj sleeve gpio property Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too Jack Yu (4): ASoC: rt722-sdca: modify channel number to support 4 channels ASoC: rt722-sdca: add headset microphone vrefo setting ASoC: rt715: add vendor clear control register ASoC: rt715-sdca: volume step modification Jerome Brunet (5): ASoC: meson: axg-fifo: use threaded irq to check periods ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger ASoC: meson: axg-tdm: add continuous clock support ASoC: meson: cards: select SND_DYNAMIC_MINORS Joao Paulo Goncalves (1): ASoC: ti: davinci-mcasp: Fix race condition during probe Mark Brown (4): ASoC: SOF: misc fixes ASoC: SOF: ipc4-pcm: Do not reset ChainDMA if it is ASoC: cs35l56: Fixes to handling of ASP1 config ASoC: meson: tdm fixes Peter Ujfalusi (5): ASoC: SOF: ipc4-pcm: Use consistent name for snd_sof_pcm_stream pointer ASoC: SOF: ipc4-pcm: Use consistent name for sof_ipc4_timestamp_info pointer ASoC: SOF: ipc4-pcm: Introduce generic sof_ipc4_pcm_stream_priv ASoC: SOF: ipc4-pcm: Do not reset the ChainDMA if it has not been allocated ASoC: SOF: Core: Handle error returned by sof_select_ipc_and_paths Pierre-Louis Bossart (4): ASoC: SOF: Intel: add default firmware library path for LNL ASoC: SOF: debug: show firmware/topology prefix/names ASoC: da7219-aad: fix usage of device_get_named_child_node() ASoC: cs35l56: fix usages of device_get_named_child_node() Ranjani Sridharan (1): ASoC: SOF: pcm: Restrict DSP D0i3 during S0ix to IPC3 Richard Fitzgerald (4): regmap: Add regmap_read_bypassed() ALSA: hda: cs35l56: Exit cache-only after cs35l56_wait_for_firmware_boot() ASoC: cs35l56: Fix unintended bus access while resetting amp ASoC: cs35l56: Prevent overwriting firmware ASP config Sameer Pujar (1): ASoC: tegra: Fix DSPK 16-bit playback Simon Trimmer (1): ASoC: cs35l56: Avoid static analysis warning of uninitialised variable Srinivas Kandagatla (1): ASoC: codecs: wsa881x: set clk_stop_mode1 flag Stefan Binding (1): ASoC: cs35l41: Update DSP1RX5/6 Sources for DSP config end.to.start (1): ASoC: acp: Support microphone from device Acer 315-24p Documentation/devicetree/bindings/sound/rt5645.txt | 6 ++ drivers/base/regmap/regmap.c | 37 +++++++ include/linux/regmap.h | 8 ++ include/sound/cs35l56.h | 2 + sound/pci/hda/cs35l56_hda.c | 4 + sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/cs35l41.c | 26 +++-- sound/soc/codecs/cs35l56-sdw.c | 2 - sound/soc/codecs/cs35l56-shared.c | 85 +++++++++------ sound/soc/codecs/cs35l56.c | 39 ++++++- sound/soc/codecs/da7219-aad.c | 6 +- sound/soc/codecs/rt5645.c | 25 +++++ sound/soc/codecs/rt715-sdca.c | 8 +- sound/soc/codecs/rt715-sdw.c | 1 + sound/soc/codecs/rt722-sdca.c | 27 +++-- sound/soc/codecs/rt722-sdca.h | 3 + sound/soc/codecs/wsa881x.c | 1 + sound/soc/intel/avs/icl.c | 2 +- sound/soc/intel/avs/topology.c | 2 + sound/soc/intel/boards/bytcr_rt5640.c | 14 +-- sound/soc/meson/Kconfig | 1 + sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-fifo.c | 29 ++++-- sound/soc/meson/axg-tdm-formatter.c | 40 +++++++ sound/soc/meson/axg-tdm-interface.c | 38 +++++-- sound/soc/meson/axg-tdm.h | 5 + sound/soc/sof/core.c | 4 +- sound/soc/sof/debug.c | 18 ++++ sound/soc/sof/intel/pci-lnl.c | 3 + sound/soc/sof/ipc3-pcm.c | 1 + sound/soc/sof/ipc4-pcm.c | 115 ++++++++++++++------- sound/soc/sof/pcm.c | 13 ++- sound/soc/sof/sof-audio.h | 2 + sound/soc/tegra/tegra186_dspk.c | 7 +- sound/soc/ti/davinci-mcasp.c | 12 +-- 35 files changed, 460 insertions(+), 134 deletions(-)

2 1

[PATCH v2] ALSA: ASoc/tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 01 May '24

01 May '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. 'Fixes: 0a0877812628 ("ASoc: tas2781: Fix spelling mistake "calibraiton" -> "calibration"")' Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - In the Subject, fixed --> Fix - In tas2781-fmwlib.c, tasdevice-fmw.c ---> tas2781-fmwlib.c - dsp --> DSP - Remove unneeded parentheses for & (dereference) operator - Add Fixes tag - Changed the copyright year to 2024 in the related files - In tas2781-dsp.h, __TASDEVICE_DSP_H__ --> __TAS2781_DSP_H__ v1: - Download calibrated data after loading the new DSP config params - call tasdevice_prmg_load instead of tasdevice_prmg_calibdata_load, it is unnecessary to load calibrated data after loading DSP program. Load it after loading DSP config params each time. - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading DSP program. --- include/sound/tas2781-dsp.h | 7 +-- sound/soc/codecs/tas2781-fmwlib.c | 99 +++++++------------------------ sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 28 insertions(+), 82 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..5a4095c6a954 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tasdevice-fmw.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -2151,6 +2151,20 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, return ret; } +static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, + int i) +{ + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(priv, &cal->dev_data); + return; + } +} + int tasdevice_select_tuningprm_cfg(void *context, int prm_no, int cfg_no, int rca_conf_no) { @@ -2210,21 +2224,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2247,9 +2249,13 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, if (tas_priv->tasdevice[i].is_loaderr == true) { status |= 1 << (i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + tasdev_load_calibrated_data(tas_priv, i); tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2314,6 @@ int tasdevice_prmg_load(void *context, int prm_no) } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { -- 2.34.1

2 2

[PATCH 0/5] ASoc: SOF: misc updates for 6.10
by Pierre-Louis Bossart 01 May '24

01 May '24

Fixes when fw_lib_prefix is not set, updated error messages, improved dmesg logs to SoundWire configurations not supported by ACPI tables/topology and support for IEC61937 passthrough. Peter Ujfalusi (2): ASoC: SOF: debug: Handle cases when fw_lib_prefix is not set, NULL ASoC: SOF: ipc4-topology: Advertise passthrough capable PCMs (using ChainDMA) Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda: simplify and reduce indentation for hda_sdw_machine_select() ASoC: SOF: Intel: hda: list SoundWire peripherals on mismatch Yong Zhi (1): ASoC: SOF: topology: remove incorrect widget id in error message sound/soc/sof/debug.c | 23 ++++- sound/soc/sof/intel/hda.c | 154 +++++++++++++++++++--------------- sound/soc/sof/ipc4-topology.c | 50 +++++++++++ sound/soc/sof/topology.c | 7 +- 4 files changed, 160 insertions(+), 74 deletions(-) -- 2.40.1

2 6

[PATCH 0/4] ASoC: meson: tdm fixes
by Jerome Brunet 01 May '24

01 May '24

This patchset fixes 2 problems on TDM which both find a solution by properly implementing the .trigger() callback for the TDM backend. ATM, enabling the TDM formatters is done by the .prepare() callback because handling the formatter is slow due to necessary calls to CCF. The first problem affects the TDMIN. Because .prepare() is called on DPCM backend first, the formatter are started before the FIFOs and this may cause a random channel shifts if the TDMIN use multiple lanes with more than 2 slots per lanes. Using trigger() allows to set the FE/BE order, solving the problem. There has already been an attempt to fix this 3y ago [1] and reverted [2] It triggered a 'sleep in irq' error on the period IRQ. The solution is to just use the bottom half of threaded IRQ. This is patch #1. Patch #2 and #3 remain mostly the same as 3y ago. For TDMOUT, the problem is on pause. ATM pause only stops the FIFO and the TDMOUT just starves. When it does, it will actually repeat the last sample continuously. Depending on the platform, if there is no high-pass filter on the analog path, this may translate to a constant position of the speaker membrane. There is no audible glitch but it may damage the speaker coil. Properly stopping the TDMOUT in pause solves the problem. There is behaviour change associated with that fix. Clocks used to be continuous on pause because of the problem above. They will now be gated on pause by default, as they should. The last change introduce the proper support for continuous clocks, if needed. [1]: https://lore.kernel.org/linux-amlogic/20211020114217.133153-1-jbrunet@bayli… [2]: https://lore.kernel.org/linux-amlogic/20220421155725.2589089-1-narmstrong@b… Jerome Brunet (4): ASoC: meson: axg-fifo: use threaded irq to check periods ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger ASoC: meson: axg-tdm: add continuous clock support sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-fifo.c | 29 +++++++++++++-------- sound/soc/meson/axg-tdm-formatter.c | 40 +++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-interface.c | 38 +++++++++++++++++++-------- sound/soc/meson/axg-tdm.h | 5 ++++ 5 files changed, 93 insertions(+), 20 deletions(-) -- 2.43.0

2 5

[PATCH 00/23] ASoC: Intel: updates for 6.10 - part6
by Pierre-Louis Bossart 01 May '24

01 May '24

Last batch of cleanups from Brent Lu, with Chromebooks now supported with fewer modular machine drivers. Bard Liao (1): ASoC: Intel: sof_rt5682: add missing MAX98357A config Brent Lu (22): ASoC: Intel: nau8825/rt5682: move speaker widget to common modules ASoC: Intel: sof_maxim_common: support MAX98390 on cml boards ASoC: Intel: sof_da7219: support MAX98390 ASoC: Intel: sof_da7219: add cml_da7219_def for cml boards ASoC: Intel: sof_da7219: support MAX98390 on cml boards ASoC: Intel: bxt_da7219_max98357a: remove MAX98390 support ASoC: Intel: sof_da7219: support MAX98357A ASoC: Intel: sof_da7219: support MAX98357A on cml boards ASoC: Intel: bxt_da7219_max98357a: remove cml support ASoC: Intel: sof_da7219: add glk_da7219_def for glk boards ASoC: Intel: sof_da7219: support MAX98357A on glk boards ASoC: Intel: bxt_da7219_max98357a: remove glk support ASoC: Intel: sof_rt5682: add glk_rt5682_def for glk boards ASoC: Intel: sof_rt5682: support MAX98357A on glk boards ASoC: Intel: glk_rt5682_max98357a: delete driver ASoC: Intel: sof-rt5682: add mclk_en to sof_rt5682_private ASoC: Intel: sof-rt5682: remove SOF_RT5682_MCLK_BYTCHT_EN ASoC: Intel: sof_rt5682: add icl_rt5682_def for icl boards ASoC: Intel: sof-rt5682: add driver_data to sof_rt5682 board ASoC: Intel: sof-rt5682: setup pll_id only when needed ASoC: Intel: sof-rt5682: get bclk frequency from topology ASoC: Intel: sof-rt5682: support bclk as PLL source on rt5682s sound/soc/intel/boards/Kconfig | 41 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 179 +---- sound/soc/intel/boards/glk_rt5682_max98357a.c | 693 ------------------ sound/soc/intel/boards/sof_board_helpers.h | 2 + sound/soc/intel/boards/sof_da7219.c | 110 ++- sound/soc/intel/boards/sof_maxim_common.c | 108 ++- sound/soc/intel/boards/sof_nau8825.c | 4 - sound/soc/intel/boards/sof_realtek_common.c | 93 ++- sound/soc/intel/boards/sof_rt5682.c | 233 +++--- .../intel/common/soc-acpi-intel-cml-match.c | 4 +- .../intel/common/soc-acpi-intel-glk-match.c | 4 +- .../intel/common/soc-acpi-intel-icl-match.c | 2 +- 13 files changed, 461 insertions(+), 1016 deletions(-) delete mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c -- 2.40.1

2 24

[PATCH 00/12] ASoC: Intel: updates for 6.10 - part5
by Pierre-Louis Bossart 01 May '24

01 May '24

This patchset corrects a couple of mistakes corrected, improves snd_soc_card allocation. The new functionality is mostly for SoundWire platforms, with new SKUs for Dell and Acer, and support for the Cirrus Logic bridge/sidecar amplifier topology. Bard Liao (1): ASoC: Intel: sof_sdw: add a space before cfg-amp in components Charles Keepax (2): ASoC: Intel: sof_sdw: Delay update of the codec_conf array ASoC: Intel: sof_sdw: Add callbacks to register sidecar devices Mac Chiang (1): ASoC: Intel: soc-acpi-intel-lnl-match: adds RT714 and RT1318 support Maciej Strozek (1): ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps Peter Ujfalusi (3): ASoC: Intel: skl_hda_dsp_generic: Allocate snd_soc_card dynamically ASoC: Intel: skl_hda_dsp_generic: Use devm_kasprintf for the components string ASoC: Intel: sof_sdw: Allocate snd_soc_card dynamically Pierre-Louis Bossart (4): ASoC: Intel: soc-acpi: mtl: add Dell SKU 0C64 and 0CC6 ASoC: Intel: soc-acpi: mtl: add support for Acer Swift Go 14 ASoC: Intel: sof-sdw: don't set card long_name ASoC: Intel: sof-sdw: really remove FOUR_SPEAKER quirk sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/bridge_cs35l56.c | 137 ++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 50 +++---- sound/soc/intel/boards/sof_sdw.c | 106 ++++++++------ sound/soc/intel/boards/sof_sdw_common.h | 27 ++++ sound/soc/intel/boards/sof_sdw_cs42l43.c | 14 +- .../intel/common/soc-acpi-intel-lnl-match.c | 52 +++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 83 ++++++++++- 10 files changed, 397 insertions(+), 75 deletions(-) create mode 100644 sound/soc/intel/boards/bridge_cs35l56.c -- 2.40.1

2 13

[PATCH 0/8] ASoC: Intel: avs: PCM code cleanup
by Cezary Rojewski 01 May '24

01 May '24

A set of changes that aims to improve readability of cohesiveness of the pcm code for the avs-driver. Start off with a change that synchronizes DAI open/close - DAIs are started up in ascending order yet their shutdown does not follow the scheme - it is done in the ascending order too, rather than desceding one. This patch is a dependency for the next one in line. To align the HDAudio DAI startup/shutdown with the non-HDAudio equivalents, relocate the code from component to DAI. The reason above is a dependency stems from codec driver requirements - HDAudio code found in sound/pci/hda/ expects substream->runtime->private_data to point to a valid stream (HOST) pointer. With the hard part done, the follow up changes update the existing code to reduce it is complexity - removal of duplicates, renaming of ambiguous functions and adding new fields to DAI-data object so that the number of local variables and casts is reduced. Cezary Rojewski (8): ASoC: pcm: Reverse iterate DAIs when shutting them down ASoC: Intel: avs: Relocate HDA BE DAI specific operations ASoC: Intel: avs: Remove redundancy around DAI shutdown ASoC: Intel: avs: Store pointer to adev in DAI dma_data ASoC: Intel: avs: Remove redundancy around DAI startup ASoC: Intel: avs: Remove redundancy around DAI prepare ASoC: Intel: avs: Store pointer to link_stream in dma_data ASoC: Intel: avs: Clean up hw constraints initialization include/sound/soc.h | 4 + sound/soc/intel/avs/pcm.c | 250 +++++++++++++++++--------------------- sound/soc/soc-pcm.c | 2 +- 3 files changed, 117 insertions(+), 139 deletions(-) -- 2.25.1

2 9

[PATCH 1/1] ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with wait_for_completion_timeout()
by Wolfram Sang 01 May '24

01 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Fix to the proper variable type 'unsigned long' while here. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- sound/aoa/soundbus/i2sbus/pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 07df5cc0f2d7..98b812ffbde6 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, { unsigned long flags; DECLARE_COMPLETION_ONSTACK(done); - long timeout; + unsigned long time_left; spin_lock_irqsave(&i2sdev->low_lock, flags); if (pi->dbdma_ring.stopping) { pi->stop_completion = &done; spin_unlock_irqrestore(&i2sdev->low_lock, flags); - timeout = wait_for_completion_timeout(&done, HZ); + time_left = wait_for_completion_timeout(&done, HZ); spin_lock_irqsave(&i2sdev->low_lock, flags); pi->stop_completion = NULL; - if (timeout == 0) { + if (time_left == 0) { /* timeout expired, stop dbdma forcefully */ printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); /* make sure RUN, PAUSE and S0 bits are cleared */ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); pi->dbdma_ring.stopping = 0; - timeout = 10; + time_left = 10; while (in_le32(&pi->dbdma->status) & ACTIVE) { - if (--timeout <= 0) + if (--time_left <= 0) break; udelay(1); } -- 2.43.0

2 1

Hacker/stalkerALERT
by GitHub issues - opened 01 May '24

01 May '24

alsa-project/alsa-tools issue #24 was opened from interstellar08: same problems no matter what i do. new mail ... everything new phone .. com ALL i cannot get rid of this!!! Issue URL : https://github.com/alsa-project/alsa-tools/issues/24 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

topology: Add SND_PCM_RATE_24000 for 24 kHz rate support
by GitHub pull_request - opened 01 May '24

01 May '24

alsa-project/alsa-lib pull request #396 was opened from yongzhi1: The Low Complexity Communication Codec (LC3) supports 24 kHz sample rate, so add SND_PCM_RATE_24000 to the supported rates. Request URL : https://github.com/alsa-project/alsa-lib/pull/396 Patch URL : https://github.com/alsa-project/alsa-lib/pull/396.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
by Joao Schim 30 Apr '24

30 Apr '24

The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike Signed-off-by: Joao Schim <joao(a)schimsalabim.eu> --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c6..dd32780fb6a4 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tlv.h> #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) -- 2.25.1

3 2

[PATCH] ASoC: da7219-aad: fix usage of device_get_named_child_node()
by Pierre-Louis Bossart 30 Apr '24

30 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/da7219-aad.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6bc068cdcbe2..15e5e3eb592b 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -671,8 +671,10 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) return NULL; aad_pdata = devm_kzalloc(dev, sizeof(*aad_pdata), GFP_KERNEL); - if (!aad_pdata) + if (!aad_pdata) { + fwnode_handle_put(aad_np); return NULL; + } aad_pdata->irq = i2c->irq; @@ -753,6 +755,8 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) else aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + fwnode_handle_put(aad_np); + return aad_pdata; } -- 2.40.1

2 1

[PATCH] ASoC: cs35l56: fix usages of device_get_named_child_node()
by Pierre-Louis Bossart 30 Apr '24

30 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid leaked references. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/cs35l56.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index dfd703d9e12f..758dfdf9d3ea 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1361,6 +1361,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 "spk-id-gpios", ACPI_TYPE_PACKAGE, &obj); if (ret) { dev_dbg(cs35l56->base.dev, "Could not get spk-id-gpios package: %d\n", ret); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1368,6 +1369,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 if (obj->package.count != 4) { dev_warn(cs35l56->base.dev, "Unexpected spk-id element count %d\n", obj->package.count); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1382,6 +1384,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 */ ret = acpi_dev_add_driver_gpios(adev, cs35l56_af01_spkid_gpios_mapping); if (ret) { + fwnode_handle_put(af01_fwnode); return dev_err_probe(cs35l56->base.dev, ret, "Failed to add gpio mapping to AF01\n"); } @@ -1389,14 +1392,17 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 ret = devm_add_action_or_reset(cs35l56->base.dev, cs35l56_acpi_dev_release_driver_gpios, adev); - if (ret) + if (ret) { + fwnode_handle_put(af01_fwnode); return ret; + } dev_dbg(cs35l56->base.dev, "Added spk-id-gpios mapping to AF01\n"); } desc = fwnode_gpiod_get_index(af01_fwnode, "spk-id", 0, GPIOD_IN, NULL); if (IS_ERR(desc)) { + fwnode_handle_put(af01_fwnode); ret = PTR_ERR(desc); return dev_err_probe(cs35l56->base.dev, ret, "Get GPIO from AF01 failed\n"); } @@ -1405,9 +1411,12 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 gpiod_put(desc); if (ret < 0) { + fwnode_handle_put(af01_fwnode); dev_err_probe(cs35l56->base.dev, ret, "Error reading spk-id GPIO\n"); return ret; - } + } + + fwnode_handle_put(af01_fwnode); dev_info(cs35l56->base.dev, "Got spk-id from AF01\n"); -- 2.40.1

2 1

[PATCH] ASoC: meson: cards: select SND_DYNAMIC_MINORS
by Jerome Brunet 30 Apr '24

30 Apr '24

Amlogic sound cards do create a lot of pcm interfaces, possibly more than 8. Some pcm interfaces are internal (like DPCM backends and c2c) and not exposed to userspace. Those interfaces still increase the number passed to snd_find_free_minor(), which eventually exceeds 8 causing -EBUSY error on card registration if CONFIG_SND_DYNAMIC_MINORS=n and the interface is exposed to userspace. select CONFIG_SND_DYNAMIC_MINORS for Amlogic cards to avoid the problem. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- NOTE: This problem was found while toying with a minimal configuration which disabled CONFIG_SND_DYNAMIC_MINORS and DT overlays. Overlays prepend nodes instead of doing append. This changes the order in which the links are parsed by the card driver, which in turn changes the pcm numbers. The userpace pcm number then got over 8 triggering the problem. sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index b93ea33739f2..6458d5dc4902 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -99,6 +99,7 @@ config SND_MESON_AXG_PDM config SND_MESON_CARD_UTILS tristate + select SND_DYNAMIC_MINORS config SND_MESON_CODEC_GLUE tristate -- 2.43.0

2 1

[DNM] common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - edited 30 Apr '24

30 Apr '24

alsa-project/alsa-ucm-conf pull request #413 was edited from ujfalusi: Hi, The PR will introduce a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[DNM] common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - opened 30 Apr '24

30 Apr '24

alsa-project/alsa-ucm-conf pull request #413 was opened from ujfalusi: Hi, The PR will introduce a a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: Thinkpad with ALC1318 has a chance of damaging the IC
by Kailang 30 Apr '24

30 Apr '24

Hi Takashi, Please don't apply this patch. They want to add stream open to enable GPIO3 functions. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Wednesday, April 10, 2024 5:23 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Thinkpad with ALC1318 has a chance of damaging the IC > > Hi Takashi, > > Thinkpad with ALC1318 has a chance of damaging the IC at S4 resume. > > BR, > Kailang

2 5

[PATCH v1 0/2] ALSA: hda: cs35l41: Ignore errors when configuring interrupts to allow laptops with bad ACPI to play audio
by Stefan Binding 30 Apr '24

30 Apr '24

Some laptops have a bad _CRS defined for its interrupt. This errors out inside the driver probe for CS35L41 HDA. However, there is no fix in software for this bad configuration, and it is unlikely to get a new BIOS, therefore it is better to allow the laptop to continue probe, instead of erroring out inside the probe. The interrupt is only used for Error Detection and recovery, without the interrupt the driver will continue to function, but errors will not be detected, and recovery will require a reboot. Also add support for ASUS ROG 2024 laptops. These laptops were released without _DSD, so need to be added into the CS35L41 config table. Quirks for these laptops already exist. Stefan Binding (2): ALSA: hda: cs35l41: Ignore errors when configuring IRQs ALSA: hda: cs35l41: Add support for ASUS ROG 2024 Laptops sound/pci/hda/cs35l41_hda.c | 69 +++++++++++++++++++--------- sound/pci/hda/cs35l41_hda_property.c | 10 ++++ 2 files changed, 57 insertions(+), 22 deletions(-) -- 2.34.1

2 3

[PATCH 0/2][next] firewire: Avoid -Wflex-array-member-not-at-end warnings
by Gustavo A. R. Silva 29 Apr '24

29 Apr '24

-Wflex-array-member-not-at-end is coming in GCC-14, and we are getting ready to enable it globally. So, we are deprecating flexible-array members in the middle of another struct. There are currently a couple of local structures (`u` and `template`) that are using a flexible `struct fw_iso_packet` as header for a couple of on-stack arrays. We make use of the `struct_group_tagged()` helper to separate the flexible array from the rest of the members in the flexible structure, and, with this, we can now declare objects of the type of the tagged struct, without embedding the flexible array in the middle of another struct. We also use `container_of()` whenever we need to retrieve a pointer to the flexible structure, through which the flexible-array member can be accessed. With these changes, we fix a couple of -Wflex-array-member-not-at-end warnings. Gustavo A. R. Silva (2): firewire: Avoid -Wflex-array-member-not-at-end warning ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning drivers/firewire/core-cdev.c | 9 +++++---- include/linux/firewire.h | 16 +++++++++------- sound/firewire/amdtp-stream.c | 8 +++++--- 3 files changed, 19 insertions(+), 14 deletions(-) -- 2.34.1

4 7

[PATCH] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
by Joao Schim 29 Apr '24

29 Apr '24

The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c6..f8613d8c3462 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tlv.h> #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const DECLARE_TLV_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) -- 2.25.1

2 5

[PATCH 0/3] sysfs: Group visibility fixups
by Dan Williams 29 Apr '24

29 Apr '24

Hey Greg, Marc was able to get me a backtrace for that bootup hang scenario that Pierre noted here [1]. A Tested-by is still pending, but I am certain this is the issue, and it may impact more than just the one platform if that "empty_attrs" pattern has been repeated anywhere else in the kernel. I also took some time to document how to use the helpers a bit better, and introduce DEFINE_SIMPLE_SYSFS_GROUP_VISIBLE() for cases where all that matters is group visibility and not per attribute visibility. [1]: http://lore.kernel.org/r/b93ec9c2-23f5-486b-a3dc-ed9b960df359@linux.intel.c… --- Dan Williams (3): sysfs: Fix crash on empty group attributes array sysfs: Document new "group visible" helpers sysfs: Introduce DEFINE_SIMPLE_SYSFS_GROUP_VISIBLE() fs/sysfs/group.c | 4 +- include/linux/sysfs.h | 85 +++++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 84 insertions(+), 5 deletions(-) base-commit: 70317fd24b419091aa0a6dc3ea3ec7bb50c37c32

4 18

[PATCH] ALSA: hda: intel-sdw-acpi: fix usage of device_get_named_child_node()
by Pierre-Louis Bossart 29 Apr '24

29 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-sdw-acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index 5f60658c6051..d7417a40392b 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -45,6 +45,8 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) "intel-quirk-mask", &quirk_mask); + fwnode_handle_put(link); + if (quirk_mask & SDW_INTEL_QUIRK_MASK_BUS_DISABLE) return false; -- 2.40.1

2 3

[PATCH] ALSA: hda: intel-dsp-config: harden I2C/I2S codec detection
by Pierre-Louis Bossart 29 Apr '24

29 Apr '24

The SOF driver is selected whenever specific I2C/I2S HIDs are reported as 'present' in the ACPI DSDT. In some cases, an HID is reported but the hardware does not actually rely on I2C/I2S. This false positive leads to an invalid selection of the SOF driver and as a result an invalid topology is loaded. This patch hardens the detection with a check that the NHLT table is consistent with the report of an I2S-based codec in DSDT. This table should expose at least one SSP endpoint configured for an I2S-codec connection. Tested on Huawei Matebook D14 (NBLB-WAX9N) using an HDaudio codec with an invalid ES8336 ACPI HID reported: [ 7.858249] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 7.858312] snd_hda_intel 0000:00:1f.3: snd_intel_dsp_find_config: no valid SSP found for HID ESSX8336, skipped Reported-by: Mauro Carvalho Chehab <mchehab(a)kernel.org> Tested-by: Mauro Carvalho Chehab <mchehab(a)kernel.org> Closes: https://github.com/thesofproject/linux/issues/4934 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a384b922e4f..d1f6cdcf1866 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -557,9 +557,32 @@ static const struct config_entry *snd_intel_dsp_find_config if (table->codec_hid) { int i; - for (i = 0; i < table->codec_hid->num_codecs; i++) - if (acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + for (i = 0; i < table->codec_hid->num_codecs; i++) { + struct nhlt_acpi_table *nhlt; + bool ssp_found = false; + + if (!acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + continue; + + nhlt = intel_nhlt_init(&pci->dev); + if (!nhlt) { + dev_warn(&pci->dev, "%s: NHLT table not found, skipped HID %s\n", + __func__, table->codec_hid->codecs[i]); + continue; + } + + if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_SSP) && + intel_nhlt_ssp_endpoint_mask(nhlt, NHLT_DEVICE_I2S)) + ssp_found = true; + + intel_nhlt_free(nhlt); + + if (ssp_found) break; + + dev_warn(&pci->dev, "%s: no valid SSP found for HID %s, skipped\n", + __func__, table->codec_hid->codecs[i]); + } if (i == table->codec_hid->num_codecs) continue; } -- 2.40.1

2 1

ALSA: Digital Microphone
by Muni Sekhar 29 Apr '24

29 Apr '24

Hi all, We have a Digital Microphone block on a Xilinx FPGA board connected to x86 host CPU through PCIe, it basically emulates a Microphone sending data to DUT(device under test) in PDM format. Input to the Digital Microphone block is Signed 32-bit PCM sample. Xilinx FPGA code has logic to convert PCM samples to PDM with a Sigma-Delta Converter. Could you kindly recommend any specific ALSA driver within the Linux kernel that matches with my hardware setup. Additionally, if there are any relevant documentation or community discussions that you could point me to, it would be immensely helpful. -- Thanks, Sekhar

1 0

[PATCH] ALSA: usb-audio: Add sampling rates support for Mbox3
by ManuLinares 29 Apr '24

29 Apr '24

This adds support for all sample rates supported by the hardware, Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: ManuLinares <mbarriolinares(a)gmail.com> --- sound/usb/quirks-table.h | 37 +++++++++++------- sound/usb/quirks.c | 84 +++++++++++++++++++++++++++++++--------- 2 files changed, 88 insertions(+), 33 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8441cb..5228afbc00e3 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..279b80874f6a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; - if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + if (err < 0) + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); - if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + if (err < 0) + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,52 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + // Set rate only for one interface + //u8 iface = subs->data_endpoint->iface; + //if (iface != 2) return; + + u8 buffer[4] = {0}; + u32 new_rate = subs->data_endpoint->cur_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buffer, 4); + //u32 current_rate = le32_to_cpu(*(u32 *)buffer); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", le32_to_cpu(*(u32 *)buffer)); + if (le32_to_cpu(*(u32 *)buffer) == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed, current rate:%d == new rate:%d", + le32_to_cpu(*(u32 *)buffer), new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + __le32 set_rate = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &set_rate, 4); + + // Set clock source to Internal + buffer[0] = 0x01; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buffer[0], 1); + + // Check whether the change was successful + buffer[0] = 0; buffer[1] = 0; buffer[2] = 0; buffer[3] = 0; + snd_usb_ctl_msg(subs->dev, + usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buffer, 4); + //set_rate = *(int *)buffer; + set_rate = le32_to_cpu(*(u32 *)buffer); + if (new_rate != set_rate) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1798,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } -- 2.44.0

2 2

[PATCH v1] ALSA: ASoc/tas2781: fixed wrong loading calibrated data sequence
by Shenghao Ding 29 Apr '24

29 Apr '24

Calibrated data will be set to default after loading dsp config params, which will cause speaker protection work abnormally. Reload calibrated data after loading dsp config params. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v1: - Download calibrated data after loading the new dsp config params - call tasdevice_prmg_load instead of tasdevice_prmg_calibdata_load, it is unnecessary to load calibrated data after loading dsp program. Load it after loading dsp config params. - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading dsp program. - work in bypass-dsp mode even if no dsp fw loaded. --- include/sound/tas2781-dsp.h | 3 +- sound/soc/codecs/tas2781-fmwlib.c | 96 +++++++------------------------ sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 23 insertions(+), 80 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..14a4a254ad2a 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..c21bce6f09d6 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tasdevice-fmw.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -2210,21 +2210,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2247,9 +2235,24 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, if (tas_priv->tasdevice[i].is_loaderr == true) { status |= 1 << (i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2311,6 @@ int tasdevice_prmg_load(void *context, int prm_no) } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { -- 2.34.1

2 1

[PATCH] ASoC: intel: skylake: Add check for dma_set_max_seg_size
by Chen Ni 29 Apr '24

29 Apr '24

Add check for the return value of dma_set_max_seg_size() and return the error if it fails in order to catch the error. Fixes: c22a8086b384 ("ASoC: intel: skylake: Set max DMA segment size") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/skylake/skl.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 117125187793..8edc13795462 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -943,7 +943,9 @@ static int skl_first_init(struct hdac_bus *bus) /* allow 64bit DMA address if supported by H/W */ if (dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(64))) dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(32)); - dma_set_max_seg_size(bus->dev, UINT_MAX); + err = dma_set_max_seg_size(bus->dev, UINT_MAX); + if (err) + return err; /* initialize streams */ snd_hdac_ext_stream_init_all -- 2.25.1

1 0

[PATCH v3 00/23] ASoC: Replace dpcm_playback/capture to playback/capture_assertion
by Kuninori Morimoto 26 Apr '24

26 Apr '24

Hi Mark This is v3 patch-set When we use DPCM, we need to set dpcm_playback/capture flag. If these flag are set, soc_get_playback_capture() will check its availability, but non DPCM doesn't need such special flags. OTOH, it cares playback/capture_only flag. It is needed. This patch adds new assertion flag, and convert current dpcm_playback dpcm_capture flag and playback/capture_only flag to it. v2 -> v3 - re-order patches - add new playback/capture_assertion flag and covert both dpcm_xxx and xxx_only flag into it v1 -> v2 - based on latest ASoC branch - keep comment on Intel - tidyup patch title - tidyup DPCM BE warning output condition - Add new patch for Document Link: https://lore.kernel.org/r/87o7b353of.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87zfuesz8y.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (23): ASoC: soc-pcm: cleanup soc_get_playback_capture() ASoC: soc-pcm: indicate warning if DPCM BE Codec has no settings ASoC: soc-dai: remove snd_soc_dai_link_set_capabilities() ASoC: amd: Replace dpcm_playback/capture to playback/capture_assertion ASoC: fsl: Replace dpcm_playback/capture to playback/capture_assertion ASoC: sof: Replace dpcm_playback/capture to playback/capture_assertion ASoC: meson: Replace dpcm_playback/capture to playback/capture_assertion ASoC: Intel: Replace dpcm_playback/capture to playback/capture_assertion ASoC: mediatek: Replace dpcm_playback/capture to playback/capture_assertion ASoC: soc-core: Replace dpcm_playback/capture to playback/capture_assertion ASoC: soc-topology: Replace dpcm_playback/capture to playback/capture_assertion ASoC: soc-compress: Replace dpcm_playback/capture to playback/capture_assertion ASoC: Intel: avs: boards: Replace dpcm_playback/capture to playback/capture_assertion ASoC: ti: Replace playback/capture_only to playback/capture_assertion ASoC: amd: Replace playback/capture_only to playback/capture_assertion ASoC: fsl: Replace playback/capture_only to playback/capture_assertion ASoC: mxs: Replace playback/capture_only to playback/capture_assertion ASoC: atmel: Replace playback/capture_only to playback/capture_assertion ASoC: Intel: Replace playback/capture_only to playback/capture_assertion ASoC: samsung: Replace playback/capture_only to playback/capture_assertion ASoC: generic: Replace playback/capture_only to playback/capture_assertion ASoC: soc-pcm: remove dpcm_playback/capture and playback/capture_only ASoC: doc: Replace dpcm_playback/capture to playback/capture_assertion Documentation/sound/soc/dpcm.rst | 10 +- include/sound/simple_card_utils.h | 2 +- include/sound/soc-dai.h | 1 - include/sound/soc.h | 15 +- sound/soc/amd/acp-da7219-max98357a.c | 20 +-- sound/soc/amd/acp-es8336.c | 4 +- sound/soc/amd/acp/acp-mach-common.c | 34 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 8 +- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/renoir/acp3x-rn.c | 2 +- sound/soc/amd/vangogh/acp5x-mach.c | 14 +- sound/soc/amd/yc/acp6x-mach.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 20 +-- sound/soc/fsl/imx-audmix.c | 8 +- sound/soc/fsl/imx-card.c | 7 +- sound/soc/fsl/imx-hdmi.c | 12 +- sound/soc/fsl/imx-rpmsg.c | 10 +- sound/soc/fsl/imx-spdif.c | 10 +- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 10 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/generic/simple-card.c | 2 - sound/soc/intel/avs/boards/da7219.c | 4 +- sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 4 +- sound/soc/intel/avs/boards/hdaudio.c | 8 +- sound/soc/intel/avs/boards/i2s_test.c | 4 +- sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 4 +- sound/soc/intel/avs/boards/max98927.c | 4 +- sound/soc/intel/avs/boards/nau8825.c | 4 +- sound/soc/intel/avs/boards/rt274.c | 4 +- sound/soc/intel/avs/boards/rt286.c | 4 +- sound/soc/intel/avs/boards/rt298.c | 4 +- sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 4 +- sound/soc/intel/avs/boards/rt5682.c | 4 +- sound/soc/intel/avs/boards/ssm4567.c | 4 +- sound/soc/intel/boards/bdw-rt5650.c | 8 +- sound/soc/intel/boards/bdw-rt5677.c | 10 +- sound/soc/intel/boards/bdw_rt286.c | 14 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++--- sound/soc/intel/boards/bxt_rt298.c | 28 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +- sound/soc/intel/boards/bytcht_da7213.c | 10 +- sound/soc/intel/boards/bytcht_es8316.c | 10 +- sound/soc/intel/boards/bytcht_nocodec.c | 10 +- sound/soc/intel/boards/bytcr_rt5640.c | 10 +- sound/soc/intel/boards/bytcr_rt5651.c | 10 +- sound/soc/intel/boards/bytcr_wm5102.c | 10 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 10 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 10 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 10 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 10 +- sound/soc/intel/boards/ehl_rt5660.c | 16 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 32 ++--- sound/soc/intel/boards/hsw_rt5640.c | 14 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 28 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 60 ++++---- sound/soc/intel/boards/kbl_rt5660.c | 20 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 48 +++---- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 26 ++-- sound/soc/intel/boards/skl_hda_dsp_common.c | 18 +-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 28 ++-- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 30 ++-- sound/soc/intel/boards/skl_rt286.c | 28 ++-- sound/soc/intel/boards/sof_board_helpers.c | 18 +-- sound/soc/intel/boards/sof_es8336.c | 10 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_wm8804.c | 4 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 30 ++-- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 8 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 34 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 8 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 8 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 8 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 12 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 40 +++--- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 40 +++--- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 108 +++++++-------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 60 ++++---- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 86 ++++++------ sound/soc/mediatek/mt8195/mt8195-mt6359.c | 62 ++++----- sound/soc/meson/axg-card.c | 9 +- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/samsung/odroid.c | 12 +- sound/soc/soc-compress.c | 8 +- sound/soc/soc-core.c | 15 +- sound/soc/soc-dai.c | 38 ----- sound/soc/soc-pcm.c | 130 ++++++++++-------- sound/soc/soc-topology-test.c | 4 +- sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 +- sound/soc/ti/j721e-evm.c | 8 +- 100 files changed, 789 insertions(+), 832 deletions(-) -- 2.25.1

3 38

[Remove Mbox3 UCM] Request to revert pull 237
by GitHub issues - opened 26 Apr '24

26 Apr '24

alsa-project/alsa-ucm-conf issue #412 was opened from ManuLinares: https://github.com/alsa-project/alsa-ucm-conf/pull/237 The SplitPCM adds not only latency, https://github.com/alsa-project/alsa-ucm-conf/issues/238 but other general issues when streaming, etc. So I propose removing the UCM profile. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/412 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 00/25] virtio: store owner from modules with register_virtio_driver()
by Krzysztof Kozlowski 25 Apr '24

25 Apr '24

Changes in v2: - Three new patches: virtio mem+input+balloon - Minor commit msg adjustments - Add tags - Link to v1: https://lore.kernel.org/r/20240327-module-owner-virtio-v1-0-0feffab77d99@li… Merging ======= All further patches depend on the first virtio patch, therefore please ack and this should go via one tree: maybe virtio? Description =========== Modules registering driver with register_virtio_driver() often forget to set .owner field. Solve the problem by moving this task away from the drivers to the core virtio code, just like we did for platform_driver in commit 9447057eaff8 ("platform_device: use a macro instead of platform_driver_register"). Best regards, Krzysztof --- Krzysztof Kozlowski (25): virtio: store owner from modules with register_virtio_driver() virtio: balloon: drop owner assignment virtio: input: drop owner assignment virtio: mem: drop owner assignment um: virt-pci: drop owner assignment virtio_blk: drop owner assignment bluetooth: virtio: drop owner assignment hwrng: virtio: drop owner assignment virtio_console: drop owner assignment crypto: virtio - drop owner assignment firmware: arm_scmi: virtio: drop owner assignment gpio: virtio: drop owner assignment drm/virtio: drop owner assignment iommu: virtio: drop owner assignment misc: nsm: drop owner assignment net: caif: virtio: drop owner assignment net: virtio: drop owner assignment net: 9p: virtio: drop owner assignment vsock/virtio: drop owner assignment wifi: mac80211_hwsim: drop owner assignment nvdimm: virtio_pmem: drop owner assignment rpmsg: virtio: drop owner assignment scsi: virtio: drop owner assignment fuse: virtio: drop owner assignment sound: virtio: drop owner assignment Documentation/driver-api/virtio/writing_virtio_drivers.rst | 1 - arch/um/drivers/virt-pci.c | 1 - drivers/block/virtio_blk.c | 1 - drivers/bluetooth/virtio_bt.c | 1 - drivers/char/hw_random/virtio-rng.c | 1 - drivers/char/virtio_console.c | 2 -- drivers/crypto/virtio/virtio_crypto_core.c | 1 - drivers/firmware/arm_scmi/virtio.c | 1 - drivers/gpio/gpio-virtio.c | 1 - drivers/gpu/drm/virtio/virtgpu_drv.c | 1 - drivers/iommu/virtio-iommu.c | 1 - drivers/misc/nsm.c | 1 - drivers/net/caif/caif_virtio.c | 1 - drivers/net/virtio_net.c | 1 - drivers/net/wireless/virtual/mac80211_hwsim.c | 1 - drivers/nvdimm/virtio_pmem.c | 1 - drivers/rpmsg/virtio_rpmsg_bus.c | 1 - drivers/scsi/virtio_scsi.c | 1 - drivers/virtio/virtio.c | 6 ++++-- drivers/virtio/virtio_balloon.c | 1 - drivers/virtio/virtio_input.c | 1 - drivers/virtio/virtio_mem.c | 1 - fs/fuse/virtio_fs.c | 1 - include/linux/virtio.h | 7 +++++-- net/9p/trans_virtio.c | 1 - net/vmw_vsock/virtio_transport.c | 1 - sound/virtio/virtio_card.c | 1 - 27 files changed, 9 insertions(+), 30 deletions(-) --- base-commit: 7fdcff3312e16ba8d1419f8a18f465c5cc235ecf change-id: 20240327-module-owner-virtio-546763b3ca22 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

12 40

[PATCH] ALSA: hda/ASOC tas2781: align the path of firmware
by Aaron Ma 25 Apr '24

25 Apr '24

TI upstream the files in ti/tas2781 for linux-firmware. Make the driver find the correct files. Signed-off-by: Aaron Ma <aaron.ma(a)canonical.com> --- include/sound/tas2781.h | 1 + sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 2 +- 3 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..72119cc32c0f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -21,6 +21,7 @@ /* version number */ #define TAS2781_DRV_VER 1 #define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_FIRMWARE_ROOT "ti/tas2781/" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 75f7674c66ee..72dc1469655f 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -649,7 +649,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; - scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + scnprintf(tas_priv->coef_binaryname, 64, "%sTAS2XXX%04X.bin", TAS2781_FIRMWARE_ROOT, codec->core.subsystem_id & 0xffff); ret = tasdevice_dsp_parser(tas_priv); if (ret) { diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 3aa81514dad7..96fe1b855ec9 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -277,7 +277,7 @@ int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, */ mutex_lock(&tas_priv->codec_lock); - scnprintf(tas_priv->rca_binaryname, 64, "%sRCA%d.bin", + scnprintf(tas_priv->rca_binaryname, 64, "%s%sRCA%d.bin", TAS2781_FIRMWARE_ROOT, tas_priv->dev_name, tas_priv->ndev); crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); tas_priv->codec = codec; -- 2.34.1

2 5

[PATCH v4 68/71] ASoC: Intel: avs: es8336: Switch to new Intel CPU model defines
by Tony Luck 25 Apr '24

25 Apr '24

New CPU #defines encode vendor and family as well as model. Signed-off-by: Tony Luck <tony.luck(a)intel.com> --- sound/soc/intel/avs/boards/es8336.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 5c90a6007577..3d5d4685a668 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -18,7 +18,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-acpi.h> -#include <asm/intel-family.h> +#include <asm/cpu_device_id.h> #include "../utils.h" #define ES8336_CODEC_DAI "ES8316 HiFi" @@ -153,9 +153,9 @@ static int avs_es8336_hw_params(struct snd_pcm_substream *substream, int clk_freq; int ret; - switch (boot_cpu_data.x86_model) { - case INTEL_FAM6_KABYLAKE_L: - case INTEL_FAM6_KABYLAKE: + switch (boot_cpu_data.x86_vfm) { + case INTEL_KABYLAKE_L: + case INTEL_KABYLAKE: clk_freq = 24000000; break; default: -- 2.44.0

4 4

[PATCH v2] ASoC: dt-bindings: fsl,ssi: Convert to YAML
by Shengjiu Wang 25 Apr '24

25 Apr '24

Convert the fsl,ssi binding to YAML. Add below compatible strings which were not listed in document: fsl,imx50-ssi fsl,imx53-ssi fsl,imx25-ssi fsl,imx27-ssi fsl,imx6q-ssi fsl,imx6sl-ssi fsl,imx6sx-ssi Add below fsl,mode strings which were not listed. i2s-slave i2s-master lj-slave lj-master rj-slave rj-master Add 'ac97-gpios' property which were not listed. Then dtbs_check can pass. And remove the 'codec' description which should be in the 'codec' binding doc. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - change fallback string to const. - add dai-common.yaml - add ac97-gpios property .../devicetree/bindings/sound/fsl,ssi.txt | 87 -------- .../devicetree/bindings/sound/fsl,ssi.yaml | 192 ++++++++++++++++++ 2 files changed, 192 insertions(+), 87 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt deleted file mode 100644 index 7e15a85cecd2..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ /dev/null @@ -1,87 +0,0 @@ -Freescale Synchronous Serial Interface - -The SSI is a serial device that communicates with audio codecs. It can -be programmed in AC97, I2S, left-justified, or right-justified modes. - -Required properties: -- compatible: Compatible list, should contain one of the following - compatibles: - fsl,mpc8610-ssi - fsl,imx51-ssi - fsl,imx35-ssi - fsl,imx21-ssi -- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on. -- reg: Offset and length of the register set for the device. -- interrupts: <a b> where a is the interrupt number and b is a - field that represents an encoding of the sense and - level information for the interrupt. This should be - encoded based on the information in section 2) - depending on the type of interrupt controller you - have. -- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. - This number is the maximum allowed value for SFCSR[TFWM0]. - - clocks: "ipg" - Required clock for the SSI unit - "baud" - Required clock for SSI master mode. Otherwise this - clock is not used - -Required are also ac97 link bindings if ac97 is used. See -Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary -bindings. - -Optional properties: -- codec-handle: Phandle to a 'codec' node that defines an audio - codec connected to this SSI. This node is typically - a child of an I2C or other control node. -- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to - filter the codec stream. This is necessary for some boards - where an incompatible codec is connected to this SSI, e.g. - on pca100 and pcm043. -- dmas: Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. -- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq - is not defined. -- fsl,mode: The operating mode for the AC97 interface only. - "ac97-slave" - AC97 mode, SSI is clock slave - "ac97-master" - AC97 mode, SSI is clock master -- fsl,ssi-asynchronous: - If specified, the SSI is to be programmed in asynchronous - mode. In this mode, pins SRCK, STCK, SRFS, and STFS must - all be connected to valid signals. In synchronous mode, - SRCK and SRFS are ignored. Asynchronous mode allows - playback and capture to use different sample sizes and - sample rates. Some drivers may require that SRCK and STCK - be connected together, and SRFS and STFS be connected - together. This would still allow different sample sizes, - but not different sample rates. -- fsl,playback-dma: Phandle to a node for the DMA channel to use for - playback of audio. This is typically dictated by SOC - design. See the notes below. - Only used on Power Architecture. -- fsl,capture-dma: Phandle to a node for the DMA channel to use for - capture (recording) of audio. This is typically dictated - by SOC design. See the notes below. - Only used on Power Architecture. - -Child 'codec' node required properties: -- compatible: Compatible list, contains the name of the codec - -Child 'codec' node optional properties: -- clock-frequency: The frequency of the input clock, which typically comes - from an on-board dedicated oscillator. - -Notes on fsl,playback-dma and fsl,capture-dma: - -On SOCs that have an SSI, specific DMA channels are hard-wired for playback -and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for -playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for -playback and DMA channel 3 for capture. The developer can choose which -DMA controller to use, but the channels themselves are hard-wired. The -purpose of these two properties is to represent this hardware design. - -The device tree nodes for the DMA channels that are referenced by -"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with -"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g. -"fsl,mpc8610-dma-channel") can remain. If these nodes are left as -"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA -drivers (fsldma) will attempt to use them, and it will conflict with the -sound drivers. diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.yaml b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml new file mode 100644 index 000000000000..d22911b0e9ef --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml @@ -0,0 +1,192 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Serial Interface + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: + Notes on fsl,playback-dma and fsl,capture-dma + On SOCs that have an SSI, specific DMA channels are hard-wired for playback + and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for + playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for + playback and DMA channel 3 for capture. The developer can choose which + DMA controller to use, but the channels themselves are hard-wired. The + purpose of these two properties is to represent this hardware design. + + The device tree nodes for the DMA channels that are referenced by + "fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with + "fsl,ssi-dma-channel". The SOC-specific compatible string (e.g. + "fsl,mpc8610-dma-channel") can remain. If these nodes are left as + "fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA + drivers (fsldma) will attempt to use them, and it will conflict with the + sound drivers. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx50-ssi + - fsl,imx53-ssi + - enum: + - fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx25-ssi + - fsl,imx27-ssi + - fsl,imx35-ssi + - fsl,imx51-ssi + - fsl,imx6q-ssi + - fsl,imx6sl-ssi + - fsl,imx6sx-ssi + - enum: + - fsl,imx21-ssi + - fsl,imx51-ssi + - items: + - const: fsl,mpc8610-ssi + + cell-index: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: The SSI index + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + + clocks: + items: + - description: The ipg clock for register access + - description: clock for SSI master mode + minItems: 1 + + clock-names: + items: + - const: ipg + - const: baud + minItems: 1 + + dmas: + oneOf: + - items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + - items: + - description: DMA controller phandle and request line for RX0 + - description: DMA controller phandle and request line for TX0 + - description: DMA controller phandle and request line for RX1 + - description: DMA controller phandle and request line for TX1 + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - items: + - const: rx0 + - const: tx0 + - const: rx1 + - const: tx1 + + codec-handle: + $ref: /schemas/types.yaml#/definitions/phandle + description: + Phandle to a 'codec' node that defines an audio + codec connected to this SSI. This node is typically + a child of an I2C or other control node. + + fsl,fiq-stream-filter: + type: boolean + description: + Disabled DMA and use FIQ instead to filter the codec stream. + This is necessary for some boards where an incompatible codec + is connected to this SSI, e.g. on pca100 and pcm043. + + fsl,mode: + $ref: /schemas/types.yaml#/definitions/string + enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master, + lj-slave, lj-master, rj-slave, rj-master ] + description: | + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - Left justified mode, SSI is clock slave + "lj-master" - Left justified mode, SSI is clock master + "rj-slave" - Right justified mode, SSI is clock slave + "rj-master" - Right justified mode, SSI is clock master + + fsl,ssi-asynchronous: + type: boolean + description: If specified, the SSI is to be programmed in asynchronous + mode. In this mode, pins SRCK, STCK, SRFS, and STFS must + all be connected to valid signals. In synchronous mode, + SRCK and SRFS are ignored. Asynchronous mode allows + playback and capture to use different sample sizes and + sample rates. Some drivers may require that SRCK and STCK + be connected together, and SRFS and STFS be connected + together. This would still allow different sample sizes, + but not different sample rates. + + fsl,playback-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. Only used on Power Architecture. + + fsl,capture-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. Only used on Power Architecture. + + ac97-gpios: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: Please refer to soc-ac97link.txt + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +required: + - compatible + - reg + - interrupts + - fsl,fifo-depth + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx6qdl-clock.h> + ssi1: ssi@2028000 { + compatible = "fsl,imx6q-ssi", + "fsl,imx51-ssi"; + reg = <0x02028000 0x4000>; + interrupts = <GIC_SPI 46 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX6QDL_CLK_SSI1_IPG>, + <&clks IMX6QDL_CLK_SSI1>; + clock-names = "ipg", "baud"; + dmas = <&sdma 37 1 0>, <&sdma 38 1 0>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + fsl,fifo-depth = <15>; + }; -- 2.34.1

3 2

[PATCH v9 00/13] Introduce STM32 Firewall framework
by Gatien Chevallier 24 Apr '24

24 Apr '24

Introduce STM32 Firewall framework for STM32MP1x and STM32MP2x platforms. STM32MP1x(ETZPC) and STM32MP2x(RIFSC) Firewall controllers register to the framework to offer firewall services such as access granting. This series of patches is a new approach on the previous STM32 system bus, history is available here: https://lore.kernel.org/lkml/20230127164040.1047583/ The need for such framework arises from the fact that there are now multiple hardware firewalls implemented across multiple products. Drivers are shared between different products, using the same code. When it comes to firewalls, the purpose mostly stays the same: Protect hardware resources. But the implementation differs, and there are multiple types of firewalls: peripheral, memory, ... Some hardware firewall controllers such as the RIFSC implemented on STM32MP2x platforms may require to take ownership of a resource before being able to use it, hence the requirement for firewall services to take/release the ownership of such resources. On the other hand, hardware firewall configurations are becoming more and more complex. These mecanisms prevent platform crashes or other firewall-related incoveniences by denying access to some resources. The stm32 firewall framework offers an API that is defined in firewall controllers drivers to best fit the specificity of each firewall. For every peripherals protected by either the ETZPC or the RIFSC, the firewall framework checks the firewall controlelr registers to see if the peripheral's access is granted to the Linux kernel. If not, the peripheral is configured as secure, the node is marked populated, so that the driver is not probed for that device. The firewall framework relies on the access-controller device tree binding. It is used by peripherals to reference a domain access controller. In this case a firewall controller. The bus uses the ID referenced by the access-controller property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. The STM32 ETZPC device is responsible for filtering accesses based on security level, or co-processor isolation for any resource connected to it. The RIFSC is responsible for filtering accesses based on Compartment ID / security level / privilege level for any resource connected to it. STM32MP13/15/25 SoC device tree files are updated in this series to implement this mecanism. Changes in V9: - Added Rob's reviews tags on bindings/of files - Added "simple-bus" to RIFSC/ETZPC compatible item list Changes in V8: - Add missing "simple-bus" compatible in STM32MP13/25 SoC device tree files for the ETZPC/RIFSC nodes - Add missing dependency on OF for OF_DYNAMIC that is selected by STM32_FIREWALL Changes in V7: - Separate indentation changes from access-controllers changes in the device tree file commits - Select OF_DYNAMIC when STM32_FIREWALL is set in order to use of_detach_node() in the firewall framework - Handle previously non-present RNG and HASH nodes in the STM32MP13 device tree file Changes in V6: - Rename access-controller to access-controllers - Remove access-controller-provider - Update device trees and other bindings accordingly - Rework ETZPC/RIFSC bindings to define what access-controllers cells contain inside #access-controller-cells - Some other minor fixes Changes in V5: - Integrate and rework the "feature-domains" binding patch in this patchset. The binding is renamed to "access-controller" - Rename every feature-domain* reference to access-control* ones - Correct loop bug and missing select STM32_FIREWALL in 32-bit platform Kconfig Changes in V4: - Fix typo in commit message and YAML check errors in "dt-bindings: Document common device controller bindings" Note: This patch should be ignored as stated in the cover letter. I've done this to avoid errors on this series of patch - Correct code syntax/style issues reported by Simon Horman - Added Jonathan's tag for IIO on the treewide patch Changes in V3: Change incorrect ordering for bindings commits leading to an error while running "make DT_CHECKER_FLAGS=-m dt_binding_check" Changes in V2: generic: - Add fw_devlink dependency for "feature-domains" property. bindings: - Corrected YAMLS errors highlighted by Rob's robot - Firewall controllers YAMLs no longer define the maxItems for the "feature-domains" property - Renamed st,stm32-rifsc.yaml to st,stm32mp25-rifsc.yaml - Fix examples in YAML files - Change feature-domains maxItems to 2 in firewall consumer files as there should not be more than 2 entries for now - Declare "feature-domain-names" as an optional property for firewall controllers child nodes. - Add missing "feature-domains" property declaration in bosch,m_can.yaml and st,stm32-cryp.yaml files firewall framework: - Support multiple entries for "feature-domains" property - Better handle the device-tree parsing using phandle+args APIs - Remove "resource firewall" type - Add a field for the name of the firewall entry - Fix licenses RIFSC: - Add controller name - Driver is now a module_platform_driver - Fix license ETZPC: - Add controller name - Driver is now a module_platform_driver - Fix license Device trees: - Fix rifsc node name - Move the "ranges" property under the "feature-domains" one Gatien Chevallier (12): dt-bindings: treewide: add access-controllers description dt-bindings: bus: document RIFSC dt-bindings: bus: document ETZPC firewall: introduce stm32_firewall framework of: property: fw_devlink: Add support for "access-controller" bus: rifsc: introduce RIFSC firewall controller driver arm64: dts: st: add RIFSC as an access controller for STM32MP25x boards bus: etzpc: introduce ETZPC firewall controller driver ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: put ETZPC as an access controller for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards ARM: dts: stm32: put ETZPC as an access controller for STM32MP13x boards Oleksii Moisieiev (1): dt-bindings: document generic access controllers .../access-controllers.yaml | 84 + .../bindings/bus/st,stm32-etzpc.yaml | 96 + .../bindings/bus/st,stm32mp25-rifsc.yaml | 105 + .../bindings/crypto/st,stm32-cryp.yaml | 4 + .../bindings/crypto/st,stm32-hash.yaml | 4 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 4 + .../bindings/dma/st,stm32-dmamux.yaml | 4 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 4 + .../bindings/iio/adc/st,stm32-adc.yaml | 4 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 + .../bindings/iio/dac/st,stm32-dac.yaml | 4 + .../bindings/media/cec/st,stm32-cec.yaml | 4 + .../bindings/media/st,stm32-dcmi.yaml | 4 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 4 + .../bindings/mfd/st,stm32-lptimer.yaml | 4 + .../bindings/mfd/st,stm32-timers.yaml | 4 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 4 + .../bindings/net/can/bosch,m_can.yaml | 4 + .../devicetree/bindings/net/stm32-dwmac.yaml | 4 + .../bindings/phy/phy-stm32-usbphyc.yaml | 4 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 4 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 4 + .../bindings/serial/st,stm32-uart.yaml | 4 + .../bindings/sound/st,stm32-i2s.yaml | 4 + .../bindings/sound/st,stm32-sai.yaml | 4 + .../bindings/sound/st,stm32-spdifrx.yaml | 4 + .../bindings/spi/st,stm32-qspi.yaml | 4 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 4 + .../devicetree/bindings/usb/dwc2.yaml | 4 + MAINTAINERS | 7 + arch/arm/boot/dts/st/stm32mp131.dtsi | 1063 ++++--- arch/arm/boot/dts/st/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/st/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp13xf.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp151.dtsi | 2756 +++++++++-------- arch/arm/boot/dts/st/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/st/stm32mp15xc.dtsi | 19 +- arch/arm/mach-stm32/Kconfig | 1 + arch/arm64/Kconfig.platforms | 1 + arch/arm64/boot/dts/st/stm32mp251.dtsi | 7 +- drivers/bus/Kconfig | 10 + drivers/bus/Makefile | 1 + drivers/bus/stm32_etzpc.c | 141 + drivers/bus/stm32_firewall.c | 294 ++ drivers/bus/stm32_firewall.h | 83 + drivers/bus/stm32_rifsc.c | 252 ++ drivers/of/property.c | 2 + include/linux/bus/stm32_firewall_device.h | 141 + 48 files changed, 3370 insertions(+), 1938 deletions(-) create mode 100644 Documentation/devicetree/bindings/access-controllers/access-controllers.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-etzpc.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32mp25-rifsc.yaml create mode 100644 drivers/bus/stm32_etzpc.c create mode 100644 drivers/bus/stm32_firewall.c create mode 100644 drivers/bus/stm32_firewall.h create mode 100644 drivers/bus/stm32_rifsc.c create mode 100644 include/linux/bus/stm32_firewall_device.h -- 2.35.3

5 19

[PATCH alsa-ucm-conf] sof-soundwire: Add basic support for cs42l43's speaker
by Charles Keepax 24 Apr '24

24 Apr '24

Previous support was added for the mics and headset on cs42l43, expand this support with additional configuration for the speaker output. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/sof-soundwire/cs42l43-spk.conf | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) create mode 100644 ucm2/sof-soundwire/cs42l43-spk.conf diff --git a/ucm2/sof-soundwire/cs42l43-spk.conf b/ucm2/sof-soundwire/cs42l43-spk.conf new file mode 100644 index 0000000..73f1299 --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43-spk.conf @@ -0,0 +1,25 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Speaker" { + Comment "Speaker" + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='cs42l43 Speaker L Input 1' 'DP5RX1'" + cset "name='cs42l43 Speaker R Input 1' 'DP5RX2'" + ] + + DisableSequence [ + cset "name='cs42l43 Speaker L Input 1' 'None'" + cset "name='cs42l43 Speaker R Input 1' 'None'" + ] + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId},0" + PlaybackMixerElem "cs42l43 Speaker Digital" + } +} -- 2.39.2

2 1

UCM Path condition type does not support variables as file path
by GitHub issues - opened 24 Apr '24

24 Apr '24

alsa-project/alsa-lib issue #395 was opened from singalsu: To control audio processing in SOF I want to do do this: ``` Define.SpeakerIirBlob "${ConfTopDir}/sof/ipc4/eq_iir/highpass_100hz_0db_48khz.blob" Define.SpeakerFirBlob "${ConfTopDir}/sof/ipc4/eq_fir/pass.blob" Define.SpeakerDrcBlob "${ConfTopDir}/sof/ipc4/drc/spk_default.blob" Define.HeadphoneIirBlob "${ConfTopDir}/sof/ipc4/eq_iir/pass.blob" Define.HeadphoneFirBlob "${ConfTopDir}/sof/ipc4/eq_fir/pass.blob" Define.ProductConfigSOF "/sof/product_configs/${CardLongName}.conf" Define.FullPathToIt "${ConfTopDir}${var:ProductConfigSOF}" If.HasProductConfig { Condition { Type Path Mode read Path "${var:FullPathToIt}" } True { Include.product-config.File "${var:ProductConfigSOF}" } } ``` However if I replace line as "Path "/usr/share/alsa/ucm2/sof/product_configs/ACME-2000--foobar.conf" (product name obfuscated for confidentiality) the True condition is achieved. It looks like Path is not evaluating variables. This feature was added in commit 2f57b404b1c1c376316ca0c6bb5e789e2066298f where the usage is explained. Issue URL : https://github.com/alsa-project/alsa-lib/issues/395 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 0/2] Add support for Lenovo Thinkbook 13X
by Stefan Binding 24 Apr '24

24 Apr '24

This laptop does not contain _DSD required to support CS35L41 HDA. To workaround this, add support for this laptop into the configuration table in cs35l41_hda_property.c. Also add the required entry into patch_realtek for this SSID. Stefan Binding (2): ALSA: hda: cs35l41: Support Lenovo 13X laptop without _DSD ALSA: hda/realtek: Add quirks for Lenovo 13X sound/pci/hda/cs35l41_hda_property.c | 4 ++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 6 insertions(+) -- 2.34.1

2 3

[PATCH] ASoC: rt715-sdca: volume step modification
by Jack Yu 24 Apr '24

24 Apr '24

Volume step (dB/step) modification to fix format error which shown in amixer control. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715-sdca.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 3a6dfe1f6c48..7e10fd913812 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -316,7 +316,7 @@ static int rt715_sdca_set_amp_gain_8ch_get(struct snd_kcontrol *kcontrol, return 0; } -static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, @@ -477,7 +477,7 @@ static const struct snd_kcontrol_new rt715_sdca_snd_controls[] = { RT715_SDCA_FU_VOL_CTRL, CH_01), SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC7_27_VOL, RT715_SDCA_FU_VOL_CTRL, CH_02), - 0x2f, 0x7f, 0, + 0x2f, 0x3f, 0, rt715_sdca_set_amp_gain_get, rt715_sdca_set_amp_gain_put, in_vol_tlv), RT715_SDCA_EXT_TLV("FU02 Capture Volume", @@ -485,13 +485,13 @@ static const struct snd_kcontrol_new rt715_sdca_snd_controls[] = { RT715_SDCA_FU_VOL_CTRL, CH_01), rt715_sdca_set_amp_gain_4ch_get, rt715_sdca_set_amp_gain_4ch_put, - in_vol_tlv, 4, 0x7f), + in_vol_tlv, 4, 0x3f), RT715_SDCA_EXT_TLV("FU06 Capture Volume", SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC10_11_VOL, RT715_SDCA_FU_VOL_CTRL, CH_01), rt715_sdca_set_amp_gain_4ch_get, rt715_sdca_set_amp_gain_4ch_put, - in_vol_tlv, 4, 0x7f), + in_vol_tlv, 4, 0x3f), /* MIC Boost Control */ RT715_SDCA_BOOST_EXT_TLV("FU0E Boost", SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_DMIC_GAIN_EN, -- 2.34.1

2 1

[PATCH v19 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 24 Apr '24

24 Apr '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Requesting to see if we can get some Acked-By tags, and merge on usb-next. Changelog -------------------------------------------- Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (39): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 439 ++++ drivers/usb/host/xhci.c | 48 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 182 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6306 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 43

Unhandled asynchronous SIGIO
by GitHub issues - opened 23 Apr '24

23 Apr '24

alsa-project/alsa-lib issue #394 was opened from jansenmd: When using `libasound.so` on embedded systems, I've come across several instances where an unhandled `SIGIO` is detected. This can occur when closing a hardware microphone device that was set up to use asynchronous calls backs via `snd_async_add_pcm_handler`. Example: when opening a hardware device with asynchronous callbacks: * Sets up a signal handler for `SIGIO` via `sigaction`. * Enable generation of signals from the `fd` by setting the `O_ASYNC` flag: `fcntl(fd, F_SETFL, flags | O_ASYNC)` * Sets the signal to be generated: `fcntl(fd, F_SETSIG, (long)SIGIO)` Now, when data is available to be read from the `fd`, `SIGIO` should be generated and handled by the signal handler set in `sigaction`. That appears to work fine. The current teardown process looks like this: * Clear/reset the signal handler via `sigaction`. * Disable generation of signals from the `fd` by clearing the `O_ASYNC` flag: `fcntl(fd, F_SETFL, flags & ~O_ASYNC)` * Close the `fd`. This teardown process seems to set up a race condition where an unhandled `SIGIO` can be generated between unregistering the signal handler and actually stopping any more signals from occurring. Since the default action for `SIGIO` is termination, this can cause an application to unexpectedly terminate. Issue URL : https://github.com/alsa-project/alsa-lib/issues/394 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 0/2] ASoC: qcom: add support for qcm6490 machine
by Mohammad Rafi Shaik 23 Apr '24

23 Apr '24

This patchset adds support for QCM6490 SoC machine driver. Mohammad Rafi Shaik (2): ASoC: dt-bindings: qcom,qcm6490: Add qcm6490 snd qcs6490 sound card ASoC: qcom: qcm6490: Add machine driver for qcm6490 .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 13 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qcm6490.c | 173 ++++++++++++++++++ 4 files changed, 190 insertions(+) create mode 100644 sound/soc/qcom/qcm6490.c -- 2.25.1

5 11

Missed review / mailing list submission?
by Andy Shevchenko 23 Apr '24

23 Apr '24

https://git.kernel.org/pub/scm/linux/kernel/git/broonie/ci.git/commit/?h=as… I can't find this in the mailing list. Moreover, it has an obstacle to be applied, i.e. as per https://elixir.bootlin.com/linux/v6.9-rc5/source/include/linux/gpio.h#L5 the gpio.h must not be included in a new code and brief looking into that driver shows that legacy APIs are being used without _any_ good reason. This has to be fixed (by the author). -- With Best Regards, Andy Shevchenko

2 4

[PATCH v4 0/7] Patches to update for rockchip pdm
by Sugar Zhang 23 Apr '24

23 Apr '24

These patches fixup or update for rockchip pdm. Changes in v4: - Acked by Rob Herring Changes in v3: - Fix property 'path-map' suggested by Rob Herring. Changes in v2: - Fix yamllint errors. Sugar Zhang (7): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml .../devicetree/bindings/sound/rockchip,pdm.txt | 46 -------- .../devicetree/bindings/sound/rockchip,pdm.yaml | 120 +++++++++++++++++++++ sound/soc/rockchip/rockchip_pdm.c | 112 +++++++++++++++++-- sound/soc/rockchip/rockchip_pdm.h | 6 ++ 4 files changed, 232 insertions(+), 52 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml -- 2.7.4

3 9

[PATCH] ASoC: cs35l41: Delete unnecessary condition in cs35l41_pcm_hw_params()
by Dan Carpenter 23 Apr '24

23 Apr '24

This code returns -EINVAL if "i" is out of bounds a few lines earlier. Delete this unnecessary check and pull the code in a tab. Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/cs35l41.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 2799ccd6b5c7..d3da6a9f7f2b 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -772,10 +772,9 @@ static int cs35l41_pcm_hw_params(struct snd_pcm_substream *substream, asp_wl = params_width(params); - if (i < ARRAY_SIZE(cs35l41_fs_rates)) - regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL, - CS35L41_GLOBAL_FS_MASK, - cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT); + regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL, + CS35L41_GLOBAL_FS_MASK, + cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(cs35l41->regmap, CS35L41_SP_FORMAT, -- 2.43.0

2 1

conf: aliases: hdmi: Include unconditionally the pcm/hdmi.conf
by GitHub pull_request - opened 23 Apr '24

23 Apr '24

alsa-project/alsa-lib pull request #393 was opened from ujfalusi: The hdmi.conf contains the high level macro to be used by cards to create the hdmi: device. Instead of including it in different config files, include it in the main aliases.conf and remove it's inclusion by other config files. This change is needed to add support for the hdmi: device mapping via UCM. Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Request URL : https://github.com/alsa-project/alsa-lib/pull/393 Patch URL : https://github.com/alsa-project/alsa-lib/pull/393.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: cs35l56: Avoid static analysis warning of uninitialised variable
by Richard Fitzgerald 23 Apr '24

23 Apr '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> Static checkers complain that the silicon_uid variable passed by pointer to cs35l56_read_silicon_uid() could later be used uninitialised when calling cs_amp_get_efi_calibration_data(). cs35l56_read_silicon_uid() must have succeeded to call cs_amp_get_efi_calibration_data() and that would have populated the variable. However, initialise the value so we are not haunted by it forevermore. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 3aa39f5923eb..8af89a263594 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -707,7 +707,7 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, SND_SOC_CS35L56_SHARED); int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base) { - u64 silicon_uid; + u64 silicon_uid = 0; int ret; /* Driver can't apply calibration to a secured part, so skip */ -- 2.39.2

2 1

CM106L, CM6206, CM6206-LX
by James Lehman 22 Apr '24

22 Apr '24

Hello everyone. I have been using CMedia USB sound devices to make laser control signals for many years. I have sound devices with CM106L, CM6206 and CM6206-LX. Some are populated for 6 channels and some for 8. The CM106L chip works perfectly. I can send it a stream of 16-bit data that uses the full range of -32767 to +32767 and at 100% volume in either alsamixer or Pulseaudio Volume Control, I get a perfect image. https://laserboy.org/piDAC/perfect_circle.jpg If I use a device with a CM6206, I get a very near perfect image (only clipped a tiny bit). But If I use a CM6206-LX chip, it clips hard. https://laserboy.org/piDAC/CM6206-LX_100.jpg Even if I reduce the volume to 50%, it just seems to reduce the clipped signals. https://laserboy.org/piDAC/CM6206-LX_50.jpg I'm not sure what is going on here, but it seems like there is some kind of internal digital gain of more than 1.0. I also notice that lsusb shows exactly the same device ID for both the CM6206 and the CM6206-LX. .... 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device And what's really weird is that a genuine CM106L shows up as something else! .... 0d8c:0006 C-Media Electronics, Inc. Storm HP-USB500 5.1 Headset Is there something I can do in alsa config to fix this issue? The CM106L is long since out of production and it seems like any device that uses the CM6206 might have the LX version (or not). There's no way to know until I get them and open them up. I'm using this on an Orange Pi Zero 3 4gb running dietpi with the 6.6.16-current-sunxi64 kernel, but that doesn't seem to matter. I get the same results any other way I try it. The LX version of this chip clips the same way in Windows too! Thanks for your time. James.

1 0

[tiwai-sound:for-next 34/51] sound/usb/mixer_scarlett2.c:3690 scarlett2_ag_target_ctl_get() error: uninitialized symbol 'err'.
by Dan Carpenter 22 Apr '24

22 Apr '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 32cb23a0f911317cdb5030035e49a204aa86fef5 commit: e30ea5340c25d41484d551f61d8c93106989019b [34/51] ALSA: scarlett2: Add autogain target controls config: csky-randconfig-r081-20240420 (https://download.01.org/0day-ci/archive/20240420/202404201509.mWdaL3xF-lkp@…) compiler: csky-linux-gcc (GCC) 13.2.0 If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> | Closes: https://lore.kernel.org/r/202404201509.mWdaL3xF-lkp@intel.com/ New smatch warnings: sound/usb/mixer_scarlett2.c:3690 scarlett2_ag_target_ctl_get() error: uninitialized symbol 'err'. vim +/err +3690 sound/usb/mixer_scarlett2.c e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3665 static int scarlett2_ag_target_ctl_get( e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3666 struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3667 { e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3668 struct usb_mixer_elem_info *elem = kctl->private_data; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3669 struct usb_mixer_interface *mixer = elem->head.mixer; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3670 struct scarlett2_data *private = mixer->private_data; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3671 int err; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3672 e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3673 mutex_lock(&private->data_mutex); e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3674 e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3675 if (private->hwdep_in_use) { e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3676 err = -EBUSY; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3677 goto unlock; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3678 } e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3679 e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3680 if (private->autogain_updated) { e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3681 err = scarlett2_update_autogain(mixer); e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3682 if (err < 0) e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3683 goto unlock; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3684 } err uninitialized on else path. e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3685 e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3686 ucontrol->value.integer.value[0] = private->ag_targets[elem->control]; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3687 e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3688 unlock: e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3689 mutex_unlock(&private->data_mutex); e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 @3690 return err; e30ea5340c25d41 Geoffrey D. Bennett 2024-03-13 3691 } -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

snd_hda_intel with buffer_size set to 32768 sounds horrible. Kernel driver bug?
by Jaime 22 Apr '24

22 Apr '24

Fresh Debian Bookworm (v12) installation with alsa but no sound server. aplay with no .asoundrc sounds great (default buffer_size is 16384). If I force the buffer_size to 32768 ([1]), then aplay sounds horrible ([2]). Is this a kernel driver bug? (If yes, I'll log it at https://bugzilla.kernel.org) I discovered this bug because pipewire sets the buffer_size to 32768 hence it triggers this bug - I've logged a bug against pipewire at [3]. Output from alsa-info at [4]. TIA. [1] To set aplay's buffer_size to 32768, I use the following ~/.asoundrc: pcm.!default { type plug slave { pcm { type dmix ipc_key 1234 slave { period_size 1024 buffer_size 32768 pcm "hw:0" format S32_LE } } } } [2] Upload of horrible sound at https://audio.com/jt-1796478503182974/audio/alsaonlybuf-32768nopipewire [3] https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3966 [4] http://alsa-project.org/db/?f=5dbfb02c81bb1523505c725a189b6f0133f2b3d7

1 0

[PATCH v2 0/8] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 22 Apr '24

22 Apr '24

This patchset adds support for Qualcomm WCD937X codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v8: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (8): ASoC: dt-bindings: wcd937x: add bindings for wcd937x ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: dt-bindings: wcd937x-sdw: add bindings for wcd937x-sdw ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 71 + .../bindings/sound/qcom,wcd937x.yaml | 119 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1148 +++++++ sound/soc/codecs/wcd937x.c | 3036 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 655 ++++ 7 files changed, 5056 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 6bd343537461b57f3efe5dfc5fc193a232dfef1e -- 2.25.1

5 18

sof-hda-dsp/sof-soundwire: Create "hdmi:" mapping PCMs to allow passthrough if supported
by GitHub pull_request - opened 22 Apr '24

22 Apr '24

alsa-project/alsa-ucm-conf pull request #411 was opened from ujfalusi: Hi, SOF with IPC4 can use 'ChainDMA' on selected PCMs (HDMI/DP currently) which allows them to be used for bytestream passthrough since the data is passed through unmodified. The kernel will list the PCMs with ChainDMA to the card's components list: https://github.com/thesofproject/linux/pull/4921 For example for sof-dsp-hda cards `iec61937-pcm:5,4,3` will be added. For user space to use HDMI passthrough, the `hdmi:` PCM device should be present correctly mapping to the machine, for example for sof-dsp-hda: ``` aplay -L | grep hdmi hdmi:CARD=sofhdadsp,DEV=0 hdmi:CARD=sofhdadsp,DEV=1 hdmi:CARD=sofhdadsp,DEV=2 ``` This PR does this by 1. Checking the `iec61937-pcm` indexes against the expected HDMI devices (sof-dsp-hda: 3-5, sof-soundwire: 5-7) 2. If there is a match (passthrough can be supported) then we will create three conf files: The generated files are: [1] /var/lib/alsa/conf.d/42-sof-hdmi.conf [2] /var/lib/alsa/card[card_number].conf.d/30-sof-hdmi-common.conf [3] /var/lib/alsa/card[card_number].conf.d/31-sof-hdmi.conf [1] includes the pcm/iec958.conf and pcm/hdmi.conf to global space of alsaconf to be used by the card macros [2] Card specific macros for hdmi PCM definition, mapping [3] Card specific definitions of the three HDMI port Note for [1]: I needed to use `shell "echo '...` since the `cfg-save` would expand the includes into the saved config file and that just does not result a working alsa configuration. @perexg, I'm sure all this can be done in a cleaner way... We cannot do this unconditionally as if the HDMI is not using ChainDMA (and SOF is not IPC4) then the passthrough is not possible since the firmware might touch the data. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/411 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/411.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/3] ASoC: nau8821: Add delay control for ADC
by Seven Lee 22 Apr '24

22 Apr '24

Change the original fixed delay to the assignment from the property. It will make it more flexible to different platforms to avoid pop noise at the beginning of recording. Change: V1 -> V2: - Revise adc-delay-ms properties description. - Fix examples of bugs found by bots. Seven Lee (3): ASoC: dt-bindings: nau8821: Add delay control for ADC ASoC: nau8821: Add delay control for ADC ASoC: nau8821: Remove redundant ADC controls .../bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ sound/soc/codecs/nau8821.c | 19 +++++++++---------- sound/soc/codecs/nau8821.h | 1 + 3 files changed, 17 insertions(+), 10 deletions(-) -- 2.25.1

3 5

[PATCH] ALSA: scarlett2: Zero initialize ret in scarlett2_ag_target_ctl_get()
by Nathan Chancellor 21 Apr '24

21 Apr '24

Clang warns (or errors with CONFIG_WERROR): sound/usb/mixer_scarlett2.c:3697:6: error: variable 'err' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] 3697 | if (private->autogain_updated) { | ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/usb/mixer_scarlett2.c:3707:9: note: uninitialized use occurs here 3707 | return err; | ^~~ sound/usb/mixer_scarlett2.c:3697:2: note: remove the 'if' if its condition is always true 3697 | if (private->autogain_updated) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/usb/mixer_scarlett2.c:3688:9: note: initialize the variable 'err' to silence this warning 3688 | int err; | ^ | = 0 1 error generated. Initialize ret to zero to ensure ret is initialized in all paths within scarlett2_ag_target_ctl_get(), which matches the style of other functions in this driver. Fixes: e30ea5340c25 ("ALSA: scarlett2: Add autogain target controls") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- FWIW, a lot of the functions that end in just 'mutex_unlock()' before returning could probably benefit from the cleanup.h infrastructure, which would help with the scope of err in this particular function, but I decided to go with the most minimal fix here. --- sound/usb/mixer_scarlett2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a2de31a0371b..b251d25f2a85 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3685,7 +3685,7 @@ static int scarlett2_ag_target_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err; + int err = 0; mutex_lock(&private->data_mutex); --- base-commit: 32cb23a0f911317cdb5030035e49a204aa86fef5 change-id: 20240419-alsa-scarlett2-fix-wsometimes-uninitialized-d44f7659b52a Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

2 1

[PATCH] ASoC: codecs: wsa881x: set clk_stop_mode1 flag
by srinivas.kandagatla＠linaro.org 21 Apr '24

21 Apr '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> WSA881x codecs do not retain the state while clock is stopped, so mark this with clk_stop_mode1 flag. Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wsa881x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 3c025dabaf7a..1253695bebd8 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1155,6 +1155,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + pdev->prop.clk_stop_mode1 = true; gpiod_direction_output(wsa881x->sd_n, !wsa881x->sd_n_val); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); -- 2.43.0

2 1

[PATCH 0/2] ASoC: Intel: avs: Refactor IRQ handling
by Cezary Rojewski 21 Apr '24

21 Apr '24

The existing code can be both improved and simplified. To make this change easier to manage, first add new implementation and then remove deadcode in a separate patch. Simplification achieved with: - reduce the amount of resources requested by the driver i.e.: IPC and CLDMA request_irq() merged into one - reduce the number of DSP ops from 2 to 1: irq_handler/thread() vs dsp_interrupt() - drop ambiguity around CLDMA interrupt, let skl.c handle that explicitly as it is the only user With that done, switch to the new implementation and remove unused members. While the change is non-trivial, from functional perspective status quo is achieved. Cezary Rojewski (2): ASoC: Intel: avs: New IRQ handling implementation ASoC: Intel: avs: Remove unused IRQ-related code sound/soc/intel/avs/apl.c | 20 +++++++- sound/soc/intel/avs/avs.h | 8 ++-- sound/soc/intel/avs/cldma.c | 42 ++++------------- sound/soc/intel/avs/cldma.h | 1 + sound/soc/intel/avs/cnl.c | 91 ++++++++++++++++++++++++------------- sound/soc/intel/avs/core.c | 85 ++++++++++++++++------------------ sound/soc/intel/avs/icl.c | 3 +- sound/soc/intel/avs/ipc.c | 48 ------------------- sound/soc/intel/avs/skl.c | 73 +++++++++++++++++++++-------- sound/soc/intel/avs/tgl.c | 3 +- 10 files changed, 186 insertions(+), 188 deletions(-) -- 2.25.1

2 3

[PATCH 0/4] ASoC: qcom: display port changes
by srinivas.kandagatla＠linaro.org 20 Apr '24

20 Apr '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset adds support for. 1. parse Display Port module tokens from ASoC topology 2. add support to DP/HDMI Jack events. 3. fixes a typo in function name in sm8250 Verified these patches on X13s along with changes to tplg in https://git.codelinaro.org/linaro/qcomlt/audioreach-topology/-/tree/topic/x… and ucm chagnes from https://github.com/Srinivas-Kandagatla/alsa-ucm-conf/tree/topic/x13s-dp Thanks, Srini Srinivas Kandagatla (4): ASoC: qcom: q6dsp: parse Display port tokens ASoC: qcom: common: add Display port Jack function ASoC: qcom: sc8280xp: add Display port Jack ASoC: qcom: sm8250: fix a typo in function name sound/soc/qcom/common.c | 30 ++++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 3 +++ sound/soc/qcom/qdsp6/topology.c | 26 ++++++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 13 +++++++++++++ sound/soc/qcom/sm8250.c | 4 ++-- 5 files changed, 74 insertions(+), 2 deletions(-) -- 2.43.0

2 6

RE: [PATCH v1 0/7] Add support for various laptops using CS35L41 HDA without _DSD
by Niclas Söderlund 19 Apr '24

19 Apr '24

Hi, amazingly high level of laymanship here and equally high level of dispear. But model after model is added to fix the problem with speakers not working, but nowhere is my model seen. I have the Lenovo 13X G2 model and even at kernel 6.8.7 my speakers still doesnt work. Where can I add a request to also include this model in the patches, so I have a hope of having the speakers start working? I do apologise in advance, Mvh, Niclas

2 1

[PATCH v2] spi: cs42l43: Correct name of ACPI property
by Maciej Strozek 19 Apr '24

19 Apr '24

Fixes: 439fbc97502a ("spi: cs42l43: Add bridged cs35l56 amplifiers") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2: Corrected the Fixes tag to point to right commit drivers/spi/spi-cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/spi/spi-cs42l43.c b/drivers/spi/spi-cs42l43.c index cdc61cd089ad..9d747ea69926 100644 --- a/drivers/spi/spi-cs42l43.c +++ b/drivers/spi/spi-cs42l43.c @@ -267,7 +267,7 @@ static bool cs42l43_has_sidecar(struct fwnode_handle *fwnode) continue; ret = fwnode_property_read_u32(ext_fwnode, - "01fa-cirrus-sidecar-instances", + "01fa-sidecar-instances", &val); fwnode_handle_put(ext_fwnode); -- 2.34.1

3 2

[PATCH] ASoC: ti: davinci-mcasp: Fix race condition during probe
by Joao Paulo Goncalves 19 Apr '24

19 Apr '24

From: Joao Paulo Goncalves <joao.goncalves(a)toradex.com> When using davinci-mcasp as CPU DAI with simple-card, there are some conditions that cause simple-card to finish registering a sound card before davinci-mcasp finishes registering all sound components. This creates a non-working sound card from userspace with no problem indication apart from not being able to play/record audio on a PCM stream. The issue arises during simultaneous probe execution of both drivers. Specifically, the simple-card driver, awaiting a CPU DAI, proceeds as soon as davinci-mcasp registers its DAI. However, this process can lead to the client mutex lock (client_mutex in soc-core.c) being held or davinci-mcasp being preempted before PCM DMA registration on davinci-mcasp finishes. This situation occurs when the probes of both drivers run concurrently. Below is the code path for this condition. To solve the issue, defer davinci-mcasp CPU DAI registration to the last step in the audio part of it. This way, simple-card CPU DAI parsing will be deferred until all audio components are registered. Fail Code Path: simple-card.c: probe starts simple-card.c: simple_dai_link_of: simple_parse_node(..,cpu,..) returns EPROBE_DEFER, no CPU DAI yet davinci-mcasp.c: probe starts davinci-mcasp.c: devm_snd_soc_register_component() register CPU DAI simple-card.c: probes again, finish CPU DAI parsing and call devm_snd_soc_register_card() simple-card.c: finish probe davinci-mcasp.c: *dma_pcm_platform_register() register PCM DMA davinci-mcasp.c: probe finish Cc: stable(a)vger.kernel.org Fixes: 9fbd58cf4ab0 ("ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller") Signed-off-by: Joao Paulo Goncalves <joao.goncalves(a)toradex.com> --- sound/soc/ti/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b892d66f78470..1e760c3155213 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2417,12 +2417,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp_reparent_fck(pdev); - ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, - &davinci_mcasp_dai[mcasp->op_mode], 1); - - if (ret != 0) - goto err; - ret = davinci_mcasp_get_dma_type(mcasp); switch (ret) { case PCM_EDMA: @@ -2449,6 +2443,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; } + ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, + &davinci_mcasp_dai[mcasp->op_mode], 1); + + if (ret != 0) + goto err; + no_audio: ret = davinci_mcasp_init_gpiochip(mcasp); if (ret) { -- 2.34.1

4 3

Re: [PATCH v1 0/7] Add support for various laptops using CS35L41 HDA without _DSD
by Niclas Söderlund 19 Apr '24

19 Apr '24

Hi, amazingly high level of laymanship here and equally high level of dispear. But model after model is added to fix the problem with speakers not working, but nowhere is my model seen. I have the Lenovo 13X G2 model and even at kernel 6.8.7 my speakers still doesnt work. Where can I add a request to also include this model in the patches, so I have a hope of having the speakers start working? I do apologise in advance but it would be a huge loss if this laptop is not appended as being one of the models being fixed. Mvh, Niclas Söderlund

1 0

Re: [PATCH v1 0/7] Add support for various laptops using CS35L41 HDA without _DSD
by Niclas Söderlund 19 Apr '24

19 Apr '24

Hi, amazingly high level of laymanship here and equally high level of dispear. But model after model is added to fix the problem with speakers not working, but nowhere is my model seen. I have the Lenovo 13X G2 model and even at kernel 6.8.7 my speakers still doesnt work. Where can I add a request to also include this model in the patches, so I have a hope of having the speakers start working? I do apologise in advance, Mvh, Niclas

1 0

[PATCH v1 0/1] soundwire: qcom: disable stop clock on 1.3.0 and below
by Anton Bambura 19 Apr '24

19 Apr '24

This patch fixes audio on sdm845 + wcd9340 with soundwire. The affected devices are AYN Odin and Lenovo Yoga C630. If this gets applied it should also be backported to stable and 6.6.x LTS. Anton Bambura (1): soundwire: qcom: disable stop clock on 1.3.0 and below drivers/soundwire/qcom.c | 3 +++ 1 file changed, 3 insertions(+) -- 2.43.0

3 4

Question: layout of a patchset dependent on another tree
by Cezary Rojewski 19 Apr '24

19 Apr '24

Hello Mark, I'd like to send changes which do modify the avs-driver but are dependent on ACPI/NHLT changes I've recently upstream [1] to Rafael's linux-acpi tree. Thanks to Rafael, there's an immutable branch [2] that has all relevant commits on the top. Should I send both the NHLT+avs changes and have broonie/for-next as a base for that or provide only the avs changes and state the dependency within the cover-letter? The latter will fail to compile without its dependency obviously. [1]: https://lore.kernel.org/linux-acpi/20240319083018.3159716-1-cezary.rojewski… [2]: https://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm.git/log/?h=… Kind regards, Czarek

2 1

FIXES: alsamixer lowering volume control widget past 0 causes hostile volume spike to 100
by GitHub pull_request - opened 19 Apr '24

19 Apr '24

alsa-project/alsa-utils pull request #266 was opened from genbtc: bug: lowering volume below 0 causes volume to overflow and spike to 100% volume hurting my ears. to reproduce: when holding down z,x,c to lower the volume, the volume spikes to 100% and then down to 0% over and over background: when volume is already 0 and direction is -1 (lowering volume) - but lower than 0 is impossible the function `set_normalized_volume` @ volume-mapping.c is eventually still called anyway `value = lrint_dir(6000.0 * log10(volume), dir) + max;` and volume is 0, so log10(0) resolves to negative infinity, so value overflows past infinity. 9223372036854775802 ![alsamixer-alsa-utils-volumemap-bug3_2024-04-18_13-27-23](https://github.com/alsa-project/alsa-utils/assets/3683744/d382c59b-0e71-420f-8495-4d5d863f3734) my fix: this function `change_volume_relative` function will now skip execution of the set_normalized_playback_volume function when volume is 0 and the dir is -1 TLDR: (stop trying to set a lower volume than 0) Request URL : https://github.com/alsa-project/alsa-utils/pull/266 Patch URL : https://github.com/alsa-project/alsa-utils/pull/266.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 18 Apr '24

18 Apr '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Modification for add tas2781 driver for SPI ALSA: hda/tas2781: Main code of tas2781 driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 driver for SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 10 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 15 + sound/pci/hda/tas2781-spi.h | 162 ++ sound/pci/hda/tas2781_hda_spi.c | 1336 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2289 +++++++++++++++++ 8 files changed, 3830 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

4 12

[PATCH] spi: cs42l43: Correct name of ACPI property
by Maciej Strozek 18 Apr '24

18 Apr '24

Fixes: 1f05252a3a95 ("Add bridged amplifiers to cs42l43") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- drivers/spi/spi-cs42l43.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/spi/spi-cs42l43.c b/drivers/spi/spi-cs42l43.c index cdc61cd089ad..9d747ea69926 100644 --- a/drivers/spi/spi-cs42l43.c +++ b/drivers/spi/spi-cs42l43.c @@ -267,7 +267,7 @@ static bool cs42l43_has_sidecar(struct fwnode_handle *fwnode) continue; ret = fwnode_property_read_u32(ext_fwnode, - "01fa-cirrus-sidecar-instances", + "01fa-sidecar-instances", &val); fwnode_handle_put(ext_fwnode); -- 2.34.1

2 3

[PATCH 0/3] adding 705.6/768kHz rates; DSD and 768kHz in aloop
by Pavel Hofman 18 Apr '24

18 Apr '24

This patch series allows using the loopback driver in modern chains with samplerates up to 768kHz and DSD formats. The patch "ALSA:pcm: add support for 705.6kHz and 768kHz sample rates" is basically a copy of commit 4cc4531c310e ("ALSA: pcm: add support for 352.8KHz and 384KHz sample rate"). As of loopback performance at 768kHz - in my tests aloop runs OK event at 12MHz samplerate. Pavel Hofman (3): ALSA:aloop: add DSD formats ALSA:pcm: add support for 705.6kHz and 768kHz sample rates ALSA:aloop: add support for up to 768kHz sample rate include/sound/pcm.h | 5 +++++ sound/core/pcm_native.c | 2 +- sound/drivers/aloop.c | 9 ++++++--- 3 files changed, 12 insertions(+), 4 deletions(-) -- 2.25.1

3 5

[PATCH v2 0/2] Add Cadence I2S-MC controller driver
by Xingyu Wu 18 Apr '24

18 Apr '24

The Cadence Multi-channel I2S (I2S-MC) Controller implements a function of the multi-channel (up to 8-channel) bus. Each stereo channel combines functions of a transmitter and a receiver, and can switch freely between them. Each channel has independent gating, clock and interruption control. It also support some of these channels are used as playback and others can also be used as record in the same time. Four I2S controllers are used on the StarFive JH8100 SoC. Two of the I2S controllers use two stereo channels, one of them use four channels, and one use eight. It had tested on the fpga with WM8960 and PDM. Changes since v1: - Added new compatible for StarFive JH8100 SoC and a special property to be got as the max channels number in the bindings. - Dropped the useless '|' in the bindings. - Moved the drivers to a new folder named 'cdns' and modified the name of functions. v1: https://lore.kernel.org/all/20231221033223.73201-1-xingyu.wu@starfivetech.c… Xingyu Wu (2): ASoC: dt-bindings: Add bindings for Cadence I2S-MC controller ASoC: cdns: Add drivers of Cadence Multi-Channel I2S Controller .../bindings/sound/cdns,i2s-mc.yaml | 110 +++ MAINTAINERS | 6 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/cdns/Kconfig | 18 + sound/soc/cdns/Makefile | 3 + sound/soc/cdns/cdns-i2s-mc-pcm.c | 262 +++++++ sound/soc/cdns/cdns-i2s-mc.c | 724 ++++++++++++++++++ sound/soc/cdns/cdns-i2s-mc.h | 157 ++++ 9 files changed, 1282 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,i2s-mc.yaml create mode 100644 sound/soc/cdns/Kconfig create mode 100644 sound/soc/cdns/Makefile create mode 100644 sound/soc/cdns/cdns-i2s-mc-pcm.c create mode 100644 sound/soc/cdns/cdns-i2s-mc.c create mode 100644 sound/soc/cdns/cdns-i2s-mc.h -- 2.25.1

4 20

[PATCH] Fix volumn control of ThinkBook 16P Gen4
by zhanghuayu1233＠qq.com 18 Apr '24

18 Apr '24

From: Huayu Zhang <zhanghuayu1233(a)qq.com> change HDA & AMP configuration from ALC287_FIXUP_CS35L41_I2C_2 to ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD for ThinkBook 16P Gen4 models to fix volumn control issue (cannot fully mute). Signed-off-by: Huayu Zhang <zhanghuayu1233(a)qq.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6940bc4ec39..1aef223307bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10396,8 +10396,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), -- 2.34.1

2 1

[PATCH v2 0/7] Add features, fixes and laptops for CS35L41 HDA
by Stefan Binding 18 Apr '24

18 Apr '24

This chain adds a number of improvements for CS35L41, including a new firmware loading feature, as well as some additional cleanup. Patch 1 adds support for a new firmware tuning file which is used to set special tuning parameters used by the firmware. The HP Omen laptop added in patches 2 and 3 use such a feature. Patch 4 changes the DSP1RX5/6 Source used by the firmware to depend on the boost type. A similar change in the ASoC driver will be needed later. Patches 5 and 6 are cleanup patches, with the first of those re-using cs-amp-lib for calibration which as added for CS35L56 drivers, ensuring that all laptops are using the correct calibration values. Patch 7 fixes a small error in the CS35L41 Property table for the Lenovo Legion slim 7 16ARHA7. Changes since v1: - Fixed print statements based on warnings from kernel test robot Richard Fitzgerald (1): ALSA: hda: cs35l41: Remove redundant argument to cs35l41_request_firmware_file() Stefan Binding (6): ALSA: hda: cs35l41: Set the max PCM Gain using tuning setting ALSA: hda: cs35l41: Support HP Omen models without _DSD ALSA: hda/realtek: Add quirks for HP Omen models using CS35L41 ALSA: hda: cs35l41: Update DSP1RX5/6 Sources for DSP config ALSA: hda: cs35l41: Use shared cs-amp-lib to apply calibration ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7 16ARHA7 include/sound/cs35l41.h | 5 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l41_hda.c | 347 +++++++++++++++++++-------- sound/pci/hda/cs35l41_hda.h | 6 + sound/pci/hda/cs35l41_hda_property.c | 8 +- sound/pci/hda/patch_realtek.c | 4 + 6 files changed, 265 insertions(+), 107 deletions(-) -- 2.34.1

2 8

[PATCH v1] ALSA: hda/tas2781: Add new vendor_id and subsystem_id to support ThinkPad ICE-1
by Shenghao Ding 18 Apr '24

18 Apr '24

Add new vendor_id and subsystem_id to support new Lenovo laptop ThinkPad ICE-1 Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cdcb28aa9d7b..051ca23c8cfa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10333,6 +10333,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x222e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), + SND_PCI_QUIRK(0x17aa, 0x2234, "Thinkpad ICE-1", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x2245, "Thinkpad T470", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2246, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2247, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), -- 2.34.1

2 1

[PATCH v1 RESEND] slimbus: stream: Add null pointer check for client functions
by Viken Dadhaniya 18 Apr '24

18 Apr '24

There is a possible scenario where client driver is calling slimbus stream APIs in incorrect sequence and it might lead to invalid null access of the stream pointer in slimbus enable/disable/prepare/unprepare/free function. Fix this by checking validity of the stream before accessing in all function API’s exposed to client. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/stream.c | 37 +++++++++++++++++++++++++++++++++---- 1 file changed, 33 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..c5a436fd0952 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -202,10 +202,16 @@ static int slim_get_prate_code(int rate) int slim_stream_prepare(struct slim_stream_runtime *rt, struct slim_stream_config *cfg) { - struct slim_controller *ctrl = rt->dev->ctrl; + struct slim_controller *ctrl; struct slim_port *port; int num_ports, i, port_id, prrate; + if (!rt || !cfg) { + pr_err("%s: Stream or cfg is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = rt->dev->ctrl; if (rt->ports) { dev_err(&rt->dev->dev, "Stream already Prepared\n"); return -EINVAL; @@ -358,9 +364,15 @@ int slim_stream_enable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = stream->dev->ctrl; if (ctrl->enable_stream) { ret = ctrl->enable_stream(stream); if (ret) @@ -411,12 +423,18 @@ int slim_stream_disable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; + ctrl = stream->dev->ctrl; if (ctrl->disable_stream) ctrl->disable_stream(stream); @@ -448,6 +466,11 @@ int slim_stream_unprepare(struct slim_stream_runtime *stream) { int i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; @@ -476,8 +499,14 @@ EXPORT_SYMBOL_GPL(slim_stream_unprepare); */ int slim_stream_free(struct slim_stream_runtime *stream) { - struct slim_device *sdev = stream->dev; + struct slim_device *sdev; + + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + sdev = stream->dev; spin_lock(&sdev->stream_list_lock); list_del(&stream->node); spin_unlock(&sdev->stream_list_lock); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

3 5

[PATCH v2] slimbus: stream: Add null pointer check for client APIs
by Viken Dadhaniya 17 Apr '24

17 Apr '24

Due to incorrect sequrnce of slimbus APIs called by client driver, facing system crash in slimbus framework APIs. To prevent system crash, add NULL pointer check in enable/disable/prepare/unprepare/free APIs. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- v1 -> v2: - Update commit log. - Update debug logs. v1 Link: https://lore.kernel.org/linux-arm-msm/ZhgIHHP1FUeCr+vx@hu-bjorande-lv.qualc… --- --- drivers/slimbus/stream.c | 37 +++++++++++++++++++++++++++++++++---- 1 file changed, 33 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..76d0579cbf53 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -202,10 +202,16 @@ static int slim_get_prate_code(int rate) int slim_stream_prepare(struct slim_stream_runtime *rt, struct slim_stream_config *cfg) { - struct slim_controller *ctrl = rt->dev->ctrl; + struct slim_controller *ctrl; struct slim_port *port; int num_ports, i, port_id, prrate; + if (!rt || !cfg) { + pr_err("Invalid stream rt or cfg\n"); + return -EINVAL; + } + + ctrl = rt->dev->ctrl; if (rt->ports) { dev_err(&rt->dev->dev, "Stream already Prepared\n"); return -EINVAL; @@ -358,9 +364,15 @@ int slim_stream_enable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("Invalid stream\n"); + return -EINVAL; + } + + ctrl = stream->dev->ctrl; if (ctrl->enable_stream) { ret = ctrl->enable_stream(stream); if (ret) @@ -411,12 +423,18 @@ int slim_stream_disable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("Invalid stream"); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; + ctrl = stream->dev->ctrl; if (ctrl->disable_stream) ctrl->disable_stream(stream); @@ -448,6 +466,11 @@ int slim_stream_unprepare(struct slim_stream_runtime *stream) { int i; + if (!stream) { + pr_err("Invalid stream\n"); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; @@ -476,8 +499,14 @@ EXPORT_SYMBOL_GPL(slim_stream_unprepare); */ int slim_stream_free(struct slim_stream_runtime *stream) { - struct slim_device *sdev = stream->dev; + struct slim_device *sdev; + + if (!stream) { + pr_err("Invalid stream\n"); + return -EINVAL; + } + sdev = stream->dev; spin_lock(&sdev->stream_list_lock); list_del(&stream->node); spin_unlock(&sdev->stream_list_lock); -- 2.17.1

1 0

[PATCH 0/7] Xperia 1 V support
by Konrad Dybcio 17 Apr '24

17 Apr '24

DTS for the phone and some fly-by fixes Patch 1 for Mark/sound Rest for qcom Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (7): dt-bindings: ASoC: cs35l45: Add interrupts dt-bindings: arm: qcom: Add Xperia 1 V arm64: dts: qcom: sm8550: Mark QUPs and GPI dma-coherent arm64: dts: qcom: sm8550: Mark APPS SMMU as dma-coherent arm64: dts: qcom: sm8550: Add missing DWC3 quirks arm64: dts: qcom: sm8550: Mark DWC3 as dma-coherent arm64: dts: qcom: Add SM8550 Xperia 1 V Documentation/devicetree/bindings/arm/qcom.yaml | 1 + .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 3 + arch/arm64/boot/dts/qcom/Makefile | 1 + .../dts/qcom/sm8550-sony-xperia-yodo-pdx234.dts | 779 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/sm8550.dtsi | 20 +- 5 files changed, 801 insertions(+), 3 deletions(-) --- base-commit: ae00c445390b349e070a64dc62f08aa878db7248 change-id: 20240210-topic-1v-18e9697d2056 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

8 20

[PATCH ver.2 6.1.y 0/6] ASoC: codecs: es8326: fix support (backport from v6.6)
by kovalev＠altlinux.org 17 Apr '24

17 Apr '24

Without these patches, all tested laptop models using the es8326 audio codec have no sound from the speakers and headphones, and the headset microphone does not work. Although the initialization of the sound card is successful. -- ver.2: drop a commit that does not affect the fix of functionality; added an explanation that does not work on the current version of the kernel without patches. -- Patches have been successfully tested on the latest 6.1.86 kernel. [PATCH 6.1.y 1/6] ASoC: codecs: ES8326: Add es8326_mute function [PATCH 6.1.y 2/6] ASoC: codecs: ES8326: Change Hp_detect register names [PATCH 6.1.y 3/6] ASoC: codecs: ES8326: Change Volatile Reg function [PATCH 6.1.y 4/6] ASoC: codecs: ES8326: Fix power-up sequence [PATCH 6.1.y 5/6] ASOC: codecs: ES8326: Add calibration support for [PATCH 6.1.y 6/6] ASoC: codecs: ES8326: Update jact detection function

1 6

[PATCH v3 71/74] x86/cpu/vfm: Update sound/soc/intel/avs/boards/es8336.c
by Tony Luck 17 Apr '24

17 Apr '24

New CPU #defines encode vendor and family as well as model. Signed-off-by: Tony Luck <tony.luck(a)intel.com> --- sound/soc/intel/avs/boards/es8336.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 5c90a6007577..6c86c581877e 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -18,6 +18,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-acpi.h> +#include <asm/cpu_device_id.h> #include <asm/intel-family.h> #include "../utils.h" @@ -153,9 +154,9 @@ static int avs_es8336_hw_params(struct snd_pcm_substream *substream, int clk_freq; int ret; - switch (boot_cpu_data.x86_model) { - case INTEL_FAM6_KABYLAKE_L: - case INTEL_FAM6_KABYLAKE: + switch (boot_cpu_data.x86_vfm) { + case INTEL_KABYLAKE_L: + case INTEL_KABYLAKE: clk_freq = 24000000; break; default: -- 2.44.0

2 1

ucm2: HDA: Use Master volume control for dual speakers
by GitHub pull_request - opened 16 Apr '24

16 Apr '24

alsa-project/alsa-ucm-conf pull request #410 was opened from tuxedoxt: This PR extends the existing 'activate Bass Speaker' case with additional configuration - Use Master mixer for volume control (instead of only using Speaker) - Configure Speaker Volume and Bass Speaker volume to max since Master volume regulation is used It fixes a case where only the first speaker volume is changed using the system volume control. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/410 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/410.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt715: add vendor clear control register
by Jack Yu 16 Apr '24

16 Apr '24

Add vendor clear control register in readable register's callback function. This prevents an access failure reported in Intel CI tests. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715-sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 24b3443aeb02..ec255ada44e0 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -111,6 +111,7 @@ static bool rt715_readable_register(struct device *dev, unsigned int reg) case 0x839d: case 0x83a7: case 0x83a9: + case 0x752001: case 0x752039: return true; default: -- 2.34.1

3 2

[PATCH 0/2] ASoC: fsl-asoc-card: add wm8904 codec support
by Shengjiu Wang 16 Apr '24

16 Apr '24

add wm8904 codec support in fsl-asoc-card. Shengjiu Wang (2): ASoC: fsl-asoc-card: add wm8904 codec support ASoC: dt-bindings: fsl-asoc-card: Add compatbile string for wm8904 codec .../devicetree/bindings/sound/fsl-asoc-card.yaml | 1 + sound/soc/fsl/fsl-asoc-card.c | 8 ++++++++ 2 files changed, 9 insertions(+) -- 2.34.1

2 3

[PATCH v2.5] ASoC: Replace dpcm_playback/capture to playback/capture_assertion
by Kuninori Morimoto 16 Apr '24

16 Apr '24

Hi Mark, Amadeusz, Jerome, Pierre-Louis This is v2.5 patch which is preparation for v3. Because full-patch-set has over 20 patches, to avoid posting huge patch-bomb, I try to post main patch only for preparation. In v2.5, it will use xxx_assertion flag, and exising dpcm_xxx and xxx_only flags will be converted to it. I think it can keep compatibility and possible to indicate link error same as before which was conserned by Pierre-Louis. Please let me know if it still not good match. One of my big consern is Amadeusz/Jerome's idea which uses "XXX | YYY" style. To avoid confusion, let's name each style as below here. [Flag style] unsigned int flags; #define SND_SOC_FLAGS_ASSERTION_PLAYBACK BIT(x) #define SND_SOC_FLAGS_ASSERTION_CAPTURE BIT(x) ... [BitField style] unsigned int playback_assertion:1; unsigned int capture_assertion:1; ... This v2.5 patch is using [BitField style] and I think it is not a big problem from "code point of view", but I think [Flag style] is better for "human understandable point of view", because we can define like below, for example. #define SND_SOC_FLAGS_AVAILABLE_PLAYBACK /* no flag is needed */ #define SND_SOC_FLAGS_AVAILABLE_PLAYBACK_WITH_ASSERTION SND_SOC_FLAGS_ASSERTION_PLAYBACK #define SND_SOC_FLAGS_AVAILABLE_CAPTURE /* no flag is needed */ #define SND_SOC_FLAGS_AVAILABLE_CAPTURE_WITH_ASSERTION SND_SOC_FLAGS_ASSERTION_CAPTURE #define SND_SOC_FLAGS_AVAILABLE_BIDIRECTIONAL /* no flag is needed */ #define SND_SOC_FLAGS_AVAILABLE_BIDIRECTIONAL_WITH_ASSERTION (SND_SOC_FLAGS_ASSERTION_PLAYBACK | SND_SOC_FLAGS_ASSERTION_CAPTURE) #define SND_SOC_FLAGS_AVAILABLE_PLAYBACK_ONLY SND_SOC_FLAGS_ASSERTION_PLAYBACK #define SND_SOC_FLAGS_AVAILABLE_CAPTURE_ONLY SND_SOC_FLAGS_ASSERTION_CAPTURE Switch to [Flag style] is OK for me, but one consern is that in such case, people will wonder "why ASoC is using both [Flag style] and [BitField style] in the same time ?", because we are using [BitField style] for other flags. So, my suggestion is that next v3 patch uses [Flag style]. And after that, post new patch-set to switch [BitField style] to [Flag style] for other flags. But I wonder is this good approach ? v2 -> v2.5 - use xxx_assertion flag - dpcm_xxx -> xxx_assertion - xxx_only -> xxx_assertion - only [01/xx] patch v1 -> v2 - based on latest ASoC branch - keep comment on Intel - tidyup patch title - tidyup DPCM BE warning output condition - Add new patch for Document Link: https://lore.kernel.org/r/87o7b353of.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87zfuesz8y.wl-kuninori.morimoto.gx@renesas.com Thank you for your help !! Best regards --- Renesas Electronics Ph.D. Kuninori Morimoto

3 5

alsa-info.sh: log SoundWire devices reported in ACPI
by GitHub pull_request - opened 15 Apr '24

15 Apr '24

alsa-project/alsa-utils pull request #264 was opened from plbossart: Track SoundWire devices reported in ACPI, this will help detect missing or invalid configurations. For now we only filter information from Realtek, Cirrus Logic and TI, based on the manufacturer ID of their devices. Example log for the classic RT711/RT1316x2/RT714 configuration. !!ACPI SoundWire Device Status Information !!--------------- Realtek 0x000030025d071101 Realtek 0x000331025d131601 Realtek 0x000230025d131601 Realtek 0x000130025d071401 Request URL : https://github.com/alsa-project/alsa-utils/pull/264 Patch URL : https://github.com/alsa-project/alsa-utils/pull/264.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: dt-bindings: omap-mcpdm: Convert to DT schema
by Mighty 15 Apr '24

15 Apr '24

From: Mithil Bavishi <bavishimithil(a)gmail.com> Convert the OMAP4+ McPDM bindings to DT schema. Signed-off-by: Mithil Bavishi <bavishimithil(a)gmail.com> --- .../devicetree/bindings/sound/omap-mcpdm.txt | 30 ---------- .../bindings/sound/ti,omap4-mcpdm.yaml | 58 +++++++++++++++++++ 2 files changed, 58 insertions(+), 30 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt deleted file mode 100644 index ff98a0cb5..000000000 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ /dev/null @@ -1,30 +0,0 @@ -* Texas Instruments OMAP4+ McPDM - -Required properties: -- compatible: "ti,omap4-mcpdm" -- reg: Register location and size as an array: - <MPU access base address, size>, - <L3 interconnect address, size>; -- interrupts: Interrupt number for McPDM -- ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" - -Example: - -mcpdm: mcpdm@40132000 { - compatible = "ti,omap4-mcpdm"; - reg = <0x40132000 0x7f>, /* MPU private access */ - <0x49032000 0x7f>; /* L3 Interconnect */ - interrupts = <0 112 0x4>; - interrupt-parent = <&gic>; - ti,hwmods = "mcpdm"; -}; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml b/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml new file mode 100644 index 000000000..73fcfaf6e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,omap-mcpdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: OMAP McPDM + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +description: + OMAP ALSA SoC DAI driver using McPDM port used by TWL6040 + +properties: + compatible: + const: ti,omap4-mcpdm + + reg: + items: + - description: MPU access base address + - description: L3 interconnect address + + interrupts: + maxItems: 1 + + ti,hwmods: + description: Name of the hwmod associated to the McPDM, likely "mcpdm" + + clocks: + description: phandle for the pdmclk provider, likely <&twl6040> + + clock-names: + description: Must be "pdmclk" + + +required: + - compatible + - reg + - interrupts + - ti,hwmods + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + pdm@0 { + compatible = "ti,omap4-mcpdm"; + reg = <0x40132000 0x7f>, /* MPU private access */ + <0x49032000 0x7f>; /* L3 Interconnect */ + interrupts = <0 112 0x4>; + interrupt-parent = <&gic>; + ti,hwmods = "mcpdm"; + clocks = <&twl6040>; + clock-names = "pdmclk"; + }; -- 2.34.1

3 2

DSD formats in snd-aloop?
by Pavel Hofman 15 Apr '24

15 Apr '24

Hi, Please would you accept a patch which adds the 16/32bit alsa DSD formats to the list of supported audio formats in snd-aloop? The loopback device does not modify or access the actual samples in any way, defines no volume or mute controls, it's strictly bitperfect. Thanks a lot for considering, Pavel.

2 1

[PATCH 1/2] ASoC: wcd934x: Drop unneeded MODULE_ALIAS
by Krzysztof Kozlowski 15 Apr '24

15 Apr '24

The ID table already has respective entry and MODULE_DEVICE_TABLE and creates proper alias for platform driver. Having another MODULE_ALIAS causes the alias to be duplicated. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 6813268e6a19..de870c7819ca 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5967,7 +5967,6 @@ static struct platform_driver wcd934x_codec_driver = { } }; -MODULE_ALIAS("platform:wcd934x-codec"); module_platform_driver(wcd934x_codec_driver); MODULE_DESCRIPTION("WCD934x codec driver"); MODULE_LICENSE("GPL v2"); -- 2.34.1

2 2

[PATCH] ASoC: cs35l41: Fix error code in cs35l41_dsp_init()
by Dan Carpenter 15 Apr '24

15 Apr '24

Set the error code on this error path. Don't return success. Fixes: eefb831d2e4d ("ASoC: cs35l41: Update DSP1RX5/6 Sources for DSP config") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/cs35l41.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index f8e57a2fc3e3..2799ccd6b5c7 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1126,6 +1126,7 @@ static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) default: dev_err(cs35l41->dev, "wm_halo_init failed - Invalid Boost Type: %d\n", cs35l41->hw_cfg.bst_type); + ret = -EINVAL; goto err_dsp; } -- 2.43.0

1 0

ucm2: soundwire: add rt722 SDCA device
by GitHub pull_request - opened 15 Apr '24

15 Apr '24

alsa-project/alsa-ucm-conf pull request #409 was opened from jack-cy-yu: Add support of ucm to rt722 SDCA. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/409 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/3] ASoC: dt-bindings: nau8821: Add delay control for ADC
by Seven Lee 15 Apr '24

15 Apr '24

Change the original fixed delay to the assignment from the property. It will make it more flexible to different platforms to avoid pop noise at the beginning of recording. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 054b53954ac3..a726c5a9b067 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -103,6 +103,13 @@ properties: just limited to the left adc for design demand. type: boolean + nuvoton,adc-delay-ms: + description: Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + '#sound-dai-cells': const: 0 @@ -136,6 +143,7 @@ examples: nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; nuvoton,dmic-slew-rate = <0>; + nuvoton,nuvoton,adc-delay-ms = <125>; #sound-dai-cells = <0>; }; }; -- 2.25.1

3 5

[PATCH] ASoC: dt-bindings: renesas: add R8A779H0 V4M
by Kuninori Morimoto 15 Apr '24

15 Apr '24

Add document for R-Car V4M (R8A779H0). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 996a13f21b63..b3da278cf141 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -49,6 +49,7 @@ properties: # for Gen4 SoC - items: - const: renesas,rcar_sound-r8a779g0 # R-Car V4H + - const: renesas,rcar_sound-r8a779h0 # R-Car V4M - const: renesas,rcar_sound-gen4 # for Generic - enum: -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: renesas,rsnd: add missing renesas,rcar_sound-gen4
by Kuninori Morimoto 15 Apr '24

15 Apr '24

It is missing generic compatible for R-Car Gen4 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 0d7a6b576d88..996a13f21b63 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -55,6 +55,7 @@ properties: - renesas,rcar_sound-gen1 - renesas,rcar_sound-gen2 - renesas,rcar_sound-gen3 + - renesas,rcar_sound-gen4 reg: minItems: 1 -- 2.25.1

2 1

[PATCH 00/12] ASoC: Intel: updates for 6.10 - part4
by Pierre-Louis Bossart 15 Apr '24

15 Apr '24

More cleanups from Brent, notably the removal of the redundant cml_rt1011_rt5682 machine driver, fixes for SoundWire platforms and changes to sof_rt5682 to allow for 96+ sampling rates. For the rest of this kernel cycle, we are still working on SoundWire updates for MeteorLake (usual missing ACPI signature required for topology selection and jack detection information). We'll provide those patches as soon as they are reviewed/validated. Bard Liao (2): ASoC: Intel: sof_sdw: add missing sof_sdw_rt_amp_init for Realtek multi-function codecs ASoC: Intel: sof_rt5682: use RT5682S_PLL1 if needed Brent Lu (6): ASoC: Intel: sof_da7219: mach cleanup for adl boards ASoC: Intel: sof_da7219: mach cleanup for rpl boards ASoC: Intel: sof_da7219: add mtl_da7219_def for mtl boards ASoC: Intel: sof_realtek_common: support 4xALC1011 amplifier ASoC: Intel: sof_rt5682: support ALC1011 on cml boards ASoC: Intel: cml_rt1011_rt5682: delete driver Pierre-Louis Bossart (4): ASoC: Intel: soc-acpi: add support for HP Omen14 SoundWire configuration ASoC: Intel: sof_sdw: add JD2 quirk for HP Omen 14 ASoC: Intel: sof_sdw: add quirk for Dell SKU 0C0F ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks sound/soc/intel/boards/Kconfig | 12 +- sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/cml_rt1011_rt5682.c | 609 ------------------ sound/soc/intel/boards/sof_da7219.c | 8 + sound/soc/intel/boards/sof_realtek_common.c | 178 ++++- sound/soc/intel/boards/sof_realtek_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 37 +- sound/soc/intel/boards/sof_sdw.c | 87 ++- sound/soc/intel/boards/sof_sdw_common.h | 1 + .../intel/common/soc-acpi-intel-adl-match.c | 19 +- .../intel/common/soc-acpi-intel-cml-match.c | 2 +- .../intel/common/soc-acpi-intel-mtl-match.c | 36 ++ .../intel/common/soc-acpi-intel-rpl-match.c | 19 +- 13 files changed, 297 insertions(+), 717 deletions(-) delete mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- 2.40.1

2 13

weird behaviour with alsactl restore under pipewire
by GitHub issues - edited 14 Apr '24

14 Apr '24

alsa-project/alsa-utils issue #263 was edited from lucasreis1: So I'm at the point where I need to manually disable auto-mute after every reboot on my card. I've tried storing info using `alsactl store` and enabling `alsa-restore.service`, but I get errors when calling restore after boot (either manually or via the systemd service): ```sh Found hardware: "HDA-Intel" "ATI R6xx HDMI" "HDA:1002aa01,00aa0100,00100700" "0x1043" "0x8809" Hardware is initialized using a generic method alsactl: set_control:1339: failed to obtain info for control #1 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #2 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #3 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #4 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #5 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #6 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #7 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #8 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #9 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #10 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #11 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #12 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #13 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #14 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #15 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #16 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #17 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #18 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #19 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #20 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #21 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #22 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #23 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #24 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #25 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #26 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #27 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #28 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #33 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #34 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #35 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #36 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #37 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #38 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #39 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #40 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #41 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #42 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #43 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #44 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #45 (No such file or directory) Found hardware: "HDA-Intel" "Realtek ALC897" "HDA:10ec0897,104387fb,00100402" "0x1043" "0x87fb" Hardware is initialized using a generic method alsactl: set_control:1339: failed to obtain info for control #1 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #6 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #7 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #8 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #9 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #10 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #11 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #12 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #13 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #14 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #15 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #16 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #17 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #18 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #19 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #20 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #21 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #22 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #23 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #24 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #25 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #26 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #27 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #28 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #29 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #30 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #31 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #32 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #33 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #34 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #35 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #36 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #37 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #38 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #39 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #40 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #41 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #42 (No such file or directory) ``` If I store configs again via `alsactl store` and restore them using `alsactl restore`, everything works, but the issue returns after another reboot. Here's the output of `alsa-info.sh` if it helps: [alsa-info.txt](https://github.com/alsa-project/alsa-utils/files/14971854/al… Note that this only happens with my speakers, I have headphonse connected which work fine and are not auto-muted after every boot. Issue URL : https://github.com/alsa-project/alsa-utils/issues/263 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

weird behaviour with alsactl restore under pipewire
by GitHub issues - opened 14 Apr '24

14 Apr '24

alsa-project/alsa-utils issue #263 was opened from lucasreis1: So I'm at the point where I need to manually disable auto-mute after every reboot on my card. I've tried storing info using `alsactl store` and enabling `alsa-restore.service`, but I get errors when calling restore after boot (either manually or via the systemd service): ```sh Found hardware: "HDA-Intel" "ATI R6xx HDMI" "HDA:1002aa01,00aa0100,00100700" "0x1043" "0x8809" Hardware is initialized using a generic method alsactl: set_control:1339: failed to obtain info for control #1 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #2 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #3 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #4 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #5 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #6 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #7 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #8 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #9 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #10 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #11 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #12 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #13 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #14 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #15 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #16 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #17 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #18 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #19 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #20 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #21 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #22 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #23 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #24 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #25 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #26 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #27 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #28 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #33 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #34 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #35 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #36 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #37 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #38 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #39 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #40 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #41 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #42 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #43 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #44 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #45 (No such file or directory) Found hardware: "HDA-Intel" "Realtek ALC897" "HDA:10ec0897,104387fb,00100402" "0x1043" "0x87fb" Hardware is initialized using a generic method alsactl: set_control:1339: failed to obtain info for control #1 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #6 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #7 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #8 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #9 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #10 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #11 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #12 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #13 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #14 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #15 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #16 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #17 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #18 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #19 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #20 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #21 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #22 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #23 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #24 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #25 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #26 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #27 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #28 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #29 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #30 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #31 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #32 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #33 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #34 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #35 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #36 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #37 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #38 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #39 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #40 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #41 (No such file or directory) alsactl: set_control:1339: failed to obtain info for control #42 (No such file or directory) ``` If I store configs again via `alsactl store` and restore them using `alsactl restore`, everything works, but the issue returns after another reboot. Here's the output of `alsa-info.sh` if it helps: [alsa-info.txt](https://github.com/alsa-project/alsa-utils/files/14971854/al… Issue URL : https://github.com/alsa-project/alsa-utils/issues/263 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Compatibility with FFMPEG 7.0
by GitHub pull_request - opened 14 Apr '24

14 Apr '24

alsa-project/alsa-plugins pull request #58 was opened from eclipseo: channel_layout has been replace with ch_layout avcodec_close has been deprecated in favor of avcodec_free_context Fix #57 Request URL : https://github.com/alsa-project/alsa-plugins/pull/58 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/58.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

Re:Re:Re:Re: [PATCH] cpufreq: exit() callback is optional
by lizhe 13 Apr '24

13 Apr '24

HI I urge you to pay close attention to this incident once again. thanks Please don't let the Linux kernel community be filled with robbers and thieves! On April 12th, 2024, at 00:57, I sent the vulnerability and explained the situation to the maintainer at rafael(a)kernel.org and viresh.kumar(a)linaro.org. At 02:11 on April 12th, 2024, Rafael replied to me, asking me to submit it to the main branch. At 07:18 on April 12th, 2024, I submitted the vulnerability to the main branch. At 11:24 on April 12th, 2024, Viresh Kumar replied to me, confirming that the vulnerability was legitimate. At 13:49 on April 12th, 2024, Viresh Kumar released the patch to the main list. Here is the patchs I submitted to the main line . cpufreq: Remove redundant exit() functions https://lore.kernel.org/all/20240409141909.4413-1-sensor1010@163.com/ cpufreq: Convert to exit callback returning void https://lore.kernel.org/all/20240410132247.3587-1-sensor1010@163.com/ cpufreq: Covert to offline callback returning void https://lore.kernel.org/all/20240411150620.19262-1-sensor1010@163.com/ cpufreq: Fixed kernel crash caused by cpufreq issues https://lore.kernel.org/all/20240411231818.2471-1-sensor1010@163.com/ Viresh Kumar, are you not a robber and thief? As a maintainer, do you really not care about the healthy development of the community and only defend your own interests? It was me who first released the patch and publicly disclosed the vulnerability to the main branch. Viresh Kumar only understood the principle of the vulnerability after seeing my submitted patch. Moreover, I communicated with you later and requested to add my signature to the patch, but you refused to communicate. Your behavior fully demonstrates that you are a selfish person. I don't understand why there can be such a selfish person in a country where Buddhism was born, and this person is even a maintainer! At 2024-04-12 22:41:27, "lizhe" <sensor1010(a)163.com> wrote: Let's both take a step back and add my signature to the patch, since I was the one who discovered and reported the vulnerability. HI i will review how it all started. 1. i reported the vulnerability to the maintainer. 2. manitaner replay me 3. i report to main line 4. you reply me 5 . you report to main line You submitted the patch to the mainline after you learned about the vulnerability from the patch I submitted At 2024-04-12 21:54:00, "lizhe" <sensor1010(a)163.com> wrote: Hi Viresh Kumar please add my name to the signature. because i discovered this vulnerability. please reply me. thanks At 2024-04-12 18:28:56, "Greg KH" <gregkh(a)linuxfoundation.org> wrote: >On Fri, Apr 12, 2024 at 02:51:08PM +0530, Viresh Kumar wrote: >> On 12-04-24, 17:05, lizhe wrote: >> > I found it first and submitted it to the main line first. Please be fair and just. Let him withdraw his patch. >> >> Well, I decided not to reply to your emails anymore but this needs to >> be clarified a bit now. >> >> You sent a lot of patches, over and over again and it was a mess. I >> saw the this [1] series first and went over to read the code and fixed >> an issue which I found (by the $subject patch). >> >> Later I read your other patch [2], which I Acked roughly two hours >> back and yes you did send a patch that fixed the problem partially. I >> never saw it earlier, which is okay and it happens. Despite me giving >> an Ack to your patch, you have sent half-a-dozen more emails.. >> >> Then I posted a newer version of my patch some time back, removing the >> bits you already fixed [3]. >> >> That is all one side of the story. But all the noise you have created >> here has really demotivated people to review your stuff now. >> >> -- >> Viresh >> >> [1] https://lore.kernel.org/all/20240410132132.3526-1-sensor1010@163.com/ >> [2] https://lore.kernel.org/all/20240411231818.2471-1-sensor1010@163.com/ >> [3] https://lore.kernel.org/all/68294ce534668c6ab3b71a1b3e6650227c6e1f20.171291… > >Thanks for the links, I don't see that you did anything wrong here at >all. > >Lizhe, you seem to be confused as to how kernel development works. I >suggest you take some time off and read up on how this all is supposed >to happen and then work with some local people, in person, to get this >figured out first, before submitting changes again. > >thanks, > >greg k-h

1 0

[PATCH] Fix volumn control of ThinkBook 16P Gen4
by zhanghuayu1233＠qq.com 13 Apr '24

13 Apr '24

From: Huayu Zhang <zhanghuayu1233(a)qq.com> change HDA & AMP configuration from ALC287_FIXUP_CS35L41_I2C_2 to ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD for ThinkBook 16P Gen4 models to fix volumn control issue (cannot fully mute). Signed-off-by: Huayu Zhang <zhanghuayu1233(a)qq.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6940bc4ec39..1aef223307bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10396,8 +10396,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), -- 2.34.1

1 0

Re:Re:Re:Re: [PATCH] cpufreq: exit() callback is optional
by lizhe 13 Apr '24

13 Apr '24

HI I urge you to pay close attention to this incident once again. thanks Please don't let the Linux kernel community be filled with robbers and thieves! On April 12th, 2024, at 00:57, I sent the vulnerability and explained the situation to the maintainer at rafael(a)kernel.org and viresh.kumar(a)linaro.org. At 02:11 on April 12th, 2024, Rafael replied to me, asking me to submit it to the main branch. At 07:18 on April 12th, 2024, I submitted the vulnerability to the main branch. At 11:24 on April 12th, 2024, Viresh Kumar replied to me, confirming that the vulnerability was legitimate. At 13:49 on April 12th, 2024, Viresh Kumar released the patch to the main list. Viresh Kumar, are you not a robber and thief? As a maintainer, do you really not care about the healthy development of the community and only defend your own interests? It was me who first released the patch and publicly disclosed the vulnerability to the main branch. Viresh Kumar only understood the principle of the vulnerability after seeing my submitted patch. Even if my patch was not of high quality, I could immediately improve it. Moreover, I communicated with you later and requested to add my signature to the patch, but you refused to communicate. Your behavior fully demonstrates that you are a selfish person. I don't understand why there can be such a selfish person in a country where Buddhism was born, and this person is even a maintainer! At 2024-04-12 22:41:27, "lizhe" <sensor1010(a)163.com> wrote: Let's both take a step back and add my signature to the patch, since I was the one who discovered and reported the vulnerability. HI i will review how it all started. 1. i reported the vulnerability to the maintainer. 2. manitaner replay me 3. i report to main line 4. you reply me 5 . you report to main line You submitted the patch to the mainline after you learned about the vulnerability from the patch I submitted At 2024-04-12 21:54:00, "lizhe" <sensor1010(a)163.com> wrote: Hi Viresh Kumar please add my name to the signature. because i discovered this vulnerability. please reply me. thanks At 2024-04-12 18:28:56, "Greg KH" <gregkh(a)linuxfoundation.org> wrote: >On Fri, Apr 12, 2024 at 02:51:08PM +0530, Viresh Kumar wrote: >> On 12-04-24, 17:05, lizhe wrote: >> > I found it first and submitted it to the main line first. Please be fair and just. Let him withdraw his patch. >> >> Well, I decided not to reply to your emails anymore but this needs to >> be clarified a bit now. >> >> You sent a lot of patches, over and over again and it was a mess. I >> saw the this [1] series first and went over to read the code and fixed >> an issue which I found (by the $subject patch). >> >> Later I read your other patch [2], which I Acked roughly two hours >> back and yes you did send a patch that fixed the problem partially. I >> never saw it earlier, which is okay and it happens. Despite me giving >> an Ack to your patch, you have sent half-a-dozen more emails.. >> >> Then I posted a newer version of my patch some time back, removing the >> bits you already fixed [3]. >> >> That is all one side of the story. But all the noise you have created >> here has really demotivated people to review your stuff now. >> >> -- >> Viresh >> >> [1] https://lore.kernel.org/all/20240410132132.3526-1-sensor1010@163.com/ >> [2] https://lore.kernel.org/all/20240411231818.2471-1-sensor1010@163.com/ >> [3] https://lore.kernel.org/all/68294ce534668c6ab3b71a1b3e6650227c6e1f20.171291… > >Thanks for the links, I don't see that you did anything wrong here at >all. > >Lizhe, you seem to be confused as to how kernel development works. I >suggest you take some time off and read up on how this all is supposed >to happen and then work with some local people, in person, to get this >figured out first, before submitting changes again. > >thanks, > >greg k-h

1 0

qcom: sdm845: MM1: fixes
by GitHub pull_request - opened 13 Apr '24

13 Apr '24

alsa-project/alsa-ucm-conf pull request #408 was opened from jenneron: I'm testing it on a device which is not in upstream yet, but it uses this configuration It seems that the only device in upstream using it is Lenovo Yoga C630 laptop and we got it tested as well https://oftc.irclog.whitequark.org/aarch64-laptops/2024-04-06#33084827 The changes are pretty straightforward, check commit messages of each commit Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/408 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/408.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] dt-bindings: omap-mcpdm: Convert to DT schema
by Mighty 12 Apr '24

12 Apr '24

From: Mithil Bavishi <bavishimithil(a)gmail.com> Convert the OMAP4+ McPDM bindings to DT schema. Signed-off-by: Mighty <bavishimithil(a)gmail.com> --- .../devicetree/bindings/sound/omap-mcpdm.txt | 30 ---------- .../bindings/sound/ti,omap-mcpdm.yaml | 59 +++++++++++++++++++ 2 files changed, 59 insertions(+), 30 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,omap-mcpdm.yaml diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt deleted file mode 100644 index ff98a0cb5..000000000 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ /dev/null @@ -1,30 +0,0 @@ -* Texas Instruments OMAP4+ McPDM - -Required properties: -- compatible: "ti,omap4-mcpdm" -- reg: Register location and size as an array: - <MPU access base address, size>, - <L3 interconnect address, size>; -- interrupts: Interrupt number for McPDM -- ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" - -Example: - -mcpdm: mcpdm@40132000 { - compatible = "ti,omap4-mcpdm"; - reg = <0x40132000 0x7f>, /* MPU private access */ - <0x49032000 0x7f>; /* L3 Interconnect */ - interrupts = <0 112 0x4>; - interrupt-parent = <&gic>; - ti,hwmods = "mcpdm"; -}; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/Documentation/devicetree/bindings/sound/ti,omap-mcpdm.yaml b/Documentation/devicetree/bindings/sound/ti,omap-mcpdm.yaml new file mode 100644 index 000000000..4d5d37e98 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,omap-mcpdm.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,omap-mcpdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: OMAP McPDM + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +description: + OMAP ALSA SoC DAI driver using McPDM port used by TWL6040 + +properties: + compatible: + const: ti,omap4-mcpdm + + reg: + description: + Register location and size as an array + <MPU access base address, size>, + <L3 interconnect address, size>; + + interrupts: + maxItems: 1 + + ti,hwmods: + maxItems: 1 + + clocks: + description: phandle for the pdmclk provider, likely <&twl6040> + + clock-names: + description: Must be "pdmclk" + + +required: + - compatible + - reg + - interrupts + - ti,hwmods + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + mcpdm@0 { + compatible = "ti,omap4-mcpdm"; + reg = <0x40132000 0x7f>, /* MPU private access */ + <0x49032000 0x7f>; /* L3 Interconnect */ + interrupts = <0 112 0x4>; + interrupt-parent = <&gic>; + ti,hwmods = "mcpdm"; + clocks = <&twl6040>; + clock-names = "pdmclk"; + }; -- 2.34.1

3 10

UCM vs SOF vs HDMI passthrough
by Péter Ujfalusi 12 Apr '24

12 Apr '24

Hi, afaik for HDMI passthrough there should be a usable PCM mapping present for user space: aplay -L | grep hdmi hdmi:CARD=PCH,DEV=0 hdmi:CARD=PCH,DEV=1 hdmi:CARD=PCH,DEV=2 or similar. SOF with IPC4 (Meteor Lake and newer or some TGL/ADL setups with 2024.03 release) it is possible to use passthrough via HDMI technically but we are lacking the "hdmi:..." PCMs, so in reality it is not usable for applications. Afaik. So far I managed to cook up two ways of doing this, but neither one is good enough as they will create the "hdmi:..." for both IPC3 and IPC4 setups and IPC3 firmware cannot support passthrough. Setups: sof-hda-dsp cards have HDMI PCMs: pcm3-5 sof-soundwire cards have HDMI PCMs: pcm5-7 [A] use card profile Add these to /usr/share/alsa/cards/aliases.conf: sof-hda-dsp cards.sof-hda-dsp sof-soundwire cards.sof-soundwire Create the minimal sof-hda-ds.conf and sof-soundwire.conf to handle the two types of HDMI PCM numbering. [B] Using UCM https://github.com/ujfalusi/alsa-ucm-conf/tree/topic/sof-hdmi It is using the BootSequence[] to create three files: [1] /var/lib/alsa/conf.d/42-sof-hdmi.conf [2] /var/lib/alsa/card[card_number].conf.d/30-sof-hdmi-common.conf [3] /var/lib/alsa/card[card_number].conf.d/31-sof-hdmi.conf [1] includes the pcm/iec958.conf and pcm/hdmi.conf to global space of alsaconf to be used by the card macros [2] Card specific macros for hdmi PCM definition, ampping [3] Card specific definitions of the three HDMI port In both case I have the needed PCMs: aplay -L | grep hdmi hdmi:CARD=sofhdadsp,DEV=0 hdmi:CARD=sofhdadsp,DEV=1 hdmi:CARD=sofhdadsp,DEV=2 or hdmi:CARD=sofsoundwire,DEV=0 hdmi:CARD=sofsoundwire,DEV=1 hdmi:CARD=sofsoundwire,DEV=2 But it is unconditional and they will be created for all sof-hda-dsp and sof-soundwire cards. I would really appreciate some guidance on how we can achieve this correctly (the code for the UCM way leaves much to be desired, I know). How can this be done conditionally? We can set the HDMI PCMs to advertise IEC958_SUBFRAME_LE for IPC4 only, but how can we use that? Should we add some parameter to 'amixer -c0 info', like ipc_version:4/3, so if the ipc_version is 3 or missing we assume that passthrough is not supported? Thanks for the help, Péter

2 2

[PATCH RESEND] soundwire: qcom: allow multi-link on newer devices
by Krzysztof Kozlowski 12 Apr '24

12 Apr '24

Newer Qualcomm SoCs like X1E80100 might come with four speakers spread over two Soundwire controllers, thus they need a multi-link Soundwire stream runtime. Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Almost four months, so let's resend this one. Add Rb tag. --- drivers/soundwire/qcom.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 82b3ca5a405a..a97412c8671e 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -920,6 +920,18 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) return 0; } +static int qcom_swrm_read_prop(struct sdw_bus *bus) +{ + struct qcom_swrm_ctrl *ctrl = to_qcom_sdw(bus); + + if (ctrl->version >= SWRM_VERSION_2_0_0) { + bus->multi_link = true; + bus->hw_sync_min_links = 3; + } + + return 0; +} + static enum sdw_command_response qcom_swrm_xfer_msg(struct sdw_bus *bus, struct sdw_msg *msg) { @@ -1078,6 +1090,7 @@ static const struct sdw_master_port_ops qcom_swrm_port_ops = { }; static const struct sdw_master_ops qcom_swrm_ops = { + .read_prop = qcom_swrm_read_prop, .xfer_msg = qcom_swrm_xfer_msg, .pre_bank_switch = qcom_swrm_pre_bank_switch, .post_bank_switch = qcom_swrm_post_bank_switch, @@ -1196,6 +1209,15 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { + /* + * For streams with multiple masters: + * Allocate ports only for devices connected to this master. + * Such devices will have ports allocated by their own master + * and its qcom_swrm_stream_alloc_ports() call. + */ + if (ctrl->bus.id != m_rt->bus->id) + continue; + if (m_rt->direction == SDW_DATA_DIR_RX) { maxport = ctrl->num_dout_ports; port_mask = &ctrl->dout_port_mask; -- 2.34.1

2 5

[PATCH v1] ASoC: cs35l41: Update DSP1RX5/6 Sources for DSP config
by Stefan Binding 12 Apr '24

12 Apr '24

Currently, all ASoC systems are set to use VPMON for DSP1RX5_SRC, however, this is required only for internal boost systems. External boost systems require VBSTMON instead of VPMON to be the input to DSP1RX5_SRC. Shared Boost Active acts like Internal boost (requires VPMON). Shared Boost Passive acts like External boost (requires VBSTMON) All systems require DSP1RX6_SRC to be set to VBSTMON. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index dfb4ce53491b..f8e57a2fc3e3 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -1094,6 +1094,7 @@ static int cs35l41_handle_pdata(struct device *dev, struct cs35l41_hw_cfg *hw_cf static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) { struct wm_adsp *dsp; + uint32_t dsp1rx5_src; int ret; dsp = &cs35l41->dsp; @@ -1113,16 +1114,29 @@ static int cs35l41_dsp_init(struct cs35l41_private *cs35l41) return ret; } - ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX5_SRC, - CS35L41_INPUT_SRC_VPMON); + switch (cs35l41->hw_cfg.bst_type) { + case CS35L41_INT_BOOST: + case CS35L41_SHD_BOOST_ACTV: + dsp1rx5_src = CS35L41_INPUT_SRC_VPMON; + break; + case CS35L41_EXT_BOOST: + case CS35L41_SHD_BOOST_PASS: + dsp1rx5_src = CS35L41_INPUT_SRC_VBSTMON; + break; + default: + dev_err(cs35l41->dev, "wm_halo_init failed - Invalid Boost Type: %d\n", + cs35l41->hw_cfg.bst_type); + goto err_dsp; + } + + ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX5_SRC, dsp1rx5_src); if (ret < 0) { - dev_err(cs35l41->dev, "Write INPUT_SRC_VPMON failed: %d\n", ret); + dev_err(cs35l41->dev, "Write DSP1RX5_SRC: %d failed: %d\n", dsp1rx5_src, ret); goto err_dsp; } - ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX6_SRC, - CS35L41_INPUT_SRC_CLASSH); + ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX6_SRC, CS35L41_INPUT_SRC_VBSTMON); if (ret < 0) { - dev_err(cs35l41->dev, "Write INPUT_SRC_CLASSH failed: %d\n", ret); + dev_err(cs35l41->dev, "Write CS35L41_INPUT_SRC_VBSTMON failed: %d\n", ret); goto err_dsp; } ret = regmap_write(cs35l41->regmap, CS35L41_DSP1_RX7_SRC, -- 2.34.1

3 2

[PATCH][next] ASoC: codecs: tas2780: remove redundant assignments to variable ret
by Colin Ian King 12 Apr '24

12 Apr '24

Variable ret is being assigned a value that is never read in a couple of places. The variable is being re-assigned later on. The assignments are redundant and can be removed. Cleans up clang scan build warning: sound/soc/codecs/tas2780.c:84:2: warning: Value stored to 'ret' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/tas2780.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 41076be23854..972e8ea5ebde 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -71,7 +71,7 @@ static int tas2780_codec_resume(struct snd_soc_component *component) { struct tas2780_priv *tas2780 = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret; ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_ACTIVE); @@ -81,7 +81,6 @@ static int tas2780_codec_resume(struct snd_soc_component *component) __func__, ret); goto err; } - ret = 0; regcache_cache_only(tas2780->regmap, false); ret = regcache_sync(tas2780->regmap); err: -- 2.39.2

2 1

[PATCH] MAINTAINERS: adjust file entry in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
by Lukas Bulwahn 12 Apr '24

12 Apr '24

Commit 8167bd1c8a45 ("ASoC: dt-bindings: ti,pcm1681: Convert to dtschema") converts ti,pcm1681.txt to ti,pcm1681.yaml, but misses to adjust the file entry in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS. Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about a broken reference. Adjust the file entry in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS after this conversion. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)redhat.com> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index e5431f06ab55..d39e0f2556f7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -21976,7 +21976,7 @@ F: Documentation/devicetree/bindings/sound/tas2552.txt F: Documentation/devicetree/bindings/sound/tas2562.yaml F: Documentation/devicetree/bindings/sound/tas2770.yaml F: Documentation/devicetree/bindings/sound/tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.txt +F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/tlv320adcx140.yaml -- 2.44.0

2 1

[PATCH v2] soundwire: qcom: allow multi-link on newer devices
by Krzysztof Kozlowski 11 Apr '24

11 Apr '24

Newer Qualcomm SoCs like X1E80100 might come with four speakers spread over two Soundwire controllers, thus they need a multi-link Soundwire stream runtime. Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Only rebase (slightly different context) --- drivers/soundwire/qcom.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb70afe64fcc..ce5cf3ecceb5 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -905,6 +905,18 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) return 0; } +static int qcom_swrm_read_prop(struct sdw_bus *bus) +{ + struct qcom_swrm_ctrl *ctrl = to_qcom_sdw(bus); + + if (ctrl->version >= SWRM_VERSION_2_0_0) { + bus->multi_link = true; + bus->hw_sync_min_links = 3; + } + + return 0; +} + static enum sdw_command_response qcom_swrm_xfer_msg(struct sdw_bus *bus, struct sdw_msg *msg) { @@ -1056,6 +1068,7 @@ static const struct sdw_master_port_ops qcom_swrm_port_ops = { }; static const struct sdw_master_ops qcom_swrm_ops = { + .read_prop = qcom_swrm_read_prop, .xfer_msg = qcom_swrm_xfer_msg, .pre_bank_switch = qcom_swrm_pre_bank_switch, }; @@ -1173,6 +1186,15 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { + /* + * For streams with multiple masters: + * Allocate ports only for devices connected to this master. + * Such devices will have ports allocated by their own master + * and its qcom_swrm_stream_alloc_ports() call. + */ + if (ctrl->bus.id != m_rt->bus->id) + continue; + if (m_rt->direction == SDW_DATA_DIR_RX) { maxport = ctrl->num_dout_ports; port_mask = &ctrl->dout_port_mask; -- 2.34.1

2 4

[PATCH v2] slimbus: qcom-ngd-ctrl: Add timeout for wait operation
by Viken Dadhaniya 11 Apr '24

11 Apr '24

In current driver qcom_slim_ngd_up_worker() indefinitely waiting for ctrl->qmi_up completion object. This is resulting in workqueue lockup on Kthread. Added wait_for_completion_interruptible_timeout to allow the thread to wait for specific timeout period and bail out instead waiting infinitely. Fixes: a899d324863a ("slimbus: qcom-ngd-ctrl: add Sub System Restart support") Cc: stable(a)vger.kernel.org Reviewed-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- v1 -> v2: - Remove macro and add value inline. - add fix, cc and review tag. --- --- drivers/slimbus/qcom-ngd-ctrl.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..a09a26bf4988 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1451,7 +1451,11 @@ static void qcom_slim_ngd_up_worker(struct work_struct *work) ctrl = container_of(work, struct qcom_slim_ngd_ctrl, ngd_up_work); /* Make sure qmi service is up before continuing */ - wait_for_completion_interruptible(&ctrl->qmi_up); + if (!wait_for_completion_interruptible_timeout(&ctrl->qmi_up, + msecs_to_jiffies(MSEC_PER_SEC))) { + dev_err(ctrl->dev, "QMI wait timeout\n"); + return; + } mutex_lock(&ctrl->ssr_lock); qcom_slim_ngd_enable(ctrl, true); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 1

[PATCH] slimbus: qcom-ctrl: fix module autoloading
by Krzysztof Kozlowski 11 Apr '24

11 Apr '24

Add MODULE_DEVICE_TABLE(), so the module could be properly autoloaded based on the alias from of_device_id table. Pin controllers are considered core components, so usually they are built-in, however these Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- drivers/slimbus/qcom-ctrl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index 400b7b385a44..f7d04b50c973 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -718,6 +718,7 @@ static const struct of_device_id qcom_slim_dt_match[] = { { .compatible = "qcom,slim", }, {} }; +MODULE_DEVICE_TABLE(of, qcom_slim_dt_match); static struct platform_driver qcom_slim_driver = { .probe = qcom_slim_probe, -- 2.34.1

3 2

[PATCH] slimbus: Convert to platform remove callback returning void
by Uwe Kleine-König 11 Apr '24

11 Apr '24

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new(), which already returns void. Eventually after all drivers are converted, .remove_new() will be renamed to .remove(). Trivially convert the slimbus drivers from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- drivers/slimbus/qcom-ctrl.c | 5 ++--- drivers/slimbus/qcom-ngd-ctrl.c | 11 ++++------- 2 files changed, 6 insertions(+), 10 deletions(-) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index 400b7b385a44..7d632fad1300 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -626,7 +626,7 @@ static int qcom_slim_probe(struct platform_device *pdev) return ret; } -static int qcom_slim_remove(struct platform_device *pdev) +static void qcom_slim_remove(struct platform_device *pdev) { struct qcom_slim_ctrl *ctrl = platform_get_drvdata(pdev); @@ -635,7 +635,6 @@ static int qcom_slim_remove(struct platform_device *pdev) clk_disable_unprepare(ctrl->rclk); clk_disable_unprepare(ctrl->hclk); destroy_workqueue(ctrl->rxwq); - return 0; } /* @@ -721,7 +720,7 @@ static const struct of_device_id qcom_slim_dt_match[] = { static struct platform_driver qcom_slim_driver = { .probe = qcom_slim_probe, - .remove = qcom_slim_remove, + .remove_new = qcom_slim_remove, .driver = { .name = "qcom_slim_ctrl", .of_match_table = qcom_slim_dt_match, diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..a5aa364f562e 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1675,14 +1675,12 @@ static int qcom_slim_ngd_ctrl_probe(struct platform_device *pdev) return ret; } -static int qcom_slim_ngd_ctrl_remove(struct platform_device *pdev) +static void qcom_slim_ngd_ctrl_remove(struct platform_device *pdev) { platform_driver_unregister(&qcom_slim_ngd_driver); - - return 0; } -static int qcom_slim_ngd_remove(struct platform_device *pdev) +static void qcom_slim_ngd_remove(struct platform_device *pdev) { struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); @@ -1697,7 +1695,6 @@ static int qcom_slim_ngd_remove(struct platform_device *pdev) kfree(ctrl->ngd); ctrl->ngd = NULL; - return 0; } static int __maybe_unused qcom_slim_ngd_runtime_idle(struct device *dev) @@ -1740,7 +1737,7 @@ static const struct dev_pm_ops qcom_slim_ngd_dev_pm_ops = { static struct platform_driver qcom_slim_ngd_ctrl_driver = { .probe = qcom_slim_ngd_ctrl_probe, - .remove = qcom_slim_ngd_ctrl_remove, + .remove_new = qcom_slim_ngd_ctrl_remove, .driver = { .name = "qcom,slim-ngd-ctrl", .of_match_table = qcom_slim_ngd_dt_match, @@ -1749,7 +1746,7 @@ static struct platform_driver qcom_slim_ngd_ctrl_driver = { static struct platform_driver qcom_slim_ngd_driver = { .probe = qcom_slim_ngd_probe, - .remove = qcom_slim_ngd_remove, + .remove_new = qcom_slim_ngd_remove, .driver = { .name = QCOM_SLIM_NGD_DRV_NAME, .pm = &qcom_slim_ngd_dev_pm_ops, base-commit: 33e1d31873f87d119e5120b88cd350efa68ef276 -- 2.43.0

2 2

[PATCH v2] slimbus: qcom-ngd-ctrl: Reduce auto suspend delay
by Viken Dadhaniya 11 Apr '24

11 Apr '24

Currently we have auto suspend delay of 1s which is very high and it takes long time to driver for runtime suspend after use case is done. Hence to optimize runtime PM ops, reduce auto suspend delay to 100ms. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- v1 -> v2: - Remove macro and add value inline. - Link to v1: https://lore.kernel.org/linux-arm-msm/d784e46d-974d-4bf3-a2d3-491e7ad19701@… --- --- drivers/slimbus/qcom-ngd-ctrl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..5e132273c6c4 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -81,7 +81,6 @@ #define SLIM_USR_MC_DISCONNECT_PORT 0x2E #define SLIM_USR_MC_REPEAT_CHANGE_VALUE 0x0 -#define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 #define SLIM_LA_MGR 0xFF @@ -1571,7 +1570,7 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); - pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); + pm_runtime_set_autosuspend_delay(dev, 100); pm_runtime_set_suspended(dev); pm_runtime_enable(dev); pm_runtime_get_noresume(dev); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

3 2

[PATCH v1 0/7] Add features, fixes and laptops for CS35L41 HDA
by Stefan Binding 11 Apr '24

11 Apr '24

This chain adds a number of improvements for CS35L41, including a new firmware loading feature, as well as some additional cleanup. Patch 1 adds support for a new firmware tuning file which is used to set special tuning parameters used by the firmware. The HP Omen laptop added in patches 2 and 3 use such a feature. Patch 4 changes the DSP1RX5/6 Source used by the firmware to depend on the boost type. A similar change in the ASoC driver will be needed later. Patches 5 and 6 are cleanup patches, with the first of those re-using cs-amp-lib for calibration which as added for CS35L56 drivers, ensuring that all laptops are using the correct calibration values. Patch 7 fixes a small error in the CS35L41 Property table for the Lenovo Legion slim 7 16ARHA7. Richard Fitzgerald (1): ALSA: hda: cs35l41: Remove redundant argument to cs35l41_request_firmware_file() Stefan Binding (6): ALSA: hda: cs35l41: Set the max PCM Gain using tuning setting ALSA: hda: cs35l41: Support HP Omen models without _DSD ALSA: hda/realtek: Add quirks for HP Omen models using CS35L41 ALSA: hda: cs35l41: Update DSP1RX5/6 Sources for DSP config ALSA: hda: cs35l41: Use shared cs-amp-lib to apply calibration ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7 16ARHA7 include/sound/cs35l41.h | 5 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l41_hda.c | 347 +++++++++++++++++++-------- sound/pci/hda/cs35l41_hda.h | 6 + sound/pci/hda/cs35l41_hda_property.c | 8 +- sound/pci/hda/patch_realtek.c | 4 + 6 files changed, 265 insertions(+), 107 deletions(-) -- 2.34.1

2 9

[PATCH] ASoC: wm_adsp: Include array_size.h
by Richard Fitzgerald 10 Apr '24

10 Apr '24

Explicitly #include array_size.h for the ARRAY_SIZE() macro. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7d5c096e06cd..9a89682ebc1f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -7,6 +7,7 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ +#include <linux/array_size.h> #include <linux/ctype.h> #include <linux/module.h> #include <linux/moduleparam.h> -- 2.39.2

2 1

[PATCH] ASoC: cs35l56: Include array_size.h
by Richard Fitzgerald 10 Apr '24

10 Apr '24

Explicitly #include array_size.h for the source files that use ARRAY_SIZE(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index ec1d95e57061..3aa39f5923eb 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -5,6 +5,7 @@ // Copyright (C) 2023 Cirrus Logic, Inc. and // Cirrus Logic International Semiconductor Ltd. +#include <linux/array_size.h> #include <linux/firmware/cirrus/wmfw.h> #include <linux/gpio/consumer.h> #include <linux/regmap.h> diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 6331b8c6136e..dfd703d9e12f 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -6,6 +6,7 @@ // Cirrus Logic International Semiconductor Ltd. #include <linux/acpi.h> +#include <linux/array_size.h> #include <linux/completion.h> #include <linux/debugfs.h> #include <linux/delay.h> -- 2.39.2

2 1

[PATCH v4 00/18] SoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 10 Apr '24

10 Apr '24

Changes in v4: - Rebased over next-20240409 - Dropped the first 4 patches from v3 as were already picked by Mark Changes in v3: - Added audio-routing names in enum in all yaml files - Added mention of disallowing old and new properties together in commit message of bindings patches - Fixed validation errors with sound-card-common.yaml inclusion due to missing model property in examples - Removed `else` enforcing headset-codec/speaker-codecs requirement if xxx-dai-link not present to avoid future commit noise as the deprecated statement will disallow deprecated properties as required Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (18): ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +- .../bindings/sound/mt8195-mt6359.yaml | 134 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 199 +++ .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 25 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 27 files changed, 1613 insertions(+), 2195 deletions(-) delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

2 32

Thinkpad with ALC1318 has a chance of damaging the IC
by Kailang 10 Apr '24

10 Apr '24

Hi Takashi, Thinkpad with ALC1318 has a chance of damaging the IC at S4 resume. BR, Kailang

1 0

[PATCH v2 00/13] ASoC: ti: davinci-i2s: Add features to McBSP driver
by Bastien Curutchet 10 Apr '24

10 Apr '24

This series aims to add some features to McBSP driver. Convert bindings from .txt to .yaml. Add possibility to use an external clock as sample rate generator's input. Add handling of new formats (TDM, S24_LE, BP_FC). Enable the detection of unexpected frame pulses. Set the clock free-running mode according to SND_SOC_DAIFMT_[GATED/CONT] configuration in DAI format. Add ti,T1-framing[tx/rx] properties in DT. They allow to set the data delay to two bit-clock periods. This has been tested on a platform designed off of the DAVINCI/OMAP-L138 connected to 3 daisy-chained AD7767. An external clock drives the sample rate generator through the CLKS pin. The hardware I have only allowed me to test acquisition side of McBSP. It is connected to a 6 channels TDM and acts as Bit clock provider and Frame clock consumer. Change log v1 -> v2: PATCH 1 (bindings): * Drop power-domains property's description * Drop the unused label 'mcbsp0' in example * Add <> around each entry of the 'dmas' property * Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' PATCH 2 (bindings): * Drop the 'ti,enable-sync-err' flag * Drop the 'ti,disable-free-run' flag * Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' PATCH 4: * In probe() use dev_err for fixed error PATCH 7 (TDM): * set playback.max_channels to 128 * Add a check on tx_mask as the one done for rx_mask * Allow TDM with BP_FP format PATCH 9: * Detection of unexpected frame pulses is enabled by default PATCH 10: * Free-running mode is selected by the DAI format through SND_SOC_DAIFMT_[CONT/GATED] PATCH 12: * drop the 'ti,drive-dx' property * add 'ti,T1-framing-[rx/tx]' properties PATCH 13: * Drop the drive_dx part * Add support for 'T1 framing' with data delay set to 2 bit-clock periods Bastien Curutchet (13): ASoC: dt-bindings: davinci-mcbsp: convert McBSP bindings to yaml schema ASoC: dt-bindings: davinci-mcbsp: Add optional clock ASoC: ti: davinci-i2s: Remove the unused clk_input_pin attribute ASoC: ti: davinci-i2s: Replace dev_err with dev_err_probe ASoC: ti: davinci-i2s: Use external clock to drive sample rate generator ASoC: ti: davinci-i2s: Delete unnecessary assignment ASoC: ti: davinci-i2s: Add TDM support ASoC: ti: davinci-i2s: Add handling of BP_FC format ASoC: ti: davinci-i2s: Enable unexpected frame pulses detection ASoC: ti: davinci-i2s: Link free-run mode to SND_SOC_DAIFMT_[GATED/CONT] ASoC: ti: davinci-i2s: Add S24_LE to supported formats ASoC: dt-bindings: davinci-mcbsp: Add the 'ti,T1-framing-{rx/tx}' flags ASoC: ti: davinci-i2s: Add T1 framing support .../bindings/sound/davinci-mcbsp.txt | 50 ---- .../bindings/sound/davinci-mcbsp.yaml | 113 +++++++ include/linux/platform_data/davinci_asp.h | 15 - sound/soc/ti/davinci-i2s.c | 278 ++++++++++++++---- 4 files changed, 333 insertions(+), 123 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml -- 2.44.0

4 16

[PATCH 0/3] ASoC: SOF: misc fixes
by Pierre-Louis Bossart 10 Apr '24

10 Apr '24

We somehow missed the default path for DSP libraries for LNL, and need to restrict support for D0i3 w/ IPC4. Also add debugfs support for firmware profile information so that sof-test scripts can show what is being tested. Pierre-Louis Bossart (2): ASoC: SOF: Intel: add default firmware library path for LNL ASoC: SOF: debug: show firmware/topology prefix/names Ranjani Sridharan (1): ASoC: SOF: pcm: Restrict DSP D0i3 during S0ix to IPC3 sound/soc/sof/debug.c | 18 ++++++++++++++++++ sound/soc/sof/intel/pci-lnl.c | 3 +++ sound/soc/sof/ipc3-pcm.c | 1 + sound/soc/sof/pcm.c | 13 ++++++------- sound/soc/sof/sof-audio.h | 2 ++ 5 files changed, 30 insertions(+), 7 deletions(-) -- 2.40.1

2 4

[PATCH 0/4] ASoC: cs35l56: Fixes to handling of ASP1 config registers
by Richard Fitzgerald 10 Apr '24

10 Apr '24

This chain fixes some problems with some previous patches for handling the ASP1 config registers. The root of the problem is that the ownership of these registers can be either with the firmware or the driver, and that the chip has to be soft-reset after downloading the firmware. This chain adds and uses a regmap_read_bypassed() function so that the driver can leave the regmap in cache-only until the chip has rebooted, but still poll a register to detect when the chip has rebooted. Richard Fitzgerald (4): regmap: Add regmap_read_bypassed() ALSA: hda: cs35l56: Exit cache-only after cs35l56_wait_for_firmware_boot() ASoC: cs35l56: Fix unintended bus access while resetting amp ASoC: cs35l56: Prevent overwriting firmware ASP config drivers/base/regmap/regmap.c | 37 ++++++++++++++ include/linux/regmap.h | 8 +++ include/sound/cs35l56.h | 2 + sound/pci/hda/cs35l56_hda.c | 4 ++ sound/soc/codecs/cs35l56-sdw.c | 2 - sound/soc/codecs/cs35l56-shared.c | 83 ++++++++++++++++++++----------- sound/soc/codecs/cs35l56.c | 26 +++++++++- 7 files changed, 130 insertions(+), 32 deletions(-) -- 2.39.2

2 6

[PATCH 2/2] ASoC: rt722-sdca: modify channel number to support 4 channels
by Jack Yu 10 Apr '24

10 Apr '24

Channel numbers of dmic supports 4 channels, modify channels_max regarding to this issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 87bd50e761fc..e5bd9ef812de 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1330,7 +1330,7 @@ static struct snd_soc_dai_driver rt722_sdca_dai[] = { .capture = { .stream_name = "DP6 DMic Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 4, .rates = RT722_STEREO_RATES, .formats = RT722_FORMATS, }, -- 2.34.1

2 1

[PATCH 1/2] ASoC: rt722-sdca: add headset microphone vrefo setting
by Jack Yu 10 Apr '24

10 Apr '24

Add vrefo settings to fix jd and headset mic recording issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca.c | 25 +++++++++++++++++++------ sound/soc/codecs/rt722-sdca.h | 3 +++ 2 files changed, 22 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index e0ea3a23f7cc..87bd50e761fc 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1439,9 +1439,12 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) int loop_check, chk_cnt = 100, ret; unsigned int calib_status = 0; - /* Read eFuse */ - rt722_sdca_index_write(rt722, RT722_VENDOR_SPK_EFUSE, RT722_DC_CALIB_CTRL, - 0x4808); + /* Config analog bias */ + rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_ANALOG_BIAS_CTL3, + 0xa081); + /* GE related settings */ + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_GE_RELATED_CTL2, + 0xa009); /* Button A, B, C, D bypass mode */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_UMP_HID_CTL4, 0xcf00); @@ -1475,9 +1478,6 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) if ((calib_status & 0x0040) == 0x0) break; } - /* Release HP-JD, EN_CBJ_TIE_GL/R open, en_osw gating auto done bit */ - rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_DIGITAL_MISC_CTRL4, - 0x0010); /* Set ADC09 power entity floating control */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ADC0A_08_PDE_FLOAT_CTL, 0x2a12); @@ -1490,8 +1490,21 @@ static void rt722_sdca_jack_preset(struct rt722_sdca_priv *rt722) /* Set DAC03 and HP power entity floating control */ rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_DAC03_HP_PDE_FLOAT_CTL, 0x4040); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_ENT_FLOAT_CTRL_1, + 0x4141); + rt722_sdca_index_write(rt722, RT722_VENDOR_HDA_CTL, RT722_FLOAT_CTRL_1, + 0x0101); /* Fine tune PDE40 latency */ regmap_write(rt722->regmap, 0x2f58, 0x07); + regmap_write(rt722->regmap, 0x2f03, 0x06); + /* MIC VRefo */ + rt722_sdca_index_update_bits(rt722, RT722_VENDOR_REG, + RT722_COMBO_JACK_AUTO_CTL1, 0x0200, 0x0200); + rt722_sdca_index_update_bits(rt722, RT722_VENDOR_REG, + RT722_VREFO_GAT, 0x4000, 0x4000); + /* Release HP-JD, EN_CBJ_TIE_GL/R open, en_osw gating auto done bit */ + rt722_sdca_index_write(rt722, RT722_VENDOR_REG, RT722_DIGITAL_MISC_CTRL4, + 0x0010); } int rt722_sdca_io_init(struct device *dev, struct sdw_slave *slave) diff --git a/sound/soc/codecs/rt722-sdca.h b/sound/soc/codecs/rt722-sdca.h index 44af8901352e..2464361a7958 100644 --- a/sound/soc/codecs/rt722-sdca.h +++ b/sound/soc/codecs/rt722-sdca.h @@ -69,6 +69,7 @@ struct rt722_sdca_dmic_kctrl_priv { #define RT722_COMBO_JACK_AUTO_CTL2 0x46 #define RT722_COMBO_JACK_AUTO_CTL3 0x47 #define RT722_DIGITAL_MISC_CTRL4 0x4a +#define RT722_VREFO_GAT 0x63 #define RT722_FSM_CTL 0x67 #define RT722_SDCA_INTR_REC 0x82 #define RT722_SW_CONFIG1 0x8a @@ -127,6 +128,8 @@ struct rt722_sdca_dmic_kctrl_priv { #define RT722_UMP_HID_CTL6 0x66 #define RT722_UMP_HID_CTL7 0x67 #define RT722_UMP_HID_CTL8 0x68 +#define RT722_FLOAT_CTRL_1 0x70 +#define RT722_ENT_FLOAT_CTRL_1 0x76 /* Parameter & Verb control 01 (0x1a)(NID:20h) */ #define RT722_HIDDEN_REG_SW_RESET (0x1 << 14) -- 2.34.1

2 1

[PATCH] ASoC: amd: fix for soundwire build dependencies for legacy stack
by Vijendar Mukunda 09 Apr '24

09 Apr '24

The SND_SOC_AMD_SOUNDWIRE Kconfig symbol has build dependency on SOUNDWIRE_AMD. It gets it wrong for a configuration involving SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y,SND_SOC_AMD_PS=y and SOUNDWIRE_AMD=m, which results in a link failure: ld: vmlinux.o: in function `amd_sdw_probe': >> sound/soc/amd/ps/pci-ps.c:271:(.text+0x1d51eff): undefined reference to `sdw_amd_probe' ld: vmlinux.o: in function `acp63_sdw_machine_select': >> sound/soc/amd/ps/pci-ps.c:294:(.text+0x1d525d5): undefined reference to `sdw_amd_get_slave_info' ld: vmlinux.o: in function `amd_sdw_exit': >> sound/soc/amd/ps/pci-ps.c:280:(.text+0x1d538ce): undefined reference to `sdw_amd_exit' Add a top level check config that forbids any of the AMD ACP drivers with version >= 6.3 from being built-in with CONFIG_SOUNDWIRE_AMD=m. Move SND_AMD_ACP_CONFIG common dependency config to SND_SOC_AMD_ACP63_TOPLEVEL config. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202404061257.khJml82D-lkp@intel.com/ Fixes: a806793f231e ("ASoC: amd: simplify soundwire dependencies for legacy stack") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 3508f5a96b75..6dec44f516c1 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -132,6 +132,19 @@ config SND_SOC_AMD_RPL_ACP6x Say m if you have such a device. If unsure select "N". +config SND_SOC_AMD_ACP63_TOPLEVEL + tristate "support for AMD platforms with ACP version >= 6.3" + default SND_AMD_ACP_CONFIG + depends on SND_AMD_ACP_CONFIG + depends on SOUNDWIRE_AMD || !SOUNDWIRE_AMD + depends on X86 || COMPILE_TEST + help + This adds support for AMD platforms with ACP version >= 6.3. + Say Y if you have such a device. + If unsure select "N". + +if SND_SOC_AMD_ACP63_TOPLEVEL + config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate select SND_AMD_SOUNDWIRE_ACPI if ACPI @@ -149,7 +162,6 @@ config SND_SOC_AMD_SOUNDWIRE config SND_SOC_AMD_PS tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" - select SND_AMD_ACP_CONFIG select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on X86 && PCI && ACPI help @@ -170,3 +182,5 @@ config SND_SOC_AMD_PS_MACH DMIC can be connected directly to ACP IP. Say m if you have such a device. If unsure select "N". + +endif -- 2.34.1

2 1

[PATCH 1/2] ASoC: rt5645: Fix the electric noise due to the CBJ contacts floating
by derek.fang＠realtek.com 09 Apr '24

09 Apr '24

From: Derek Fang <derek.fang(a)realtek.com> The codec leaves tie combo jack's sleeve/ring2 to floating status default. It would cause electric noise while connecting the active speaker jack during boot or shutdown. This patch requests a gpio to control the additional jack circuit to tie the contacts to the ground or floating. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e3ba04484813..d0d24a53df74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -444,6 +444,7 @@ struct rt5645_priv { struct regmap *regmap; struct i2c_client *i2c; struct gpio_desc *gpiod_hp_det; + struct gpio_desc *gpiod_cbj_sleeve; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; @@ -3186,6 +3187,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, 0); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 1); + msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); val &= 0x7; @@ -3202,6 +3206,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, @@ -3229,6 +3235,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } return rt5645->jack_type; @@ -4012,6 +4021,16 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) return ret; } + rt5645->gpiod_cbj_sleeve = devm_gpiod_get_optional(&i2c->dev, "cbj-sleeve", + GPIOD_OUT_LOW); + + if (IS_ERR(rt5645->gpiod_cbj_sleeve)) { + ret = PTR_ERR(rt5645->gpiod_cbj_sleeve); + dev_info(&i2c->dev, "failed to initialize gpiod, ret=%d\n", ret); + if (ret != -ENOENT) + return ret; + } + for (i = 0; i < ARRAY_SIZE(rt5645->supplies); i++) rt5645->supplies[i].supply = rt5645_supply_names[i]; @@ -4259,6 +4278,9 @@ static void rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); + regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); } @@ -4274,6 +4296,9 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) 0); msleep(20); regmap_write(rt5645->regmap, RT5645_RESET, 0); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } static int __maybe_unused rt5645_sys_suspend(struct device *dev) -- 2.34.1

2 2

[PATCH] ASoC: Intel: sof_rt5682: Fix uninitialized variable in probe
by Dan Carpenter 09 Apr '24

09 Apr '24

Initialize "is_legacy_cpu" to false to prevent an uninitialized variable bug. Fixes: 8efcd4864652 ("ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/intel/boards/sof_rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index aadd341a202c..966bcd3e8bdc 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -606,7 +606,7 @@ static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; - bool is_legacy_cpu; + bool is_legacy_cpu = false; int ret; if (pdev->id_entry && pdev->id_entry->driver_data) -- 2.43.0

3 2

[PATCH] ASoC: Intel: bytcr_rt5640: Apply Asus T100TA quirk to Asus T100TAM too
by Hans de Goede 09 Apr '24

09 Apr '24

The Asus T100TA quirk has been using an exact match on a product-name of "T100TA" but there are also T100TAM variants with a slightly higher clocked CPU and a metal backside which need the same quirk. Sort the existing T100TA (stereo speakers) below the more specific T100TAF (mono speaker) quirk and switch from exact matching to substring matching so that the T100TA quirk will also match on the T100TAM models. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 24 +++++++++++++----------- 1 file changed, 13 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 05f38d1f7d82..b41a1147f1c3 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -636,17 +636,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_USE_AMCR0F28), }, { - .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TA"), - }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_JD_SRC_JD2_IN4N | - BYT_RT5640_OVCD_TH_2000UA | - BYT_RT5640_OVCD_SF_0P75 | - BYT_RT5640_MCLK_EN), - }, - { + /* Asus T100TAF, unlike other T100TA* models this one has a mono speaker */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), @@ -660,6 +650,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { + /* Asus T100TA and T100TAM, must come after T100TAF (mono spk) match */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.44.0

3 2

[PATCH] ASoC: Intel: avs: Fix debug window description
by Cezary Rojewski 09 Apr '24

09 Apr '24

Recent changes addressed PAGE_SIZE ambiguity in 2/3 locations for struct avs_icl_memwnd2. The unaddressed one causes build errors when PAGE_SIZE != SZ_4K. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202404070100.i3t3Jf7d-lkp@intel.com/ Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Follow up to the recent series for the avs-driver [1]. With static_asserts() added [2], build problems occur when PAGE_SIZE != SZ_4K. Patch fixing memory window description [3] fixed only 2/3 offending spots. [1]: https://lore.kernel.org/alsa-devel/20240405090929.1184068-1-cezary.rojewski… [2]: https://lore.kernel.org/alsa-devel/20240405090929.1184068-13-cezary.rojewsk… [3]: https://lore.kernel.org/alsa-devel/20240405090929.1184068-3-cezary.rojewski… sound/soc/intel/avs/icl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index e8b4983e03e9..7a1ecf241856 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -66,7 +66,7 @@ static_assert(sizeof(struct avs_icl_memwnd2_desc) == 12); struct avs_icl_memwnd2 { union { struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; - u8 rsvd[PAGE_SIZE]; + u8 rsvd[SZ_4K]; }; u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K]; } __packed; base-commit: 1f58d8580e5da64669d923ced99cb3e29832273b -- 2.25.1

2 1

[RESEND PATCH v2] ASoC: tegra: Fix DSPK 16-bit playback
by Sameer Pujar 09 Apr '24

09 Apr '24

DSPK configuration is wrong for 16-bit playback and this happens because the client config is always fixed at 24-bit in hw_params(). Fix this by updating the client config to 16-bit for the respective playback. Fixes: 327ef6470266 ("ASoC: tegra: Add Tegra186 based DSPK driver") Cc: stable(a)vger.kernel.org Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- changes in v2: * moved common setting to S32_LE switch case. sound/soc/tegra/tegra186_dspk.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index aa37c4ab0adb..21cd41fec7a9 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -1,8 +1,7 @@ // SPDX-License-Identifier: GPL-2.0-only +// SPDX-FileCopyrightText: Copyright (c) 2020-2024 NVIDIA CORPORATION & AFFILIATES. All rights reserved. // // tegra186_dspk.c - Tegra186 DSPK driver -// -// Copyright (c) 2020 NVIDIA CORPORATION. All rights reserved. #include <linux/clk.h> #include <linux/device.h> @@ -241,14 +240,14 @@ static int tegra186_dspk_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - cif_conf.client_bits = TEGRA_ACIF_BITS_24; - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: cif_conf.audio_bits = TEGRA_ACIF_BITS_16; + cif_conf.client_bits = TEGRA_ACIF_BITS_16; break; case SNDRV_PCM_FORMAT_S32_LE: cif_conf.audio_bits = TEGRA_ACIF_BITS_32; + cif_conf.client_bits = TEGRA_ACIF_BITS_24; break; default: dev_err(dev, "unsupported format!\n"); -- 2.25.1

3 2

aseqdump -u1 can't receive MT4 messages
by GitHub issues - opened 09 Apr '24

09 Apr '24

alsa-project/alsa-utils issue #262 was opened from masahirokakishita: Hello, aseqdump version 1.2.11 karnel 6.5.0 Sending MT4 messages from ProtoZoA to Linux PC, the command, aseqdump -u1 can't receive messages. The command, aseqdump -u0 and aseqdump -u2 can receive these messages. ``` aseqdump -u 2 -p 24:0 24:1 Group 1, Note on 0, note 67, velocity 0xfedc, attr type = 66, data = 0xba98 24:1 Group 1, Note off 0, note 67, velocity 0x1234, attr type = 66, data = 0x5678 24:1 Group 1, Note on 0, note 65, velocity 0x2200, attr type = 1, data = 0x3344 24:1 Group 1, Note off 0, note 65, velocity 0x5500, attr type = 0, data = 0x6677 24:1 Group 1, Note on 0, note 63, velocity 0xffff, attr type = 0, data = 0x0 24:1 Group 1, Note off 0, note 63, velocity 0x8000, attr type = 0, data = 0x0 aseqdump -u 0 -p 24:0 24:1 Note on 0, note 67, velocity 127 24:1 Note off 0, note 67, velocity 9 24:1 Note on 0, note 65, velocity 17 24:1 Note off 0, note 65, velocity 42 24:1 Note on 0, note 63, velocity 127 24:1 Note off 0, note 63, velocity 64 ``` ``` aseqdump -u 1 -p 24:0 ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/262 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 00/16] ASoC: Replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 09 Apr '24

09 Apr '24

Hi Mark This is v2 patch-set When we use DPCM, we need to set dpcm_playback/capture flag. If these flag are set, soc_get_playback_capture() will check its availability, but non DPCM doesn't need such special flags. OTOH, it cares playback/capture_only flag. It is needed. This patch remove DPCM special flag, and replace it playback/capture_only flag if needed. v1 -> v2 - based on latest ASoC branch - keep comment on Intel - tidyup patch title - tidyup DPCM BE warning output condition - Add new patch for Document Link: https://lore.kernel.org/r/87o7b353of.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (16): ASoC: soc-pcm.c: cleanup soc_get_playback_capture() ASoC: amd: Replace dpcm_playback/capture to playback/capture_only ASoC: fsl: Replace dpcm_playback/capture to playback/capture_only ASoC: sof: Replace dpcm_playback/capture to playback/capture_only ASoC: meson: Replace dpcm_playback/capture to playback/capture_only ASoC: Intel: Replace dpcm_playback/capture to playback/capture_only ASoC: samsung: Replace dpcm_playback/capture to playback/capture_only ASoC: mediatek: Replace dpcm_playback/capture to playback/capture_only ASoC: soc-core: Replace dpcm_playback/capture to playback/capture_only ASoC: soc-topology: Replace dpcm_playback/capture to playback/capture_only ASoC: soc-compress: Replace dpcm_playback/capture to playback/capture_only ASoC: Intel: avs: boards: Replace dpcm_playback/capture to playback/capture_only ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: soc-pcm: remove dpcm_playback/capture ASoC: soc-pcm: indicate warning if DPCM BE Codec has no settings ASoC: doc: remove .dpcm_playback/capture flags Documentation/sound/soc/dpcm.rst | 14 ++- include/sound/soc-dai.h | 1 - include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++--- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 16 ++-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 7 +- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 ++++--- sound/soc/intel/boards/bxt_rt298.c | 26 +++--- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 15 ++-- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++---- sound/soc/intel/boards/hsw_rt5640.c | 10 +-- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 +++--- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 +++++------- sound/soc/intel/boards/kbl_rt5660.c | 18 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++--- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 ++- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 +++--- sound/soc/intel/boards/skl_rt286.c | 26 +++--- sound/soc/intel/boards/sof_board_helpers.c | 13 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 ++--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +-- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +-- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++----- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++----- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 +++++++----------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 +++++++----------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +++++++------ sound/soc/meson/axg-card.c | 9 +- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/samsung/odroid.c | 11 ++- sound/soc/soc-compress.c | 10 ++- sound/soc/soc-core.c | 20 +---- sound/soc/soc-dai.c | 38 -------- sound/soc/soc-pcm.c | 87 ++++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 91 files changed, 502 insertions(+), 863 deletions(-) -- 2.25.1

4 45

[PATCH v3 00/17] ALSA: emu10k1 & emux: fixes related to wavetable playback
by Oswald Buddenhagen 09 Apr '24

09 Apr '24

--- v3: - re-send with safer transfer encoding v2: - rebased on top of reverting problematic commit first Oswald Buddenhagen (17): ALSA: emux: fix /proc teardown at module unload ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch() ALSA: emux: fix validation of snd_emux.num_ports ALSA: emux: fix init of patch_info.truesize in load_data() ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support ALSA: emux: centralize & improve patch info validation ALSA: emux: improve patch ioctl data validation ALSA: emu10k1: move patch loader assertions into low-level functions ALSA: emu10k1: fix sample signedness issues in wavetable loader ALSA: emu10k1: fix playback of 8-bit wavetable samples ALSA: emu10k1: merge conditions in patch loader ALSA: emu10k1: fix wavetable offset recalculation ALSA: emu10k1: de-duplicate size calculations for 16-bit samples ALSA: emu10k1: improve cache behavior documentation ALSA: emu10k1: fix wavetable playback position and caching, take 2 ALSA: emu10k1: shrink blank space in front of wavetable samples ALSA: emux: simplify snd_sf_list.callback handling include/sound/emu10k1.h | 32 +++-- include/sound/soundfont.h | 2 +- sound/isa/sb/emu8000_patch.c | 13 -- sound/pci/emu10k1/emu10k1_callback.c | 13 +- sound/pci/emu10k1/emu10k1_patch.c | 207 +++++++++++---------------- sound/pci/emu10k1/memory.c | 55 +++++-- sound/synth/emux/emux.c | 6 +- sound/synth/emux/emux_hwdep.c | 3 +- sound/synth/emux/emux_oss.c | 3 +- sound/synth/emux/emux_proc.c | 1 + sound/synth/emux/emux_seq.c | 6 +- sound/synth/emux/soundfont.c | 73 +++++++--- 12 files changed, 219 insertions(+), 195 deletions(-) -- 2.44.0.701.g2cf7baacf3.dirty

3 18

[PATCH v1 0/2] ASoC: meson: implement link-name optional property in meson audio card
by Dmitry Rokosov 08 Apr '24

08 Apr '24

The 'link-name' is optional feature in the Device Tree that allows users to customize the name associated with the DAI link and PCM stream. This feature provides enhanced flexibility in DAI naming conventions, leading to improved system integration and a better user experience. Dmitry Rokosov (2): ASoC: dt-bindings: meson: introduce link-name optional property ASoC: meson: implement link-name optional property in meson card utils .../bindings/sound/amlogic,axg-sound-card.yaml | 6 ++++++ .../bindings/sound/amlogic,gx-sound-card.yaml | 6 ++++++ sound/soc/meson/meson-card-utils.c | 12 ++++++++---- 3 files changed, 20 insertions(+), 4 deletions(-) -- 2.43.0

3 10

[RESEND v3 0/2] Add support for QCM6490 and QCS6490
by Mohammad Rafi Shaik 08 Apr '24

08 Apr '24

This patchset adds support for sound card on Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 boards. Changes since v2: - Modify qcm6490 compatible name as qcm6490-idp. Suggested by Dmitry Changes since v1: - Use existing sc8280xp machine driver instead of separate driver. - Modify qcs6490 compatible name as qcs6490-rb3gen2. Mohammad Rafi Shaik (2): ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++ sound/soc/qcom/sc8280xp.c | 2 ++ 2 files changed, 4 insertions(+) -- 2.25.1

5 10

[PATCH v2] dt-bindings: ti,pcm1681: Convert to dtschema
by Animesh Agarwal 08 Apr '24

08 Apr '24

Convert the Texas Instruments PCM1681 bindings to DT schema. Make bindings complete by adding #sound-dai-cells. Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- Changes in v2: - Added existing maintainers to the list of maintainers. - Added #sound-dai-cells, $ref to dai-common and unevaluatedProperties: false. - Added "audio-codec" in example. --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 ------- .../devicetree/bindings/sound/ti,pcm1681.yaml | 43 +++++++++++++++++++ 2 files changed, 43 insertions(+), 15 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt deleted file mode 100644 index 4df17185ab80..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt +++ /dev/null @@ -1,15 +0,0 @@ -Texas Instruments PCM1681 8-channel PWM Processor - -Required properties: - - - compatible: Should contain "ti,pcm1681". - - reg: The i2c address. Should contain <0x4c>. - -Examples: - - i2c_bus { - pcm1681@4c { - compatible = "ti,pcm1681"; - reg = <0x4c>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 000000000000..5aa00617291c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel PWM Processor + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + - Kevin Lu <kevin-lu(a)ti.com> + - Baojun Xu <baojun.xu(a)ti.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681: audio-codec@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; -- 2.44.0

3 2

Add support for opus to aplay
by GitHub issues - opened 08 Apr '24

08 Apr '24

alsa-project/alsa-utils issue #261 was opened from bmariuszb: I would like to use aplay to play Opus audio files. Issue URL : https://github.com/alsa-project/alsa-utils/issues/261 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL FOR v6.10] Add imx-asrc m2m audio rate control driver
by Hans Verkuil 08 Apr '24

08 Apr '24

Hi Mauro, This adds support for the imx-asrc m2m audio rate control driver. Most of these patches add support for the new v4l-audioX device nodes and related types, and adding support for fixed point control types. The ASoC patches have been acked by Mark Brown. It also adds a vim2m-audio test driver so we can test the v4l-audio infrastructure. Since V4L2 is fourcc-based all the way, directly using snd_pcm_format_t values is not an option: fourcc's are expected to be printable characters, and it is pretty much certain that there are applications that show it like that to the end-user. The utilities in v4l-utils certainly do, and it is in fact a perfectly reasonable thing to do. So instead we map the snd_pcm_format_t value to a fourcc with v4l2_audfmt_to_fourcc and map it back with v4l2_fourcc_to_audfmt. This PR is using v15 of the patch series: https://patchwork.linuxtv.org/project/linux-media/list/?series=12460 And the corresponding v4l-utils patch series is here: https://patchwork.linuxtv.org/user/todo/linux-media/?series=12074 Regards, Hans The following changes since commit b14257abe7057def6127f6fb2f14f9adc8acabdb: media: rcar-isp: Disallow unbind of devices (2024-03-07 16:35:13 +0100) are available in the Git repository at: git://linuxtv.org/hverkuil/media_tree.git tags/br-audio for you to fetch changes up to af06c8792653c42d1f126505ec9180255091d94e: media: vim2m-audio: add virtual driver for audio memory to memory (2024-03-19 15:55:38 +0100) ---------------------------------------------------------------- Tag branch ---------------------------------------------------------------- Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory Documentation/userspace-api/media/mediactl/media-types.rst | 11 + Documentation/userspace-api/media/v4l/buffer.rst | 6 + Documentation/userspace-api/media/v4l/common.rst | 1 + Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst | 71 +++ Documentation/userspace-api/media/v4l/devices.rst | 1 + Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst | 59 +++ Documentation/userspace-api/media/v4l/pixfmt-audio.rst | 100 ++++ Documentation/userspace-api/media/v4l/pixfmt.rst | 1 + Documentation/userspace-api/media/v4l/vidioc-enum-fmt.rst | 2 + Documentation/userspace-api/media/v4l/vidioc-g-ext-ctrls.rst | 4 + Documentation/userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + Documentation/userspace-api/media/v4l/vidioc-querycap.rst | 3 + Documentation/userspace-api/media/v4l/vidioc-queryctrl.rst | 11 +- Documentation/userspace-api/media/videodev2.h.rst.exceptions | 3 + MAINTAINERS | 17 + drivers/media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++++++++++++++++++++++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 ++++++++++++++++++++++++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + drivers/media/test-drivers/vivid/vivid-ctrls.c | 26 ++ drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +++- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 +++ drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 ++ {sound/soc/fsl => include/sound}/fsl_asrc_common.h | 60 +++ include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 50 +- sound/soc/fsl/fsl_asrc.c | 144 ++++++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 ++++++++++ sound/soc/fsl/fsl_easrc.h | 6 +- 43 files changed, 3145 insertions(+), 27 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%)

2 2

[PATCH v3 0/2] Add support for QCM6490 and QCS6490
by Mohammad Rafi Shaik 08 Apr '24

08 Apr '24

This patchset adds support for sound card on Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 boards. Changes since v2: - Modify qcm6490 compatible name as qcs6490-idp. Suggested by Dmitry Changes since v1: - Use existing sc8280xp machine driver instead of separate driver. - Modify qcs6490 compatible name as qcs6490-rb3gen2. Mohammad Rafi Shaik (2): ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++ sound/soc/qcom/sc8280xp.c | 2 ++ 2 files changed, 4 insertions(+) -- 2.25.1

1 2

[PATCH v3] ALSA: hda/tas2781: correct the register for pow calibrated data
by Shenghao Ding 07 Apr '24

07 Apr '24

Calibrated data was written into an incorrect register, which cause speaker protection sometimes malfuctions Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add a bit more info desciption - No blank line between Fixes and Singed-off-by tags - https://lore.kernel.org/all/20240404122530.1049-1-shenghao-ding@ti.com/ --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 48dae3339305..75f7674c66ee 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -514,10 +514,10 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x0d, 0x18 + 0x17, 0x18, 0x18, 0x13, 0x18, }; static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x3c, 0x7c + 0x74, 0x0c, 0x14, 0x70, 0x7c, }; unsigned char *data; int i, j, rc; -- 2.34.1

2 1

Fwd: No speaker output on lenovo legion s7 16arha7
by Bagas Sanjaya 06 Apr '24

06 Apr '24

Hi, On Bugzilla, františek <Oddy38(a)seznam.cz> reported speaker problem on his Lenovo laptop [1]: > Hello after troubleshooting the issue on linux mint support we have come to the fact that it report a bug > my laptop's repeaters don't work (headphones work) > I updated to kernel 6.5.0-26-generic before I had the old version and it also did not work > I updated this firmware https://git.kernel.org/pub/scm/linux/kernel/git/firmware/linux-firmware.git… > in the sound settings it detects sound for example from youtube but still nothing comes out of the speakers > here is a link to how we troubleshot it > https://forums.linuxmint.com/viewtopic.php?p=2450183#p2450183 See Bugzilla for details. Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=218667 -- An old man doll... just what I always wanted! - Clara

1 0

[PATCH v1 1/8] ALSA: hda/tas2781: Add tas2781 SPI-based driver
by Baojun Xu 06 Apr '24

06 Apr '24

Add TXNW2781 support in smi. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- drivers/acpi/scan.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 7c157bf92..226bd9097 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1765,6 +1765,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, -- 2.40.1

6 20

[PATCH v2 00/17] ALSA: emu10k1 & emux: fixes related to wavetable playback
by Oswald Buddenhagen 06 Apr '24

06 Apr '24

--- This patch series needs to be applied on top of the patch titled "Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching"". Oswald Buddenhagen (17): ALSA: emux: fix /proc teardown at module unload ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch() ALSA: emux: fix validation of snd_emux.num_ports ALSA: emux: fix init of patch_info.truesize in load_data() ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support ALSA: emux: centralize & improve patch info validation ALSA: emux: improve patch ioctl data validation ALSA: emu10k1: move patch loader assertions into low-level functions ALSA: emu10k1: fix sample signedness issues in wavetable loader ALSA: emu10k1: fix playback of 8-bit wavetable samples ALSA: emu10k1: merge conditions in patch loader ALSA: emu10k1: fix wavetable offset recalculation ALSA: emu10k1: de-duplicate size calculations for 16-bit samples ALSA: emu10k1: improve cache behavior documentation ALSA: emu10k1: fix wavetable playback position and caching, take 2 ALSA: emu10k1: shrink blank space in front of wavetable samples ALSA: emux: simplify snd_sf_list.callback handling include/sound/emu10k1.h | 32 +++-- include/sound/soundfont.h | 2 +- sound/isa/sb/emu8000_patch.c | 13 -- sound/pci/emu10k1/emu10k1_callback.c | 13 +- sound/pci/emu10k1/emu10k1_patch.c | 207 +++++++++++---------------- sound/pci/emu10k1/memory.c | 55 +++++-- sound/synth/emux/emux.c | 6 +- sound/synth/emux/emux_hwdep.c | 3 +- sound/synth/emux/emux_oss.c | 3 +- sound/synth/emux/emux_proc.c | 1 + sound/synth/emux/emux_seq.c | 6 +- sound/synth/emux/soundfont.c | 73 +++++++--- 12 files changed, 219 insertions(+), 195 deletions(-) base-commit: ed93395844979f6bf2e1fbfcda38d1718289b426 prerequisite-patch-id: b7769e1c8649b86fd9e0b259e11bfd8e468393e5 -- 2.42.0.419.g70bf8a5751

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[PATCH 00/13] ASoC: Intel: avs: Fixes and cleanups for 6.10
by Cezary Rojewski 06 Apr '24

06 Apr '24

Set of changes targeting the avs-driver only. No new features, patchset either fixes or fortifies existing code. Patchset starts off with a fix for debugbility on ICL+ platforms which I have forgotten to fixup when providing support for these initially. The next two address copier module initialization, most importantly, silence the gcc 'field-spanning write' false-positive. The following four: 6/13 ASoC: Intel: avs: Replace risky functions with safer variants 7/13 ASoC: Intel: avs: Fix potential integer overflow 8/13 ASoC: Intel: avs: Test result of avs_get_module_entry() 9/13 ASoC: Intel: avs: Remove dead code address problems found out by Coverity static analysis tool. The last two worth mentioning are: recommendation from the firmware team to wake subsystem from D0ix when starting any pipeline -and- shielding against invalid period/buffer sizes. Audio format shall be taken into consideration when calculating either of these. Amadeusz Sławiński (2): ASoC: Intel: avs: Restore stream decoupling on prepare ASoC: Intel: avs: Add assert_static to guarantee ABI sizes Cezary Rojewski (11): ASoC: Intel: avs: Fix debug-slot offset calculation ASoC: Intel: avs: Silence false-positive memcpy() warnings ASoC: Intel: avs: Fix config_length for config-less copiers ASoC: Intel: avs: Fix ASRC module initialization ASoC: Intel: avs: Replace risky functions with safer variants ASoC: Intel: avs: Fix potential integer overflow ASoC: Intel: avs: Test result of avs_get_module_entry() ASoC: Intel: avs: Remove dead code ASoC: Intel: avs: Wake from D0ix when starting streaming ASoC: Intel: avs: Init debugfs before booting firmware ASoC: Intel: avs: Rule invalid buffer and period sizes out sound/soc/intel/avs/avs.h | 1 + sound/soc/intel/avs/cldma.c | 2 +- sound/soc/intel/avs/core.c | 4 +-- sound/soc/intel/avs/icl.c | 12 ++++++--- sound/soc/intel/avs/loader.c | 6 +++-- sound/soc/intel/avs/messages.h | 47 ++++++++++++++++++++++++++++++++-- sound/soc/intel/avs/path.c | 13 ++++------ sound/soc/intel/avs/pcm.c | 34 +++++++++++++++++++++++- sound/soc/intel/avs/probes.c | 14 ++++++---- 9 files changed, 109 insertions(+), 24 deletions(-) -- 2.25.1

2 14

[PATCH 0/4] ASoC: SOF: Intel: improve and extend HDaudio-based wakes
by Pierre-Louis Bossart 06 Apr '24

06 Apr '24

For LunarLake, the SoundWire in-band wake detection is reported with the HDAudio WAKE_EN/WAKE_STS registers. In the existing code, these registers are only handled for HDaudio codecs. Now the same registers have to be handled with care as shared resources. The in-band wake detection mainly used for jack detection. Without this patchset, the SoundWire headset codecs signal an event that would be ignored and not reported. Pierre-Louis Bossart (4): ASoC: SOF: Intel: hda-ctrl: add missing WAKE_STS clear ASoC: SOF: Intel: lnl: add helper to detect SoundWire wakes ASoC: SOF: Intel: hda-codec: preserve WAKEEN values ASoC: SOF: Intel: hda-ctrl: only clear WAKESTS for HDaudio codecs sound/soc/sof/intel/hda-codec.c | 15 ++++++++++++--- sound/soc/sof/intel/hda-ctrl.c | 9 ++++++++- sound/soc/sof/intel/lnl.c | 18 ++++++++++++++++++ 3 files changed, 38 insertions(+), 4 deletions(-) -- 2.40.1

2 5

[PATCH 0/4] ASoC: SOF: Intel: extend hda-loader for BTP/BRA transfers
by Pierre-Louis Bossart 06 Apr '24

06 Apr '24

For SoundWire BPT/BRA transfers, we need to reuse the HDaudio DMA infrastructure. This patchset extends the code used by the code loader. The BPT/BRA functionality will be provided in a follow-up contribution. Pierre-Louis Bossart (4): ASoC: SOF: Intel: hda-loader: add flag to tag ICCMAX stream ASoC: SOF: Intel: hda-loader: change signature of code loader helpers ASoC: SOF: Intel: don't ignore IOC interrupts for non-audio transfers ASoC: SOF: Intel: hda-stream: clarify comment sound/soc/sof/intel/hda-loader.c | 80 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda-stream.c | 24 ++++++++-- sound/soc/sof/intel/hda.h | 13 ++++-- 3 files changed, 85 insertions(+), 32 deletions(-) -- 2.40.1

2 5

[PATCH] ASoC: qcom: Fix out of bounds access
by Mikhail Kobuk 05 Apr '24

05 Apr '24

Case values introduced in commit 5f78e1fb7a3e ("ASoC: qcom: Add driver support for audioreach solution") cause out of bounds access in arrays of sc7280 driver data (e.g. in case of RX_CODEC_DMA_RX_0 in sc7280_snd_hw_params()). Redefine LPASS_MAX_PORTS to consider the maximum possible port id for q6dsp as sc7280 driver utilizes some of those values. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 77d0ffef793d ("ASoC: qcom: Add macro for lpass DAI id's max limit") Cc: <stable(a)vger.kernel.org> Signed-off-by: Mikhail Kobuk <m.kobuk(a)ispras.ru> --- sound/soc/qcom/lpass.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 27a2bf9a6613..10a507c95312 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -13,10 +13,11 @@ #include <linux/platform_device.h> #include <linux/regmap.h> #include <dt-bindings/sound/qcom,lpass.h> +#include <dt-bindings/sound/qcom,q6dsp-lpass-ports.h> #include "lpass-hdmi.h" #define LPASS_AHBIX_CLOCK_FREQUENCY 131072000 -#define LPASS_MAX_PORTS (LPASS_CDC_DMA_VA_TX8 + 1) +#define LPASS_MAX_PORTS (QUINARY_MI2S_TX + 1) #define LPASS_MAX_MI2S_PORTS (8) #define LPASS_MAX_DMA_CHANNELS (8) #define LPASS_MAX_HDMI_DMA_CHANNELS (4) -- 2.44.0

1 0

[PATCH v2] ASoC: au1x: declare that FIFO is reported in frames
by Oswald Buddenhagen 05 Apr '24

05 Apr '24

Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Is this right? The comment on the field says "fifo entries of AC97/I2S PSC", which doesn't suggest bytes. The data sheet speaks of "words" and "byte masks", but without digging into it I can't tell how it would behave with different sample widths and channel counts (which the driver does not seem to limit _at all_? what am I missing?). This is just something I stumbled over while surveying how different drivers set the field. If this is actually a bug, it is minor. --- sound/soc/au1x/dbdma2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index ea01d6490cec..2c8c01a5fb8f 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -66,7 +66,8 @@ struct au1xpsc_audio_dmadata { /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_FIFO_IN_FRAMES, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, .periods_min = 2, -- 2.42.0.419.g70bf8a5751

2 2

[PATCH v2 0/2] Add support for QCM6490 and QCS6490
by Mohammad Rafi Shaik 05 Apr '24

05 Apr '24

This patchset adds support for sound card on Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 boards. Changes since v1: - Use existing sc8280xp machine driver instead of separate driver. - Modify qcs6490 compatible name as qcs6490-rb3gen2. Mohammad Rafi Shaik (2): ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++ sound/soc/qcom/sc8280xp.c | 2 ++ 2 files changed, 4 insertions(+) -- 2.25.1

2 5

[GIT PULL] ASoC fixes for v6.9-rc2
by Mark Brown 05 Apr '24

05 Apr '24

The following changes since commit 4cece764965020c22cff7665b18a012006359095: Linux 6.9-rc1 (2024-03-24 14:10:05 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.9-rc2 for you to fetch changes up to 90f8917e7a15f6dd508779048bdf00ce119b6ca0: ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path (2024-04-04 19:51:51 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.9 A relatively large set of fixes here, the biggest piece of it is a series correcting some problems with the delay reporting for Intel SOF cards but there's a bunch of other things. Everything here is driver specific except for a fix in the core for an issue with sign extension handling volume controls. ---------------------------------------------------------------- Amadeusz Sławiński (1): ASoC: Intel: avs: boards: Add modules description Chaitanya Kumar Borah (1): ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path Charles Keepax (1): ASoC: cs42l43: Correct extraction of data pointer in suspend/resume Mark Brown (4): ASoC: SOF: ipc4/Intel: Fix delay reporting ASoC: rt-sdw: fix locking and improve error logs ASoC: codecs: ES8326: solve some hp issues and ASoC: Merge up left over v6.8 fix Peter Ujfalusi (19): ASoC: SOF: Add dsp_max_burst_size_in_ms member to snd_sof_pcm_stream ASoC: SOF: ipc4-topology: Save the DMA maximum burst size for PCMs ASoC: SOF: Intel: hda-pcm: Use dsp_max_burst_size_in_ms to place constraint ASoC: SOF: Intel: hda: Implement get_stream_position (Linear Link Position) ASoC: SOF: Intel: mtl/lnl: Use the generic get_stream_position callback ASoC: SOF: Introduce a new callback pair to be used for PCM delay reporting ASoC: SOF: Intel: Set the dai/host get frame/byte counter callbacks ASoC: SOF: ipc4-pcm: Use the snd_sof_pcm_get_dai_frame_counter() for pcm_delay ASoC: SOF: Intel: hda-common-ops: Do not set the get_stream_position callback ASoC: SOF: Remove the get_stream_position callback ASoC: SOF: ipc4-pcm: Move struct sof_ipc4_timestamp_info definition locally ASoC: SOF: ipc4-pcm: Combine the SOF_IPC4_PIPE_PAUSED cases in pcm_trigger ASoC: SOF: ipc4-pcm: Invalidate the stream_start_offset in PAUSED state ASoC: SOF: sof-pcm: Add pointer callback to sof_ipc_pcm_ops ASoC: SOF: ipc4-pcm: Correct the delay calculation ALSA: hda: Add pplcllpl/u members to hdac_ext_stream ASoC: SOF: Intel: hda: Compensate LLP in case it is not reset ASoC: SOF: Intel: hda-dsp: Skip IMR boot on ACE platforms in case of S3 suspend ASoC: SOF: Intel: lnl: Disable DMIC/SSP offload on remove Pierre-Louis Bossart (6): ASoC: rt5682-sdw: fix locking sequence ASoC: rt711-sdca: fix locking sequence ASoC: rt711-sdw: fix locking sequence ASoC: rt712-sdca-sdw: fix locking sequence ASoC: rt722-sdca-sdw: fix locking sequence ASoC: rt-sdw*: add __func__ to all error logs Rander Wang (1): ASoC: SOF: mtrace: rework mtrace timestamp setting Richard Fitzgerald (1): ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl() Simon Trimmer (1): ASoC: cs-amp-lib: Check for no firmware controls when writing calibration Stephen Lee (1): ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw Vijendar Mukunda (3): ASoC: amd: acp: fix for acp pdm configuration check ASoC: amd: acp: fix for acp_init function error handling ASoC: SOF: amd: fix for false dsp interrupts Zhang Yi (4): ASoC: codecs: ES8326: Solve error interruption issue ASoC: codecs: ES8326: modify clock table ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume ASoC: codecs: ES8326: Removing the control of ADC_SCALE include/sound/hdaudio_ext.h | 3 + sound/soc/amd/acp/acp-pci.c | 13 +-- sound/soc/codecs/cs-amp-lib.c | 5 + sound/soc/codecs/cs42l43.c | 12 ++- sound/soc/codecs/es8326.c | 37 ++++--- sound/soc/codecs/es8326.h | 2 +- sound/soc/codecs/rt1316-sdw.c | 8 +- sound/soc/codecs/rt1318-sdw.c | 8 +- sound/soc/codecs/rt5682-sdw.c | 16 +-- sound/soc/codecs/rt700.c | 16 +-- sound/soc/codecs/rt711-sdca-sdw.c | 6 +- sound/soc/codecs/rt711-sdca.c | 18 ++-- sound/soc/codecs/rt711-sdw.c | 8 +- sound/soc/codecs/rt711.c | 16 +-- sound/soc/codecs/rt712-sdca-dmic.c | 24 +++-- sound/soc/codecs/rt712-sdca-sdw.c | 7 +- sound/soc/codecs/rt712-sdca.c | 20 ++-- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 ++++---- sound/soc/codecs/rt715-sdw.c | 4 +- sound/soc/codecs/rt715.c | 24 ++--- sound/soc/codecs/rt722-sdca-sdw.c | 4 +- sound/soc/codecs/rt722-sdca.c | 21 ++-- sound/soc/codecs/wm_adsp.c | 3 +- sound/soc/intel/avs/boards/da7219.c | 1 + sound/soc/intel/avs/boards/dmic.c | 1 + sound/soc/intel/avs/boards/es8336.c | 1 + sound/soc/intel/avs/boards/i2s_test.c | 1 + sound/soc/intel/avs/boards/max98357a.c | 1 + sound/soc/intel/avs/boards/max98373.c | 1 + sound/soc/intel/avs/boards/max98927.c | 1 + sound/soc/intel/avs/boards/nau8825.c | 1 + sound/soc/intel/avs/boards/probe.c | 1 + sound/soc/intel/avs/boards/rt274.c | 1 + sound/soc/intel/avs/boards/rt286.c | 1 + sound/soc/intel/avs/boards/rt298.c | 1 + sound/soc/intel/avs/boards/rt5514.c | 1 + sound/soc/intel/avs/boards/rt5663.c | 1 + sound/soc/intel/avs/boards/rt5682.c | 1 + sound/soc/intel/avs/boards/ssm4567.c | 1 + sound/soc/soc-ops.c | 2 +- sound/soc/sof/amd/acp.c | 8 +- sound/soc/sof/core.c | 14 +-- sound/soc/sof/intel/hda-common-ops.c | 3 + sound/soc/sof/intel/hda-dai-ops.c | 11 ++ sound/soc/sof/intel/hda-dsp.c | 20 +++- sound/soc/sof/intel/hda-pcm.c | 29 +++++ sound/soc/sof/intel/hda-stream.c | 70 ++++++++++++ sound/soc/sof/intel/hda.h | 6 ++ sound/soc/sof/intel/lnl.c | 34 ++++-- sound/soc/sof/intel/mtl.c | 14 --- sound/soc/sof/intel/mtl.h | 10 -- sound/soc/sof/ipc4-mtrace.c | 11 +- sound/soc/sof/ipc4-pcm.c | 191 ++++++++++++++++++++++++++------- sound/soc/sof/ipc4-priv.h | 14 --- sound/soc/sof/ipc4-topology.c | 22 +++- sound/soc/sof/ops.h | 24 ++++- sound/soc/sof/pcm.c | 8 ++ sound/soc/sof/sof-audio.h | 9 +- sound/soc/sof/sof-priv.h | 24 +++-- 60 files changed, 596 insertions(+), 267 deletions(-)

2 1

[PATCH] ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path
by Pierre-Louis Bossart 05 Apr '24

05 Apr '24

From: Chaitanya Kumar Borah <chaitanya.kumar.borah(a)intel.com> In cases where the sof driver is unable to find the firmware and/or topology file [1], it exits without releasing the i915 runtime pm wakeref [2]. This results in dmesg warnings[3] during suspend/resume or driver unbind. Add remove_late() to the failure path of sof_init_environment so that i915 wakeref is released appropriately [1] [ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found. [ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles [ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested): [ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri [ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg [ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed. [ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from: [ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/ [ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2 [2] ref_tracker: 0000:00:02.0@ffff9b8511b6a378 has 1/5 users at track_intel_runtime_pm_wakeref.part.0+0x36/0x70 [i915] __intel_runtime_pm_get+0x51/0xb0 [i915] intel_runtime_pm_get+0x17/0x20 [i915] intel_display_power_get+0x2f/0x70 [i915] i915_audio_component_get_power+0x23/0x120 [i915] snd_hdac_display_power+0x89/0x130 [snd_hda_core] hda_codec_i915_init+0x3f/0x50 [snd_sof_intel_hda] hda_dsp_probe_early+0x170/0x250 [snd_sof_intel_hda_common] snd_sof_device_probe+0x224/0x320 [snd_sof] sof_pci_probe+0x15b/0x220 [snd_sof_pci] hda_pci_intel_probe+0x30/0x70 [snd_sof_intel_hda_common] local_pci_probe+0x4c/0xb0 pci_device_probe+0xcc/0x250 really_probe+0x18e/0x420 __driver_probe_device+0x7e/0x170 driver_probe_device+0x23/0xa0 [3] [ 484.105070] ------------[ cut here ]------------ [ 484.108238] thunderbolt 0000:00:0d.2: PM: pci_pm_suspend_late+0x0/0x50 returned 0 after 0 usecs [ 484.117106] i915 0000:00:02.0: i915 raw-wakerefs=1 wakelocks=1 on cleanup [ 484.792005] WARNING: CPU: 2 PID: 2405 at drivers/gpu/drm/i915/intel_runtime_pm.c:444 intel_runtime_pm_driver_release+0x6c/0x80 Tested-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> Reviewed-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/core.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9b00ede2a486..cc84d4c81be9 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -339,8 +339,7 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = snd_sof_probe(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to probe DSP %d\n", ret); - sof_ops_free(sdev); - return ret; + goto err_sof_probe; } /* check machine info */ @@ -358,15 +357,18 @@ static int sof_init_environment(struct snd_sof_dev *sdev) ret = validate_sof_ops(sdev); if (ret < 0) { snd_sof_remove(sdev); + snd_sof_remove_late(sdev); return ret; } } + return 0; + err_machine_check: - if (ret) { - snd_sof_remove(sdev); - sof_ops_free(sdev); - } + snd_sof_remove(sdev); +err_sof_probe: + snd_sof_remove_late(sdev); + sof_ops_free(sdev); return ret; } -- 2.40.1

2 1

[PATCH 0/4] ASoC: Drop soc-topology ABI v4 support
by Cezary Rojewski 04 Apr '24

04 Apr '24

This patchset impacts UAPI. The only known users of the soc-topology ABI v4 are Chromebook configurations. Starting from kernel v5.4, all of them are making use of soc-topology ABI v5. The patchset first removes obsolete code from the Intel's skylake-driver - the driver of choice for the mentioned Chromebooks - and then proceeds with removal of relevant soc-topology.c and uapi bits. Cezary Rojewski (4): ASoC: Intel: Skylake: Remove soc-topology ABI v4 support ASoC: topology: Remove ABI v4 support ASoC: topology: Cleanup after ABI v4 support removal ASoC: topology: Remove obsolete ABI v4 structs include/uapi/sound/asoc.h | 56 ------ include/uapi/sound/skl-tplg-interface.h | 74 -------- sound/soc/intel/skylake/skl-topology.c | 169 ----------------- sound/soc/soc-topology.c | 241 ++---------------------- 4 files changed, 18 insertions(+), 522 deletions(-) -- 2.25.1

2 7

[PATCH] ASoC: SOF: amd: fix for false dsp interrupts
by Vijendar Mukunda 04 Apr '24

04 Apr '24

Before ACP firmware loading, DSP interrupts are not expected. Sometimes after reboot, it's observed that before ACP firmware is loaded false DSP interrupt is reported. Registering the interrupt handler before acp initialization causing false interrupts sometimes on reboot as ACP reset is not applied. Correct the sequence by invoking acp initialization sequence prior to registering interrupt handler. Fixes: 738a2b5e2cc9 ("ASoC: SOF: amd: Add IPC support for ACP IP block") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index be7dc1e02284..c12c7f820529 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -704,6 +704,10 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto unregister_dev; } + ret = acp_init(sdev); + if (ret < 0) + goto free_smn_dev; + sdev->ipc_irq = pci->irq; ret = request_threaded_irq(sdev->ipc_irq, acp_irq_handler, acp_irq_thread, IRQF_SHARED, "AudioDSP", sdev); @@ -713,10 +717,6 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) goto free_smn_dev; } - ret = acp_init(sdev); - if (ret < 0) - goto free_ipc_irq; - /* scan SoundWire capabilities exposed by DSDT */ ret = acp_sof_scan_sdw_devices(sdev, chip->sdw_acpi_dev_addr); if (ret < 0) { -- 2.34.1

2 1

[PATCH v2] ALSA: hda/tas2781: correct the register for pow calibrated data
by Shenghao Ding 04 Apr '24

04 Apr '24

Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add Fixes tag - To add a trailing comma will help in case of extending this in the future. --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 48dae3339305..75f7674c66ee 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -514,10 +514,10 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x0d, 0x18 + 0x17, 0x18, 0x18, 0x13, 0x18, }; static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x3c, 0x7c + 0x74, 0x0c, 0x14, 0x70, 0x7c, }; unsigned char *data; int i, j, rc; -- 2.34.1

2 1

[PATCH] ASoC: rt5645: Fix the electric noise due to the CBJ contacts floating
by derek.fang＠realtek.com 04 Apr '24

04 Apr '24

From: Derek Fang <derek.fang(a)realtek.com> The codec leaves tie combo jack's sleeve/ring2 to floating status default. It would cause electric noise while connecting the active speaker jack during boot or shutdown. This patch requests a gpio to control the additional jack circuit to tie the contacts to the ground or floating. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e3ba04484813..d0d24a53df74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -444,6 +444,7 @@ struct rt5645_priv { struct regmap *regmap; struct i2c_client *i2c; struct gpio_desc *gpiod_hp_det; + struct gpio_desc *gpiod_cbj_sleeve; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; @@ -3186,6 +3187,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, 0); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 1); + msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); val &= 0x7; @@ -3202,6 +3206,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, @@ -3229,6 +3235,9 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse if (rt5645->pdata.level_trigger_irq) regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } return rt5645->jack_type; @@ -4012,6 +4021,16 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) return ret; } + rt5645->gpiod_cbj_sleeve = devm_gpiod_get_optional(&i2c->dev, "cbj-sleeve", + GPIOD_OUT_LOW); + + if (IS_ERR(rt5645->gpiod_cbj_sleeve)) { + ret = PTR_ERR(rt5645->gpiod_cbj_sleeve); + dev_info(&i2c->dev, "failed to initialize gpiod, ret=%d\n", ret); + if (ret != -ENOENT) + return ret; + } + for (i = 0; i < ARRAY_SIZE(rt5645->supplies); i++) rt5645->supplies[i].supply = rt5645_supply_names[i]; @@ -4259,6 +4278,9 @@ static void rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); + regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); } @@ -4274,6 +4296,9 @@ static void rt5645_i2c_shutdown(struct i2c_client *i2c) 0); msleep(20); regmap_write(rt5645->regmap, RT5645_RESET, 0); + + if (rt5645->gpiod_cbj_sleeve) + gpiod_set_value(rt5645->gpiod_cbj_sleeve, 0); } static int __maybe_unused rt5645_sys_suspend(struct device *dev) -- 2.34.1

2 1

[PATCH 0/5] ALSA: cirrus: Tidy up of firmware control read/write
by Richard Fitzgerald 04 Apr '24

04 Apr '24

This set of patches factors out some repeated code to clean up firmware control read/write functions, and removes some redundant control notification code. Simon Trimmer (5): firmware: cs_dsp: Add locked wrappers for coeff read and write ASoC: wm_adsp: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() ALSA: hda: hda_cs_dsp_ctl: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() ASoC: wm_adsp: Remove notification of driver write ALSA: hda: hda_cs_dsp_ctl: Remove notification of driver write drivers/firmware/cirrus/cs_dsp.c | 54 ++++++++++++++++++++++++++ include/linux/firmware/cirrus/cs_dsp.h | 4 ++ sound/pci/hda/hda_cs_dsp_ctl.c | 22 +---------- sound/soc/codecs/wm_adsp.c | 32 ++++----------- 4 files changed, 67 insertions(+), 45 deletions(-) -- 2.39.2

3 9

[PATCH] dt-bindings: omap-mcpdm: Convert to DT schema
by Mighty 03 Apr '24

03 Apr '24

Convert the OMAP4+ McPDM bindings to DT schema. Signed-off-by: Mighty <bavishimithil(a)gmail.com> --- .../devicetree/bindings/sound/omap-mcpdm.txt | 30 ---------- .../devicetree/bindings/sound/omap-mcpdm.yaml | 58 +++++++++++++++++++ 2 files changed, 58 insertions(+), 30 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.txt create mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.yaml diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt deleted file mode 100644 index ff98a0cb5..000000000 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ /dev/null @@ -1,30 +0,0 @@ -* Texas Instruments OMAP4+ McPDM - -Required properties: -- compatible: "ti,omap4-mcpdm" -- reg: Register location and size as an array: - <MPU access base address, size>, - <L3 interconnect address, size>; -- interrupts: Interrupt number for McPDM -- ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" - -Example: - -mcpdm: mcpdm@40132000 { - compatible = "ti,omap4-mcpdm"; - reg = <0x40132000 0x7f>, /* MPU private access */ - <0x49032000 0x7f>; /* L3 Interconnect */ - interrupts = <0 112 0x4>; - interrupt-parent = <&gic>; - ti,hwmods = "mcpdm"; -}; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.yaml b/Documentation/devicetree/bindings/sound/omap-mcpdm.yaml new file mode 100644 index 000000000..8c9ff9d90 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/omap-mcpdm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: OMAP McPDM + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +description: + OMAP ALSA SoC DAI driver using McPDM port used by TWL6040 + +properties: + compatible: + const: ti,omap4-mcpdm + + reg: + description: 'Register location and size as an array: + <MPU access base address, size>, + <L3 interconnect address, size>;' + + interrupts: + description: Interrupt number for McPDM + + ti,hwmods: + description: Name of the hwmod associated to the McPDM + + clocks: + description: phandle for the pdmclk provider, likely <&twl6040> + + clock-names: + description: Must be "pdmclk" + + +required: + - compatible + - reg + - interrupts + - ti,hwmods + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + mcpdm: mcpdm@40132000 { + compatible = "ti,omap4-mcpdm"; + reg = <0x40132000 0x7f>, /* MPU private access */ + <0x49032000 0x7f>; /* L3 Interconnect */ + interrupts = <0 112 0x4>; + interrupt-parent = <&gic>; + ti,hwmods = "mcpdm"; + clocks = <&twl6040>; + clock-names = "pdmclk"; + }; -- 2.34.1

2 1

[PATCH 00/10] Use const where applicable
by Amadeusz Sławiński 03 Apr '24

03 Apr '24

Use devm_kstrdup_const in few places to possibly save a bit of memory. Mark some of the structs as const, as they are not modified and used as is. Amadeusz Sławiński (10): ASoC: Intel: avs: hdaudio: Use devm_kstrdup_const ASoC: Intel: avs: Use devm_kstrdup_const ASoC: Intel: avs: hdaudio: Constify probing_link ASoC: Intel: avs: es8336: Constify card_headset_pins ASoC: Intel: avs: nau8825: Constify card_headset_pins ASoC: Intel: avs: rt274: Constify card_headset_pins ASoC: Intel: avs: rt286: Constify card_headset_pins ASoC: Intel: avs: rt298: Constify card_headset_pins ASoC: Intel: avs: rt5663: Constify card_headset_pins ASoC: Intel: avs: rt5682: Constify card_headset_pins sound/soc/intel/avs/avs.h | 2 +- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/hdaudio.c | 6 +++--- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 2 +- sound/soc/intel/avs/boards/rt286.c | 2 +- sound/soc/intel/avs/boards/rt298.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 +- sound/soc/intel/avs/boards/rt5682.c | 2 +- sound/soc/intel/avs/utils.c | 8 ++++---- 10 files changed, 15 insertions(+), 15 deletions(-) base-commit: 36d464cba8bdfd268282a97e9fdc701aca2ed8cb -- 2.34.1

2 11

[PATCH 0/5] ASoC: rsnd: reg cleanup
by Kuninori Morimoto 03 Apr '24

03 Apr '24

Hi Mark These are Renesas Sound driver cleanups for Gen3/Gen4. Kuninori Morimoto (5): ASoC: rsnd: cleanup regmap table ASoC: rsnd: don't get resource from ID ASoC: rsnd: rename rsnd_is_e3() to rsnd_is_gen3_e3() ASoC: rsnd: R-Car Gen1/Gen2 exception ASoC: rsnd: no exception for SCU sound/soc/sh/rcar/cmd.c | 6 +- sound/soc/sh/rcar/core.c | 4 +- sound/soc/sh/rcar/ctu.c | 6 +- sound/soc/sh/rcar/dma.c | 6 +- sound/soc/sh/rcar/dvc.c | 6 +- sound/soc/sh/rcar/gen.c | 519 +++++++++++++++++---------------------- sound/soc/sh/rcar/mix.c | 6 +- sound/soc/sh/rcar/rsnd.h | 22 +- sound/soc/sh/rcar/src.c | 12 +- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rcar/ssiu.c | 2 +- 11 files changed, 247 insertions(+), 344 deletions(-) -- 2.25.1

2 6

[PATCH v1] ALSA: hda/tas2781: correct the register for pow calibrated data
by Shenghao Ding 03 Apr '24

03 Apr '24

fixed tas2781 calibrated data - pow was written into a wrong register. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 48dae3339305..116821ae3cb3 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -514,10 +514,10 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x0d, 0x18 + 0x17, 0x18, 0x18, 0x13, 0x18 }; static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x3c, 0x7c + 0x74, 0x0c, 0x14, 0x70, 0x7c }; unsigned char *data; int i, j, rc; -- 2.34.1

2 1

[PATCH 00/17] ASoC: SOF: Intel: improve SoundWire support for LunarLake
by Pierre-Louis Bossart 03 Apr '24

03 Apr '24

This patchset contains important updates for SoundWire support. We initially implemented support for multiple amplifiers on different links using a single HDaudio DMA transfer. To align with the other OS, the 'aggregation' is now supported by the firmware. This change in directions has kernel impacts, since we now have multiple HDaudio DMAs to program and start, but since there are no platforms released so far there's no end-user impact. In addition, the behavior in case of xruns is improved by clearing the PCM states and better handling of the hw_free case. Note that the hw_free support will compile but will only be functional with the companion patch "soundwire: intel: add intel_free_stream() back" already applied in the SoundWire tree. Bard Liao (5): Revert "ASoC: SOF: Intel: hda-dai-ops: reset device count for SoundWire DAIs" Revert "ASoC: SOF: Intel: hda-dai-ops: only allocate/release streams for first CPU DAI" ASoC: SOF: make dma_config_tlv be an array ASoC: SOF: Intel: hda-dai: set lowest N bits in ch_mask ASoC: SOF: Intel: hda-dai: set dma_stream_channel_map device Pierre-Louis Bossart (10): ASoC: SOF: Intel: hda-dai: fix channel map configuration for aggregated dailink ASoC: SOF: Intel: hda-dai: add helpers to set dai config ASoC: SOF: Intel: set the DMA TLV device as dai_index ASoC: SOF: Intel: hda: extend signature of sdw_hda_dai_hw_params() ASoC: SOF: IPC4: extend dai_data with node_id ASoC: SOF: Intel: hda: move helper to static inline ASoC: SOF: sof-audio: revisit sof_pcm_stream_free() error handling and logs ASoC: SOF: pcm: simplify sof_pcm_hw_free() with helper ASoC: SOF: pcm: add pending_stop state variable ASoC: SOF: pcm: reset all PCM sources in case of xruns Ranjani Sridharan (2): ASoC: SOF: Intel: hda: Clear Soundwire node ID during BE DAI hw_free ASoC: SOF: ipc4-topology: Save the ALH DAI index during hw_params sound/soc/sof/intel/hda-dai-ops.c | 41 +--------- sound/soc/sof/intel/hda-dai.c | 124 +++++++++++++++++++++++------- sound/soc/sof/intel/hda.c | 29 ++++++- sound/soc/sof/intel/hda.h | 11 ++- sound/soc/sof/ipc4-topology.c | 62 ++++++++++----- sound/soc/sof/ipc4-topology.h | 3 +- sound/soc/sof/pcm.c | 58 +++++++------- sound/soc/sof/sof-audio.c | 29 +++++-- sound/soc/sof/sof-audio.h | 2 + 9 files changed, 232 insertions(+), 127 deletions(-) -- 2.40.1

2 18

[PATCH 0/4] ASoC: SOF: cppcheck fixes and debugfs addition
by Pierre-Louis Bossart 03 Apr '24

03 Apr '24

Small changes with 3 cppcheck fixes and the firmware version now visible with debugfs instead of only via dmesg logs. Pierre-Louis Bossart (3): ASoC: SOF: amd: acp-loader: abort firmware download on write error ASoC: SOF: ipc4-priv: align prototype and function declaration ASoC: SOF: ipc4-topology: remove shadowed variable Yong Zhi (1): ASoC: SOF: ipc4-loader: save FW version info to debugfs sound/soc/sof/amd/acp-loader.c | 2 ++ sound/soc/sof/ipc4-loader.c | 8 ++++++++ sound/soc/sof/ipc4-priv.h | 2 +- sound/soc/sof/ipc4-topology.c | 1 - 4 files changed, 11 insertions(+), 2 deletions(-) -- 2.40.1

2 5

[PATCH v2 0/3] ASoC: Use snd_ctl_find_id_mixer() instead of open-coding
by Richard Fitzgerald 03 Apr '24

03 Apr '24

The first two patches change snd_soc_card_get_kcontrol() to use the core snd_ctl_find_id_mixer() functionality instead of open-coding its own list walk. The last patch adds a KUnit test for this, which was tested on the original and modified code. Changes in V2: Only change is in the KUnit test (patch #3) to make the const strings more consty. Richard Fitzgerald (3): ALSA: control: Introduce snd_ctl_find_id_mixer_locked() ASoC: soc-card: Use snd_ctl_find_id_mixer() instead of open-coding ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol include/sound/control.h | 23 +++++ sound/soc/Kconfig | 8 ++ sound/soc/Makefile | 4 + sound/soc/soc-card-test.c | 184 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-card.c | 21 +---- 5 files changed, 223 insertions(+), 17 deletions(-) create mode 100644 sound/soc/soc-card-test.c -- 2.39.2

3 5

[PATCH 0/4] ASoC: codecs: ES8326: solve some hp issues and
by Zhang Yi 02 Apr '24

02 Apr '24

We solved some issues related to headphone detection.And for using the same configuration in different power conditions,we modified the clock table Zhang Yi (4): ASoC: codecs: ES8326: Solve error interruption issue ASoC: codecs: ES8326: modify clock table ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume ASoC: codecs: ES8326: Removing the control of ADC_SCALE sound/soc/codecs/es8326.c | 37 +++++++++++++++++++++---------------- sound/soc/codecs/es8326.h | 2 +- 2 files changed, 22 insertions(+), 17 deletions(-) -- 2.17.1

2 5

[PATCH] ASoC: Intel: avs: boards: Add modules description
by Amadeusz Sławiński 02 Apr '24

02 Apr '24

Modpost warns about missing module description, add it. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/avs/boards/da7219.c | 1 + sound/soc/intel/avs/boards/dmic.c | 1 + sound/soc/intel/avs/boards/es8336.c | 1 + sound/soc/intel/avs/boards/i2s_test.c | 1 + sound/soc/intel/avs/boards/max98357a.c | 1 + sound/soc/intel/avs/boards/max98373.c | 1 + sound/soc/intel/avs/boards/max98927.c | 1 + sound/soc/intel/avs/boards/nau8825.c | 1 + sound/soc/intel/avs/boards/probe.c | 1 + sound/soc/intel/avs/boards/rt274.c | 1 + sound/soc/intel/avs/boards/rt286.c | 1 + sound/soc/intel/avs/boards/rt298.c | 1 + sound/soc/intel/avs/boards/rt5514.c | 1 + sound/soc/intel/avs/boards/rt5663.c | 1 + sound/soc/intel/avs/boards/rt5682.c | 1 + sound/soc/intel/avs/boards/ssm4567.c | 1 + 16 files changed, 16 insertions(+) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index c018f84fe0252..fc072dc58968c 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -296,5 +296,6 @@ static struct platform_driver avs_da7219_driver = { module_platform_driver(avs_da7219_driver); +MODULE_DESCRIPTION("Intel da7219 machine driver"); MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski(a)intel.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index ba2bc7f689eb6..d9e5e85f52335 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -96,4 +96,5 @@ static struct platform_driver avs_dmic_driver = { module_platform_driver(avs_dmic_driver); +MODULE_DESCRIPTION("Intel DMIC machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 1090082e7d5bf..5c90a60075773 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -326,4 +326,5 @@ static struct platform_driver avs_es8336_driver = { module_platform_driver(avs_es8336_driver); +MODULE_DESCRIPTION("Intel es8336 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 282256d18cc69..d419556856643 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -125,4 +125,5 @@ static struct platform_driver avs_i2s_test_driver = { module_platform_driver(avs_i2s_test_driver); +MODULE_DESCRIPTION("Intel i2s test machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index a83b95f25129f..1ff85e4d8e160 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -154,4 +154,5 @@ static struct platform_driver avs_max98357a_driver = { module_platform_driver(avs_max98357a_driver) +MODULE_DESCRIPTION("Intel max98357a machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index 3b980a025e6f6..8d31586b73eae 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -211,4 +211,5 @@ static struct platform_driver avs_max98373_driver = { module_platform_driver(avs_max98373_driver) +MODULE_DESCRIPTION("Intel max98373 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index 86dd2b228df3a..572ec58073d06 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -208,4 +208,5 @@ static struct platform_driver avs_max98927_driver = { module_platform_driver(avs_max98927_driver) +MODULE_DESCRIPTION("Intel max98927 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 1c1e2083f474d..55db75efae414 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -313,4 +313,5 @@ static struct platform_driver avs_nau8825_driver = { module_platform_driver(avs_nau8825_driver) +MODULE_DESCRIPTION("Intel nau8825 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index a9469b5ecb402..8be6887bbc6e8 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -69,4 +69,5 @@ static struct platform_driver avs_probe_mb_driver = { module_platform_driver(avs_probe_mb_driver); +MODULE_DESCRIPTION("Intel probe machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index bfcb8845fd15d..1cf5242160875 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -276,4 +276,5 @@ static struct platform_driver avs_rt274_driver = { module_platform_driver(avs_rt274_driver); +MODULE_DESCRIPTION("Intel rt274 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 28d7d86b1cc99..4740bba105703 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -247,4 +247,5 @@ static struct platform_driver avs_rt286_driver = { module_platform_driver(avs_rt286_driver); +MODULE_DESCRIPTION("Intel rt286 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 80f490b9e1184..6e409e29f6974 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -266,4 +266,5 @@ static struct platform_driver avs_rt298_driver = { module_platform_driver(avs_rt298_driver); +MODULE_DESCRIPTION("Intel rt298 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index 60105f453ae23..097ae5f73241e 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -192,4 +192,5 @@ static struct platform_driver avs_rt5514_driver = { module_platform_driver(avs_rt5514_driver); +MODULE_DESCRIPTION("Intel rt5514 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index b4762c2a7bf2d..1880c315cc4d1 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -265,4 +265,5 @@ static struct platform_driver avs_rt5663_driver = { module_platform_driver(avs_rt5663_driver); +MODULE_DESCRIPTION("Intel rt5663 machine driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 243f979fda98a..594a971ded9eb 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -341,5 +341,6 @@ static struct platform_driver avs_rt5682_driver = { module_platform_driver(avs_rt5682_driver) +MODULE_DESCRIPTION("Intel rt5682 machine driver"); MODULE_AUTHOR("Cezary Rojewski <cezary.rojewski(a)intel.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index abb87bb88fffe..9e78a3b912087 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -197,4 +197,5 @@ static struct platform_driver avs_ssm4567_driver = { module_platform_driver(avs_ssm4567_driver) +MODULE_DESCRIPTION("Intel ssm4567 machine driver"); MODULE_LICENSE("GPL"); -- 2.34.1

2 1

[PATCH v12] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 02 Apr '24

02 Apr '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v12: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name - use __func__ in the dev_err message in two identical logs intas2783_apply_calibv1/v2. - In tas2783_apply_calibv2, the very least move initializations to a different line. - No need for separate j variable, reuse i in tasdevice_read_prop. - remove unused "struct sdw_bus_params params", unsigned char dev_name[32] pstream, and chip_id - define dspfw_binaryname as a local variable in tasdevice_comp_probe. - reduce a lot of indentation in tasdevice_mute and spilte tasdevice_mute into tasdevice_enable_echo_ref and tasdevice_playback - remove tas2783_reg_defaults - use REGCACHE_MAPLE instead of REGCACHE_RBTREE - define fw_node_sz to store the node size - remove "struct snd_soc_component *component" - Add some SDCA registers as readable - remove unnecessary dev_dbg log for timestamp - playback still work without calibrated data stored in UEFI, for example bypass mode. Even if in case of dsp mode, algo can still work with default calibrated data. So, not an error, use dev_info for them --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 892 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1016 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f78ea2f86fa6..939384837d5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1880,6 +1881,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7c075539dc47..6f27f810affc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -287,6 +287,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -672,6 +673,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..429191b79f8d --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,892 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +struct calibdatav2_info { + unsigned int number_of_devices; + unsigned int crc32_pos; + unsigned int byt_sz; + unsigned int *cali_data; +}; + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0): + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0): + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data V1 in UEFI(total 256 bytes): + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "%s: Cali failed %x:%d\n", + __func__, tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and uniqie_id for + * multi-link solutions + * 2. Support flexible number of devices instead of fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *tmp_val) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + const struct calibdatav2_info calib_info = { + .number_of_devices = tmp_val[1], + .crc32_pos = 3 + tmp_val[1] * 6, + .byt_sz = 12 + tmp_val[1] * 24, + .cali_data = &tmp_val[3] + }; + struct regmap *map = tas_dev->regmap; + unsigned int i, j, k, crc; + u8 *cali_start; + u16 dev_indx; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + if (calib_info.number_of_devices > TAS2783_MAX_DEV_NUM || + calib_info.number_of_devices == 0) { + dev_info(tas_dev->dev, + "using default calibration data, wrong data in EFI."); + return; + } + crc = crc32(~0, tmp_val, calib_info.byt_sz) ^ ~0; + if (crc != tmp_val[calib_info.crc32_pos]) { + dev_info(tas_dev->dev, + "using default calibration data, crc not match."); + return; + } + + dev_indx = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + for (i = 0; i < calib_info.number_of_devices; i++) { + k = i * (arr_size + 1); + + if (dev_indx != calib_info.cali_data[k]) + continue; + + /* +1: skip 4 byte dev-indx info */ + cali_start = (u8 *)&calib_info.cali_data[k + 1]; + + for (j = 0; j < arr_size; j++) { + k = 4 * j; + ret = regmap_bulk_write(map, tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "%s: Cali data writing failed %x:%d\n", + __func__, tas2783_cali_reg[j], ret); + break; + } + } + return; + } +} + +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + /* Check Calibrated Data V1 */ + crc = crc32(~0, cali_data.data, TAS2783_CALIDATAV1_BYTE_SIZE) ^ ~0; + if (crc == tmp_val[TAS2783_CALIDATAV1_CRC32_INDX]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + return; + } + + /* Check Calibrated Data V2 */ + if (tmp_val[0] != 2783) { + dev_err(tas_dev->dev, "No tas2783 calibration data\n"); + return; + } + + tas2783_apply_calibv2(tas_dev, tmp_val); +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + unsigned char *buf; + int fw_node_sz = sizeof(struct tas2783_firmware_node); + int img_sz, ret; + int offset = 0; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read fw, work in bypass-dsp mode.\n", + __func__); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + fw_node_sz, p->length); + else + ret = regmap_write(map, p->download_addr, + *(buf + offset + fw_node_sz)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += fw_node_sz + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +/* Capture stream is the echo ref data for voice mode.*/ +static int tasdevice_enable_echo_ref(struct tasdevice_priv *tas_dev, + int mute) +{ + struct regmap *map = tas_dev->regmap; + unsigned int val; + int ret; + + if (mute) + val = TAS2783_REG_AEF_INACTIVE; + else + val = TAS2783_REG_AEF_ACTIVE; + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, "%s: Disable AEF failed.\n", __func__); + + return ret; +} + +static int tasdevice_playback(struct tasdevice_priv *tas_dev, int mute) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + if (mute) { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, "%s: FU23 mute failed.\n", + __func__); + goto out; + } + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, "%s: shutdown failed.\n", + __func__); + } + } else { + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + } + } +out: + return ret; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (direction == SNDRV_PCM_STREAM_CAPTURE) + ret = tasdevice_enable_echo_ref(tas_dev, mute); + else + ret = tasdevice_playback(tas_dev, mute); + + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-tas2783.bin", comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, dspfw_binaryname, tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, tasdevice_dai_driver, + ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..1b91eae47f29 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +#define TAS2783_CALIDATAV1_TIMESTAMP_INDX 40 +#define TAS2783_CALIDATAV1_CRC32_INDX 41 +/* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ +#define TAS2783_CALIDATAV1_BYTE_SIZE (8 * 20 + 4) + +#define TAS2783_CALIDATAV2_TIMESTAMP_INDX 2 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct sdw_slave *sdw_peripheral; + struct regmap *regmap; + struct device *dev; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

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[PATCH] ALSA: aaci: report FIFO size in frames
by Oswald Buddenhagen 02 Apr '24

02 Apr '24

We already have frames, so don't convert them to bytes - the mid-layer would convert them to frames again anyway. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Is this actually right? What about stereo? cf. 5d350cba486de34eff99d0394d8fb436af54522e Cc: Russell King <rmk(a)arm.linux.org.uk> --- sound/arm/aaci.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index c3340b8ff3da..3655e88f3fca 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -350,6 +350,7 @@ static const struct snd_pcm_hardware aaci_hw_info = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_FIFO_IN_FRAMES | SNDRV_PCM_INFO_RESUME, /* @@ -430,12 +431,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) snd_ac97_pcm_double_rate_rules(runtime); } - /* - * ALSA wants the byte-size of the FIFOs. As we only support - * 16-bit samples, this is twice the FIFO depth irrespective - * of whether it's in compact mode or not. - */ - runtime->hw.fifo_size = aaci->fifo_depth * 2; + runtime->hw.fifo_size = aaci->fifo_depth; mutex_lock(&aaci->irq_lock); if (!aaci->users++) { -- 2.42.0.419.g70bf8a5751

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[PATCH] ALSA: hda/realtek: cs35l41: Support ASUS ROG G634JYR
by Luke D. Jones 02 Apr '24

02 Apr '24

Fixes the realtek quirk to initialise the Cirrus amp correctly and adds related quirk for missing DSD properties. This model laptop has slightly updated internals compared to the previous version with Realtek Codec ID of 0x1caf. Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..25c117db3317 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -108,6 +108,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 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 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, @@ -496,6 +497,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, { "CSC3551", "17AA38AB", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..8da2827bb3c3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10133,7 +10133,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 2.44.0

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