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[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs, speaker don't work for a HP platform
by Jeremy Szu 18 Jan '23

18 Jan '23

There is a HP platform needs ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED quirk to make mic-mute/audio-mute/speaker working. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.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 6fab7c8fc19a..02905336781a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9432,6 +9432,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.38.1

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[PATCH] ALSA: fireface: fix locking bug in ff400_copy_msg_to_user()
by Dan Carpenter 18 Jan '23

18 Jan '23

The ff400_copy_msg_to_user() function drops the spin lock to call copy_to_user(). However, if the copy_to_user() fails, then it must take the lock again before returning. Failure to take the lock leads to a double unlock in the caller, hwdep_read(). Fixes: acdebd8b4c0c ("ALSA: fireface: implement message parser for Fireface 400") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- sound/firewire/fireface/ff-protocol-former.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-former.c b/sound/firewire/fireface/ff-protocol-former.c index f58008762fe6..fa41de978756 100644 --- a/sound/firewire/fireface/ff-protocol-former.c +++ b/sound/firewire/fireface/ff-protocol-former.c @@ -680,28 +680,30 @@ static long ff400_copy_msg_to_user(struct snd_ff *ff, char __user *buf, long cou struct ff400_msg_parser *parser = ff->msg_parser; u32 type = SNDRV_FIREWIRE_EVENT_FF400_MESSAGE; long consumed = 0; + int ret = 0; if (count < 8) return 0; spin_unlock_irq(&ff->lock); - if (copy_to_user(buf, &type, sizeof(type))) - return -EFAULT; - + ret = -EFAULT; spin_lock_irq(&ff->lock); + if (ret) + return ret; count -= sizeof(type); consumed += sizeof(type); while (count >= sizeof(*parser->msgs) && parser->pull_pos != parser->push_pos) { spin_unlock_irq(&ff->lock); - if (copy_to_user(buf + consumed, parser->msgs + parser->pull_pos, sizeof(*parser->msgs))) - return -EFAULT; - + ret = -EFAULT; spin_lock_irq(&ff->lock); + if (ret) + return ret; + ++parser->pull_pos; if (parser->pull_pos >= FF400_QUEUE_SIZE) parser->pull_pos = 0; -- 2.35.1

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[PATCH v3 0/3] ASoC: SOF: sof-audio: Fixes for widget prepare and unprepare (for 6.2)
by Peter Ujfalusi 18 Jan '23

18 Jan '23

Hi, Changes since v2: - re-based on v6.2-rc4 (and tested) - tags added from AngeloGioacchino Del Regno for patch 2 Changes since v1: - patches got re-ordered to make them (hopefully) apply on stable when picked - Added stable tag for 6.1 for the patches - Added Fixes tag for the swidget NULL check on unprepare This series contains one fix (first patch) followed by a nice to have safety belts in case we get a widget from topology which is not handled by SOF and will not have corresponding swidget associated with. Mark: these patches now on top of 6.2-rc4 and will not apply without conflict on next (patch 2 and 3 will ahve conflict) Regards, Peter --- Bard Liao (2): ASoC: SOF: sof-audio: unprepare when swidget->use_count > 0 ASoC: SOF: keep prepare/unprepare widgets in sink path Ranjani Sridharan (1): ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL sound/soc/sof/sof-audio.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) -- 2.39.0

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[PATCH v2 0/3] ASoC: SOF: sof-audio: Fixes for widget prepare and unprepare
by Peter Ujfalusi 18 Jan '23

18 Jan '23

Hi, Changes since v1: - patches got re-ordered to make them (hopefully) apply on stable when picked - Added stable tag for 6.1 for the patches - Added Fixes tag for the swidget NULL check on unprepare This series contains one fix (first patch) followed by a nice to have safety belts in case we get a widget from topology which is not handled by SOF and will not have corresponding swidget associated with. Mark: can these be picked for 6.2? Regards, Peter --- Bard Liao (2): ASoC: SOF: sof-audio: Unprepare when swidget->use_count > 0 ASoC: SOF: sof-audio: keep prepare/unprepare widgets in sink path Ranjani Sridharan (1): ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL sound/soc/sof/sof-audio.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) -- 2.39.0

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[PATCH v2] ASoC: dt-bindings: simple-card: Document simple-audio-card, plat
by Faiz Abbas 18 Jan '23

18 Jan '23

The simple card driver already has support for a simple-audio-card,plat property but its not reflected in the documentation. Add documentation for this plat property. Signed-off-by: Faiz Abbas <faiz.abbas(a)arm.com> --- Changes since v1: 1. Updated commit message to indicate the property already exists 2. Removed redundant example 3. Fixed formatting issue found by 'make DT_CHECKER_FLAGS=-m dt_binding_check' Documentation/devicetree/bindings/sound/simple-card.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index ed19899bc94b..9abe99f15763 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -205,6 +205,8 @@ patternProperties: $ref: "#/definitions/dai" "^simple-audio-card,codec(@[0-9a-f]+)?$": $ref: "#/definitions/dai" + "^simple-audio-card,plat(@[0-9a-f]+)?$": + $ref: "#/definitions/dai" "^simple-audio-card,dai-link(@[0-9a-f]+)?$": description: | -- 2.25.1

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[PATCH v4 0/5] ASoC: samsung: fsd: audio support for FSD SoC
by Padmanabhan Rajanbabu 18 Jan '23

18 Jan '23

This patch series enables audio support on FSD SoC. Changes in v4: 1. Rebased and addressed review comments provided for v3. Changes in v3: 1. Addressed all the review comments provided for v2 patch. 2. Fixed compilation warnings reported by kernel test robot. Changes in v2: 1. New compatible added in Exynos I2S driver for FSD platform. 2. Added Fixup support for Exynos I2S CPU DAI. 3. Migration of manual PSR, OPCLK configuration to Exynos CPU DAI driver as fixup. 4. Migrated from dedicated sound card to simple audio card. 5. Support added for tlv320aic3x-i2c codec on FSD platform. Changes in v1: 1. Add TDM support on samsung I2S interface. 2. Allow sound card to directly configure I2S prescaler divider instead of calculating it from frame clock. 3. The sound card support for FSD SoC which utilizes samsung I2S interface as CPU DAI. Padmanabhan Rajanbabu (5): ASoC: dt-bindings: Add FSD I2S controller bindings ASoC: samsung: i2s: add support for FSD I2S arm64: dts: fsd: Add I2S DAI node for Tesla FSD arm64: dts: fsd: Add codec node for Tesla FSD arm64: dts: fsd: Add sound card node for Tesla FSD .../bindings/sound/samsung-i2s.yaml | 8 +++ arch/arm64/boot/dts/tesla/fsd-evb.dts | 53 +++++++++++++++++++ arch/arm64/boot/dts/tesla/fsd-pinctrl.dtsi | 14 +++++ arch/arm64/boot/dts/tesla/fsd.dtsi | 34 ++++++++++++ sound/soc/samsung/i2s-regs.h | 1 + sound/soc/samsung/i2s.c | 53 +++++++++++++++++++ 6 files changed, 163 insertions(+) -- 2.17.1

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[PATCH] ASoC: SOF: sof-audio: start with the right widget type
by Peter Ujfalusi 18 Jan '23

18 Jan '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> If there is a connection between a playback stream and a capture stream, all widgets that are connected to the playback stream and the capture stream will be in the list. So, we have to start with the exactly right widget type. snd_soc_dapm_aif_out is for capture stream and a playback stream should start with a snd_soc_dapm_aif_in widget. Contrarily, snd_soc_dapm_dai_in is for playback stream, and a capture stream should start with a snd_soc_dapm_dai_out widget. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index e1ab8380e7d8..785d797b73d0 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -436,11 +436,11 @@ sof_walk_widgets_in_order(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget_l for_each_dapm_widgets(list, i, widget) { /* starting widget for playback is AIF type */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK && !WIDGET_IS_AIF(widget->id)) + if (dir == SNDRV_PCM_STREAM_PLAYBACK && widget->id != snd_soc_dapm_aif_in) continue; /* starting widget for capture is DAI type */ - if (dir == SNDRV_PCM_STREAM_CAPTURE && !WIDGET_IS_DAI(widget->id)) + if (dir == SNDRV_PCM_STREAM_CAPTURE && widget->id != snd_soc_dapm_dai_out) continue; switch (op) { -- 2.39.0

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[PATCH 1/3] ASoC: dt-bindings: qcom,wsa881x: Allow sound-name-prefix
by Krzysztof Kozlowski 18 Jan '23

18 Jan '23

Reference common DAI properties to fix: sdm845-db845c.dtb: speaker@0,1: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml index d702b489320f..ac03672ebf6d 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml @@ -15,6 +15,9 @@ description: | Their primary operating mode uses a SoundWire digital audio interface. This binding is for SoundWire interface. +allOf: + - $ref: dai-common.yaml# + properties: compatible: const: sdw10217201000 @@ -39,7 +42,7 @@ required: - "#thermal-sensor-cells" - "#sound-dai-cells" -additionalProperties: false +unevaluatedProperties: false examples: - | -- 2.34.1

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[PATCH] ALSA: hda: Do not unset preset when cleaning up codec
by Cezary Rojewski 18 Jan '23

18 Jan '23

Several functions that take part in codec's initialization and removal are re-used by ASoC codec drivers implementations. Drivers mimic the behavior of hda_codec_driver_probe/remove() found in sound/pci/hda/hda_bind.c with their component->probe/remove() instead. One of the reasons for that is the expectation of snd_hda_codec_device_new() to receive a valid struct snd_card pointer what cannot be fulfilled on ASoC side until a card is attempted to be bound and its component probing is triggered. As ASoC sound card may be unbound without codec device being actually removed from the system, unsetting ->preset in snd_hda_codec_cleanup_for_unbind() interferes with module unload -> load scenario causing null-ptr-deref. Preset is assigned only once, during device/driver matching whereas ASoC codec driver's module reloading may occur several times throughout the lifetime of an audio stack. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- This is a continuation of a discussion that begun in the middle of 2022 [1] and was part of a larger series addressing several HDAudio topics. Single rmmod on ASoC's codec driver module is enough to cause a panic. Given our results, no regression shows up with modprobe/rmmod on snd_hda_intel side with this patch applied. [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-2-cezary.rojewski@… sound/pci/hda/hda_codec.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index edd653ece70d..ac1cc7c5290e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -795,7 +795,6 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->cvt_setups); snd_array_free(&codec->spdif_out); snd_array_free(&codec->verbs); - codec->preset = NULL; codec->follower_dig_outs = NULL; codec->spdif_status_reset = 0; snd_array_free(&codec->mixers); -- 2.25.1

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Question about amidi -l output
by Jeroen Baten 18 Jan '23

18 Jan '23

Hi, I hope someone can point me in the right direction. So the Ardour application uses the output of amidi -l to name USB devices. In my case I now have 3 devices all named "Boutique MIDI 1", although they are 3 different types of synth. I already added the devices to usb-list and system-hwdb so lsusb shows up with correct synth names. What can I do to make amidi -l show better device names? Where do these names come from? What lib or db should I look at to patch and submit? I hope someone can help me. Regards, Jeroen Baten -- Jeroen Baten | EMAIL : JBATEN(a)I2RS.NL ____ _ __ | web : www.i2rs.nl | )|_)(_ | tel : +31 (0)648519096 _|_/_| \__) | Frisolaan 16, 4101 JK, Culemborg, the Netherlands

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