January 2023 - Alsa-devel - mailman.alsa-project.org

[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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[PATCH v2 00/10] ASoC: dt-bindings: audio-graph-port related update
by Kuninori Morimoto 18 Jan '23

18 Jan '23

Hi ASoC ML, DT ML Cc Geert These v2 patches fixups audio-graph-port, and its related DT schema. Audio-Graph-Card and Simple-Audio-Card are similar Card and are sharing same utils. Thus we can also sharing same schema. This patch-set fixup some Renesas's "make dtbs_check". v1 -> v2 - remove patch for Nvidia - tidyup ti,pcm3168a schema - maintainers - ports - explain more on git-log Link: https://lore.kernel.org/r/87v8mepyoy.wl-kuninori.morimoto.gx@renesas.com Geert Uytterhoeven (1): ASoC: dt-bindings: ti,pcm3168a: Convert to json-schema Kuninori Morimoto (9): ASoC: dt-bindings: audio-graph-port: use definitions for port/endpoint ASoC: dt-bindings: audio-graph-port: add definitions/ports ASoC: dt-bindings: audio-graph-port: add missing mclk-fs ASoC: dt-bindings: audio-graph-port: add clocks on endpoint ASoC: dt-bindings: audio-graph-port: remove prefix ASoC: dt-bindings: ak4613: enable Of-graph (Audio-Graph-Card) style ASoC: dt-bindings: renesas,rsnd: add missing playback/capture ASoC: dt-bindings: renesas,rsnd: #sound-dai-cells is not mandatory ASoC: dt-bindings: simple-card: add missing #address-cells/#size-cells .../devicetree/bindings/sound/ak4613.yaml | 7 ++ .../bindings/sound/audio-graph-port.yaml | 65 +++++++---- .../bindings/sound/renesas,rsnd.yaml | 27 ++++- .../bindings/sound/simple-card.yaml | 4 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 56 --------- .../bindings/sound/ti,pcm3168a.yaml | 107 ++++++++++++++++++ 6 files changed, 184 insertions(+), 82 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -- 2.25.1

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[PATCH 0/1] firewire: fix memory leak for payload of request subaction to IEC 61883-1 FCP region
by Takashi Sakamoto 17 Jan '23

17 Jan '23

Hi, This single patch fixes a bug of Linux FireWire subsystem for kernel v2.6.33 or later. In detail, please refer patch comment. My intension of this cover-letter is to request sound subsystem maintainer to send it to mainline since upstream of Linux FireWire subsystem is already inactive. To Iwai-san, I would be pleased if you alternatively send it to mainline. Thanks Takashi Sakamoto (1): firewire: fix memory leak for payload of request subaction to IEC 61883-1 FCP region drivers/firewire/core-cdev.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) -- 2.37.2

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[PATCH] ALSA: hda/ca0132: minor fix for allocation size
by Alexey V. Vissarionov 17 Jan '23

17 Jan '23

Although the "dma_chan" pointer occupies more or equal space compared to "*dma_chan", the allocation size should use the size of variable itself. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 01ef7dbffb411d9d ("ALSA: hda - Update CA0132 codec to load DSP firmware binary") Signed-off-by: Alexey V. Vissarionov <gremlin(a)altlinux.org> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0a292bf271f2e59a..acde4cd58785e0cb 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2455,7 +2455,7 @@ static int dspio_set_uint_param(struct hda_codec *codec, int mod_id, static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan) { int status = 0; - unsigned int size = sizeof(dma_chan); + unsigned int size = sizeof(*dma_chan); codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n"); status = dspio_scp(codec, MASTERCONTROL, 0x20, -- Alexey V. Vissarionov gremlin ПРИ altlinux ТЧК org; +vii-cmiii-ccxxix-lxxix-xlii GPG: 0D92F19E1C0DC36E27F61A29CD17E2B43D879005 @ hkp://keys.gnupg.net

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[PATCH 0/9] ALSA: hda: Codec-reload bug fixes and cleanups
by Cezary Rojewski 17 Jan '23

17 Jan '23

Total of 6 fixes and 3 cleanups - cleanups are last. All of the fixes address problems that present themselves in situation when user engages in codec driver reload. Second condition to reproduce is two-step initialization of HDAudio codec - this is the case only for ASoC HDAudio bus driver as snd_hda_intel calls only compound function snd_hda_codec_new(). Once these conditions are met, several reload/unload scenarios end with null-ptr-deref and page faults. Goal of the series is to allow codec/bus driver reloading without any errors. Amadeusz Sławiński (2): ALSA: hda: Reset all SIE bits in INTCTL ALSA: hda: Remove unused macro definition Cezary Rojewski (7): ALSA: hda: Do not unset preset when cleaning up codec ALSA: hda: Fix null-ptr-deref when i915 fails and hdmi is denylisted ALSA: hda: Make device usage_count consistent across subsequent probing ALSA: hda: Fix put_device() inconsistency in error path ALSA: hda: Skip event processing for unregistered codecs ALSA: hda: Fix page fault in snd_hda_codec_shutdown() ALSA: hda/realtek: Remove redundant init_hook() in alc_default_init() include/sound/hda_codec.h | 1 - include/sound/hdaudio.h | 1 + sound/hda/ext/hdac_ext_controller.c | 7 --- sound/hda/hdac_bus.c | 2 +- sound/hda/hdac_controller.c | 7 +-- sound/pci/hda/hda_bind.c | 7 +++ sound/pci/hda/hda_codec.c | 83 +++++++++++++++-------------- sound/pci/hda/patch_realtek.c | 3 -- sound/soc/codecs/hda.c | 4 +- 9 files changed, 56 insertions(+), 59 deletions(-) -- 2.25.1

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[PATCH 0/4] Add support to compress API to ipc_msg_data / set_stream_data_offset
by Daniel Baluta 17 Jan '23

17 Jan '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds compress API support to ipc_msg_data / set_stream_data_offset callbacks. Daniel Baluta (4): ASoC: SOF: Prepare ipc_msg_data to be used with compress API ASoC: SOF: Prepare set_stream_data_offset for compress API ASoC: SOF: Add support for compress API for stream data/offset ASoC: SOF: compress: Set compress data offset sound/soc/sof/amd/acp-ipc.c | 8 ++-- sound/soc/sof/amd/acp.h | 5 ++- sound/soc/sof/compress.c | 9 +++++ sound/soc/sof/intel/hda-ipc.c | 8 ++-- sound/soc/sof/intel/hda.h | 4 +- sound/soc/sof/ipc3-pcm.c | 3 +- sound/soc/sof/ipc3.c | 4 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sof/ops.h | 8 ++-- sound/soc/sof/sof-priv.h | 11 ++++-- sound/soc/sof/stream-ipc.c | 53 +++++++++++++++++++------- 12 files changed, 81 insertions(+), 36 deletions(-) -- 2.25.1

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