January 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks
by Takashi Iwai 29 Jan '22

29 Jan '22

The recent change for DPCM locking caused spurious lockdep warnings. Actually the warnings are false-positive, as those are triggered due to the nested stream locks for FE and BE. Since both locks belong to the same lock class, lockdep sees it as if a deadlock. For fixing this, we need take PCM stream locks for BE with the nested lock primitives. Since currently snd_pcm_stream_lock*() helper assumes only the top-level single locking, a new helper function snd_pcm_stream_lock_irqsave_nested() is defined for a single-depth nested lock, which is now used in the BE DAI trigger that is always performed inside a FE stream lock. Fixes: b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking") Reported-and-tested-by: Hans de Goede <hdegoede(a)redhat.com> Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Link: https://lore.kernel.org/r/73018f3c-9769-72ea-0325-b3f8e2381e30@redhat.com Link: https://lore.kernel.org/alsa-devel/9a0abddd-49e9-872d-2f00-a1697340f786@sam… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 15 +++++++++++++++ sound/core/pcm_native.c | 13 +++++++++++++ sound/soc/soc-pcm.c | 6 +++--- 3 files changed, 31 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 9b187d86e1bd..36da42cd0774 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -617,6 +617,7 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream); void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream); /** * snd_pcm_stream_lock_irqsave - Lock the PCM stream @@ -635,6 +636,20 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags); +/** + * snd_pcm_stream_lock_irqsave_nested - Single-nested PCM stream locking + * @substream: PCM substream + * @flags: irq flags + * + * This locks the PCM stream like snd_pcm_stream_lock_irqsave() but with + * the single-depth lockdep subclass. + */ +#define snd_pcm_stream_lock_irqsave_nested(substream, flags) \ + do { \ + typecheck(unsigned long, flags); \ + flags = _snd_pcm_stream_lock_irqsave_nested(substream); \ + } while (0) + /** * snd_pcm_group_for_each_entry - iterate over the linked substreams * @s: the iterator diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..a056b3ef3c84 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -172,6 +172,19 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream) +{ + unsigned long flags = 0; + if (substream->pcm->nonatomic) + mutex_lock_nested(&substream->self_group.mutex, + SINGLE_DEPTH_NESTING); + else + spin_lock_irqsave_nested(&substream->self_group.lock, flags, + SINGLE_DEPTH_NESTING); + return flags; +} +EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested); + /** * snd_pcm_stream_unlock_irqrestore - Unlock the PCM stream * @substream: PCM substream diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7abfc48b26ca..e8876e65c649 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -46,8 +46,8 @@ static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); } -#define snd_soc_dpcm_stream_lock_irqsave(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave(snd_soc_dpcm_get_substream(rtd, stream), flags) +#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ + snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, int stream) @@ -2094,7 +2094,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave(be, stream, flags); + snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) -- 2.31.1

3 4

[PATCH] ASoC: SOF: Intel: hda: Remove link assignment limitation
by Peter Ujfalusi 28 Jan '22

28 Jan '22

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The limitation to assign a link DMA channel for a BE iff the corresponding host DMA channel is assigned to a connected FE is only applicable if the PROCEN_FMT_QUIRK is set. So, remove it for platforms that do not enable the quirk. Complements: a792bfc1c2bc ("ASoC: SOF: Intel: hda-stream: limit PROCEN workaround") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index cd12589355ef..28a54145c150 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -59,6 +59,8 @@ static struct hdac_ext_stream * { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct sof_intel_hda_stream *hda_stream; + const struct sof_intel_dsp_desc *chip; + struct snd_sof_dev *sdev; struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; @@ -77,9 +79,20 @@ static struct hdac_ext_stream * continue; hda_stream = hstream_to_sof_hda_stream(hstream); + sdev = hda_stream->sdev; + chip = get_chip_info(sdev->pdata); /* check if link is available */ if (!hstream->link_locked) { + /* + * choose the first available link for platforms that do not have the + * PROCEN_FMT_QUIRK set. + */ + if (!(chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK)) { + res = hstream; + break; + } + if (stream->opened) { /* * check if the stream tag matches the stream -- 2.35.0

2 1

[PATCH] ASoC: SOF: trace: Simplify count adjustment in trace_read
by Peter Ujfalusi 28 Jan '22

28 Jan '22

The first count check and fixup against "buffer - lpos" can be removed as we will do the adjustment later against the "avail" in sof_dfsentry_trace_read() Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/trace.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index f13024c8ebf2..9b505c4fe794 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -308,9 +308,6 @@ static ssize_t sof_dfsentry_trace_read(struct file *file, char __user *buffer, lpos_64 = lpos; lpos = do_div(lpos_64, buffer_size); - if (count > buffer_size - lpos) /* min() not used to avoid sparse warnings */ - count = buffer_size - lpos; - /* get available count based on current host offset */ avail = sof_wait_trace_avail(sdev, lpos, buffer_size); if (sdev->dtrace_error) { @@ -319,7 +316,8 @@ static ssize_t sof_dfsentry_trace_read(struct file *file, char __user *buffer, } /* make sure count is <= avail */ - count = avail > count ? count : avail; + if (count > avail) + count = avail; /* copy available trace data to debugfs */ rem = copy_to_user(buffer, ((u8 *)(dfse->buf) + lpos), count); -- 2.35.0

2 1

[PATCH v2 0/5] Tegra234 APE support
by Sameer Pujar 28 Jan '22

28 Jan '22

This adds APE support on Tegra234 and enables sound card on Jetson AGX Orin platform. Changelog ========= v1->v2 ------ * Rebased patches on latest linux-next Mohan Kumar (1): ASoC: tegra: Update AHUB driver for Tegra234 Sameer Pujar (4): dt-bindings: Add Tegra234 APE support dt-bindings: Document Tegra234 APE support arm64: tegra: Add audio devices on Tegra234 arm64: tegra: APE sound card for Jetson AGX Orin .../bindings/bus/nvidia,tegra210-aconnect.yaml | 1 + .../bindings/dma/nvidia,tegra210-adma.yaml | 4 +- .../bindings/interrupt-controller/arm,gic.yaml | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 4 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 4 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 1 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 1 + .../bindings/sound/nvidia,tegra210-amx.yaml | 3 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 1 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 1 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 1 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 1 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 1 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 1783 ++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 432 +++++ include/dt-bindings/clock/tegra234-clock.h | 74 +- include/dt-bindings/memory/tegra234-mc.h | 7 + include/dt-bindings/power/tegra234-powergate.h | 9 + sound/soc/tegra/tegra210_ahub.c | 146 +- sound/soc/tegra/tegra210_ahub.h | 4 +- 20 files changed, 2473 insertions(+), 6 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.7.4

2 6

[PATCH] ASoC: dt-bindings: realtek, rt5682s: Drop Tegra specifics from example
by Rob Herring 28 Jan '22

28 Jan '22

There's no need to complicate examples with a platform specific macro. It also complicates example parsing to figure out the number of interrupt cells in examples (based on bracketing). Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/sound/realtek,rt5682s.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index d65c0ed5060c..ca5b8987b749 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -21,6 +21,7 @@ properties: description: I2C address of the device. interrupts: + maxItems: 1 description: The CODEC's interrupt output. realtek,dmic1-data-pin: @@ -94,7 +95,7 @@ required: examples: - | - #include <dt-bindings/gpio/tegra-gpio.h> + #include <dt-bindings/gpio/gpio.h> #include <dt-bindings/interrupt-controller/irq.h> i2c { @@ -104,10 +105,9 @@ examples: codec@1a { compatible = "realtek,rt5682s"; reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + interrupts = <6 IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = - <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + <&gpio 2 GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; -- 2.32.0

2 1

[PATCH] ASoC: fsl: Add missing error handling in pcm030_fabric_probe
by Miaoqian Lin 28 Jan '22

28 Jan '22

Add the missing platform_device_put() and platform_device_del() before return from pcm030_fabric_probe in the error handling case. Fixes: c912fa913446 ("ASoC: fsl: register the wm9712-codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index af3c3b90c0ac..02c4ae20c673 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -93,15 +93,22 @@ static int pcm030_fabric_probe(struct platform_device *op) dev_err(&op->dev, "platform_device_alloc() failed\n"); ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + goto exit_device_put; + } ret = snd_soc_register_card(card); if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); platform_set_drvdata(op, pdata); + return ret; +exit_release_dev: + platform_device_del(pdata->codec_device); +exit_device_put: + platform_device_put(pdata->codec_device); return ret; } -- 2.17.1

2 3

[PATCH 0/4] ASoC: qcom: fixes for Qualcomm codecs and q6apm
by Srinivas Kandagatla 28 Jan '22

28 Jan '22

Hi Mark, Some recent testing found few issues with wcd938x and lpass-tx codec drivers. WCD938x was accessing array out of boundaries resulting in corruption and system crashes along with not handling kcontrol put return values correctly and rx-macro had incorrect sidetone registers offsets. One final fix in q6apm to add a check if graph started before stopping it. Thanks, srini Srinivas Kandagatla (4): ASoC: codecs: wcd938x: fix incorrect used of portid ASoC: codecs: lpass-rx-macro: fix sidetone register offsets ASoC: codecs: wcd938x: fix return value of mixer put function ASoC: qdsp6: q6apm-dai: only stop graphs that are started sound/soc/codecs/lpass-rx-macro.c | 8 ++++---- sound/soc/codecs/wcd938x.c | 31 +++++++++++++++++-------------- sound/soc/qcom/qdsp6/q6apm-dai.c | 7 +++++-- 3 files changed, 26 insertions(+), 20 deletions(-) -- 2.21.0

2 7

[PATCH 1/2] ASoC: ops: fix signedness bug in snd_soc_put_volsw()
by Dan Carpenter 28 Jan '22

28 Jan '22

The "val" and "val2" variables need to signed for the checking to work as intended. Fixes: 817f7c9335ec ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-ops.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index dc0e7c8d31f3..0091fa96eb48 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -310,8 +310,9 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int invert = mc->invert; int err; bool type_2r = false; - unsigned int val2 = 0; - unsigned int val, val_mask; + unsigned int val_mask; + int val2 = 0; + int val; if (sign_bit) mask = BIT(sign_bit + 1) - 1; -- 2.20.1

2 5

[PATCH] ASoC: dt-bindings: Centralize the 'sound-dai' definition
by Rob Herring 28 Jan '22

28 Jan '22

'sound-dai' is a common property, but has duplicate type definitions. Create a new common definition to define the type and then update all the other occurrences to just define how many entries there are just like other phandle+arg properties. The constraints on the number of entries is based on the examples and could be wrong. Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> Signed-off-by: Rob Herring <robh(a)kernel.org> --- Please ack, this depends on commit abf0fee97313 ("dt-bindings: Improve phandle-array schemas") in my tree. --- .../bindings/sound/amlogic,gx-sound-card.yaml | 4 ++-- .../bindings/sound/google,sc7180-trogdor.yaml | 6 ++++-- .../bindings/sound/imx-audio-card.yaml | 7 +++++-- .../bindings/sound/qcom,sm8250.yaml | 10 +++++++--- .../bindings/sound/samsung,aries-wm8994.yaml | 5 +---- .../bindings/sound/samsung,midas-audio.yaml | 2 -- .../bindings/sound/samsung,odroid.yaml | 9 +++------ .../devicetree/bindings/sound/sound-dai.yaml | 20 +++++++++++++++++++ 8 files changed, 42 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-dai.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index 2e35aeaa8781..8b5be4b92f35 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -57,7 +57,7 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle of the CPU DAI patternProperties: @@ -71,7 +71,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle of the codec DAI required: diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 837e3faa63a9..233caa0ade87 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -62,13 +62,15 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 required: - link-name diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index d1816dd061cf..bb3a435722c7 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -59,13 +59,16 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + minItems: 1 + maxItems: 2 fsl,mclk-equal-bclk: description: Indicates mclk can be equal to bclk, especially for sai interface diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 4bfda04b4608..4ecd4080bb96 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -69,19 +69,23 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 platform: description: Holds subnode which indicates platform dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + minItems: 1 + maxItems: 4 required: - link-name diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index eb487ed3ca3b..4ffa275b3c49 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -27,9 +27,6 @@ properties: sound-dai: minItems: 2 maxItems: 2 - items: - maxItems: 1 - $ref: /schemas/types.yaml#/definitions/phandle-array description: | phandles to the I2S controller and bluetooth codec, in that order @@ -38,7 +35,7 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the WM8994 CODEC samsung,audio-routing: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 095775c598fa..ec50bcb4af5f 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -21,7 +21,6 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array maxItems: 1 description: phandle to the I2S controller required: @@ -31,7 +30,6 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array maxItems: 1 description: phandle to the WM1811 CODEC required: diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index e8122bc87362..db2513f3e168 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -37,18 +37,15 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array description: phandles to the I2S controllers codec: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array - description: | - List of phandles to the CODEC nodes, - first entry must be corresponding to the MAX98090 CODEC and - the second entry must be the phandle of the HDMI IP block node. + items: + - description: phandle of the MAX98090 CODEC + - description: phandle of the HDMI IP block node samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000000..61c6f7abc4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface consumer Device Tree Bindings + +maintainers: + - Rob Herring <robh(a)kernel.org> + +select: true + +properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A phandle plus args to digital audio interface provider(s) + +additionalProperties: true +... -- 2.32.0

4 5

Fix pre-processor defs parsing with recursive includes
by GitHub pull_request - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-utils pull request #140 was opened from ranj063: Since the Define {} config has been removed form the input_cfg, use the tplg_pp->define_cfg_merged for search for variable definitions. Also, fix the parsing logic to first merge all variables from the conf files recursively and then override them with the command line definitions. Request URL : https://github.com/alsa-project/alsa-utils/pull/140 Patch URL : https://github.com/alsa-project/alsa-utils/pull/140.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0