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

HDA ATI HDMI not supported
by GitHub issues - edited 29 Jul '22

29 Jul '22

alsa-project/alsa-ucm-conf issue #194 was edited from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… The `alsaucm reload` command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` and the HDMI is not working. despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 29 Jul '22

29 Jul '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 215 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 215 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..3e3d99febd69 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,215 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - const: fsl,vf610-sai + - const: fsl,imx6sx-sai + - const: fsl,imx6ul-sai + - const: fsl,imx7ulp-sai + - const: fsl,imx8mq-sai + - const: fsl,imx8qm-sai + - const: fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will output the SAI MCLK clock. + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Interrupt is shared with other modules. + + "#sound-dai-cells": + const: 0 + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + #sound-dai-cells = <0>; + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - 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". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

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[GIT PULL] ASoC fixes for v5.19-rc8
by Mark Brown 28 Jul '22

28 Jul '22

The following changes since commit 1e347f861da8ddb17e1d1b3113cb6c188e0de3e5: ASoC: rockchip-i2s: Undo BCLK pinctrl changes (2022-07-14 13:25:52 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc8 for you to fetch changes up to 30097967e0566cac817273ef76add100f6b0f463: ASoC: codecs: va-macro: use fsgen as clock (2022-07-27 14:46:41 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more small fixes, they could all wait for the merge window if you prefer. ---------------------------------------------------------------- Ryan Lee (1): ASoC: max98373: Removing 0x203E from the volatile reg Srinivas Kandagatla (1): ASoC: codecs: va-macro: use fsgen as clock Wang Jiada (1): ASoC: rsnd: ssiu: add missing .quit callback for gen2 sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- sound/soc/codecs/max98373-i2c.c | 1 - sound/soc/sh/rcar/ssiu.c | 1 + 3 files changed, 10 insertions(+), 3 deletions(-)

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[PATCH] ASoC: qcom: SC7280: Add support for external DMIC bias supply
by Srinivasa Rao Mandadapu 28 Jul '22

28 Jul '22

Update SC7280 machine driver for enabling external mic bias supply, Which is required for villager rev boards. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/sc7280.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index da7469a..6404e94 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -33,6 +33,7 @@ struct sc7280_snd_data { struct snd_soc_jack hdmi_jack; bool jack_setup; bool stream_prepared[LPASS_MAX_PORTS]; + struct regulator *vdd_supply; }; static void sc7280_jack_free(struct snd_jack *jack) @@ -345,6 +346,24 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) return ret; } +static int sc7280_dmic_micbias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(data->vdd_supply); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(data->vdd_supply); + break; + } + return ret; +} + static const struct snd_soc_ops sc7280_ops = { .startup = sc7280_snd_startup, .hw_params = sc7280_snd_hw_params, @@ -356,6 +375,7 @@ static const struct snd_soc_ops sc7280_ops = { static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("PMIC BIAS", sc7280_dmic_micbias), }; static int sc7280_snd_platform_probe(struct platform_device *pdev) @@ -389,6 +409,10 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev) link->ops = &sc7280_ops; } + data->vdd_supply = devm_regulator_get(dev, "vdd-dmic-bias"); + if (IS_ERR(data->vdd_supply) + return PTR_ERR(data->vdd_supply); + return devm_snd_soc_register_card(dev, card); } -- 2.7.4

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[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

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[PATCH v2] ASoC: amd: acp: Initialize list to store acp_stream during pcm_open
by Venkata Prasad Potturu 28 Jul '22

28 Jul '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> We are currently allocating acp_stream during pcm_open and saving it in static array corresponds to array index calculated based on cpu dai->driver id. This approach will fail if we have single dai linked to multiple pcm device as we will have same dai->driver id or array index for multiple pcm open. Initialize new linked list stream_list to store opened pcm stream info dynamically. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-platform.c | 33 ++++++++++++-------------------- sound/soc/amd/acp/amd.h | 3 ++- 2 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 10730d33c3b0..20c0f75f7c97 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,7 +94,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 ext_intr_stat, ext_intr_stat1, i; + u32 ext_intr_stat, ext_intr_stat1; if (!adata) return IRQ_NONE; @@ -104,8 +104,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); - for (i = 0; i < ACP_MAX_STREAM; i++) { - stream = adata->stream[i]; + list_for_each_entry(stream, &adata->stream_list, list) { if (stream && (ext_intr_stat & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); @@ -146,9 +145,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } -static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) +static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size) { - struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; @@ -174,13 +172,10 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; int ret; stream = kzalloc(sizeof(*stream), GFP_KERNEL); @@ -188,7 +183,8 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + list_add_tail(&stream->list, &adata->stream_list); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +208,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); struct acp_dev_data *adata = snd_soc_component_get_drvdata(component); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct acp_stream *stream = substream->runtime->private_data; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; u64 size = params_buffer_bytes(params); /* Configure ACP DMA block with params */ config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream_id, size); + config_acp_dma(adata, stream, size); return 0; } @@ -261,16 +254,11 @@ static int acp_dma_new(struct snd_soc_component *component, static int acp_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); - struct device *dev = component->dev; - struct acp_dev_data *adata = dev_get_drvdata(dev); - struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; + struct acp_stream *stream = substream->runtime->private_data; - stream = adata->stream[stream_id]; + /* Remove entry from list */ + list_del(&stream->list); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +293,9 @@ int acp_platform_register(struct device *dev) dev_err(dev, "Fail to register acp i2s component\n"); return status; } + + INIT_LIST_HEAD(&adata->stream_list); + return 0; } EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index af9603724a68..148a9ab6206d 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -91,6 +91,7 @@ struct acp_chip_info { }; struct acp_stream { + struct list_head list; struct snd_pcm_substream *substream; int irq_bit; int dai_id; @@ -123,7 +124,7 @@ struct acp_dev_data { struct snd_soc_dai_driver *dai_driver; int num_dai; - struct acp_stream *stream[ACP_MAX_STREAM]; + struct list_head stream_list; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; -- 2.25.1

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[PATCH v2 0/4] Add support for CLSA0101
by Lucas Tanure 28 Jul '22

28 Jul '22

Add Support for the CLSA0101 laptop, an Intel version of CLSA0100. This patch has been tested using the CLSA0100, ensuring it doesn't break the sound for it. We appreciate it if someone with CLSA0101 could verify that this the patch works for them. Changes from V1: - Add CLSA0101 id into scan.c, serial-multi-instantiate.c and cs35l41_hda_i2c.c Lucas Tanure (4): ALSA: hda: cs35l41: Use the CS35L41 HDA internal define ALSA: hda: cs35l41: Support CLSA0101 ACPI: scan: Add CLSA0101 Laptop Support platform/x86: serial-multi-instantiate: Add CLSA0101 Laptop drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 1 + sound/pci/hda/cs35l41_hda.c | 67 ++++++++++++------- sound/pci/hda/cs35l41_hda_i2c.c | 3 + sound/pci/hda/patch_realtek.c | 12 ++++ 5 files changed, 58 insertions(+), 26 deletions(-) -- 2.37.1

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Add correct support for Rane SL-1
by GitHub pull_request - edited 28 Jul '22

28 Jul '22

alsa-project/alsa-ucm-conf pull request #187 was edited from christophehenry: Following [this ticket](https://bugzilla.kernel.org/show_bug.cgi?id=216082) on kernel bugzilla, I'm trying to add corect support for [Rane SL-1](https://cdn.korn.eu/pictures/product/400/040506_3.jpg), an old USB soundcard featuring 2 stereo output (RCA) and 3 stereo input (2 RCA, 1 6.3mm jack). Currently, `alsa-info.sh` does show the correct number of inputs and outputs But neither `pavucontrol` nor Mixxx do. I was told complex hardware should be described via UCM but the syntax is unfortunately very poorly documented so I can get my head around it and get `pavucontrol` to show all the inputs and outputs. Here's what I have so far which, to my disappointment, does not work. If I put only 1 `SectionDevice` section in the file, I can display 1 ouput under `pavucontrol`. With 2 `SectionDevice`, nothing shows. Without any UCM file, `pavucontrol` shows 2 outputs, 1 intputs (instead of 3) and Mixxx shows 2 outputs and not input. [The result of `alsa-info.sh` is available here](https://bugzilla.kernel.org/attachment.cgi?id=301351). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/187.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 28 Jul '22

28 Jul '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..c08ef317e3e3 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_dma; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_dma: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

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[PATCH v3 0/2] ASoC: atmel: one fix and one cleanup
by Claudiu Beznea 28 Jul '22

28 Jul '22

Hi, The series adds one fix for mchp-spdifrx and one cleanups for mchp-spdifrx and mchp-spdifrx drivers. Thank you, Claudiu Beznea Changes in v3: - changed cover letter title s/few/one, s/cleanups/cleanup - fix compilation error and warnings - keep only patch 1/5 and patch 3/5 from previous version as the rest of them were integrated Changes in v2: - s/tag/tab in the title of patch 2/5 Claudiu Beznea (2): ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdiftx: remove references to mchp_i2s_caps sound/soc/atmel/mchp-spdifrx.c | 9 ++++++--- sound/soc/atmel/mchp-spdiftx.c | 8 -------- 2 files changed, 6 insertions(+), 11 deletions(-) -- 2.34.1

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