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

[PATCH] ASoC: tlv320adcx140: Fix a typo in a comment
by Christophe JAILLET 15 Aug '22

15 Aug '22

s/TLV320ADCX104/TLV320ADCX140/ Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/tlv320adcx140.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index d7d4e3a88b5c..795b5f7194e6 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// TLV320ADCX104 Sound driver +// TLV320ADCX140 Sound driver // Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H -- 2.34.1

2 1

[PATCH 0/8] ASoC: Replace sprintf() with sysfs_emit()
by Takashi Iwai 15 Aug '22

15 Aug '22

Hi, this is a patch set for rather simple conversions from the plain sprintf() & co to the new helpers, sysfs_emit() and sysfs_emit_at(). No functional changes are expected. Takashi === Takashi Iwai (8): ASoC: cs43130: Replace scnprintf() with sysfs_emit() ASoC: tlv320aic26: Replace sprintf() with sysfs_emit() ASoC: Intel: sst: Replace sprintf() with sysfs_emit() ASoC: Intel: catpt: Replace sprintf() with sysfs_emit() ASoC: Intel: skylake: Replace sprintf() with sysfs_emit() ASoC: core: Replace sprintf() with sysfs_emit() ASoC: DAPM: Replace sprintf() calls with sysfs_emit_at() ASoC: omap: Replace sprintf() with sysfs_emit() sound/soc/codecs/cs43130.c | 11 +++++------ sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/intel/atom/sst/sst.c | 8 ++++---- sound/soc/intel/catpt/sysfs.c | 6 +++--- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 9 ++++----- sound/soc/ti/omap-mcbsp-st.c | 6 +++--- sound/soc/ti/omap-mcbsp.c | 8 ++++---- 9 files changed, 26 insertions(+), 28 deletions(-) -- 2.35.3

4 12

[PATCH v2] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use unsigned long for sysclk sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..4d99f4858a14 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned long sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

2 1

[PATCH] ASoC: mediatek: mt8186: remove unnecessary NULL check
by Dan Carpenter 15 Aug '22

15 Aug '22

The "i2s_priv" pointer cannot be NULL. Some NULL checks were deleted in commit d7bffbe9cbd3 ("ASoC: mediatek: mt8186: remove unnecessary judgments") but this one was accidentally left behind. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index ec79e2f2a54d..d7a227169548 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -968,7 +968,7 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, } /* set share i2s */ - if (i2s_priv && i2s_priv->share_i2s_id >= 0) { + if (i2s_priv->share_i2s_id >= 0) { ret = mtk_dai_i2s_config(afe, params, i2s_priv->share_i2s_id); if (ret) return ret; -- 2.35.1

2 1

[PATCH v4] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 15 Aug '22

15 Aug '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. Beside conversion, 'fsl,shared-interrupt' and '#sound-dai-cells' are added for they are already used by some dts. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v4 - remove $ref for boolean properties changes in v3 - address comments from Krzysztof changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 216 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 216 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..70c4111d59c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,216 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - 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: + 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. + type: boolean + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + 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. + type: boolean + + fsl,sai-asynchronous: + 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. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + 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: + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +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 { + 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>; + #sound-dai-cells = <0>; + }; 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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[PATCH v3] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 15 Aug '22

15 Aug '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. The UIDs for these amps in ACPI are fixed, and map to a name_prefix, where: UID 0x0 -> WL UID 0x1 -> WR UID 0x2 -> TL UID 0x3 -> TR Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 92 ++++++++++++---------- 1 file changed, 49 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..6e39eda77385 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,10 +140,51 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +/* + * Expected UIDs are integers (stored as strings). + * UID Mapping is fixed: + * UID 0x0 -> WL + * UID 0x1 -> WR + * UID 0x2 -> TL + * UID 0x3 -> TR + * Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create + * dai links for UIDs which exist, and ignore non-existant ones. Only 2 or 4 amps are expected. + * Return number of codecs found. + */ +static int cs35l41_compute_codec_conf(void) +{ + const char * const uid_strings[] = { "0", "1", "2", "3" }; + unsigned int uid, sz = 0; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, uid_strings[uid], -1); + if (!adev) { + pr_devel("Cannot find match for HID %s UID %u (%s)\n", CS35L41_HID, uid, + cs35l41_name_prefixes[uid]); + continue; + } + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[sz].name = dev_name(physdev); + cs35l41_components[sz].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[sz].dlc.name = dev_name(physdev); + cs35l41_codec_conf[sz].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + sz++; + } + + if (sz != 2 && sz != 4) + pr_warn("Invalid number of cs35l41 amps found: %d, expected 2 or 4\n", sz); + return sz; +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + link->num_codecs = cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; - link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; link->ops = &cs35l41_ops; } -- 2.34.1

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[PATCH] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..bef4bd997c98 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned int sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

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[PATCH v1] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 15 Aug '22

15 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..b9d94f3a5b99 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

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Re: [PATCH v1] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Takashi Iwai 15 Aug '22

15 Aug '22

On Mon, 15 Aug 2022 16:19:53 +0200, Stefan Binding wrote: > > These Asus Zenbook laptop use Realtek HDA codec combined with > 2xCS35L41 Amplifiers using SPI. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Thanks, applied. Takashi

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[RFC PATCH] Soundwire: Initialize multi_link with fwnode props
by Khalid Masum 15 Aug '22

15 Aug '22

According to the TODO, In sw_bus_master_add, bus->multi_link is to be populated with properties from FW node props. Make this happen by creating a new fwnode_handle flag FWNODE_FLAG_MULTI_LINKED and use the flag to store the multi_link value from intel_link_startup. Use this flag to initialize bus->multi_link. Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> --- I do not think adding a new flag for fwnode_handle is a good idea. So, what would be the best way to initialize bus->multilink with fwnode props? -- Khalid Masum drivers/soundwire/bus.c | 4 ++-- drivers/soundwire/intel.c | 1 + include/linux/fwnode.h | 1 + 3 files changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a2bfb0434a67..80df1672c60b 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -74,9 +74,9 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* * Initialize multi_link flag - * TODO: populate this flag by reading property from FW node */ - bus->multi_link = false; + bus->multi_link = (fwnode->flags & FWNODE_FLAG_MULTI_LINKED) + == FWNODE_FLAG_MULTI_LINKED; if (bus->ops->read_prop) { ret = bus->ops->read_prop(bus); if (ret < 0) { diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 505c5ef061e3..034d1c523ddf 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1347,6 +1347,7 @@ int intel_link_startup(struct auxiliary_device *auxdev) */ bus->multi_link = true; bus->hw_sync_min_links = 1; + dev->fwnode->flags |= FWNODE_FLAG_MULTI_LINKED; } /* Initialize shim, controller */ diff --git a/include/linux/fwnode.h b/include/linux/fwnode.h index 9a81c4410b9f..446a52744953 100644 --- a/include/linux/fwnode.h +++ b/include/linux/fwnode.h @@ -32,6 +32,7 @@ struct device; #define FWNODE_FLAG_NOT_DEVICE BIT(1) #define FWNODE_FLAG_INITIALIZED BIT(2) #define FWNODE_FLAG_NEEDS_CHILD_BOUND_ON_ADD BIT(3) +#define FWNODE_FLAG_MULTI_LINKED BIT(4) struct fwnode_handle { struct fwnode_handle *secondary; -- 2.37.1

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