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

[PATCH] ASoC: atmel: Add missing of_node_put() in at91sam9g20ek_audio_probe
by Miaoqian Lin 07 Mar '22

07 Mar '22

This node pointer is returned by of_parse_phandle() with refcount incremented in this function. Calling of_node_put() to avoid the refcount leak. Fixes: 531f67e41dcd ("ASoC: at91sam9g20ek-wm8731: convert to dt support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/atmel/sam9g20_wm8731.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 915da92e1ec8..33e43013ff77 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -214,6 +214,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "dai and pcm info missing\n"); + of_node_put(codec_np); return -EINVAL; } at91sam9g20ek_dai.cpus->of_node = cpu_np; -- 2.17.1

3 2

[PATCH 00/10] ASoC: SOF: updates for 5.18
by Pierre-Louis Bossart 07 Mar '22

07 Mar '22

A couple of updates for Intel and AMD hardware, along with minor cleanups Ajit Kumar Pandey (4): ASoC: SOF: amd: Flush cache after ATU_BASE_ADDR_GRP register update ASoC: SOF: amd: Use semaphore register to synchronize ipc's irq ASoC: SOF: amd: Move group register configuration to acp-loader ASoC: SOF: amd: Increase ACP_HW_SEM_RETRY_COUNT value Curtis Malainey (1): ASoC: SOF: fix 32 signed bit overflow Gongjun Song (1): ASoC: SOF: Intel: pci-tgl: add RPL-S support Peter Ujfalusi (2): ASoC: SOF: amd: acp-pcm: Take buffer information directly from runtime ASoC: SOF: amd: Do not set ipc_pcm_params ops as it is optional Pierre-Louis Bossart (2): ASoC: SOF: debug: clarify operator precedence ASoC: SOF: Intel: hda: clarify operator precedence include/sound/sof/header.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/amd/acp-dsp-offset.h | 1 + sound/soc/sof/amd/acp-ipc.c | 22 ++++++++++++++-------- sound/soc/sof/amd/acp-loader.c | 9 +++++++++ sound/soc/sof/amd/acp-pcm.c | 7 ++++--- sound/soc/sof/amd/acp-stream.c | 3 +++ sound/soc/sof/amd/acp.c | 29 ++++++++++++++--------------- sound/soc/sof/amd/acp.h | 3 +-- sound/soc/sof/amd/renoir.c | 1 - sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 2 ++ 13 files changed, 52 insertions(+), 33 deletions(-) -- 2.30.2

2 11

[PATCH 1/4] ASoC: amd: vg: fix signedness bug in acp5x_audio_probe()
by Dan Carpenter 07 Mar '22

07 Mar '22

The "adata->i2s_irq" variable is unsigned so the error handling will not work. Fixes: 87d71a128771 ("ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index e4e668593b3d..31fa166df98a 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -388,9 +388,10 @@ static int acp5x_audio_probe(struct platform_device *pdev) if (!adata->acp5x_base) return -ENOMEM; - adata->i2s_irq = platform_get_irq(pdev, 0); - if (adata->i2s_irq < 0) - return -ENODEV; + status = platform_get_irq(pdev, 0); + if (status < 0) + return status; + adata->i2s_irq = status; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, -- 2.20.1

2 7

[PATCH] ASoC: cs42l42: Add warnings about DETECT_MODE and PLL_START
by Richard Fitzgerald 07 Mar '22

07 Mar '22

DETECT_MODE and PLL_START must be zero while HP_PDN and ADC_PDN are both 1. If this condition is broken it can discharge FILT+ and it can then take up to 1 second for FILT+ to recharge. There is no workaround required for this, simply avoiding settings and sequences that would break the requirement. The driver already meets the requirement. But it is not obvious from reading the code that this requirement exists, or what is ensuring it is met. So it would not currently be obvious to someone changing the code that there is certain special behaviour that must be maintained. To avoid accidental breakage in the future: - Add comments into the register definitions to warn about this so that anyone changing the code around DETECT_MODE and PLL_START is aware of this requirement. - Add a comment where PLL_START is written to 1 to highlight the requirement and why it is satisfied. - Add a comment in cs42l42_setup_hs_type_detect() when DETECT_MODE is initialized. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 13 ++++++++++++- sound/soc/codecs/cs42l42.h | 9 ++++++++- 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index db6ef6cdce15..c8409d50e934 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1012,7 +1012,14 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) } } else { if (!cs42l42->stream_use) { - /* SCLK must be running before codec unmute */ + /* SCLK must be running before codec unmute. + * + * PLL must not be started with ADC and HP both off + * otherwise the FILT+ supply will not charge properly. + * DAPM widgets power-up before stream unmute so at least + * one of the "DAC" or "ADC" widgets will already have + * powered-up. + */ if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); @@ -1830,6 +1837,10 @@ static void cs42l42_setup_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = CS42L42_PLUG_INVALID; + /* + * DETECT_MODE must always be 0 with ADC and HP both off otherwise the + * FILT+ supply will not charge properly. + */ regmap_update_bits(cs42l42->regmap, CS42L42_MISC_DET_CTL, CS42L42_DETECT_MODE_MASK, 0); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 244b24d1f5e9..60d3bdf5d7c9 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -491,7 +491,10 @@ #define CS42L42_TS_UNPLUG 0 #define CS42L42_TS_TRANS 1 -/* Page 0x15 Fractional-N PLL Registers */ +/* + * NOTE: PLL_START must be 0 while both ADC_PDN=1 and HP_PDN=1. + * Otherwise it will prevent FILT+ from charging properly. + */ #define CS42L42_PLL_CTL1 (CS42L42_PAGE_15 + 0x01) #define CS42L42_PLL_START_SHIFT 0 #define CS42L42_PLL_START_MASK (1 << CS42L42_PLL_START_SHIFT) @@ -574,6 +577,10 @@ #define CS42L42_TIP_SENSE_CTRL_MASK (3 << \ CS42L42_TIP_SENSE_CTRL_SHIFT) +/* + * NOTE: DETECT_MODE must be 0 while both ADC_PDN=1 and HP_PDN=1. + * Otherwise it will prevent FILT+ from charging properly. + */ #define CS42L42_MISC_DET_CTL (CS42L42_PAGE_1B + 0x74) #define CS42L42_PDN_MIC_LVL_DET_SHIFT 0 #define CS42L42_PDN_MIC_LVL_DET_MASK (1 << CS42L42_PDN_MIC_LVL_DET_SHIFT) -- 2.30.2

2 1

[PATCH v2] ASoC: wm8350: Handle error for wm8350_register_irq
by Jiasheng Jiang 07 Mar '22

07 Mar '22

As the potential failure of the wm8350_register_irq(), it should be better to check it and return error if fails. Also, use 'free_' in order to avoid the same code. Fixes: a6ba2b2dabb5 ("ASoC: Implement WM8350 headphone jack detection") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2 * Change 1. Change to non-caps labels and remove the revert to 'return ret;'. --- sound/soc/codecs/wm8350.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 15d42ce3b21d..41504ce2a682 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1537,18 +1537,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_jck_det_l; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_jck_det_r; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_micscd; return 0; + +free_micscd: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_jck_det_r: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_jck_det_l: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) -- 2.25.1

3 2

[PATCH v2 0/2] ASoC: codecs: Add Awinic AW8738 audio amplifier driver
by Stephan Gerhold 07 Mar '22

07 Mar '22

This series adds a simple driver and DT schema for the Awinic AW8738 audio amplifier. It's fairly simple - the main difference to simple-amplifier is that there is a "one-wire pulse control" that allows configuring the amplifier to one of a few pre-defined modes. This can be used to configure the speaker-guard function (primarily the power limit for the amplifier). Jonathan Albrieux (1): ASoC: codecs: Add Awinic AW8738 audio amplifier driver Stephan Gerhold (1): ASoC: dt-bindings: Add schema for "awinic,aw8738" .../bindings/sound/awinic,aw8738.yaml | 54 +++++++++ sound/soc/codecs/Kconfig | 10 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw8738.c | 104 ++++++++++++++++++ 4 files changed, 170 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw8738.yaml create mode 100644 sound/soc/codecs/aw8738.c -- 2.35.1

2 3

[PATCH] ASoC: wm_adsp: Expand firmware loading search options
by Simon Trimmer 07 Mar '22

07 Mar '22

The parts supported by this driver can have product-specific firmware and tunings files. Typically these have been used on embedded systems where the manufacturer is responsible for installing the correct product-specific firmware files into /lib/firmware. However, the linux-firmware repository places all available firmwares into /lib/firmware and it is up to the driver to select the correct product-specific firmware from that directory. For example a product containing four smart amplifiers may provide firmware specific for that product and each of the amplifiers may have coefficient files containing tunings for their placement in the mechanical design. This change extends firmware (wmfw) and coefficient (bin) filenames to be of the general form: <cirrus/>part-dspN-fwtype<-system_name<-asoc_component_prefix>>.type Where the cirrus subdirectory, system_name and asoc_component_prefix are optional. New files will be placed in the cirrus subdirectory to avoid polluting the main /lib/firmware/ location. The generic name must be searched in /lib/firmware before /lib/firmware/cirrus so that a generic file in the new location does not override existing product-specific files in the legacy location. The search order for firmware files is: - cirrus/part-dspN-fwtype-system_name-asoc_component_prefix.wmfw - cirrus/part-dspN-fwtype-system_name.wmfw - part-dspN-fwtype.wmfw - cirrus/part-dspN-fwtype.wmfw - Qualifications are added to the filename so that rightwards is more specific. - The system_name is provided by the codec driver. - The asoc_component_prefix is used to identify tunings for individual parts because it would already exist to disambiguate the controls and it makes it obvious which firmware file applies to which device. The optional coefficient file must have the same filename construction as the discovered wmfw except: - where the wmfw has only system_name then the bin file can optionally include the asoc_component_prefix. This is to allow a common wmfw for all amps but separate tunings per amp. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 99 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 88 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..5ba88c5272d0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -744,21 +744,48 @@ static void wm_adsp_release_firmware_files(struct wm_adsp *dsp, } static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, - const struct firmware **firmware, - char **filename, - char *suffix) + const struct firmware **firmware, char **filename, + const char *dir, const char *system_name, + const char *asoc_component_prefix, + const char *filetype) { struct cs_dsp *cs_dsp = &dsp->cs_dsp; + char *s, c; int ret = 0; - *filename = kasprintf(GFP_KERNEL, "%s-%s-%s.%s", dsp->part, dsp->fwf_name, - wm_adsp_fw[dsp->fw].file, suffix); + if (system_name && asoc_component_prefix) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, dsp->part, + dsp->fwf_name, wm_adsp_fw[dsp->fw].file, system_name, + asoc_component_prefix, filetype); + else if (system_name) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, dsp->part, + dsp->fwf_name, wm_adsp_fw[dsp->fw].file, system_name, + filetype); + else + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, dsp->part, dsp->fwf_name, + wm_adsp_fw[dsp->fw].file, filetype); + if (*filename == NULL) return -ENOMEM; - ret = request_firmware(firmware, *filename, cs_dsp->dev); + /* + * Make sure that filename is lower-case and any non alpha-numeric + * characters except full stop and forward slash are replaced with + * hyphens. + */ + s = *filename; + while (*s) { + c = *s; + if (isalnum(c)) + *s = tolower(c); + else if ((c != '.') && (c != '/')) + *s = '-'; + s++; + } + + ret = firmware_request_nowarn(firmware, *filename, cs_dsp->dev); if (ret != 0) { - adsp_err(dsp, "Failed to request '%s'\n", *filename); + adsp_dbg(dsp, "Failed to request '%s'\n", *filename); kfree(*filename); *filename = NULL; } @@ -766,21 +793,69 @@ static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, return ret; } +static const char *cirrus_dir = "cirrus/"; static int wm_adsp_request_firmware_files(struct wm_adsp *dsp, const struct firmware **wmfw_firmware, char **wmfw_filename, const struct firmware **coeff_firmware, char **coeff_filename) { + const char *system_name = dsp->system_name; + const char *asoc_component_prefix = dsp->component->name_prefix; int ret = 0; - ret = wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, "wmfw"); - if (ret != 0) - return ret; + if (system_name && asoc_component_prefix) { + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, system_name, + asoc_component_prefix, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + asoc_component_prefix, "bin"); + return 0; + } + } - wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, "bin"); + if (system_name) { + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, system_name, + NULL, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + if (asoc_component_prefix) + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + asoc_component_prefix, "bin"); + + if (!*coeff_firmware) + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + NULL, "bin"); + return 0; + } + } - return 0; + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + "", NULL, NULL, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + "", NULL, NULL, "bin"); + return 0; + } + + ret = wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, NULL, NULL, "wmfw"); + if (!ret) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, NULL, NULL, "bin"); + return 0; + } + + adsp_err(dsp, "Failed to request firmware <%s>%s-%s-%s<-%s<%s>>.wmfw\n", + cirrus_dir, dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file, + system_name, asoc_component_prefix); + + return -ENOENT; } static int wm_adsp_common_init(struct wm_adsp *dsp) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 7f4fabbc6ad3..375009a65828 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -28,6 +28,7 @@ struct wm_adsp { struct cs_dsp cs_dsp; const char *part; const char *fwf_name; + const char *system_name; struct snd_soc_component *component; unsigned int sys_config_size; -- 2.34.1

3 2

[PATCH v3] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 07 Mar '22

07 Mar '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: ab6f6d85210c ("ASoC: fsi: add master clock control functions") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2 * Change 1. Seperate the error handler. v2 -> v3 * Change 1. Revert disable_xck and disable_ick. --- sound/soc/sh/fsi.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index cdf3b7f69ba7..e9a1eb6bdf66 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto disable_xck; + ret = clk_enable(clock->div); + if (ret) + goto disable_ick; clock->count++; } return ret; + +disable_ick: + clk_disable(clock->ick); +disable_xck: + clk_disable(clock->xck); +err: + return ret; } static int fsi_clk_disable(struct device *dev, -- 2.25.1

2 1

[PATCH 0/7] ASoC: fsl_sai: Cleanups and 1:1 bclk:mclk ratio support
by Sascha Hauer 07 Mar '22

07 Mar '22

This series has some updates for the fsl_sai driver: Some general cleanup patches, a bugfix in the ip revision checking and finally the mclk setting is made more accurate and support for 1:1 bclk:mclk setting is added. Ahmad Fatoum (2): ASoC: fsl_sai: simplify register poking in fsl_sai_set_bclk ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support Sascha Hauer (5): ASoC: fsl_sai: Drop unnecessary defines ASoC: fsl_sai: simplify irq return value ASoC: fsl_sai: store full version instead of major/minor ASoC: fsl_sai: Use better variable names ASoC: fsl_sai: use DIV_ROUND_CLOSEST() to calculate divider sound/soc/fsl/fsl_sai.c | 100 ++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 9 +--- 2 files changed, 52 insertions(+), 57 deletions(-) -- 2.30.2

3 9

[PATCH V3 0/2] ASoC: audio_graph_card2: Support variable slot widths
by Richard Fitzgerald 07 Mar '22

07 Mar '22

This adds support for I2S/TDM links where the slot width varies depending on the sample width, in a way that cannot be guessed by component hw_params(). A typical example is: - 16-bit samples use 16-bit slots - 24-bit samples use 32-bit slots There is no way for a component hw_params() to deduce from the information it is passed that 24-bit samples will be in 32-bit slots. Some audio hardware cannot support a fixed slot width or a slot width equal to the sample width in all cases. This is usually due either to limitations of the audio serial port or system clocking restrictions. These two patches add support for defining a mapping between sample widths and sample slots. This allows audio_graph_card2 to be used in these situations instead of having to write a custom machine driver. Changes since V2: - Change property type to uint32-matrix. - Put min/max ranges on cells. - Use per-item description to document the tuple entries. - Add a slot count to the tuple - it's unlikely that this would change for different sample rates but it's trivial to implement support for it. - Minor tidying of the code in asoc_simple_parse_tdm_width_map() and asoc_simple_set_tdm(). Richard Fitzgerald (2): ASoC: dt-bindings: audio-graph-port: Add dai-tdm-slot-width-map ASoC: audio_graph_card2: Add support for variable slot widths .../bindings/sound/audio-graph-port.yaml | 20 ++++ include/sound/simple_card_utils.h | 11 +++ sound/soc/generic/audio-graph-card2.c | 4 + sound/soc/generic/simple-card-utils.c | 97 +++++++++++++++++++ 4 files changed, 132 insertions(+) -- 2.30.2

2 3