December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/7] ASoC/SoundWire: improve suspend flows and use set_stream() instead of set_tdm_slots() for HDAudio
by Bard Liao 24 Dec '21

24 Dec '21

This series contains three topics. 1. SoundWire: Intel: remove pdm support 2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire 3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines The topics are independent but the changes are dependent. So please allow me to send them in one series. The code is basically not changed, so I still take Vinod Koul's Ack. v2: - ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire: rename .set_sdw_stream to .set_stream and .get_sdw_stream to .get_stream in intel_pdm_dai_ops. Pierre-Louis Bossart (6): ASOC: SOF: Intel: use snd_soc_dai_get_widget() ASoC/soundwire: intel: simplify callbacks for params/hw_free ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio soundwire: intel: remove unnecessary init soundwire: intel: remove PDM support Ranjani Sridharan (1): soundwire: intel: improve suspend flows drivers/soundwire/cadence_master.c | 36 +--- drivers/soundwire/cadence_master.h | 14 +- drivers/soundwire/intel.c | 253 ++++++++++++++-------------- drivers/soundwire/qcom.c | 8 +- drivers/soundwire/stream.c | 4 +- include/linux/soundwire/sdw_intel.h | 4 +- include/sound/soc-dai.h | 32 ++-- sound/soc/codecs/hdac_hda.c | 22 +-- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca.c | 2 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/sdw-mockup.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 7 +- sound/soc/qcom/sdm845.c | 4 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/sof/intel/hda-dai.c | 7 +- sound/soc/sof/intel/hda.c | 12 +- 26 files changed, 190 insertions(+), 247 deletions(-) -- 2.17.1

2 8

[PATCH 0/4] ASoC: mediatek: fix device_node leak
by Tzung-Bi Shih 24 Dec '21

24 Dec '21

The series is a follow up series. The 1st and 2nd patch follow [1] to fix the device_node leak. The 3rd patch follows [2] to reduce the log verbosity. The 4th patch cleans the device_node related code on MT8195. In order to align to previous platforms. [1]: https://patchwork.kernel.org/project/alsa-devel/patch/20211214040028.299262… [2]: https://patchwork.kernel.org/project/alsa-devel/patch/20211220093408.207206… Tzung-Bi Shih (4): ASoC: mediatek: mt8173: fix device_node leak ASoC: mediatek: mt8183: fix device_node leak ASoC: mediatek: mt8173: reduce log verbosity in probe() ASoC: mediatek: mt8195: release device_node after snd_soc_register_card sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 6 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 7 +- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 53 +++++---------- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 64 ++++++------------- 8 files changed, 56 insertions(+), 95 deletions(-) -- 2.34.1.448.ga2b2bfdf31-goog

2 5

[PATCH] ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback.
by V sujith kumar Reddy 24 Dec '21

24 Dec '21

Configure the speaker gpio pin based on power sequence of the DAPM speaker events. Enable speaker after widget power up and Disable before widget powerdown. Signed-off-by: V sujith kumar Reddy <vsujithkumar.reddy(a)amd.com> --- sound/soc/amd/acp/Kconfig | 6 +++--- sound/soc/amd/acp/acp-legacy-mach.c | 19 ++++++++++++++++--- sound/soc/amd/acp/acp-mach-common.c | 25 +++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 10 +++++++++- sound/soc/amd/acp/acp-sof-mach.c | 21 ++++++++++++++++++--- 5 files changed, 71 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 154be5e70821..d5838df3064b 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -32,7 +32,7 @@ config SND_AMD_ASOC_RENOIR config SND_SOC_AMD_MACH_COMMON tristate - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_DMIC @@ -44,14 +44,14 @@ config SND_SOC_AMD_MACH_COMMON config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables legacy sound card support for ACP audio. config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" - depends on X86 && PCI && I2C + depends on X86 && PCI && I2C && GPIOLIB select SND_SOC_AMD_MACH_COMMON help This option enables SOF sound card support for ACP audio. diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index de0f8024e2fb..0ad1cf41b308 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = NONE, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -41,15 +42,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -67,9 +69,20 @@ static int acp_asoc_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_legacy_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 7785f12aa006..03d8d1af14b3 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -71,6 +71,31 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct acp_card_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(drvdata->gpio_spkr_en)) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + gpio_set_value(drvdata->gpio_spkr_en, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + gpio_set_value(drvdata->gpio_spkr_en, 0); + break; + default: + dev_warn(card->dev, "%s invalid setting\n", __func__); + break; + } + return 0; +} +EXPORT_SYMBOL_NS_GPL(event_spkr_handler, SND_SOC_AMD_MACH); + /* Define card ops for RT5682 CODEC */ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index 5dc47cfbff10..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -17,6 +17,12 @@ #include <linux/input.h> #include <linux/module.h> #include <sound/soc.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> + +#define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 +#define EN_SPKR_GPIO_NONE -EINVAL enum be_id { HEADSET_BE_ID = 0, @@ -49,9 +55,11 @@ struct acp_card_drvdata { unsigned int dai_fmt; struct clk *wclk; struct clk *bclk; + unsigned int gpio_spkr_en; }; int acp_sofdsp_dai_links_create(struct snd_soc_card *card); int acp_legacy_dai_links_create(struct snd_soc_card *card); - +int event_spkr_handler(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event); #endif diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 854eb7214cea..07de46142655 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -27,6 +27,7 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static struct acp_card_drvdata sof_rt5682_max_data = { @@ -36,6 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_max_data = { @@ -45,6 +47,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -58,15 +61,16 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), }; static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; + unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -84,9 +88,20 @@ static int acp_sof_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_sofdsp_dai_links_create(card); + if (gpio_is_valid(spkr_gpio)) { + ret = devm_gpio_request(dev, spkr_gpio, "spkren"); + if (ret) { + dev_err(dev, "(%s) gpio request failed: %d\n", + __func__, ret); + return ret; + } + gpio_direction_output(spkr_gpio, 0); + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, -- 2.25.1

2 1

[PATCH] sound:soc:remove unneeded variable
by cgel.zte＠gmail.com 24 Dec '21

24 Dec '21

From: chiminghao <chi.minghao(a)zte.com.cn> return value form directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cm> Signed-off-by: chiminghao <chi.minghao(a)zte.com.cn> --- sound/soc/codecs/mt6660.c | 5 ++--- sound/soc/codecs/wcd938x.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 358c500377df..3a881523c30f 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -47,13 +47,12 @@ static int mt6660_reg_write(void *context, unsigned int reg, unsigned int val) struct mt6660_chip *chip = context; int size = mt6660_get_reg_size(reg); u8 reg_data[4]; - int i, ret; + int i; for (i = 0; i < size; i++) reg_data[size - i - 1] = (val >> (8 * i)) & 0xff; - ret = i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); - return ret; + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); } static int mt6660_reg_read(void *context, unsigned int reg, unsigned int *val) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 52de7d14b139..0033d61d42e8 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3083,7 +3083,7 @@ static int wcd938x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *compon int micb_num, bool req_en) { struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); - int rc, micb_mv; + int micb_mv; if (micb_num != MIC_BIAS_2) return -EINVAL; @@ -3097,9 +3097,7 @@ static int wcd938x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *compon micb_mv = req_en ? WCD_MBHC_THR_HS_MICB_MV : wcd938x->micb2_mv; - rc = wcd938x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); - - return rc; + return wcd938x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); } static inline void wcd938x_mbhc_get_result_params(struct wcd938x_priv *wcd938x, -- 2.25.1

2 2

"alsatplg -v 1 -o -" silently corrupts output
by GitHub issues - opened 24 Dec '21

24 Dec '21

alsa-project/alsa-utils issue #131 was opened from marc-hb: `alsatplg -v 1` works. `alsatplg -o -` works too. Using both produces silently a binary .tplg file prefixed with debug logs: ``` B0x003d3b/015675 - pcm 'DMIC': write 912 bytes B0x0040cb/016587 - pcm 'HDMI1': write 912 bytes B0x00445b/017499 - pcm 'HDMI2': write 912 bytes B0x0047eb/018411 - pcm 'HDMI3': write 912 bytes B0x004b7b/019323 - pcm 'DMIC16kHz': write 912 bytes B0x004f0b/020235 - block size for type be (11:10) is 0x2ad4/10964 B0x004f0b/020235 - header index 0CoSA^E^@^@^@^@^@^@^@^H^@^@^@$^@^@ ^@^@^@^@^@s^@^@^@^@^@^@^@^A^@^@^@p^@^@^@ ^@^@^@.^@^@^@ ^@^@^@^F^@^ @^@^F^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^ @^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^ @^@^@^@^@^@^@^@^@^... ``` I would have expected alsatplg -v to use stderr, this would have avoided this problem. Probably too late by now. I suggest simply failing with an error message when both options are used at the same time. Any other solution that does not silently produces corrupted output would be a huge improvement. Context: thesofproject/sof/pull/5132 Issue URL : https://github.com/alsa-project/alsa-utils/issues/131 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC PATCH] ASoC: mediatek: mt8192-mt6359: fix device_node leak
by Tzung-Bi Shih 24 Dec '21

24 Dec '21

The of_parse_phandle() document: >>> Use of_node_put() on it when done. The driver didn't call of_node_put(). Fixes the leak. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Hi maintainers, Need your comments. The patch is one of the partial fixes that I think it makes the most sense. Option 1. Machine driver makes sure the object is valid until registered This patch adopts the option. It needs snd_soc_register_card() to call of_node_get() somewhere to hold the reference count of of_node. However, I failed to find similar logic in soc-core.c. Option 2. Machine driver borrows the reference count This is what [1] adopts. Decreasing the reference count in device's remove() to make sure the object is valid for whole sound card's lifecycle. [1]: https://elixir.bootlin.com/linux/v5.16-rc5/source/sound/soc/mediatek/mt8195… In fact, in my test environment, CONFIG_OF_DYNAMIC is not set. The implementation is empty[2]. Per [3]: >>> Hardly any platforms need this [2]: https://elixir.bootlin.com/linux/v5.16-rc5/source/include/linux/of.h#L125 [3]: https://elixir.bootlin.com/linux/v5.16-rc5/source/drivers/of/Kconfig#L55 I am not sure if it is worthy to find somewhere to hold the reference count in soc-core.c but I think option 1 makes it more clear to drivers. sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index a606133951b7..24a5d0adec1b 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1172,7 +1172,11 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) return ret; } - return devm_snd_soc_register_card(&pdev->dev, card); + ret = devm_snd_soc_register_card(&pdev->dev, card); + + of_node_put(platform_node); + of_node_put(hdmi_codec); + return ret; } #ifdef CONFIG_OF -- 2.34.1.173.g76aa8bc2d0-goog

2 2

[PATCH 0/4] ASoC: qcom: Parse "pin-switches" and "widgets" from DT
by Stephan Gerhold 23 Dec '21

23 Dec '21

Some sound card setups might require extra pin switches to allow turning off certain audio components. simple-card supports this already using the "pin-switches" and "widgets" device tree property. This series makes it possible to use the same properties for the Qcom sound cards. To implement that, the function that parses the "pin-switches" property in simple-card-utils.c is first moved into the ASoC core. Then two simple function calls are added to the common Qcom sound card DT parser. Finally there is a small patch for the msm8916-wcd-analog codec to make it possible to model sound card setups used in some MSM8916 smartphones. (See PATCH 2/4 for an explanation of some real example use cases.) Using pin switches rather than patching codec drivers with switches was originally suggested by Mark Brown on a patch for the tfa989x codec: https://lore.kernel.org/alsa-devel/YXaMVHo9drCIuD3u@sirena.org.uk/ Stephan Gerhold (4): ASoC: core: Add snd_soc_of_parse_pin_switches() from simple-card-utils ASoC: dt-bindings: qcom: sm8250: Document "pin-switches" and "widgets" ASoC: qcom: common: Parse "pin-switches" and "widgets" from DT ASoC: msm8916-wcd-analog: Use separate outputs for HPH_L/HPH_R .../bindings/sound/qcom,sm8250.yaml | 16 ++++++ include/sound/soc.h | 1 + sound/soc/codecs/msm8916-wcd-analog.c | 7 +-- sound/soc/generic/simple-card-utils.c | 45 +---------------- sound/soc/qcom/common.c | 10 ++++ sound/soc/soc-core.c | 50 +++++++++++++++++++ 6 files changed, 82 insertions(+), 47 deletions(-) -- 2.34.1

3 6

[PATCH 00/20] ASoC: SOF: Re-visit firmware state and panic tracking/handling
by Peter Ujfalusi 23 Dec '21

23 Dec '21

Hi, this series will improve how we are tracking the firmware's state to be able to avoid communication with it when it is not going to answer due to a panic and we will attempt to force power cycle the DSP to recover at the next runtime suspend time. The state handling brings in other improvements on the way the kernel reports errors and DSP panics to reduce the printed lines for normal users, but at the same time allowing developers (or for bug reports) to have more precise information available to track down the issue. We can now place messages easily in the correct debug level and not bound to the static ERROR for some of the print chains, causing excess amount or partial information to be printed, confusing users and machines (CI). I would have prefered to split this series up, but it was developed together to achieve a single goal to reduce the noise, but also provide the details we need to be able to rootcause issues. Regards, Peter --- Peter Ujfalusi (20): ASoC: SOF: ops: Use dev_warn() if the panic offsets differ ASoC: SOF: Intel: hda-loader: Avoid re-defining the HDA_FW_BOOT_ATTEMPTS ASoC: SOF: core: Add simple wrapper to check flags in sof_core_debug ASoC: SOF: Use sof_debug_check_flag() instead of sof_core_debug directly ASoC: SOF: Add 'non_recoverable' parameter to snd_sof_dsp_panic() ASoC: SOF: Add a 'message' parameter to snd_sof_dsp_dbg_dump() ASoC: SOF: Introduce new firmware state: SOF_FW_CRASHED ASoC: SOF: Introduce new firmware state: SOF_FW_BOOT_READY_OK ASoC: SOF: Move the definition of enum snd_sof_fw_state to global header ASoC: SOF: Rename 'enum snd_sof_fw_state' to 'enum sof_fw_state' ASoC: SOF: ipc: Only allow sending of an IPC in SOF_FW_BOOT_COMPLETE state ASoC: SOF: Set SOF_FW_BOOT_FAILED in case we have failure during boot ASoC: SOF: pm: Force DSP off on suspend in BOOT_FAILED state also ASoc: SOF: core: Update the FW boot state transition diagram ASoC: SOF: ops: Always print DSP Panic message but use different message ASoC: SOF: dsp_arch_ops: add kernel log level parameter for oops and stack ASoC: SOF: Rename snd_sof_get_status() and add kernel log level parameter ASoC: SOF: Add clarifying comments for sof_core_debug and DSP dump flags ASoC: SOF: debug: Use DEBUG log level for optional prints ASoC: SOF: Intel: hda: Use DEBUG log level for optional prints include/sound/sof.h | 22 ++++++ sound/soc/sof/core.c | 119 +++++++++++++++++++------------ sound/soc/sof/debug.c | 35 +++++---- sound/soc/sof/imx/imx-common.c | 4 +- sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/atom.c | 8 +-- sound/soc/sof/intel/bdw.c | 8 +-- sound/soc/sof/intel/cnl.c | 21 +++++- sound/soc/sof/intel/hda-ipc.c | 19 ++++- sound/soc/sof/intel/hda-loader.c | 24 ++++--- sound/soc/sof/intel/hda.c | 20 +++--- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/ipc.c | 4 +- sound/soc/sof/loader.c | 16 ++--- sound/soc/sof/ops.c | 47 ++++++++---- sound/soc/sof/ops.h | 4 +- sound/soc/sof/pm.c | 10 +++ sound/soc/sof/sof-priv.h | 44 +++++------- sound/soc/sof/topology.c | 12 ++-- sound/soc/sof/xtensa/core.c | 44 +++++++----- 21 files changed, 299 insertions(+), 168 deletions(-) -- 2.34.1

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[PATCH -next] ASoC: codec: tlv320adc3xxx: Fix missing clk_disable_unprepare() on error in adc3xxx_i2c_probe()
by Yang Yingliang 23 Dec '21

23 Dec '21

Fix the missing clk_disable_unprepare() before return from adc3xxx_i2c_probe() in the error handling case. Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/tlv320adc3xxx.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index a683bda7eb36..4baf3d881633 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1232,21 +1232,21 @@ static int adc3xxx_i2c_probe(struct i2c_client *i2c, ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); if (ret < 0) - return ret; + goto err_unprepare_mclk; ret = adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); if (ret < 0) - return ret; + goto err_unprepare_mclk; adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); if (IS_ERR(adc3xxx->regmap)) { ret = PTR_ERR(adc3xxx->regmap); - return ret; + goto err_unprepare_mclk; } i2c_set_clientdata(i2c, adc3xxx); @@ -1263,9 +1263,15 @@ static int adc3xxx_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_component(dev, &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); - if (ret < 0) + if (ret < 0) { dev_err(dev, "Failed to register codec: %d\n", ret); + goto err_unprepare_mclk; + } + + return 0; +err_unprepare_mclk: + clk_disable_unprepare(adc3xxx->mclk); return ret; } -- 2.25.1

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[PATCH 0/5] ASoC: Intel: machine driver updates for 5.16
by Pierre-Louis Bossart 23 Dec '21

23 Dec '21

One quirk for a Dell TigerLake/SoundWire device, and initial support for platforms based on the ES8336 codec (aka ES8316). For full functionality, an update of the codec driver will be needed. Pierre-Louis Bossart (5): ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec ASoC: Intel: add machine driver for SOF+ES8336 ASoC: Intel: soc-acpi: add missing quirk for TGL SDCA single amp ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45 sound/hda/intel-dsp-config.c | 22 +- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_es8336.c | 569 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 10 + .../intel/common/soc-acpi-intel-bxt-match.c | 6 + .../intel/common/soc-acpi-intel-glk-match.c | 7 +- .../intel/common/soc-acpi-intel-tgl-match.c | 47 ++ 8 files changed, 674 insertions(+), 3 deletions(-) create mode 100644 sound/soc/intel/boards/sof_es8336.c -- 2.25.1

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