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

[PATCH v7 1/2] cros_ec_commands: Add EC_CODEC_I2S_RX_RESET
by Yu-Hsuan Hsu 21 Jan '21

21 Jan '21

Add the new command EC_CODEC_I2S_RX_RESET in ec_codec_i2s_rx_subcmd, which is used for resetting the EC codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; -- 2.30.0.296.g2bfb1c46d8-goog

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[PATCH 0/3] ASoC: Simplify with dma_set_mask_and_coherent()
by Takashi Iwai 21 Jan '21

21 Jan '21

Simple conversions from two dma_set_mask*() calls into the single dma_set_mask_and_coherent(). Just a minor code refactoring, no functional changes. Takashi === Takashi Iwai (3): ASoC: intel: skl: Simplify with dma_set_mask_and_coherent() ASoC: SOF: intel: Simplify with dma_set_mask_and_coherent() ASoC: tegra: Simplify with dma_set_mask_and_coherent() sound/soc/intel/skylake/skl.c | 8 ++------ sound/soc/sof/intel/hda.c | 8 ++------ sound/soc/tegra/tegra_pcm.c | 6 +----- 3 files changed, 5 insertions(+), 17 deletions(-) -- 2.26.2

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[PATCH 1/2] ALSA: usb-audio: Skip the clock selector inquiry for single connections
by Takashi Iwai 20 Jan '21

20 Jan '21

The current USB-audio driver gets an error at probing NUX MG-300 about parsing the clocks. This is because the firmware doesn't return the proper connection of the clock selector that is connected to a single clock; it's likely that the firmware was lazy^w optimized and the inquiry wasn't handled. Actually it makes little sense to inquire and set up the single connection explicitly. This patch fixes the issue by simply skipping the clock selector inquiry if it's a single connection. Reported-by: Mike Oliphant <oliphant(a)nostatic.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index dc68ed65e478..8243652d5604 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -298,6 +298,11 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (selector) { int ret, i, cur; + if (selector->bNrInPins == 1) { + ret = 1; + goto find_source; + } + /* the entity ID we are looking for is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, selector->bClockID); @@ -314,6 +319,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } + find_source: cur = ret; ret = __uac_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], -- 2.26.2

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[PATCH v3 0/5] MFD/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 20 Jan '21

20 Jan '21

Hi All, Here is v3 of my series to add support for Intel Bay Trail based devices which use a WM5102 codec for audio output/input. New in v3 is that the compile error when CONFIG_ACPI is unset should be gone. Everything else is the same as in v2, see the cover-letter of v2 below: This was developed and tested on a Lenovo Yoga Tablet 1051L. The biggest difference from v1 is that, based on all the discussions about the jack-detect stuff, I've decided to split this into 2 series. One series adding the basic support for this setup, which is this series. And a second series which reworks the extcon driver into an arizona-jackdet library and then modifies the codec drivers to use that directly, replacing the old separate extcon child-device and extcon-driver. The are 2 reasons for the split: 1. With the new jack-det rework, the series really address 2 separate (but related) enhancements. 2. I expect this series to be ready for merging, while the jack-det stuff likely will need a couple more revisions. Other then the split there are some minor changes addressing various review comments, see the individual patch changelogs. The MFD and ASoC parts can be merged independent from each-other (both must be merged to get working sound on these boards, but that is only a runtime dependency and a part missing won't have any bad side-effects). Or the entire series could be merged through the MFD tree if people prefer that. Regards, Hans

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[PATCH 0/2] ASoC: remove obsolete drivers
by Arnd Bergmann 20 Jan '21

20 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> A few Arm platforms are getting removed in v5.12, this removes the corresponding sound drivers. Link: https://lore.kernel.org/linux-arm-kernel/20210120124812.2800027-1-arnd@kern… Arnd Bergmann (2): ASoC: remove sirf prima/atlas drivers ASoC: remove zte zx drivers .../bindings/sound/sirf-audio-codec.txt | 17 - .../devicetree/bindings/sound/sirf-usp.txt | 27 - .../devicetree/bindings/sound/zte,tdm.txt | 30 - .../bindings/sound/zte,zx-aud96p22.txt | 24 - .../devicetree/bindings/sound/zte,zx-i2s.txt | 45 -- .../bindings/sound/zte,zx-spdif.txt | 27 - sound/soc/Kconfig | 2 - sound/soc/Makefile | 2 - sound/soc/codecs/Makefile | 4 - sound/soc/codecs/sirf-audio-codec.c | 575 ------------------ sound/soc/codecs/zx_aud96p22.c | 401 ------------ sound/soc/sirf/Kconfig | 21 - sound/soc/sirf/Makefile | 8 - sound/soc/sirf/sirf-audio-port.c | 86 --- sound/soc/sirf/sirf-audio.c | 160 ----- sound/soc/sirf/sirf-usp.c | 435 ------------- sound/soc/sirf/sirf-usp.h | 292 --------- sound/soc/zte/Kconfig | 26 - sound/soc/zte/Makefile | 4 - sound/soc/zte/zx-i2s.c | 452 -------------- sound/soc/zte/zx-spdif.c | 363 ----------- sound/soc/zte/zx-tdm.c | 458 -------------- 22 files changed, 3459 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt delete mode 100644 sound/soc/codecs/sirf-audio-codec.c delete mode 100644 sound/soc/codecs/zx_aud96p22.c delete mode 100644 sound/soc/sirf/Kconfig delete mode 100644 sound/soc/sirf/Makefile delete mode 100644 sound/soc/sirf/sirf-audio-port.c delete mode 100644 sound/soc/sirf/sirf-audio.c delete mode 100644 sound/soc/sirf/sirf-usp.c delete mode 100644 sound/soc/sirf/sirf-usp.h delete mode 100644 sound/soc/zte/Kconfig delete mode 100644 sound/soc/zte/Makefile delete mode 100644 sound/soc/zte/zx-i2s.c delete mode 100644 sound/soc/zte/zx-spdif.c delete mode 100644 sound/soc/zte/zx-tdm.c -- 2.29.2

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[PATCH] ALSA: usb-audio: Fix hw constraints dependencies
by Takashi Iwai 20 Jan '21

20 Jan '21

Since the recent refactoring, it's been reported that some USB-audio devices (typically webcams) are no longer detected properly by PulseAudio. The debug session revealed that it's failing at probing by PA to try the sample rate 44.1kHz while the device has discrete sample rates other than 44.1kHz. But the puzzle was that arecord works as is, and some other devices with the discrete rates work, either. After all, this turned out to be the lack of the dependencies in a few hw constraint rules: snd_pcm_hw_rule_add() has the (variable) arguments specifying the dependent parameters, and some functions didn't set the target parameter itself as the dependencies. This resulted in an invalid parameter that could be generated only in a certain call pattern. This bug itself has been present in the code, but it didn't trigger errors just because the rules were casually avoiding such a corner case. After the recent refactoring and cleanup, however, the hw constraints work "as expected", and the problem surfaced now. For fixing the problem above, this patch adds the missing dependent parameters to each snd_pcm_hw_rule() call. Fixes: bc4e94aa8e72 ("ALSA: usb-audio: Handle discrete rates properly in hw constraints") BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f71965bf815f..078bb4c94033 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -981,6 +981,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, @@ -990,6 +991,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, param_period_time_if_needed, @@ -998,6 +1000,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, -- 2.26.2

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[PATCH 2/2] ASoC: rt5682: add device property to control LDO2 power
by shumingf＠realtek.com 20 Jan '21

20 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The application circuit shall provide MICVDD power. In default, the codec driver doesn't need to enable LDO2 power. In case, the customers use VBAT for micbias, it should enable the device property "realtek,ldo2-pow-en". Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 18 +++++++++++++++++- 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 3900a07e3935..4b4c61b48380 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -41,6 +41,7 @@ struct rt5682_platform_data { unsigned int dmic_clk_rate; unsigned int dmic_delay; bool dmic_clk_driving_high; + bool ldo2_pow_en; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 42ad5bea86b5..68d29d71d6e4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1343,6 +1343,19 @@ static int is_using_asrc(struct snd_soc_dapm_widget *w, } } +static int is_using_ldo2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + if (rt5682->pdata.ldo2_pow_en) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, @@ -1854,7 +1867,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"CLKDET SYS", NULL, "CLKDET"}, - {"IN1P", NULL, "LDO2"}, + {"IN1P", NULL, "LDO2", is_using_ldo2}, {"BST1 CBJ", NULL, "IN1P"}, @@ -2999,6 +3012,9 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dmic_clk_driving_high = device_property_read_bool(dev, "realtek,dmic-clk-driving-high"); + rt5682->pdata.ldo2_pow_en = device_property_read_bool(dev, + "realtek,ldo2-pow-en"); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); -- 2.29.0

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[PATCH] ASoC: Intel: Skylake: Check the kcontrol against NULL
by Lukasz Majczak 20 Jan '21

20 Jan '21

There is no check for the kcontrol against NULL and in some cases it causes kernel to crash. Fixes: 2d744ecf2b984 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT") Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/skylake/skl-topology.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd592922..c9abbe4ff0ba3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3618,12 +3618,18 @@ static void skl_tplg_complete(struct snd_soc_component *component) int i; list_for_each_entry(dobj, &component->dobj_list, list) { - struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct snd_kcontrol *kcontrol; + struct soc_enum *se; + char **texts; char chan_text[4]; + kcontrol = dobj->control.kcontrol; + if(!kcontrol) + continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; + if (dobj->type != SND_SOC_DOBJ_ENUM || dobj->control.kcontrol->put != skl_tplg_multi_config_set_dmic) base-commit: 69fe63aa100220c8fd1f451dd54dd0895df1441d -- 2.25.1

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handling different frame sizes in snd_pcm_extplug_callback_t transfer callback
by Arsen Arsenović 20 Jan '21

20 Jan '21

Hello! I am writing an external noise cancellation plugin for ALSA, based on rnnoise, but I've hit a bit of a roadblock: rnnoise processes chunks of 480 floats at a time and does no buffering, but alsa expects the transfer callback to take all the data from the input and put it onto the output, with no way to signal that any buffering is or should be happening. I have tried to return a multiple of 480 from the transfer callback and keep a buffer with the leftover data, to be prepended to the next payload, but it looks like ALSA will, when a plugin returns too little data, just send back the same non-processed data in a new call, which breaks this idea. Any way I can work around these limitations? Another solution could be for external plugins to report the amount they consumed and produced, rather than a single number for both, but I am not aware of how to do that, if it's even possible with the filter plugin SDK. Thanks in advance! EDIT: I had previously posted this question before I confirmed my mailing list membership, but seeing as it wasn't approved for a few days I cancelled the request and reposted it. Sorry about that. -- Arsen Arsenović

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sof topology compilation fails
by GitHub issues - opened 20 Jan '21

20 Jan '21

alsa-project/alsa-lib issue #114 was opened from juimonen: sof topologies fail to compile because of following commit: commit 51e1d486ce4535923692bd1d7e59d5aa3a29cd50 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Thu Jan 7 17:40:42 2021 +0100 topology: parse_tuple_set() - remove dead condition code Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I don't completely understand the change...? we are in "case SND_SOC_TPLG_TUPLE_TYPE_WORD" and then we check for SND_SOC_TPLG_TUPLE_TYPE_WORD (and not the value anymore) and error out? Issue URL : https://github.com/alsa-project/alsa-lib/issues/114 Repository URL: https://github.com/alsa-project/alsa-lib

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