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

About ALSA dummy module: support customized mixer volume leveling
by YJ Lee 08 Sep '22

08 Sep '22

Hello Takashi (and the alsa community), This is YJ Lee from chromium.org. I'm thinking about extending the ALSA dummy module to support customized volume leveling. It will be very useful to test devices with different volume granularity. Currently this module is using hard-coded mixer volume leveling, from min=-50 to max=100. See L716: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable-rc.git/… My plan (A) is to expose 2 more additional module parameters and replace a few occurrences, from L742-750. Another plan (B) is to leverage the current existing module parameter:model, and make the dummy-module to be able to accept models with customized volume leveling. New dummy models can be provided as configuration files, and be used in the same way as how currently existing dummy models (L146-213) are used. However, it's a bit over-engineering from the original purpose. What do you think about these plans? Any insight you (and the alsa community) can provide will be greatly appreciated. Best, YJ

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[PATCH 0/2] ASoC: soc-dapm.c: random cleanup retry
by Kuninori Morimoto 08 Sep '22

08 Sep '22

Hi Mark These are remains of my previous cleanup patch-set. [1/2] adds comment why kzalloc()/kfree() are needed on snd_soc_dai_link_event_pre_pmu(). [2/2] is adjusted to [1/1] patch. Link: https://lore.kernel.org/all/8735d59zt9.wl-kuninori.morimoto.gx@renesas.com/ Kuninori Morimoto (2): ASoC: soc-dapm.c: add comment for kzalloc()/kfree() on snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: tidyup snd_soc_dai_link_event_pre_pmu() sound/soc/soc-dapm.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) -- 2.25.1

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[PATCH] ASoC: ak4458: Remove component probe() and remove()
by Shengjiu Wang 08 Sep '22

08 Sep '22

Most function in ak4458_probe() and ak4458_remove() are duplicate with dai ops, so remove them and move dsd_path setting to dai ops. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 53 ++++++--------------------------------- 1 file changed, 7 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index ea33cc83c86c..b534212096ee 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -447,6 +447,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_0B_CONTROL7, AK4458_DCHAIN_MASK, dchn); + if (ak4458->drvdata->type == AK4497) { + ret = snd_soc_component_update_bits(component, AK4458_09_DSD2, + 0x4, (ak4458->dsd_path << 2)); + if (ret < 0) + return ret; + } + ret = ak4458_rstn_control(component, 0); if (ret) return ret; @@ -629,48 +636,6 @@ static void ak4458_reset(struct ak4458_priv *ak4458, bool active) } } -static int ak4458_init(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - int ret; - - /* External Mute ON */ - if (ak4458->mute_gpiod) - gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - - ak4458_reset(ak4458, false); - - ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, - 0x80, 0x80); /* ACKS bit = 1; 10000000 */ - if (ret < 0) - return ret; - - if (ak4458->drvdata->type == AK4497) { - ret = snd_soc_component_update_bits(component, AK4458_09_DSD2, - 0x4, (ak4458->dsd_path << 2)); - if (ret < 0) - return ret; - } - - return ak4458_rstn_control(component, 1); -} - -static int ak4458_probe(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - - ak4458->fs = 48000; - - return ak4458_init(component); -} - -static void ak4458_remove(struct snd_soc_component *component) -{ - struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - - ak4458_reset(ak4458, true); -} - #ifdef CONFIG_PM static int __maybe_unused ak4458_runtime_suspend(struct device *dev) { @@ -714,8 +679,6 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) #endif /* CONFIG_PM */ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { - .probe = ak4458_probe, - .remove = ak4458_remove, .controls = ak4458_snd_controls, .num_controls = ARRAY_SIZE(ak4458_snd_controls), .dapm_widgets = ak4458_dapm_widgets, @@ -728,8 +691,6 @@ static const struct snd_soc_component_driver soc_codec_dev_ak4458 = { }; static const struct snd_soc_component_driver soc_codec_dev_ak4497 = { - .probe = ak4458_probe, - .remove = ak4458_remove, .controls = ak4497_snd_controls, .num_controls = ARRAY_SIZE(ak4497_snd_controls), .dapm_widgets = ak4497_dapm_widgets, -- 2.34.1

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[v2] ASoC: max98390: Remove unnecessary amp on/off conrtol
by Steve Lee 08 Sep '22

08 Sep '22

The Amp is already control in userspace before trigger calibrate function. Remove unnecessary control in calibrate function and add condition to check calibration is ready. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 91c0bf3d76fc..4b2ee70c29d7 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -635,10 +635,19 @@ static int max98390_dsm_calib_get(struct snd_kcontrol *kcontrol, static int max98390_dsm_calib_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + unsigned int val; struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = + snd_soc_component_get_drvdata(component); - max98390_dsm_calibrate(component); + regmap_read(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, &val); + if (val == 0x1) + max98390_dsm_calibrate(component); + else { + dev_err(component->dev, "AMP is not ready to run calibration\n"); + return -ECANCELED; + } return 0; } @@ -826,9 +835,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) struct max98390_priv *max98390 = snd_soc_component_get_drvdata(component); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x81); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x01); - regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); regmap_read(max98390->regmap, @@ -847,9 +853,6 @@ static int max98390_dsm_calibrate(struct snd_soc_component *component) dev_info(component->dev, "rdc resistance about %d.%02d ohm, reg=0x%X temp reg=0x%X\n", rdc_integer, rdc_factor, rdc_cal_result, temp); - regmap_write(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, 0x00); - regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); - return 0; } -- 2.17.1

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[PATCH linux-next] ASoC: sti-sas: Remove the unneeded result variable
by cgel.zte＠gmail.com 08 Sep '22

08 Sep '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value regmap_write() and sti_sas_init_sas_registers() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/codecs/sti-sas.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index f076878908ee..99545bcb2ba9 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -96,11 +96,8 @@ static int sti_sas_write_reg(void *context, unsigned int reg, unsigned int value) { struct sti_sas_data *drvdata = context; - int status; - - status = regmap_write(drvdata->dac.regmap, reg, value); - return status; + return regmap_write(drvdata->dac.regmap, reg, value); } static int sti_sas_init_sas_registers(struct snd_soc_component *component, @@ -385,11 +382,8 @@ static int sti_sas_resume(struct snd_soc_component *component) static int sti_sas_component_probe(struct snd_soc_component *component) { struct sti_sas_data *drvdata = dev_get_drvdata(component->dev); - int ret; - ret = sti_sas_init_sas_registers(component, drvdata); - - return ret; + return sti_sas_init_sas_registers(component, drvdata); } static struct snd_soc_component_driver sti_sas_driver = { -- 2.25.1

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[GIT PULL] ASoC fixes for v6.0-rc4
by Mark Brown 08 Sep '22

08 Sep '22

The following changes since commit 568035b01cfb107af8d2e4bd2fb9aea22cf5b868: Linux 6.0-rc1 (2022-08-14 15:50:18 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc4 for you to fetch changes up to b1cd3fd42db7593a2d24c06f1c53b8c886592080: ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable (2022-08-25 14:17:31 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 Quite a few fixes here, all driver specific and fairly small. ---------------------------------------------------------------- Bard Liao (1): ASoC: SOF: ipc4-topology: fix alh_group_ida max value Jiaxin Yu (1): ASoC: mediatek: mt8186: fix DMIC record noise Mark Brown (3): Merge remote-tracking branch 'asoc/for-5.20' into asoc-6.0 ASoC: SOF: Kconfig: Fix the dependency for client modules ASoC: nau8xxx: Implement hw constraint for rates Nathan Chancellor (1): ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion Peter Ujfalusi (2): ASoC: SOF: Kconfig: Make IPC_FLOOD_TEST depend on SND_SOC_SOF ASoC: SOF: Kconfig: Make IPC_MESSAGE_INJECTOR depend on SND_SOC_SOF Pieterjan Camerlynck (1): ASoC: fsl_sai: fix incorrect mclk number in error message Richard Fitzgerald (1): ASoC: cs42l42: Only report button state if there was a button interrupt Shengjiu Wang (3): ASoC: fsl_mqs: Fix supported clock DAI format ASoC: fsl_aud2htx: register platform component before registering cpu dai ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable Srinivas Kandagatla (1): ASoC: qcom: sm8250: add missing module owner Takashi Iwai (5): ASoC: nau8821: Implement hw constraint for rates ASoC: nau8824: Fix semaphore unbalance at error paths ASoC: nau8824: Implement hw constraint for rates ASoC: nau8825: Implement hw constraint for rates ASoC: nau8540: Implement hw constraint for rates Yang Yingliang (1): ASoC: SOF: ipc4-topology: fix wrong use of sizeof in sof_ipc4_widget_setup_comp_src() sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++ sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/codecs/cs42l42.c | 13 ++-- sound/soc/codecs/nau8540.c | 40 ++++++++---- sound/soc/codecs/nau8821.c | 66 ++++++++++--------- sound/soc/codecs/nau8824.c | 80 +++++++++++++---------- sound/soc/codecs/nau8825.c | 83 +++++++++++++----------- sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 +++++++++++++---------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 +++ sound/soc/fsl/fsl_aud2htx.c | 16 +++-- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 - sound/soc/qcom/sm8250.c | 1 + sound/soc/sh/rz-ssi.c | 26 ++++---- sound/soc/soc-pcm.c | 3 + sound/soc/sof/Kconfig | 2 + sound/soc/sof/debug.c | 6 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- sound/soc/sof/ipc4-topology.c | 4 +- 25 files changed, 299 insertions(+), 207 deletions(-)

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Alsaloop long-running session leads to CPU spikes and stalls
by GitHub issues - opened 08 Sep '22

08 Sep '22

alsa-project/alsa-utils issue #168 was opened from ebrodlic: We have a recurring issue on an embedded system where we use alsaloop to continually forward audio between: EC25E modem -> system speaker (output) system microphone (input) -> EC25E modem The two alsaloop processes run constantly in the background, and are in charge of enabling the GSM call audio functionality on our devices. What ends up happening is: - System boots up and starts the 2 processes - System usage is normal levels for 2-3 hours (very low CPU utilization) - After cca 3 hours of runtime, one alsaloop process will spike a single core of the CPU up to 100% continually - After a seemingly random amount of time of the single CPU core being fully utilized, the system may show reduced responsiveness, eventually showing recurring kernel warnings regarding CPU stalls ``` [Tue Aug 30 16:50:42 2022] rcu: INFO: rcu_preempt self-detected stall on CPU [Tue Aug 30 16:50:42 2022] rcu: 3-....: (5048 ticks this GP) idle=752/1/0x4000000000000002 softirq=1819133/1819133 fqs=2493 [Tue Aug 30 16:50:42 2022] (t=5250 jiffies g=3235733 q=26917) [Tue Aug 30 16:50:42 2022] Task dump for CPU 3: [Tue Aug 30 16:50:42 2022] alsaloop R running task 0 761 1 0x00000202 [Tue Aug 30 16:50:42 2022] Call trace: [Tue Aug 30 16:50:42 2022] dump_backtrace+0x0/0x140 [Tue Aug 30 16:50:42 2022] show_stack+0x14/0x20 [Tue Aug 30 16:50:42 2022] sched_show_task+0x110/0x138 [Tue Aug 30 16:50:42 2022] dump_cpu_task+0x40/0x4c [Tue Aug 30 16:50:42 2022] rcu_dump_cpu_stacks+0x94/0xd0 [Tue Aug 30 16:50:42 2022] rcu_sched_clock_irq+0x75c/0x9c8 [Tue Aug 30 16:50:42 2022] update_process_times+0x2c/0x68 [Tue Aug 30 16:50:42 2022] tick_sched_handle.isra.0+0x30/0x50 [Tue Aug 30 16:50:42 2022] tick_sched_timer+0x48/0x98 [Tue Aug 30 16:50:42 2022] __hrtimer_run_queues+0x110/0x1b0 [Tue Aug 30 16:50:42 2022] hrtimer_interrupt+0xe4/0x240 [Tue Aug 30 16:50:42 2022] arch_timer_handler_phys+0x30/0x40 [Tue Aug 30 16:50:42 2022] handle_percpu_devid_irq+0x80/0x140 [Tue Aug 30 16:50:42 2022] generic_handle_irq+0x24/0x38 [Tue Aug 30 16:50:42 2022] __handle_domain_irq+0x60/0xb8 [Tue Aug 30 16:50:42 2022] gic_handle_irq+0x5c/0x148 [Tue Aug 30 16:50:42 2022] el1_irq+0xb8/0x180 [Tue Aug 30 16:50:42 2022] snd_power_wait+0x0/0xe8 [Tue Aug 30 16:50:42 2022] snd_pcm_ioctl+0x2c/0x40 [Tue Aug 30 16:50:42 2022] do_vfs_ioctl+0x964/0xb48 [Tue Aug 30 16:50:42 2022] ksys_ioctl+0x78/0xa8 [Tue Aug 30 16:50:42 2022] __arm64_sys_ioctl+0x1c/0xb8 [Tue Aug 30 16:50:42 2022] el0_svc_common.constprop.0+0x68/0x160 [Tue Aug 30 16:50:42 2022] el0_svc_handler+0x20/0x80 [Tue Aug 30 16:50:42 2022] el0_svc+0x8/0x208 [Tue Aug 30 16:50:42 2022] retire_capture_urb: 307 callbacks suppressed ``` Commands used to start the two processes: `/usr/bin/alsaloop -P ec25 -C default -t 500000 -n -r 8000 -f S16_LE -c 1` `/usr/bin/alsaloop -C ec25 -P default -t 500000 -n -r 8000 -f S16_LE -c 1` Our current alsa .asoundrc conf file: ``` pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0666 # IPC permissions for multi-user sharing (octal, default 0600) slave { pcm "hw:tas5722,0" } bindings { 0 0 1 1 } } pcm.dsnooped { type dsnoop ipc_key 1027 ipc_key_add_uid false # let multiple users share ipc_perm 0666 # IPC permissions for multi-user sharing (octal, default 0600) slave { pcm "hw:ics43432,0" channels 1 format S32_LE } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "softvol" } pcm.!default { type plug slave.pcm "asymed" } pcm_slave.ec25 { pcm "hw:EC25E,0" format S16_LE channels 1 rate 8000 } pcm.ec25 { type plug slave ec25 } pcm.softvol { type softvol slave.pcm dsnooped control { name "Boost Capture Volume" card ics43432 } min_dB -3.0 max_dB 50.0 } ``` This one is not very easy to debug so it would be great if I could get some pointers in are we dealing with: - potential bugs - alsaloop misuse in approach - problems in existing configuration Any info would be appreciated! Issue URL : https://github.com/alsa-project/alsa-utils/issues/168 Repository URL: https://github.com/alsa-project/alsa-utils

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ucm2: Add support for MT8195 Cherry Tomato Chromebook with SOF
by GitHub pull_request - edited 08 Sep '22

08 Sep '22

alsa-project/alsa-ucm-conf pull request #213 was edited from kholk: Add support for the Acer Chromebook Spin 513 CP513-2H, powered by MediaTek Kompanio 1380 (MT8195) with a HiFi 4 Audio DSP running SoundOpenFirmware. This machine uses MT6359 PMIC, with RT1019 and RT5682 as headphones and speaker codecs/amps. This configuration has been successfully tested on Debian 11 with PipeWire 0.3.56 + wireplumber. `Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com>` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2: Add support for MT8195 Cherry Tomato Chromebook with SOF
by GitHub pull_request - opened 08 Sep '22

08 Sep '22

alsa-project/alsa-ucm-conf pull request #213 was opened from kholk: Add support for the Acer Chromebook Spin 513 CP513-2H, powered by MediaTek Kompanio 1380 (MT8195) with a HiFi 4 Audio DSP running SoundOpenFirmware. This machine uses MT6359 PMIC, with RT1019 and RT5682 as headphones and speaker codecs/amps. This configuration has been successfully tested on Debian 11 with PipeWire 0.3.56 + wireplumber. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/213.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 00/14] ASoC/qcom/arm64: Qualcomm ADSP DTS and binding fixes
by Krzysztof Kozlowski 08 Sep '22

08 Sep '22

Hi, Dependencies/merging ==================== 1. The DTS patches are independent. 2. The binding patches should come together, because of context changes. Could be one of: Qualcomm SoC, ASoC or DT tree. Changes since v1 ================ 1. Patch 9: New patch. 2. Patch 10: Correct also sound/qcom,q6apm-dai.yaml (Rob). 2. Patch 13: New patch. 3. Add Rb/Tb tags. Best regards, Krzysztof Krzysztof Kozlowski (14): arm64: dts: qcom: sdm630: align APR services node names with dtschema arm64: dts: qcom: sdm845: align APR services node names with dtschema arm64: dts: qcom: sm8250: align APR services node names with dtschema arm64: dts: qcom: msm8996: fix APR services nodes arm64: dts: qcom: sdm845: align dai node names with dtschema arm64: dts: qcom: msm8996: align dai node names with dtschema arm64: dts: qcom: qrb5165-rb5: align dai node names with dtschema arm64: dts: qcom: sm8250: use generic name for LPASS clock controller ASoC: dt-bindings:: qcom,q6asm: convert to dtschema dt-bindings: soc: qcom: apr: correct service children ASoC: dt-bindings: qcom,q6dsp-lpass-ports: cleanup example ASoC: dt-bindings: qcom,q6dsp-lpass-clocks: cleanup example ASoC: dt-bindings: qcom,q6apm-dai: adjust indentation in example dt-bindings: soc: qcom: apr: add missing properties .../bindings/soc/qcom/qcom,apr.yaml | 111 +++++++++++++++-- .../bindings/sound/qcom,q6apm-dai.yaml | 21 ++-- .../devicetree/bindings/sound/qcom,q6asm.txt | 70 ----------- .../devicetree/bindings/sound/qcom,q6asm.yaml | 112 ++++++++++++++++++ .../sound/qcom,q6dsp-lpass-clocks.yaml | 36 +++--- .../sound/qcom,q6dsp-lpass-ports.yaml | 64 +++++----- arch/arm64/boot/dts/qcom/msm8996.dtsi | 10 +- arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sdm630.dtsi | 8 +- arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-beryllium.dts | 2 +- .../boot/dts/qcom/sdm845-xiaomi-polaris.dts | 4 +- arch/arm64/boot/dts/qcom/sdm845.dtsi | 8 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 10 +- 14 files changed, 293 insertions(+), 169 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.yaml -- 2.34.1

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