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Re: [PATCH 09/12] ASoC: cs42l42: Split I2C identity into separate module
by Martin Povišer 08 Sep '22

08 Sep '22

> On 19. 8. 2022, at 14:52, Richard Fitzgerald <rf(a)opensource.cirrus.com> wrote: > > Split the I2C bus driver definition and probe()/remove() into a > separate module so that a Soundwire build of CS42L42 support does > not have a spurious dependency on I2C. > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> > --- > +static int cs42l42_i2c_probe(struct i2c_client *i2c_client) > +{ > + struct device *dev = &i2c_client->dev; > + struct cs42l42_private *cs42l42; > + struct regmap *regmap; > + int ret; > + > + cs42l42 = devm_kzalloc(dev, sizeof(*cs42l42), GFP_KERNEL); > + if (!cs42l42) > + return -ENOMEM; > + > + regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); > + if (IS_ERR(regmap)) { > + ret = PTR_ERR(regmap); > + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); > + return ret; > + } > + > + cs42l42->dev = dev; > + cs42l42->regmap = regmap; > + cs42l42->irq = i2c_client->irq; > + > + ret = cs42l42_common_probe(cs42l42, &cs42l42_soc_component, &cs42l42_dai); > + if (ret) > + return ret; > + > + return cs42l42_init(cs42l42); The common_remove should probably be called here if the init fails. Best, Martin > +} > + > +static int cs42l42_i2c_remove(struct i2c_client *i2c_client) > +{ > + struct cs42l42_private *cs42l42 = dev_get_drvdata(&i2c_client->dev); > + > + cs42l42_common_remove(cs42l42); > + > + return 0; > +}

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[PATCH v3 0/4] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 08 Sep '22

08 Sep '22

This patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Changes since v2: - put SoundWire specific symbols under IS_ENABLED check as not all Qcom plaforms need SoundWire that use common.c Srinivas Kandagatla (4): ASoC: qcom: common: use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 173 +++++++++++++++++- sound/soc/qcom/common.h | 35 ++++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +-------------- 7 files changed, 386 insertions(+), 146 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

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[PATCH v2 0/4] ASoC: qcom: add support for sc8280xp machine
by Srinivas Kandagatla 08 Sep '22

08 Sep '22

This patchset adds support for SC8280XP SoC machine driver. First patch moves some of the commonly used code to common from sm8250 machine driver and the follow on code adds minimal support for sc8280xp. Currently this driver is only tested with SmartSpeakers and Headset on Lenovo Thinkpad X13s. Support for sm8450 is tested and I will post the patches soon. Thanks, Srini Changes since v1: - make all the exported symbol as EXPORT_SYMBOL_GPL Srinivas Kandagatla (4): ASoC: qcom: common: use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL ASoC: dt-bindings: qcom,sm8250: add compatibles for sm8450 and sm8250 ASoC: qcom: sm8250: move some code to common ASoC: qcom: add machine driver for sc8280xp .../bindings/sound/qcom,sm8250.yaml | 2 + sound/soc/qcom/Kconfig | 11 ++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 171 +++++++++++++++++- sound/soc/qcom/common.h | 12 ++ sound/soc/qcom/sc8280xp.c | 157 ++++++++++++++++ sound/soc/qcom/sm8250.c | 152 +--------------- 7 files changed, 361 insertions(+), 146 deletions(-) create mode 100644 sound/soc/qcom/sc8280xp.c -- 2.21.0

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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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