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

[PATCH linux-next] ASoC: amd: acp: use function devm_kcalloc() instead of devm_kzalloc()
by cgel.zte＠gmail.com 20 Sep '22

20 Sep '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/amd/acp/acp-mach-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index f0c49127aad1..4c69cb6e3400 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -584,7 +584,7 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->dmic_cpu_id) num_links++; - links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL); + links = devm_kcalloc(dev, num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); if (!links) return -ENOMEM; @@ -749,7 +749,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) if (drv_data->dmic_cpu_id) num_links++; - links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL); + links = devm_kcalloc(dev, num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); if (!links) return -ENOMEM; -- 2.25.1

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[PATCH linux-next] ASoC: amd: acp: use devm_kcalloc() instead of devm_kzalloc()
by cgel.zte＠gmail.com 20 Sep '22

20 Sep '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Use 2-factor multiplication argument form devm_kcalloc() instead of devm_kzalloc(). Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/amd/acp/acp-pci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index ef2ce083521e..a0c84cd07fde 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -107,7 +107,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - res = devm_kzalloc(&pci->dev, sizeof(struct resource) * num_res, GFP_KERNEL); + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; goto unregister_dmic_dev; -- 2.25.1

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[PATCH] ASoC: max98390: Fix dsm calibration reading
by Peter Ujfalusi 20 Sep '22

20 Sep '22

With the change introduced by 6ac246105b4f, the calibration can only be done after the codec probe (but questionable if it is working since 203A_AMP_EN is 0) or when the codec is powered up for audio use, in other cases "AMP is not ready to run calibration" is printed. This changes how this worked before the patch: the codec was force powered on for the duration of the calibration readout, then shut down. So, if a calibration was asked when the codec was active, it would have powered it down? To correct the calibration logic: check if the codec is powered on and if it is not then enable it, do the readout and put it back to disabled. Do this while keeping the dapm licked to avoid interfering with normal operation wia DAPM. Fixes: 6ac246105b4f ("ASoC: max98390: Remove unnecessary amp on/off conrtol") Reported-by: Fred Oh <fred.oh(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/codecs/max98390.c | 79 ++++++++++++++++++------------------- 1 file changed, 39 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 4ef8cd1053af..7a5260ff8d6b 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -161,8 +161,6 @@ static struct reg_default max98390_reg_defaults[] = { {MAX98390_R23FF_GLOBAL_EN, 0x00}, }; -static int max98390_dsm_calibrate(struct snd_soc_component *component); - static int max98390_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_component *component = codec_dai->component; @@ -635,20 +633,49 @@ 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); + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max98390_priv *max98390 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + unsigned int rdc, rdc_cal_result, rdc_integer, rdc_factor, temp, val; + + snd_soc_dapm_mutex_lock(dapm); 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; + if (!val) { + /* Enable the codec for the duration of calibration readout */ + regmap_update_bits(max98390->regmap, MAX98390_R203A_AMP_EN, + MAX98390_AMP_EN_MASK, 1); + regmap_update_bits(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, + MAX98390_GLOBAL_EN_MASK, 1); + } + + regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE1, &rdc); + regmap_read(max98390->regmap, THERMAL_RDC_RD_BACK_BYTE0, &rdc_cal_result); + regmap_read(max98390->regmap, MAX98390_MEAS_ADC_CH2_READ, &temp); + + if (!val) { + /* Disable the codec if it was disabled */ + regmap_update_bits(max98390->regmap, MAX98390_R23FF_GLOBAL_EN, + MAX98390_GLOBAL_EN_MASK, 0); + regmap_update_bits(max98390->regmap, MAX98390_R203A_AMP_EN, + MAX98390_AMP_EN_MASK, 0); } + snd_soc_dapm_mutex_unlock(dapm); + + rdc_cal_result |= (rdc << 8) & 0x0000FFFF; + if (rdc_cal_result) + max98390->ref_rdc_value = 268435456U / rdc_cal_result; + + max98390->ambient_temp_value = temp * 52 - 1188; + + rdc_integer = rdc_cal_result * 937 / 65536; + rdc_factor = ((rdc_cal_result * 937 * 100) / 65536) - (rdc_integer * 100); + + 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); + return 0; } @@ -828,34 +855,6 @@ static int max98390_dsm_init(struct snd_soc_component *component) return ret; } -static int max98390_dsm_calibrate(struct snd_soc_component *component) -{ - unsigned int rdc, rdc_cal_result, temp; - unsigned int rdc_integer, rdc_factor; - struct max98390_priv *max98390 = - snd_soc_component_get_drvdata(component); - - regmap_read(max98390->regmap, - THERMAL_RDC_RD_BACK_BYTE1, &rdc); - regmap_read(max98390->regmap, - THERMAL_RDC_RD_BACK_BYTE0, &rdc_cal_result); - rdc_cal_result |= (rdc << 8) & 0x0000FFFF; - if (rdc_cal_result) - max98390->ref_rdc_value = 268435456U / rdc_cal_result; - - regmap_read(max98390->regmap, MAX98390_MEAS_ADC_CH2_READ, &temp); - max98390->ambient_temp_value = temp * 52 - 1188; - - rdc_integer = rdc_cal_result * 937 / 65536; - rdc_factor = ((rdc_cal_result * 937 * 100) / 65536) - - (rdc_integer * 100); - - 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); - - return 0; -} - static void max98390_init_regs(struct snd_soc_component *component) { struct max98390_priv *max98390 = -- 2.37.3

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[PATCH] ASoC: tas2770: Reinit regcache on reset
by Martin Povišer 20 Sep '22

20 Sep '22

On probe of the ASoC component, the device is reset but the regcache is retained. This means the regcache gets out of sync if the codec is rebound to a sound card for a second time. Fix it by reinitializing the regcache to defaults after the device is reset. Fixes: b0bcbe615756 ("ASoC: tas2770: Fix calling reset in probe") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/codecs/tas2770.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index bb653b664146..b6765235a4b3 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -495,6 +495,8 @@ static struct snd_soc_dai_driver tas2770_dai_driver[] = { }, }; +static const struct regmap_config tas2770_i2c_regmap; + static int tas2770_codec_probe(struct snd_soc_component *component) { struct tas2770_priv *tas2770 = @@ -508,6 +510,7 @@ static int tas2770_codec_probe(struct snd_soc_component *component) } tas2770_reset(tas2770); + regmap_reinit_cache(tas2770->regmap, &tas2770_i2c_regmap); return 0; } -- 2.33.0

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[PATCH] ASOC: nau8824: Fix semaphore is released unexpectedly
by Wallace Lin 20 Sep '22

20 Sep '22

From: SJLIN0 <SJLIN0(a)nuvoton.com> On resuming, we anticipate that the jack is detected before playback or capture. Therefore, we use semaphore to control the jack detection done without any bothering. During booting, the driver launches jack detection and releases the semaphore. However, it doesn't perceive the maniputation of semaphore is not like resuming procedure. This makes the semaphore's count value become to 2. There is more than one thread can enter into the critical section. This may get unexpected situation and make some chaos. Signed-off-by: SJLIN0 <SJLIN0(a)nuvoton.com> Signed-off-by: Wallace Lin <savagecin(a)gmail.com> --- sound/soc/codecs/nau8824.c | 17 ++++++++++++++--- sound/soc/codecs/nau8824.h | 1 + 2 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15596452ca37..4f19fd9b65d1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -901,7 +901,10 @@ static void nau8824_jdet_work(struct work_struct *work) NAU8824_IRQ_KEY_RELEASE_DIS | NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); - nau8824_sema_release(nau8824); + if (nau8824->resume_lock) { + nau8824_sema_release(nau8824); + nau8824->resume_lock = false; + } } static void nau8824_setup_auto_irq(struct nau8824 *nau8824) @@ -966,7 +969,10 @@ static irqreturn_t nau8824_interrupt(int irq, void *data) /* release semaphore held after resume, * and cancel jack detection */ - nau8824_sema_release(nau8824); + if (nau8824->resume_lock) { + nau8824_sema_release(nau8824); + nau8824->resume_lock = false; + } cancel_work_sync(&nau8824->jdet_work); } else if (active_irq & NAU8824_KEY_SHORT_PRESS_IRQ) { int key_status, button_pressed; @@ -1524,6 +1530,7 @@ static int __maybe_unused nau8824_suspend(struct snd_soc_component *component) static int __maybe_unused nau8824_resume(struct snd_soc_component *component) { struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); + int ret; regcache_cache_only(nau8824->regmap, false); regcache_sync(nau8824->regmap); @@ -1531,7 +1538,10 @@ static int __maybe_unused nau8824_resume(struct snd_soc_component *component) /* Hold semaphore to postpone playback happening * until jack detection done. */ - nau8824_sema_acquire(nau8824, 0); + nau8824->resume_lock = true; + ret = nau8824_sema_acquire(nau8824, 0); + if (ret) + nau8824->resume_lock = false; enable_irq(nau8824->irq); } @@ -1940,6 +1950,7 @@ static int nau8824_i2c_probe(struct i2c_client *i2c) nau8824->regmap = devm_regmap_init_i2c(i2c, &nau8824_regmap_config); if (IS_ERR(nau8824->regmap)) return PTR_ERR(nau8824->regmap); + nau8824->resume_lock = false; nau8824->dev = dev; nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index de4bae8281d0..5fcfc43dfc85 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -436,6 +436,7 @@ struct nau8824 { struct semaphore jd_sem; int fs; int irq; + int resume_lock; int micbias_voltage; int vref_impedance; int jkdet_polarity; -- 2.25.1

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[PATCH] ASoC: Intel: sof_sdw: add support for Dell SKU 0AFF
by Pierre-Louis Bossart 20 Sep '22

20 Sep '22

Yet another SKU that needs a quirk for jack detection and four-speaker support. BugLink: https://github.com/thesofproject/linux/issues/3777 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a49bfaab6b21f..2ff30b40a1e4c 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -266,6 +266,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFF") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.34.1

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Alsa - has invalid interface number
by GitHub issues - edited 19 Sep '22

19 Sep '22

alsa-project/alsa-lib issue #271 was edited from smangels: hi, I'm running the latest ArchLinux installation using a Behringer Flow8 as my main audio interface. Today I'm running into an issue where the USB audio hardware can not be loaded successfully. ``` [ 1690.797522] usb 1-3.1.1.4: new high-speed USB device number 14 using xhci_hcd [ 1690.891936] usb 1-3.1.1.4: config 1 has an invalid interface number: 7 but max is 5 [ 1690.891940] usb 1-3.1.1.4: config 1 has no interface number 5 [ 1690.892620] usb 1-3.1.1.4: New USB device found, idVendor=1397, idProduct=050d, bcdDevice= 2.10 [ 1690.892624] usb 1-3.1.1.4: New USB device strings: Mfr=1, Product=2, SerialNumber=6 [ 1690.892627] usb 1-3.1.1.4: Product: FLOW 8 [ 1690.892628] usb 1-3.1.1.4: Manufacturer: Behringer [ 1690.892630] usb 1-3.1.1.4: SerialNumber: 18-04-ED-FB-BE-A0 ``` The following Alsa related packages have been upgraded. ``` [2022-09-19T18:22:14+0200] [ALPM] upgraded alsa-card-profiles (1:0.3.57-1 -> 1:0.3.58-1) [2022-09-19T18:22:15+0200] [ALPM] upgraded pipewire-alsa (1:0.3.57-1 -> 1:0.3.58-1) [2022-09-19T23:07:36+0200] [PACMAN] Running 'pacman -Sy alsa-tools' [2022-09-19T23:07:38+0200] [ALPM] installed alsa-tools (1.2.5-1) ``` Kernel and UDEV version `Linux archlinux 5.19.9-arch1-1 #1 SMP PREEMPT_DYNAMIC Thu, 15 Sep 2022 16:08:26 +0000 x86_64 GNU/Linux` `udev version 251` Issue URL : https://github.com/alsa-project/alsa-lib/issues/271 Repository URL: https://github.com/alsa-project/alsa-lib

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Alsa - has invalid interface number
by GitHub issues - opened 19 Sep '22

19 Sep '22

alsa-project/alsa-lib issue #271 was opened from smangels: hi, I'm running the latest ArchLinux installation using a Behringer Flow8 as my main audio interface. Today I'm running into an issue where the USB audio hardware can not be loaded successfully. ``` [ 1690.797522] usb 1-3.1.1.4: new high-speed USB device number 14 using xhci_hcd [ 1690.891936] usb 1-3.1.1.4: config 1 has an invalid interface number: 7 but max is 5 [ 1690.891940] usb 1-3.1.1.4: config 1 has no interface number 5 [ 1690.892620] usb 1-3.1.1.4: New USB device found, idVendor=1397, idProduct=050d, bcdDevice= 2.10 [ 1690.892624] usb 1-3.1.1.4: New USB device strings: Mfr=1, Product=2, SerialNumber=6 [ 1690.892627] usb 1-3.1.1.4: Product: FLOW 8 [ 1690.892628] usb 1-3.1.1.4: Manufacturer: Behringer [ 1690.892630] usb 1-3.1.1.4: SerialNumber: 18-04-ED-FB-BE-A0 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/271 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/3] ASoC: ts3a227e control debounce times
by Astrid Rost 19 Sep '22

19 Sep '22

From: Astrid Rost <astridr(a)axis.com> PATCH 1 - convert ts3a227e bindings to yaml. PATCH 2 - add ts3a227e bindings to support debounce times. PATCH 3 - add ts3a227e driver to support debounce times. v1 -> v2 - Converted the devicetree-bindings to YAML. - The debounce-times are in ms. The closed match is used, if the given times do not match the register values. Signed-off-by: Astrid Rost <astrid.rost(a)axis.com>. Astrid Rost (3): dt-bindings: sound: ts3a227e: convert to yaml dt-bindings: sound: ts3a227e: add control of debounce times ASoC: ts3a227e: add parameters to control debounce times .../devicetree/bindings/sound/ts3a227e.txt | 30 ------ .../devicetree/bindings/sound/ts3a227e.yaml | 98 +++++++++++++++++++ sound/soc/codecs/ts3a227e.c | 58 +++++++++-- 3 files changed, 150 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.yaml -- 2.20.1

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[PATCH] ALSA: hda/hdmi: Fix the converter reuse for the silent stream
by Jaroslav Kysela 19 Sep '22

19 Sep '22

When the user space pcm stream uses the silent stream converter, it is no longer allocated for the silent stream. Clear the appropriate flag in the hdmi_pcm_open() function. The silent stream setup may be applied in hdmi_pcm_close() (and the error path - open fcn) again. If the flag is not cleared, the reuse conditions for the silent stream converter in hdmi_choose_cvt() may improperly share this converter. Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..22766810635b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1267,6 +1267,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, set_bit(pcm_idx, &spec->pcm_in_use); per_pin = get_pin(spec, pin_idx); per_pin->cvt_nid = per_cvt->cvt_nid; + per_pin->silent_stream = false; hinfo->nid = per_cvt->cvt_nid; /* flip stripe flag for the assigned stream if supported */ -- 2.35.3

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