July 2020 - Alsa-devel - mailman.alsa-project.org

snd_soc_component_driver substream ops
by Daniel Baluta 27 Jul '20

27 Jul '20

Hi Morimoto-san, Looking at snd_soc_component_driver I see there are some operations like: open, close, hw_params, hw_free. (1) Now, snd_soc_component_driver has snd_compress_ops. Do you think it is worth it to group operations from (1) in a similar structure maybe snd_<xyz>_ops. Daniel.

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Acer TravelMate 5735Z (Realtek ALC272X): Periodic crackling noises
by Paul Menzel 27 Jul '20

27 Jul '20

Dear Linux folks, On Acer TravelMate 5735Z with Debian Sid/unstable and Linux 5.7.6, there are recurring crackling noises on the internal speaker. I ran `alsa-info` from the Debian package, and the logs are uploaded [1]. It’d be awesome, if you took a look, and told me, what I can try to solve this. Kind regards, Paul [1]: https://alsa-project.org/db/?f=67bde144f2e2a4f06da89a8cc8abf93da8c91d5b

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No sound with Realtek ALC287
by GitHub issues - opened 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #76 was opened from BLuFeNiX: I get no sound via the speakers of my Lenovo Legion 7i laptop, which alsamixer tells me is using a Realtek ALC287. Here is my alsa-info.sh output: http://alsa-project.org/db/?f=4272343a3590cc08f192f98113dedfc0418afe52 I made sure to disable Auto-Mute in alsamixer, and turn all volume levels to maximum. In all cases, I get no sounds from the speakers (running speaker-test, playing music, etc.). I *am* able to get sound via headphones and HDMI (though I believe HDMI is via a different sound card). I previously reported this here: https://bugzilla.kernel.org/show_bug.cgi?id=208555 but I received no response in almost 3 weeks, so I think maybe I have reported it to the wrong place. Please let me know if I can provide any assistance or additional information. Issue URL : https://github.com/alsa-project/alsa-lib/issues/76 Repository URL: https://github.com/alsa-project/alsa-lib

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[RESEND 3/3] ASoC: max98390: update dsm param bin max size
by Steve Lee 27 Jul '20

27 Jul '20

MAX98390_DSM_PARAM_MAX_SIZE is changed to support extended register update. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index 5f444e7779b0..dff884f68e3e 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,7 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MAX_SIZE 1024 #define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { -- 2.17.1

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[RESEND 2/3] ASoC: max98390: Update dsm init sequence and condition.
by Steve Lee 27 Jul '20

27 Jul '20

Modify dsm_init sequence and dsm param bin check condition. - Move dsm_init() to after amp init setting to make sure dsm init is last setting. - dsm param bin check condition changed for extended register setting. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 44ffebac15ad..ff5cc9bbec29 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -790,7 +790,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || - param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + param_start_addr < MAX98390_IRQ_CTRL || fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); @@ -864,11 +864,11 @@ static int max98390_probe(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); /* Sleep reset settle time */ msleep(20); - /* Update dsm bin param */ - max98390_dsm_init(component); /* Amp init setting */ max98390_init_regs(component); + /* Update dsm bin param */ + max98390_dsm_init(component); /* Dsm Setting */ if (max98390->ref_rdc_value) { -- 2.17.1

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[RESEND 1/3] ASoC: max98390: Fix dac event dapm mixer.
by Steve Lee 27 Jul '20

27 Jul '20

Global EN register guide to off before AMP_EN register when amp disable sequence. - remove AMP_EN control before max98390_dac_event call Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 3e8094241645..44ffebac15ad 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", - MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_NOPM, 0, 0, max98390_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98390_dai_controls), -- 2.17.1

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headphones detected but don't show up in alsa mixer, no audio on ubuntu 18.04 with gigabyte z370 gaming wifi motherboard
by GitHub issues - opened 27 Jul '20

27 Jul '20

alsa-project/alsa-lib issue #75 was opened from rbavery: Here is my ALSA info: http://alsa-project.org/db/?f=dd3d1d8a9fa0020900a0adc03bc6adfaf1f77ee5 I tried following these remedies (including reinstalling alsa, reloading, restarting computer) but nothing has panned out: https://support.system76.com/articles/audio/ Any help or suggestions are appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/75 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 14216/14216] ALSA: usb-audio: This patch for prevent auto wakeup from s3 trig by usb disconnect signal from Lenovo Thinkcentre TI024Gen3 USB-audio.
by penghao 25 Jul '20

25 Jul '20

From: "penghao(a)deepin.com" <penghao(a)deepin.com> TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, so disabled the /sys/bus/usb/*/power/wakeup Fixesimmediately wakup form s3 state Signed-off-by: penghao(a)uniontech.com <penghao(a)uniontech.com> Signed-off-by: penghao(a)deepin.com <penghao(a)deepin.com> --- sound/usb/card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2644a5ae2b75..969c3809e051 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -640,6 +640,12 @@ static int usb_audio_probe(struct usb_interface *intf, } } dev_set_drvdata(&dev->dev, chip); + /* + * ALSA: usb-audio: Add prevent wakeup from s3 state trig by lenovo + * ThinkCentre TI024Gen3 USB-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(&dev->dev, 0); /* * For devices with more than one control interface, we assume the @@ -914,3 +920,4 @@ static struct usb_driver usb_audio_driver = { }; module_usb_driver(usb_audio_driver); + -- 2.11.0

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[PATCH] ASoC: dapm: don't call pm_runtime_* on card device
by Tzung-Bi Shih 24 Jul '20

24 Jul '20

runtime_usage of sound card has been observed to grow without bound. For example: $ cat /sys/devices/platform/sound/power/runtime_usage 46 $ sox -n -t s16 -r 48000 -c 2 - synth 1 sine 440 vol 0.1 | \ aplay -q -D hw:0,0 -f S16_LE -r 48000 -c 2 $ cat /sys/devices/platform/sound/power/runtime_usage 52 Commit 4e872a46823c ("ASoC: dapm: Don't force card bias level to be updated") stops to force update bias_level on card. If card doesn't provide set_bias_level callback, the snd_soc_dapm_set_bias_level() is equivalent to NOP for card device. As a result, dapm_pre_sequence_async() doesn't change the bias_level of card device correctly. Thus, pm_runtime_get_sync() would be called in dapm_pre_sequence_async() without symmetric pm_runtime_put() in dapm_post_sequence_async(). Don't call pm_runtime_* on card device. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) Hi, I'm not very sure if the patch makes sense. I was considering: (1) update the bias_level of card device correctly (2) skip pm_runtime_* for card device Per commit message on 4e872a46823c, I believe there is some context I'm not aware of it. Thus, the patch adopts (2). diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5076299abf37..3273161e2787 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1798,7 +1798,7 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) /* If we're off and we're not supposed to go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { - if (d->dev) + if (d->dev && cookie) pm_runtime_get_sync(d->dev); ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); @@ -1845,7 +1845,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) dev_err(d->dev, "ASoC: Failed to turn off bias: %d\n", ret); - if (d->dev) + if (d->dev && cookie) pm_runtime_put(d->dev); } -- 2.28.0.rc0.142.g3c755180ce-goog

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[PATCH v1] ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..2dd7150ff6da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,7 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, - ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA401, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7363,7 +7363,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, - [ALC289_FIXUP_ASUS_G401] = { + [ALC289_FIXUP_ASUS_GA401] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x03a11020 }, /* headset mic with jack detect */ @@ -7561,7 +7561,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), - SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

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