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

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

1 0

[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

1 0

[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

1 0

[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

1 0

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

1 0

[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

3 3

[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

2 6

[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

2 1

[PATCH v1] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch adds support for headset mic to the ASUS ROG Zephyrus G15(GA502) notebook series by adding the corresponding vendor/pci_device id, as well as adding a new fixup for the used realtek ALC289. The fixup stets the correct pin to get the headset mic correctly recognized on audio-jack. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..a2c48a7f4594 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6153,6 +6153,7 @@ enum { ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA502, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7370,6 +7371,13 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC289_FIXUP_ASUS_GA502] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7561,6 +7569,7 @@ 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, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 1

[PATCH] dt-bindings: sound: convert Everest ES8316 binding to yaml
by Katsuhiro Suzuki 24 Jul '20

24 Jul '20

This patch converts Everest Semiconductor ES8316 low power audio CODEC binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 23 --------- .../bindings/sound/everest,es8316.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 23 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt deleted file mode 100644 index 1bf03c5f2af4..000000000000 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ /dev/null @@ -1,23 +0,0 @@ -Everest ES8316 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : should be "everest,es8316" - - reg : the I2C address of the device for I2C - -Optional properties: - - - clocks : a list of phandle, should contain entries for clock-names - - clock-names : should include as follows: - "mclk" : master clock (MCLK) of the device - -Example: - -es8316: codec@11 { - compatible = "everest,es8316"; - reg = <0x11>; - clocks = <&clks 10>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..b713404dac4f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; -- 2.27.0

2 2