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

[PATCH] ALSA: hda/realtek: Add quirk for ASUS GA402
by Jason Zheng 14 Mar '22

14 Mar '22

ASUS GA402 requires a workaround to manage the routing of its 4 speakers like the other ASUS models. Add a corresponding quirk entry to fix it. Signed-off-by: Jason Zheng <jasonzheng2004(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a42457984e9..ec67b5ba6005 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9020,6 +9020,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", 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), -- 2.35.1

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[PATCH] ALSA: seq: oss: fix typo
by Julia Lawall 14 Mar '22

14 Mar '22

Fix typo in "announcement". Reported-by: Joe Perches <joe(a)perches.com> Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- This applies on top of: From: Julia Lawall <Julia.Lawall(a)inria.fr> To: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com>,alsa-devel@alsa-project.org,linux-kernel@vger.kernel.org Subject: [PATCH 3/6] ALSA: seq: oss: use kzalloc sound/core/seq/oss/seq_oss_init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 04a3376a6e66..42d4e7535a82 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -80,7 +80,7 @@ snd_seq_oss_create_client(void) system_client = rc; - /* create annoucement receiver port */ + /* create announcement receiver port */ strcpy(port->name, "Receiver"); port->addr.client = system_client; port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */

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RE: TAS27xx driver upstream to Kernel
by Xu, Yang 14 Mar '22

14 Mar '22

Hi Takashi Iwai, Any suggestion?Thanks. Regards Raphael From: Xu, Yang Sent: Friday, March 11, 2022 4:01 PM To: 'tiwai(a)suse.com' <tiwai(a)suse.com> Cc: Ding, Shenghao <shenghao-ding(a)ti.com>; Navada Kanyana, Mukund <navada(a)ti.com> Subject: TAS27xx driver upstream to Kernel Hi Takashi Iwai, This is Raphael Xu who is from Low Power Audio team,Texas Instrument. Could you help to review the attached patch and upstream them to the kernel? Thanks and expect your response. Change log: Path: /sound/soc/codecs 1. This patch to support either TAS2764 or TAS2780 device from TI 2. Fixed "no audio" issue on some platforms. Regards Raphael

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[v4 0/2] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 14 Mar '22

14 Mar '22

The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. Changes from v3: - fix build error: too many arguments for format [-Werror-format-extra-args] Changes from v2: - fix build warnings such as "data argument not used by format string" Changes from v1: - uses the snd_soc_of_get_dai_link_codecs to complete the configuration of dai_link's codecs - uses definitions to simplifies card name and compatible name Jiaxin Yu (2): ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p and rt5682 ASoC: mediatek: mt8192: support rt1015p_rt5682s .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 1 + sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 204 +++++++++++------- 3 files changed, 129 insertions(+), 77 deletions(-) -- 2.18.0

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USB audio not detected with kernels 5.15.02 and 5.16.0; snd_usb_audio and snd_usbmidi_lib not loaded.
by GitHub issues - opened 14 Mar '22

14 Mar '22

alsa-project/alsa-lib issue #216 was opened from stevesr0: System running antiX linux Sid version (based on antiX-19). Using Pipewire on top of ALSA, both internal card and usb headset detected and play with kernel 5.10.88 (antiX nosystemd kernel). With Debian 5.15.5 and 5.16.0 kernels, the usb card is not noted in aplay -l or with /proc/asound/cards. Comparison of loaded modules for working kernel and non working kernel: --Working kernel has both snd_usb_audio and snd_usbmidi_lib loaded. --Non working kernel has NEITHER of these. Instead usbcore and usb_common are loaded. I am seeking advice about fixing the nonworking kernels. Not sure if this is a bug in the kernels, but both nonworking kernels have identical outputs in /proc/asound/cards and pactl list cards. Thanks. Happy to provide files as suggested; not sure what data would be useful. Steve Rinsler Issue URL : https://github.com/alsa-project/alsa-lib/issues/216 Repository URL: https://github.com/alsa-project/alsa-lib

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Sound automatically switch from USB Speaker to HeadSet
by GitHub issues - opened 13 Mar '22

13 Mar '22

alsa-project/alsa-lib issue #215 was opened from modulusmath: Ubuntu 21.10 / Codename: impish Running LXDE if it matters I watch youtube listening on my external usb speakers and sounds plays just fine. All of a sudden they go silent and sound is played on the plugged in headset (Plantronics, Inc. DA40 Adapter) without my intervention...Even if I unplug the headset it will switch over to the 'default usb audio analog stereo'. I do see some logs via journalctl near the time of the switch: Mar 01 10:04:57 wilddog pulseaudio[1678]: ALSA woke us up to write new data to the device, but there was actually nothing to write. Mar 01 10:04:57 wilddog pulseaudio[1678]: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Mar 01 10:04:57 wilddog pulseaudio[1678]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. This is a pretty fresh install. How can I keep it stable on the external speakers? Here's my also info: http://alsa-project.org/db/?f=3589356420b912e19608eaee1960f19b3f0a4e22 Issue URL : https://github.com/alsa-project/alsa-lib/issues/215 Repository URL: https://github.com/alsa-project/alsa-lib

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[RESEND V2] ASoC: max98390: Add reset gpio control
by Steve Lee 13 Mar '22

13 Mar '22

Add reset gpio control to support RESET PIN connected to gpio. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- sound/soc/codecs/max98390.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 40fd6f363f35..9a9299e5cc5a 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1022,6 +1022,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c, struct max98390_priv *max98390 = NULL; struct i2c_adapter *adapter = i2c->adapter; + struct gpio_desc *reset_gpio; ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE @@ -1073,6 +1074,17 @@ static int max98390_i2c_probe(struct i2c_client *i2c, return ret; } + reset_gpio = devm_gpiod_get_optional(&i2c->dev, + "maxim,reset-gpios", GPIOD_OUT_LOW); + + /* Power on device */ + if (reset_gpio) { + usleep_range(1000, 2000); + /* bring out of reset */ + gpiod_set_value_cansleep(reset_gpio, 1); + usleep_range(1000, 2000); + } + /* Check Revision ID */ ret = regmap_read(max98390->regmap, MAX98390_R24FF_REV_ID, &reg); -- 2.17.1

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[PATCH] ALSA: usb-audio: Add mute TLV for playback volumes on RODE NT-USB
by Lars-Peter Clausen 12 Mar '22

12 Mar '22

For the RODE NT-USB the lowest Playback mixer volume setting mutes the audio output. But it is not reported as such causing e.g. PulseAudio to accidentally mute the device when selecting a low volume. Fix this by applying the existing quirk for this kind of issue when the device is detected. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/usb/mixer_quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e447ddd6854c..d35cf54cab33 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3360,9 +3360,10 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, if (unitid == 7 && cval->control == UAC_FU_VOLUME) snd_dragonfly_quirk_db_scale(mixer, cval, kctl); break; - /* lowest playback value is muted on C-Media devices */ - case USB_ID(0x0d8c, 0x000c): - case USB_ID(0x0d8c, 0x0014): + /* lowest playback value is muted on some devices */ + case USB_ID(0x0d8c, 0x000c): /* C-Media */ + case USB_ID(0x0d8c, 0x0014): /* C-Media */ + case USB_ID(0x19f7, 0x0003): /* RODE NT-USB */ if (strstr(kctl->id.name, "Playback")) cval->min_mute = 1; break; -- 2.30.2

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[PATCH] ASoC: mchp-spdifrx: fix typo
by Claudiu Beznea 11 Mar '22

11 Mar '22

Fix typo in log describing failure of devm_snd_dmaengine_pcm_register(). Signed-off-by: Claudiu Beznea <claudiu.beznea(a)microchip.com> --- sound/soc/atmel/mchp-spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index bcd4f3e4fb0f..5fc968483f2c 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -920,7 +920,7 @@ static int mchp_spdifrx_probe(struct platform_device *pdev) err = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (err) { - dev_err(&pdev->dev, "failed to register PMC: %d\n", err); + dev_err(&pdev->dev, "failed to register PCM: %d\n", err); return err; } -- 2.32.0

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[PATCH] ASoC: simple-card-utils: Don't reset clock of active DAI
by Sameer Pujar 11 Mar '22

11 Mar '22

Playback or capture errors are seen when clock is reset during an active stage of DAI. Presently this scenario happens when DAI has both playback and capture sessions running and one of these finishes first which will be followed by clock rate reset. The remaining active session will be affected in such case. Address this problem by allowing clock rate reset to happen only when the DAI is no more active. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/generic/simple-card-utils.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a4babfb..6a5faaf 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -272,15 +272,19 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) int i; for_each_prop_dai_cpu(props, i, dai) { - if (props->mclk_fs && !dai->clk_fixed) - snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, i), + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, i); + + if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(cpu_dai)) + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN); asoc_simple_clk_disable(dai); } for_each_prop_dai_codec(props, i, dai) { - if (props->mclk_fs && !dai->clk_fixed) - snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, i), + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, i); + + if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(codec_dai)) + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); asoc_simple_clk_disable(dai); -- 2.7.4

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