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

[PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671
by huangwenhui 11 Mar '22

11 Mar '22

On a HP 288 Pro G8, the front mic could not be detected.In order to get it working, the pin configuration needs to be set correctly, and the ALC671_FIXUP_HP_HEADSET_MIC2 fixup needs to be applied. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.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..e9c9b1d1bf0d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11067,6 +11067,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), + SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), -- 2.20.1

2 1

[v3 0/2] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 11 Mar '22

11 Mar '22

From: Jiaxin Yu <jiaxin.yu(a)mediatek.corp-partner.google.com> The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. 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

1 3

jack: implement pcm operation sw_params
by GitHub pull_request - edited 11 Mar '22

11 Mar '22

alsa-project/alsa-plugins pull request #43 was edited from aditpape: Alsa-lib commit #88e4ae27 introduced a dynamic recalculation of the avail_min value on the underlaying slave devices. This updated avail_min value is not reaching the alsa-jack plugin due to lack of the sw_params operation. This patch adds this missing operation to correctly apply an updated avail_min setting during runtime. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/43 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[RFC PATCH 0/3] Flexible codec clock configuration
by Sameer Pujar 11 Mar '22

11 Mar '22

Typically the codec drivers require setting up of Sysclk. Sometimes presence of internal PLL can provide more options of Sysclk configuration. Presently ASoC provides callbacks set_sysclk() and set_pll() in such cases. However it comes with following limitations considering generic machine drivers (simple-card or audio-graph-card): 1. The Sysclk source needs to be passed to set_sysclk() callback. Presently simple-card or audio-graph-card card rely on default source value (which is 0). If any other source needs to be used, it is currently not possible. 2. The same would be true for codec PLL configuration as well, though simple-card or audio-graph-card don't have support yet for the PLL configuration. Earlier attempt[0] to address above was not felt suitable. The suggestion was to use standard clock based bindings instead. This RFC series takes RT5659 as a reference and exposes clock relationships via DT. **This is not in the final shape yet**, but I wanted to get some valuable feedback to understand if the idea is right. If this appears fine, this can be extended to other codecs (wherever necessary). This does not completely remove the need of set_sysclk() callback because the clock requirement (MCLK * fs) would come from the machine driver. But machine driver need not worry about Sysclk source. It would be internally managed by Codec via DT clock relationships. [0] https://patchwork.kernel.org/project/alsa-devel/list/?series=438531&archive… Sameer Pujar (3): ASoC: soc-pcm: tweak DPCM BE hw_param() call order ASoC: rt5659: Expose internal clock relationships ASoC: tegra: Get clock rate in consumer mode sound/soc/codecs/rt5659.c | 257 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt5659.h | 9 ++ sound/soc/soc-pcm.c | 60 +++++++++- sound/soc/tegra/tegra210_i2s.c | 25 ++-- 4 files changed, 332 insertions(+), 19 deletions(-) -- 2.7.4

2 6

[PATCH v2] ASoC: tas5805m: fix pdn polarity
by John Keeping 11 Mar '22

11 Mar '22

The binding defines the GPIO as "pdn-gpios" so when the GPIO is active the expectation is that the power down signal is asserted and this is how all other drivers using this GPIO name interpret the value. But the tas5805m driver inverts the sense from the normal expectation so when the powerdown GPIO is logically asserted the chip is running. This is a new driver that is not yet in a released kernel and has no in-tree users of the binding so fix the sense of the GPIO so that logically asserted means that the device is powered down. Rename the variable to match so that the compiler will catch any places that should have been updated but have been missed. Fixes: ec45268467f4 ("ASoC: add support for TAS5805M digital amplifier") Signed-off-by: John Keeping <john(a)metanate.com> --- v2: - Rewrite commit message to make it more obvious that this is a change to the interpretation of the GPIO in the binding sound/soc/codecs/tas5805m.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index fa0e81ec875a..12146a860ef8 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -155,7 +155,7 @@ static const uint32_t tas5805m_volume[] = { struct tas5805m_priv { struct regulator *pvdd; - struct gpio_desc *gpio_pdn_n; + struct gpio_desc *gpio_pdn; uint8_t *dsp_cfg_data; int dsp_cfg_len; @@ -444,11 +444,11 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) dev_set_drvdata(dev, tas5805m); tas5805m->regmap = regmap; - tas5805m->gpio_pdn_n = devm_gpiod_get(dev, "pdn", GPIOD_OUT_LOW); - if (IS_ERR(tas5805m->gpio_pdn_n)) { + tas5805m->gpio_pdn = devm_gpiod_get(dev, "pdn", GPIOD_OUT_HIGH); + if (IS_ERR(tas5805m->gpio_pdn)) { dev_err(dev, "error requesting PDN gpio: %ld\n", - PTR_ERR(tas5805m->gpio_pdn_n)); - return PTR_ERR(tas5805m->gpio_pdn_n); + PTR_ERR(tas5805m->gpio_pdn)); + return PTR_ERR(tas5805m->gpio_pdn); } /* This configuration must be generated by PPC3. The file loaded @@ -505,7 +505,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) } usleep_range(100000, 150000); - gpiod_set_value(tas5805m->gpio_pdn_n, 1); + gpiod_set_value(tas5805m->gpio_pdn, 0); usleep_range(10000, 15000); /* Don't register through devm. We need to be able to unregister @@ -515,7 +515,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) &tas5805m_dai, 1); if (ret < 0) { dev_err(dev, "unable to register codec: %d\n", ret); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); regulator_disable(tas5805m->pvdd); return ret; } @@ -529,7 +529,7 @@ static int tas5805m_i2c_remove(struct i2c_client *i2c) struct tas5805m_priv *tas5805m = dev_get_drvdata(dev); snd_soc_unregister_component(dev); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); usleep_range(10000, 15000); regulator_disable(tas5805m->pvdd); return 0; -- 2.35.1

2 7

[PATCH 0/2] xen/grant-table: do some cleanup
by Juergen Gross 11 Mar '22

11 Mar '22

Cleanup grant table code by removing unused functionality. Juergen Gross (2): xen/grant-table: remove gnttab_*transfer*() functions xen/grant-table: remove readonly parameter from functions drivers/block/xen-blkfront.c | 8 +- drivers/char/tpm/xen-tpmfront.c | 2 +- drivers/gpu/drm/xen/xen_drm_front_evtchnl.c | 2 +- drivers/input/misc/xen-kbdfront.c | 4 +- drivers/net/xen-netfront.c | 13 +- drivers/pci/xen-pcifront.c | 2 +- drivers/scsi/xen-scsifront.c | 4 +- drivers/usb/host/xen-hcd.c | 4 +- drivers/xen/gntalloc.c | 2 +- drivers/xen/gntdev-dmabuf.c | 2 +- drivers/xen/grant-table.c | 151 +++----------------- drivers/xen/pvcalls-front.c | 6 +- drivers/xen/xen-front-pgdir-shbuf.c | 3 +- include/xen/grant_table.h | 13 +- net/9p/trans_xen.c | 8 +- sound/xen/xen_snd_front_evtchnl.c | 2 +- 16 files changed, 50 insertions(+), 176 deletions(-) -- 2.34.1

3 3

Re: [PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671
by huangwenhui 11 Mar '22

11 Mar '22

Hi  Takashi, if adjusting the mixer element or replugging the headset, the headphone would be unmuted. But most users hope that the default state is unmuted. Should this unconditional unmute be required? Thanks. ------------------ Original ------------------ From:  "Takashi Iwai"<tiwai(a)suse.de>; Date:  Thu, Mar 10, 2022 10:03 PM To:  "黄文辉"<huangwenhuia(a)uniontech.com>; Cc:  "perex"<perex(a)perex.cz>; "tiwai"<tiwai(a)suse.com>; "jeremy.szu"<jeremy.szu(a)canonical.com>; "hui.wang"<hui.wang(a)canonical.com>; "wse"<wse(a)tuxedocomputers.com>; "cam"<cam(a)neo-zeon.de>; "kailang"<kailang(a)realtek.com>; "tanureal"<tanureal(a)opensource.cirrus.com>; "sami"<sami(a)loone.fi>; "alsa-devel"<alsa-devel(a)alsa-project.org>; "linux-kernel"<linux-kernel(a)vger.kernel.org>; Subject:  Re: [PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP machine with alc671   On Thu, 10 Mar 2022 14:58:36 +0100, huangwenhui wrote: > > Hi  Takashi, > > Thank you for your reply. > > When booting with plugged headset, the headphone will be muted. The muted state is the default behavior.  Wouldn't it be unmuted if you adjust the corresponding mixer element? Takashi > > Thanks. >  > ------------------ Original ------------------ > From:  "Takashi Iwai"<tiwai(a)suse.de>; > Date:  Thu, Mar 10, 2022 09:29 PM > To:  "huangwenhui"<huangwenhuia(a)uniontech.com>; > Cc:  "perex"<perex(a)perex.cz>; "tiwai"<tiwai(a)suse.com>; "jeremy.szu" > <jeremy.szu(a)canonical.com>; "hui.wang"<hui.wang(a)canonical.com>; "wse" > <wse(a)tuxedocomputers.com>; "cam"<cam(a)neo-zeon.de>; "kailang" > <kailang(a)realtek.com>; "tanureal"<tanureal(a)opensource.cirrus.com>; "sami" > <sami(a)loone.fi>; "alsa-devel"<alsa-devel(a)alsa-project.org>; "linux-kernel" > <linux-kernel(a)vger.kernel.org>; > Subject:  Re: [PATCH] ALSA: hda/realtek - Fix headset mic problem for a HP > machine with alc671 >  > On Thu, 10 Mar 2022 14:03:01 +0100, > huangwenhui wrote: > > > > On a HP 288 Pro G8, the front Mic could not be detected. > > > > Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> > > Thanks for the patch.  Most of the changes look OK, but one thing I > still don't get: > > > + case HDA_FIXUP_ACT_INIT: > > + alc_write_coef_idx(codec, 0x19, 0xa054); > > + msleep(80); > > + snd_hda_codec_write(codec, hp_pin, 0, > > +     AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); > > Why this unconditional unmute is required for fixing the mic problem? > > Takashi > >

2 2

[PATCH] ASoC: max98390: Add reset gpio control
by Steve Lee 11 Mar '22

11 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 | 18 ++++++++++++++++++ sound/soc/codecs/max98390.h | 1 + 2 files changed, 19 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b392567c2b3e..574d8d5f1119 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1073,6 +1073,24 @@ static int max98390_i2c_probe(struct i2c_client *i2c, return ret; } + max98390->reset_gpio = of_get_named_gpio(i2c->dev.of_node, + "maxim,reset-gpios", 0); + + /* Power on device */ + if (gpio_is_valid(max98390->reset_gpio)) { + ret = devm_gpio_request(&i2c->dev, max98390->reset_gpio, + "MAX98390_RESET"); + if (ret) { + dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", + __func__, max98390->reset_gpio); + return -EINVAL; + } + gpio_direction_output(max98390->reset_gpio, 0); + usleep_range(1000, 2000); + gpio_direction_output(max98390->reset_gpio, 1); + usleep_range(1000, 2000); + } + /* Check Revision ID */ ret = regmap_read(max98390->regmap, MAX98390_R24FF_REV_ID, &reg); diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index c250740f73a2..5518f2340247 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -655,6 +655,7 @@ struct max98390_priv { struct regmap *regmap; + int reset_gpio; unsigned int sysclk; unsigned int master; unsigned int tdm_mode; -- 2.17.1

4 4

Presence of "default" device name crashes unity3d games
by GitHub issues - opened 11 Mar '22

11 Mar '22

alsa-project/alsa-lib issue #214 was opened from the3dfxdude: In a recent Slackware 15.0 distro release, the alsa-lib version is now 1.2.5.1. This causes a crash in Kerbal Space Program similar to issue #27. A change was made to close issue #27 in alsa-lib 1.2.3. That change did not correct the crash however in 1.2.3 or 1.2.5.1 in my case. This is the noted difference for "aplay -L" between 1.2.3 and 1.2.1: --- dump_1_2_1 2022-03-10 12:38:16.991433004 -0700 +++ dump_1_2_3 2022-03-10 12:37:53.467443226 -0700 @@ -1,5 +1,9 @@ null Discard all samples (playback) or generate zero samples (capture) +default + Default Audio Device +sysdefault + Default Audio Device default:CARD=IXP ATI IXP, ATI IXP AC97 Default Audio Device Removing this "default" device entry fixes the issue on this system. You may remove the default device name by editing src/conf/pcm/default.conf and adding under the hint section "omit_noargs true". Unity must be parsing the device name, and with certain device names, if there is no ':' doing something bad, as there is a pattern here on what it trips on. I have no idea how to get them to correct this, so opening the issue here at least for the record if anyone else has an issue with this. Also, I am not sure what triggers "default" to show. I did a survey and 3 out of 5 of my slackware 15.0 systems have it. And 2 do not. Does this have something to do with parsing the system alsa config and the available sound devices in the system? I don't define any asound.conf for any of these, so these are almost 100% stock slackware systems, there should otherwise be no difference. Issue URL : https://github.com/alsa-project/alsa-lib/issues/214 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[v2 0/2] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 11 Mar '22

11 Mar '22

The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. 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

3 5