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[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 14 Dec '20

14 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

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[PATCH alsa-lib 1/3] chtnau8824: Fix mono speaker config not working
by Hans de Goede 14 Dec '20

14 Dec '20

Fix the Speaker define / var being changed from "Speaker" to "MonoSpeaker" not having any effect because the variable gets expanded before the if making the change is interpreted. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/chtnau8824/HiFi.conf | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 3ccd6b1..d7a5f63 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -34,8 +34,15 @@ SectionVerb { } } -Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" -Include.hp.File "/codecs/nau8824/HeadPhones.conf" +# The includes using $vars in there path must be conditional otherwise the +# $var gets expanded before the other If-s above can change the vars. +If.cfg-includes { + Condition { Type String Empty "" } + True { + Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" + Include.hp.File "/codecs/nau8824/HeadPhones.conf" -Include.mic.File "/codecs/nau8824/InternalMic.conf" -Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + Include.mic.File "/codecs/nau8824/InternalMic.conf" + Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + } +} -- 2.28.0

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[RFC PATCH 0/4] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 14 Dec '20

14 Dec '20

v1: Initial patch version, to enable DMA mode on Intel Keembay platform by exposing some dmaengine api to work around DMA limitations at the client driver level. This patchset suggests an ALSA-only quirk, having other more generic means to deal with this limitation would be fine - we just wanted to have a discussion on preferred directions. The IPs used are not Intel-specific so sooner or later someone else will have similar limitations to work-around. Michael Sit Wei Hong (4): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: soc-generic-dmaengine-pcm: Add custom prepare and submit function ASoC: dmaengine_pcm: expose functions to header file for custom functions ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ include/sound/dmaengine_pcm.h | 21 ++ sound/core/pcm_dmaengine.c | 46 ++-- sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 202 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + sound/soc/soc-generic-dmaengine-pcm.c | 8 +- 7 files changed, 267 insertions(+), 35 deletions(-) -- 2.17.1

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Problem with synthesized output
by Spencer Kohan 13 Dec '20

13 Dec '20

Hello, I hope I am posting this to the correct place. I'm trying to use a Raspberry Pi as a wifi speaker, and as a first step I'm trying to just play some sound out of a buffer through the attached amp. I'm using an AMP2 with a RPi 4, and I'm getting some strange behavior when I try to play any "sine" output. So for instance, if I use `speaker-test`, I am able to play the wav test and the pink-noise test just fine, but when I try to play the sine test, it plays for like one quick instant, and then the program blocks. What's strange is I get the exact same behavior when I use other methods to try to play a sine output over ASLA. For instance, if I use sox: play -n synth sine 1000 I get the same output: one quick blip of the tone, and then the program blocks. The output to the console looks like this: play WARN alsa: can't encode 0-bit Unknown or not applicable File Size: 0 Encoding: n/a Channels: 1 @ 32-bit Samplerate: 48000Hz Replaygain: off Duration: unknown In:0.00% 00:00:00.00 [00:00:00.00] Out:0 [ | ] Clip:0 And then I get the exact same behavior when I try to do the `sio_sine` example from [libsoundio](https://github.com/andrewrk/libsoundio), with this output: Backend: ALSA ALSA lib pcm_dsnoop.c:575:(snd_pcm_dsnoop_open) The dsnoop plugin supports only capture stream Output device: snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0: Default Audio Device Software latency: 0.008000 'p\n' - pause 'u\n' - unpause 'P\n' - pause from within callback 'c\n' - clear buffer 'q\n' - quit unable to set channel layout: incompatible device Also interesting, when I try the [rsoundio]( https://github.com/klingtnet/rsoundio) sine example, it reports an underflow. My output for aplay looks like this: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device sysdefault:CARD=sndrpihifiberry snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Default Audio Device dmix:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample mixing device dsnoop:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct sample snooping device hw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Direct hardware device without any conversions plughw:CARD=sndrpihifiberry,DEV=0 snd_rpi_hifiberry_dacplus, HiFiBerry DAC+ HiFi pcm512x-hifi-0 Hardware device with all software conversions It's just strange because the pre-recorded sample and the pink noise version work just fine. How would I even go about trying to debug this?

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Bug in the ALSA driver 'snd_usb_audio'
by GitHub issues - opened 12 Dec '20

12 Dec '20

alsa-project/alsa-lib issue #105 was opened from rombru: Hello, I'm not an expert in linux systems. I will try to give a maximum of information. I've bought Bowers & Wilkins PX7 headphones. They doesn't work using USB on Ubuntu. Using Ubuntu and bluetooth, or audio jack, the headphones work well. Using Windows 10 and USB, the headphones work well. I d'like too use them with USB on Ubuntu because that recharge them in parallel. Problems while the headphons are connected : On Chrome : The videos are played too fast and there's no sound (Youtube & Twitch). On Firefox : The videos are played too fast and there's a crackling sound (Youtube & Twitch). On VLC : No sound I've tried with Ubuntu 20.04 & 18.04. They are set as default audio device I've played with PulseAudio and alsamixer (reinstall, mute/unmute, ...) The headphones are set as fallback in PulseAudio. Trying to find the issue, i've make some test found on Google : I've added pcie_aspm=off in GRUB_CMDLINE_LINUX_DEFAULT Setting devices as default in /etc/asound.conf --- The ALSA part I think the problem come from ALSA, i've found the errors : `Failed to load module "module-alsa-card` followed few lines later by `Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers.` Output of `journalctl --user -u pulseaudio` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pulseaudio -vvvvv` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Output of `pacmd ls` is [here](https://gist.githubusercontent.com/rombru/eb4ca14168494e95c62a961f92e… Please, tell me if you need something more to analyse the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/105 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH AUTOSEL 5.9 06/23] ASoC: rt5682: change SAR voltage threshold
by Sasha Levin 12 Dec '20

12 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit aa4cb898b80a28a610e26d1513e6dd42d995c225 ] To fix errors in some 4 poles headset detection cases, this patch adjusts the voltage threshold for mic detection. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20201126092759.9427-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a4713bd6508d5..e260eddb51d96 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, + {RT5682_SAR_IL_CMD_3, 0x8365}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.27.0

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[PATCH 1/2] ALSA: hda/ca0132 - Add 8051 PLL write helper functions.
by Connor McAdams 12 Dec '20

12 Dec '20

Add helper functions for the 8051 PLL PMU write verbs. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 130 ++++++++++++++--------------------- 1 file changed, 50 insertions(+), 80 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e9019e840a4d..9b6b7601a881 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1949,6 +1949,14 @@ static unsigned int chipio_8051_get_data(struct hda_codec *codec) VENDOR_CHIPIO_8051_DATA_READ, 0); } +/* PLL_PMU writes share the lower address register of the 8051 exram writes. */ +static void chipio_8051_set_data_pll(struct hda_codec *codec, unsigned int data) +{ + /* 8-bits of data. */ + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, data & 0xff); +} + static void chipio_8051_write_exram(struct hda_codec *codec, unsigned int addr, unsigned int data) { @@ -1977,6 +1985,26 @@ static void chipio_8051_read_exram(struct hda_codec *codec, *data = chipio_8051_get_data(codec); } +static void chipio_8051_write_pll_pmu(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); + + mutex_unlock(&spec->chipio_mutex); +} + +static void chipio_8051_write_pll_pmu_no_mutex(struct hda_codec *codec, + unsigned int addr, unsigned int data) +{ + chipio_8051_set_address(codec, addr & 0xff); + chipio_8051_set_data_pll(codec, data); +} + /* * Enable clocks. */ @@ -1985,18 +2013,11 @@ static void chipio_enable_clocks(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 5); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x0b); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 6); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); + + chipio_8051_write_pll_pmu_no_mutex(codec, 0x00, 0xff); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x05, 0x0b); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x06, 0xff); + mutex_unlock(&spec->chipio_mutex); } @@ -7904,10 +7925,7 @@ static void ae5_post_dsp_register_set(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); writeb(0xff, spec->mem_base + 0x304); writeb(0xff, spec->mem_base + 0x304); @@ -7949,30 +7967,11 @@ static void ae5_post_dsp_param_setup(struct hda_codec *codec) static void ae5_post_dsp_pll_setup(struct hda_codec *codec) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x45); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcc); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x40); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xcb); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); - - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x51); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x8d); + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); + chipio_8051_write_pll_pmu(codec, 0x45, 0xcc); + chipio_8051_write_pll_pmu(codec, 0x40, 0xcb); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x51, 0x8d); } static void ae5_post_dsp_stream_setup(struct hda_codec *codec) @@ -7994,10 +7993,7 @@ static void ae5_post_dsp_stream_setup(struct hda_codec *codec) chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80); @@ -8090,12 +8086,8 @@ static void ae7_post_dsp_pll_setup(struct hda_codec *codec) }; unsigned int i; - for (i = 0; i < ARRAY_SIZE(addr); i++) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]); - } + for (i = 0; i < ARRAY_SIZE(addr); i++) + chipio_8051_write_pll_pmu_no_mutex(codec, addr[i], data[i]); } static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) @@ -8111,10 +8103,7 @@ static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) mutex_lock(&spec->chipio_mutex); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7); chipio_write_no_mutex(codec, 0x189000, 0x0001f101); chipio_write_no_mutex(codec, 0x189004, 0x0001f101); @@ -8187,10 +8176,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) { chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); @@ -8207,10 +8193,7 @@ static void ae7_post_dsp_asi_setup(struct hda_codec *codec) ae7_post_dsp_pll_setup(codec); ae7_post_dsp_asi_stream_setup(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + chipio_8051_write_pll_pmu(codec, 0x43, 0xc7); ae7_post_dsp_asi_setup_ports(codec); } @@ -9367,18 +9350,11 @@ static void ae5_register_set(struct hda_codec *codec) unsigned int i, cur_addr; unsigned char tmp[3]; - if (ca0132_quirk(spec) == QUIRK_AE7) { - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); - } + if (ca0132_quirk(spec) == QUIRK_AE7) + chipio_8051_write_pll_pmu(codec, 0x41, 0xc8); chipio_8051_write_direct(codec, 0x93, 0x10); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + chipio_8051_write_pll_pmu(codec, 0x44, 0xc2); if (ca0132_quirk(spec) == QUIRK_AE7) { tmp[0] = 0x03; @@ -9454,10 +9430,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE5: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); chipio_write(codec, 0x18b030, 0x00000020); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); @@ -9465,10 +9438,7 @@ static void ca0132_alt_init(struct hda_codec *codec) break; case QUIRK_AE7: ca0132_gpio_init(codec); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); - snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, - VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + chipio_8051_write_pll_pmu(codec, 0x49, 0x88); snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); chipio_write(codec, 0x18b008, 0x000000f8); -- 2.25.1

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[PATCH alsa-utils] alsactl: Remove asound.state file check from alsa-restore.service again
by Takashi Iwai 12 Dec '20

12 Dec '20

We added the check of asound.state file presence some time ago to assure that alsactl gets called only if the state file is already present. Since then, the situation has changed significantly: e.g. now alsactl does initialize if the state file isn't present, and the same alsa-restore.service is used to save the state. This means that we should start this service no matter the state file exists at the boot time or not. So, revert the old change again. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/alsa-restore.service.in | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index a84c2e842444..80fd5fd48203 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -7,7 +7,6 @@ Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -ConditionPathExists=@asoundrcfile@ [Service] Type=oneshot -- 2.26.2

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[PATCH alsa-utils 1/2] alsactl: Fix double decrease of lock timeout
by Takashi Iwai 12 Dec '20

12 Dec '20

The state_lock() has a loop to wait for the lock file creation, and the timeout value gets decremented twice mistakenly, which leads to a half timeout (5 seconds) than expected 10 seconds. Fix it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/lock.c | 1 - 1 file changed, 1 deletion(-) diff --git a/alsactl/lock.c b/alsactl/lock.c index 4a485392b3bd..05f6e4d2a102 100644 --- a/alsactl/lock.c +++ b/alsactl/lock.c @@ -63,7 +63,6 @@ static int state_lock_(const char *file, int lock, int timeout, int _fd) if (fd < 0) { if (errno == EBUSY || errno == EAGAIN) { sleep(1); - timeout--; } else { err = -errno; goto out; -- 2.26.2

1 1

[PATCH 1/2] dt-bindings: add simple-audio-mux binding
by Alexandre Belloni 11 Dec '20

11 Dec '20

Add devicetree documentation for simple audio multiplexers Signed-off-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> --- Cc: Rob Herring <robh+dt(a)kernel.org> .../bindings/sound/simple-audio-mux.yaml | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-mux.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml new file mode 100644 index 000000000000..5986d1fcbb54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-mux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Multiplexer + +maintainers: + - Alexandre Belloni <aleandre.belloni(a)bootlin.com> + +description: | + Simple audio multiplexers are driven using gpios, allowing to select which of + their input line is connected to the output line. + +properties: + compatible: + const: simple-audio-mux + + mux-gpios: + description: | + GPIOs used to select the input line. + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: + Used as prefix for sink/source names of the component. Must be a + unique string among multiple instances of the same component. + +required: + - compatible + - mux-gpios + +additionalProperties: false + +examples: + - | + mux { + compatible = "simple-audio-mux"; + mux-gpios = <&gpio 3 0>; + }; -- 2.28.0

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