- Nov 30, 2024
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Frank Oltmanns authored
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Frank Oltmanns authored
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Frank Oltmanns authored
Also, rename it to "Dummy". With the section completely stripped except for the PlaybackPCM, the bottom mic still works. When removing the PlaybackPCM, the bottom mic stops working for some reason.
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Frank Oltmanns authored
Bottom mic still works without the headset section.
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- Nov 25, 2024
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Frank Oltmanns authored
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Frank Oltmanns authored
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Frank Oltmanns authored
VoiceCall.conf is currently broken, but HiFi.conf works even for calls. Therefore, replace the content of VoicCall.conf with the confent from HiFi.conf in order to start development of the "VoiceCall" verb. Note: This deliberately sets some unexpected behaviours: - The TQ is set to "HiFi". - The priorities are mixed up, so now calls start in speaker mode. - The "Headphone" and "Headset" section might not work (untested in calls) or not even make sense. Despite these inconsistencies, this is the first step in restoring a working voice call configuration for beryllium. Additional commits need to clean this up.
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- Sep 27, 2024
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Joel Selvaraj authored
The mic related confs are commented out as it is not working. But I have kept it as reference as it should be something similar.
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- Jun 18, 2024
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Federico Amedeo Izzo authored
Updated devices: oneplus-enchilada, oneplus-fajita and xiaomi-beryllium PulseAudio 17.0 introduced a different mapping of ucm config to pulse profiles. \ The current profile caused an `Assertion 'dev == data->device'` error, which resulted in audio being broken on sdm845 devices. The error seems to be caused by the `Mic` device having the same name in both HiFi.conf and VoiceCall.conf. Renaming `Mic` to `Mic1` in HiFi.conf, running `alsaucm reload` and rebooting the phone fixed the audio issue. Reference: * [PulseAudio 17.0 release notes](https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/17.0/) * [PulseAudio !596 introducing new ucm mapping](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596)
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- Apr 06, 2024
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jenneron authored
This one seems to be the same to Lenovo Yoga C630, so let's symlink it. Signed-off-by:
Anton Bambura <jenneron@postmarketos.org>
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jenneron authored
This has been added to kernel recently, so make use of it here. Signed-off-by:
Anton Bambura <jenneron@postmarketos.org>
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jenneron authored
Besides the improvement by using analog volume control, it also makes volume levels normal instead of insanely high levels reached by this digital mixer. Signed-off-by:
Anton Bambura <jenneron@postmarketos.org>
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- Jan 06, 2024
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Caleb Connolly authored
The kcontrol changed name when we moved to the upstream implementation. Signed-off-by:
Caleb Connolly <caleb.connolly@linaro.org>
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- Dec 09, 2023
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The volume levels in BootSequence doesn't get applied for headphones if the HP digital volume is specified. This pushes the max volume to very high levels that can even hurt the ears. So let us remove it for now until we find a way to implement is properly.
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The volume levels in BootSequence doesn't get applied for headphones if the HP digital volume is specified. This pushes the max volume to very high levels that can even hurt the ears. So let us remove it for now until we find a way to implement is properly.
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In OP6, there seems to be a race condition where q6voiced opens the voicemmode device before the routes are configured, leading to: [ 85.540643] VoiceMMode1: ASoC: no backend DAIs enabled for VoiceMMode1 [ 85.540678] VoiceMMode1: ASoC: error at dpcm_fe_dai_prepare on VoiceMMode1: -22 [ 85.540724] VoiceMMode1: ASoC: no backend DAIs enabled for VoiceMMode1 [ 85.540733] VoiceMMode1: ASoC: error at dpcm_fe_dai_prepare on VoiceMMode1: -22 So, add them to HiFi conf too, so the mixer confs are in place even if voice profile get switched to slightly later. To be safe, do the same workaround for OP6T and Poco F1 too.
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Based on enchilada.
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VoiceCall profile only supports Earpiece as both q6voice driver and callaudiod doesn't handle multiple devices properly.
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Sharing AIF1_PB on RX0 and RX1 is a problem because RX0 and RX1 are different devices. AIF1_PB is for the earpiece only.
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The SHIFT6mq (axolotl) is an SDM845-based phone with a WCD934x codec. It has the following audio devices: - Headset - Speaker - Earpiece - Headset microphone - Back microphone - Bottom microphone It exposes 6 different MultiMedia* devices for routing the audio over SLIMBUS and als a VoiceMMode1 for routing audio from and to the modem for making phone calls. To cover its use cases, a default HiFi use case is provided to play audio from music, videos, etc. and a VoiceCall use case to make phone calls.
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- Configure the earpiece volume as well with a default value like all the other output interfaces of the WCD934x codec. - The WCD934x codec has 3 microphone interfaces, configure them all. The first microphone was already configured to be used for a headset. Do the same for the other 2 microphones. This won't influence devices with less than 3 microphones because the microphones still have to be enabled in the HiFi or VoiceCall profiles. - The WCD934x codec has an earpiece output for phones. Add the config to enable and disable this earpiece. - Add enable and disable configs for the additional microphones found in the WCD934x codec.
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I removed unnecesarry csets and added comments to make it more clearer.
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Fixes mic switch issues. It now switches seamlessly between both phone and headphone mics.
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Apart from default audio routes, also configure the routes for q6voice kernel driver. Also sets Earpiece with higher priority, so its used by default during calls.
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Sets Speaker as the default playback device and Internal Mic as the default capture device
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Both channels go to the same speaker, so it all works fine still. This fixes the sound settings test button (no more static) and might improve compatibility with software that doesn't support mono outputs very well.
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The earpiece speaker causes the headphones and speaker to break a lot, it's not very reliable. As we don't have voice calls yet anyway lets just disable it for now.
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The phone uses MAX98927 as speaker amplifier and WCD934x for earpiece, headphones and mics.
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The phone uses TAS2559 as speaker amplifier and WCD934x for headphones and mics.
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- Oct 31, 2023
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Unlike all designs supported sofar the Lenovo Yoga Tab 3 YT3-X90 does not have its internal microphone (intmic) on IN3L with the headset microphone on IN1L. Instead this tablet has the intmic on IN1L and the hsmic on IN2L. Add IN1-InternalMic.conf and IN2-HeadsetMic.conf config snippets under ucm2/codecs/wm5102/ for this and check the components string to determine which microphone routes should be used. Link: https://lore.kernel.org/alsa-devel/20231021211614.115152-2-hdegoede@redhat.com/ Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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On the Lenovo Yoga Tab 3 YT3-X90 the speaker amplifiers are connected to the HPOUT2 output pins instead of the the SPK output pins. Check for "cfg-spk:hpout2" in the components string which indicates this setup and add a ucm2/codecs/wm5102/HPOut2-Speaker.conf file for this setup. Link: https://lore.kernel.org/alsa-devel/20231021211614.115152-1-hdegoede@redhat.com/ Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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2 mono speaker setup fixes: 1. Use the kernel's components string to check for a mono-speaker device when the board has a components string. 2. So far known nau8824 boards with a mono speaker where using the right speaker channel, which is unusual. Normally mono speaker setups use only the left speaker channel. The Cyberbook T116 tablet is a nau8824 based model, which indeed uses the left speaker channel for its single speaker. Modify ucm2/codecs/nau8824/MonoSpeaker.conf to send a left+right channel mix to both speaker channels, so that things will work independent of which speaker channel is used for a mono setup. Link: https://lore.kernel.org/alsa-devel/20231021143109.52210-1-hdegoede@redhat.com/ Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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When codecs/es8316/MonoSpeaker.conf's EnableSeq was run last before shutdown the 'Playback Polarity' and 'DAC Mono Mix Switch' controls for Mono Speaker output get saved and restored on the next boot. These settings are undone by the MonoSpeaker DisableSeq, but that never runs when an updated kernel switches from the default "cfg-spk:1" in components to "cfg-spk:2" for devices which do actually have 2 speakers, which causes the ucm profile to use codecs/es8316/Speaker.conf instead. Set the 'Playback Polarity' and 'DAC Mono Mix Switch' controls to their defaults in codecs/es8316/EnableSeq.conf to avoid them getting stuck in Mono mix mode in this scenario. Link: https://lore.kernel.org/alsa-devel/20231029145007.24106-1-hdegoede@redhat.com/ Signed-off-by:
Hans de Goede <hdegoede@redhat.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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- Oct 30, 2023
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Closes: https://github.com/alsa-project/alsa-ucm-conf/pull/361 Signed-off-by:
WeirdTreeThing <bradyn127@protonmail.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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For "grunt" chromebooks Closes: https://github.com/alsa-project/alsa-ucm-conf/pull/361 Signed-off-by:
WeirdTreeThing <bradyn127@protonmail.com> Signed-off-by:
Jaroslav Kysela <perex@perex.cz>
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