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Hi.
Im making a audio server that have a more or less good DAC.
So, i have there a mopidy server and a upnp server (upmpdcli). This was working with ALSA, directly. But, i think about playing also in that server the sound of other clients directly, so i read that for that you need to use PulseAudio and his protocol for nerwork.
https://wiki.archlinux.org/title/PulseA … er_network
I follow that, it is working except that i have to disable mopidy (even if it is not playing) because if not i have not sound, probably because only one can rule the sound, it seems.
sudo fuser -v /dev/snd/*
USER PID ACCESS COMMAND
/dev/snd/controlC0: panch 535 F.... pulseaudio
/dev/snd/pcmC0D0p: mopidy 1141 F...m mopidyHow i can avoid that?
I mean, that when i play something from my laptop goes to the audio server and that mopidy in the server can also play, without killing nobody?
Thanks.
Last edited by 9acca9 (2023-02-03 22:24:33)
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Your mopidy needs to play to pulseaudio so that pulseaudio can use the devices. This is easiest done by running mopidy as an user service with the same user that's running the pulse daemon. Or do a pipe to the running pulse daemon from mopidy, mopidy even has some directions of their own for that: https://docs.mopidy.com/en/latest/runni … pulseaudio
Last edited by V1del (2023-02-03 22:39:51)
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Yep, i read that, and also make it work.
But the thing is.... i dont understand how to be sure what quality of audio is pulseaudio playing. That is why i prefer ALSA.
For example if i play a 16 bit 44Khz and send to alsa and put this command:
sudo cat /proc/asound/IQaudIODAC/pcm0p/sub0/hw_params i get exactly the quality of the audio, in that case 16bit 44Khz. That parameters change with the quality of the sound (perfect).
But, with pulse audio is the same if i play a 24bit 192Khz or 16bit 44Khz, if i try that command i always get the maximum...
So, i want to be sure about the quality of the sound, but with pulse audio dont know how to be sure.
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If you can't discern the difference in bitrate with your ears by listening to the output then why does it even matter?
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You can set avoid-resampling=true in the daemon.conf which will make it aggressively try to maintain input sample/bit rate, but as soon as you play "something" that is going to be you target frequency until you stop all playback and resume again. You can't avoid this, you can only open the sound card at one specific rate at a time.
And slithery's point is an important one, you are most likely not going to hear a difference even if resampling happens. Most of this stuff happens way out of the general human hearing spectrum and if you don't do blind AB tests your brain is going to tell you what you want to hear/see despite it not being factual.
Last edited by V1del (2023-02-04 01:30:17)
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