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It is my understanding that the default Pulseaudio configuration is not adequate for my listening equipment, which consists of an odjectiveDAC, an O2 Amplifier, and a pair of Beyerdynamic DT990 600Ohm Headphones. I generally run a pure Alsa setup, however I am having a bout with Gnome, and I am attempting to find a solution.
Is Pulseaudio even able to produce the best quality possible?
If so, any information on that, and is it worth doing so, since Alsa may be superior in this regard?
If not, how would one disable pulseaudio in Gnome 3.16 without uninstalling it?
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PulseAudio is just a sound server that sits on top of ALSA.
https://wiki.archlinux.org/index.php/PulseAudio
Pulseaudio is usually configured using generic settings but can be "Tweaked" like in this example Audiophile
Software that rely on speed(low latency) tend to use Jack or direct access to ALSA...
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I figured this was a stupid question for me considering I help manage rhel servers and such as part of my job. I have read a ton of things about the audio stack online, but you just told me what I wanted to know concisely and in a way that could not be misconstrued. I appreciate that. I guess being such a niche within a niche makes things a little more obscure.
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If not, how would one disable pulseaudio in Gnome 3.16 without uninstalling it?
I think this should work: https://wiki.archlinux.org/index.php/Pu … dio_server
Only thing pulseaudio is needed for in gnome is the volume control, so you really could just remove it and install gnome-shell-extension-aslamixer. Unfortunately arch package maintainers wanted to keep pulseaudio as hard dependency for gnome-settings-daemon, so you'd have to install gnome-settings-daemon-nopulse from AUR, or figure out some other way to get rid of the dependency.
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Depending on your soundcard's supported sample rates (and bitdepths) and the sample rates of the files you want to play, you can configure pulseaudio to "copy" the audio (no resampling).
You can configure two sample rates for which pulseaudio will try not to resample, you can do this by changing /etc/pulse/daemon.conf and configuring the default-sample-rate and alternate-sample-rate options. You might also want to adjust default-sample-format. When playing some audio you can use 'pactl list sinks' and 'pactl list sink-inputs' and check what's happening with regards to resampling.
On another note, I would advise not to change the default resample algorithm, I've had problems before because of that, after an update for some reason the resample method I chose was not playing nice with some pulse clients.
You could always configure your preferred audio player to use the soundcard directly, if pulse is not using it when you start playing audio then things should work, however if your player is using the soundcard directly nothing else will be able to use it, including pulseaudio (this one is kind of obvious I suppose).
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