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Most of the 'mushiness' from Pulse comes from the default fast-and-dirty mixing and resampling methods it uses. Flat volume adds more weirdness to the mix as well. You can configure ALSA's internal mixer (dmix) and remove Pulse entirely, but even Firefox gutted ALSA sink support, so goodl luck :(
Pulse has a lot of flaws, but there's enough middleware that can only interface with a PA sink that it's a bit of a necessary evil. You'll run into a bit of the same with Jack too.
Basically, for total 'audio purity', you don't want to perform any mixing / resampling at all, which means exclusive device access.
You can, however, tweak Pulse for better quality (at the expense of slightly more CPU overhead) by avoiding resampling and using greater precision mixing -- or force pulse to share a dmix sink with other applications that can directly use ALSA (can require a bit of tinkering)
If you do try out Jack, make sure to switch to a realtime kernel.
https://old.reddit.com/r/linuxmasterrace/comments/byn028/audiophile_settings_on_linux_june_2019/
https://medium.com/@gamunu/enable-high-quality-audio-on-linux-6f16f3fe7e1f
https://wiki.archlinux.org/index.php/PulseAudio#Configuration_files
But still only real issue I have. Is Civ 6. Seems like artificial barrier. The Voice doesn't work, but than it did work. As if something not on my end was going on. But than it wasn't working any more. (Edit: I did have an issue with Civ6 but seems it's resolved now)
I've definitely run across a good number of programs which only see Pulse audio sinks, which has led me to its use, somewhat begrudgingly.
Not sure if you just want to complain (which I've done plenty of myself haha) or want solutions. Like I said though, to maintain good compatibility you can make Pulse sound better, or let PA share a dmix sink for the software which won't play nice.
How do you make pulse sound a little better? I might dig into this didn't know it was possible.
https://www.gamingonlinux.com/2020/09/bringing-together-audio-and-video-pipewire-for-linux-is-really-coming-along
Last I heard about it it was working fine when replacing ALSA in a system where original PulseAudio is running... so maybe by now its replacement for PulseAudio is good to go too? Worth investigating...
IIRC one of the things it was praised for is exactly making it easier to have higher quality and lower latency sound for software interfacing with PulseAudio instead of directly to ALSA.
(if u know what ure doing)
Else, stick with pulse (u can install pavucontrol to help you to make pulseaudio suck less)
Check the links I posted above. Also try testing with 'flat volume' enabled / disabled and see which you prefer.
It is worth noting there is also a minimal alternative to pulse available which runs like pulse without all the bulk. I'm not sure of a complete list of it's capabilities, though.
If you set your hardware playback sample rate to say 96khz, and 99% of your media is 48khz (which is a good sampling frequency, mind you), then that all has to undergo a 2x upsample, meaning every frame gets 'played' twice which doesn't at all improve the granularity of the audio signal, but still perceptibly changes it (less pure). Time stretching can occur more often depending on clock jitter.
If you play back 44khz audio and it gets upscaled to 96khz, it has to undergo a non-integer scaling (96k / 44k = 2.1818...) which stretches audio frames unevenly over the time signature. This results in constant micro-artifacting which, of course, reduces playback quality.
Letting the mixer adjust the hardware to match the input rate is really the best method where possible. Otherwise, if 99% of your media is 48khz forcing your hardware to match that rate is better than forcing upsampling / downsampling. A few lower quality (8khz-22khz) samples getting upsampled to 44khz / 48khz matter a little less since they sound like ass either way ; )
As a note, I do default sample type as s24ne (ne means use native byte order to the processor which is slightly faster. It's saves changing that entry when creating it on different platforms such as ARM)
for sample rate I use the maximum of my card (192000)
To do all this I uncoment and change the following lines in my pulsaudio daemon.conf. (/etc/pulse/daemon.conf)
resample-method = speex-float-6
default-sample-format = s24ne
default-sample-rate = 192000
But no, even integer upscaling does not improve sound quality. It will never be more pure than the source. You cannot "improve" digital audio quality over the source, you can change it by adding fake stuff to it, or taking stuff away.
Similarly, the analogue audio path in EQs, DACs, amplifiers, etc. change the signal in some way, but this is effectively distortion. Some people prefer certain types of distortion, like tube amps, but it's effectively reducing signal purity. When you're dealing with digital signals, the typical goal is not to mess them up before they get to the pre-amp ; )
There's plenty of ignorant mysticism surrounding audiophile circles -- you're guaranteed to hear type every of bs there is.
Yes, this is true. However. If sample rate is low on source, upsampling can only help a little if at all. It is 'distortion' from source yes. Upsampling to 192000 is pretty intense. But there isn't as much harm in upsampling as maybe you are saying. I look at it like Head Room, with an amp and everything you kinda want some room so things don't sound oversaturated or maxed out. On a laptop I'd actually say probably you'd lose battery life aha. Even if it was minimal.
But his sound card is probably pretty sweet.
Again ALSA I haven't seen any output higher than 100% where Pulse can actually boostt -DB output by a fair amount.