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Configuring Linux ALSA/Pulseaudio for best sound
post Mar 18 2012, 21:27
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So I have been listening to a bunch of hokum and voodoo at some other head hifi forum where abx testing is banned and I need some reeducation.

I run linux at home and I use kde. As I understand little about sound and how all the parts of the linux sound system work together, I would like to help to make sure that my sound is being output in the best fidelity possible.

So I used KDE that means I have the phonon sound system on top of the pulseaudio sound server on top of the ALSA sound system (whew!).

My onboard soundcard is an HDA-Intel Realtek ALC275 that does 16/24 bitrate and 44.1-192 sampling.

So most of my music is in 16/44.1 FLAC format, but some is 24/96 FLAC.

I use amarok and deadbeef to play my music.

I just want to make sure that I am not doing any resampling that will introduce any distortion or noise.

Sorry if this a topic that has been answered numerous times, but I searched and couldn't find anything that helped answer this.

Thanks for the input.
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post Apr 9 2012, 19:18
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I don't think ALSA does any samplerate conversion if PulseAudio (or especially Jackd) is being used, as PulseAudio uses direct access to the card

but Pulse has a few audio settings that change the sample-rate conversion algorithm, and default sample rate

Not in Linux at the moment so bare with me, /etc/pulse/daemon.conf should have settings in it

man pulse-daemon.conf should list the possible resample algorithms (and bit depth there)

there's a few options which are well documented in other parts of the internet, the 'src' ones are of good quality, Google "Secret Rabbit Code"

for the finer details, but generally the 'sinc' algorithms are the only good subset, and the "fastest" setting starts cutting off frequencies at 80% (17.5kHz*), the "medium" does so at 90% (19.8Khz*), and the "best" one does 97%(21.4kHz*), each one uses more CPU so if you're on a laptop I would avoid the "best" options as it uses about 40% CPU last time I tried, and all have a SNR of 97dB.

not sure about the speex-float or other speex algorithms, but the speex-float of quality 10 looks OK but maybe be very CPU intensive like the secret rabbit code's 'best' setting.

you should also try the SoX utility to covert the 24/96 to 16/44.1 for portable players etc, and has the ability to use SRC if you want to ABX.

* (based on the base rate of 44.1kHz)

hopefully this is useful info (also my first post blink.gif )
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post Apr 11 2012, 20:39
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QUOTE (YellowOnion @ Apr 9 2012, 14:18) *
I don't think ALSA does any samplerate conversion if PulseAudio (or especially Jackd) is being used, as PulseAudio uses direct access to the card

No, Pulseaudio actually outputs to hardware through ALSA, but it will resample because one of its function is to act as a software mixer. However, if you use ALSA directly and set it up to output to hardware (instead of going through dmix)- it will not resample.

This post has been edited by cpchan: Apr 11 2012, 20:57
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