Pros/cons of running PulseAudio at 24 Bits to match hardware?

Question

Long-time lurker, first post - please be gentle.

I am trying to get my USB audio interface to reproduce bit-perfect audio. It is a Native Instruments Audio Kontrol 1, which is plug-and-play on my Ubuntu 14.04LTS system.

The sound card runs at 24-bit/192KHz natively, and while I can edit asound.conf to make it run at other sampling rates, only 24-bit depth is supported (not 16).

I understand that by default both ALSA and PulseAudio will downsample audio material to 16-bits/48KHz. I can edit the config files of each so that they run at 24-bit/48KHz instead. I understand that this will increase CPU overhead. Sounds good right?

I have two questions/reservations:

a.) It says here that ALSA automatically resamples all audio playback to 16-bit/48KHz . Does this override my card's settings in asound.conf ?

b.) If so, is running PulseAudio at 24 bits just a waste of overhead (Since PulseAudio uses ALSA as an intermediary layer to the soundcard)?

Thanks for any help you can provide!

Solution

a) In short: no. By default pulseaudio and alsa are configured for 44.1kHz 16-bit audio. However, if you configure pulseaudio to e.g. 48kHz and 24-bit audio, see below for details, that's the minimum your device will get assuming it's supported. E.g. I configured pulseaudio, and made no changes to alsa whatsoever, to feed my external USB DAC with a 96KHz and 24-bit audio stream. LED's on the DAC confirm it's getting the 96kHz (@24-bit) after the change.

b) Given the above, NA :).

On how to configure (pulseaudio): edit with sudo and your favorite editor /etc/pulse/daemon.conf and look for the following lines:

; default-sample-format = s16le
; default-sample-rate = 44100

uncomment by removing the ';' and change settings to your liking. E.g.:

default-sample-format = s24le
default-sample-rate = 192000

Then restart pulseaudio (with your user account, no sudo):

pulseaudio -k
pulseaudio --restart

You can confirm settings changed with:

pacmd list-sinks 

and look for 'sample spec' for the 'sink' that corresponds to your Native Instruments device.

Comments

[victorlautreamont]

That's great. The post I linked definitely made it sound like even if I ran PulseAudio at 24 bits, ALSA would just downsample it in software before it ever got to the DACs. If you've confirmed at the hardware that this is not the case then I'll take your Answer to be more authoritative than theirs. Thank you!

[Bachsau]

ALSA does not resample anything if you're not using a resampler plugin like DMiX. When you're using PulseAudio it is the resampler plugin, so its configuration is what matters.