I need to change the default ALSA sample rate from 48000 to 44100. Can anyone help?
I’m using 11.2
I need to change the default ALSA sample rate from 48000 to 44100. Can anyone help?
I’m using 11.2
I’ve never done this myself. If you surf the web you will read of many ways in which users claim this can be done.
One example (ref: .asoundrc - ALSA wiki](http://alsa.opensrc.org/index.php/.asoundrc) ) is to create a custom /home/username/.asoundrc file that looks like:
pcm.rate_convert {
type plug
slave {
pcm "hw:0,0"
rate 48000
}
}
To undo you just remove the .asoundrc file.
another example from a Ubuntu forum How do I change the ALSA sample rate? - Ubuntu Forums (in which I have less confidence) suggests the command:
jackd -d alsa -r 48000
I don’t know how one would undo ? Perhaps run again with a different frequency ?
If you surf with google you will find many other suggested methods.
Good luck, and after you figure this out perhaps you could past as to what worked, and exactly why you needed to do this.
Please note sometimes pulse audio can cause problems so don’t let a pulse audio problem fool you into thinking you need to change alsa sample rate.
To oldcpu,
The jackd command starts the JACK daemon. Therefore the sample rate will only be applicable if one is using JACK-enabled software - Ardour, Hydrogen, Rosegarden for example…
(JACK | connecting a world of audio)
To rektruax
I don’t think you actually need to set the sample rate. If you are recording, you can usually choose your sample rate in the software. For example, Audacity has a menu option to set the default sample rate.
I thought that playback usually respected the sample rate of the audio file being played.
But perhaps you have some specific scenario in mind?
Paul
Thanks for all the suggestions… I’ll be busy with those for a while.
Alsa’s current default @ 48000 seems to override the settings in Hydrogen and MuseScore. I can set them up to export @ 44.100 but the resulting file is still 48 because Alsa has the I/O default. The problem is…
When I wish to use both live and computer generated music, my Tascam recording equipment only accept file imports of 44.100. That’s coupled with the fact that it only exports @ 44.100 as well. Says so right in the manual - it will only use 44100. So when I produce a string or drum part, open it with Audacity, then import a live recorded part, they’re at different sample rates and won’t sync.
I open QJackCtl>Setup>Settings, under Parameters: Sample Rate: dropdown menu, I select 44100.