Hey everyone, I’m new here so I could be making a newb error lol. Everything had been going relatively smoothly for a few weeks until I restarted my computer and audacity one day only to find that everything that I recorded sounded really distorted. I noticed if I moved the bass all the way down and the trebel all the way up, it would almost sort of fix the disortion, but it would throw the quality of the track completely off. I tried adjusting the sound controls in alsamixer, no luck there. If I move the gain down, it helps somewhat, but it just makes it sound too quiet, and doesn’t take the disortion away completely anyway. Even if I open a new project I still get the same distortion. Do any of you have any ideas why this would happen all of a sudden and what I can do about it? I’m using xfce4 if that helps at all.
Does your computer sound system use PulseAudio?
Which Linux distribution?
Honestly I’m not sure. Is there a way to find out? I’m completely new at this whole Linux thing. I really only downloaded Linux so I could use audacity since it’s not compatible with Chromebook.
Ubuntu I believe.
So you’re running some version of Linux on a Chromebook with XFCE desktop?
Could you enter this command into a terminal window, and tell us what the output is:
lsb_release -a
and then
pulseaudio --version
then open Audacity and look in “Help > About Audacity” for the full version number of Audacity.
Yes.
lsb_release: error: No arguments are permitted
(precise)bluerain@localhost:~$ lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description: Ubuntu 12.04.5 LTS
Release: 12.04
Codename: precise
bash: pulseaudio: command not found
Audacity 2.1.1-alpha-Apr 8 2015
Please note that I have never tried running Linux on a Chromebook.
I am trying to work out how the audio system is configured on your machine so that I can try to help.
In Audacity, select “Help > Audio Device Info”.
Copy and paste the full output into your reply.
Thank you, I appreciate it!
==============================
Default recording device number: 1
Default playback device number: 1
==============================
Device ID: 0
Device name: cras
Host name: ALSA
Recording channels: 2
Playback channels: 2
Low Recording Latency: 0.011610
Low Playback Latency: 0.011610
High Recording Latency: 0.046440
High Playback Latency: 0.046440
Supported Rates:
8000
9600
11025
12000
15000
16000
22050
24000
32000
44100
48000
==============================
Device ID: 1
Device name: default
Host name: ALSA
Recording channels: 2
Playback channels: 2
Low Recording Latency: 0.011610
Low Playback Latency: 0.011610
High Recording Latency: 0.046440
High Playback Latency: 0.046440
Supported Rates:
8000
9600
11025
12000
15000
16000
22050
24000
32000
44100
48000
==============================
Selected recording device: 1 - default
Selected playback device: 1 - default
Supported Rates:
8000
9600
11025
12000
15000
16000
22050
24000
32000
44100
48000
Unable to open Portmixer
It appears that Chromebooks have a unique audio device setup. https://www.chromium.org/chromium-os/chromiumos-design-docs/cras-chromeos-audio-server
What exactly are you trying to record?
Are you able to adjust the recording level so that you get a signal in Audacity that is about half the track height? If so, please make a short test recording and export a few seconds in WAV format and attach the file to your reply (see here for detailed instructions: https://forum.audacityteam.org/t/how-to-post-an-audio-sample/29851/1)
Haha yeah, Chromebooks are definitely a pain.
I’m trying to record vocals on top of a beat that I imported.
Ok here it is. I think I did it correctly.
Here’s the problem. See how the waveform has a “notch” in it where it briefly goes to the centre line. I’ve highlighted one, and there is another one visible a little earlier.
Those notches should not be there. The strange thing is that they occur at exact intervals of 128 samples.
I doubt that this is an Audacity problem. It looks more like a problem with the audio device drivers.
Try running this command in a terminal window. It will record 6 seconds of audio (starting when you press “Enter”) from the default device (should be your mic). By default it will save the file to your home folder:
arecord -d 6 -f S16_LE -t wav -c 1 test.wav
If for some reason it does not return to the normal terminal prompt after 6 seconds, press Ctrl+C to abort.
Try that and see if the “distortion” problem occurs there.
Yeah, I see what you’re talking about. That’s really strange. I tried doing the test sample and the distortion is still there. It’s like a crackling sound or something. I tried reinstalling alsa, that didn’t change anything. On bass and treble in Audacity, I disabled level control and turned the bass way down and that seemed to make it better, but it didn’t go away.
Was that with “arecord”? If so, could you post that recording so I can see if the notches are “exactly” the same as you get with Audacity.