I don’t know why it is doing this, but my last few audioblogs have all been encoded at 22 kHz sampling rates. Each of the individual tracks are 44 kHz, the project is set to 44 kHz, and the bit rate is 64 kpbs. When I encode it to MP3 with Audacity, it shows up as 22 kHz. From everything I can see in the docs, the project setting is what controls this. If I have it set at 44, that’s what the MP3 is. I tried setting the project to 48, but it still encoded at 22 kHz. This is really really annoying, and I can’t figure out why it is happening. If I export to to WAV, it stays sampled correctly. I tried exporting to WAV and then using command line lame to encode to MP3 and it sounded really funny and flangy. Audacity is using lame anyway, so somehow it is using settings that are better but I don’t know what those are.
Does anyone know of a workaround for this pain in the ass problem?
2 thoughts on “Audacity is Bugging Me”
Maybe the sampling shows up a 22 kbps because you’re recording in mono ? Could be.
Also- I think you mean kbps- kilo bits per second.
I think music shows should have higher sampling rates than voice. That said I think you’ve had decent quality.
Adam’s new setup don’t sound so good.. All the best- GREAT SHOW!
As ford says, you’re probably encoding in mono.
I suggest you to use Lame from the command line, as it gives you more control about the encoding process. I use something like this:
lame -m s –scale 1 -h -b 48 –cbr -c -p –tt “title –ta “artist” –tl “album” –ty year –clipdetect file.wav file.mp3
then look at what lame tells you, adjust –scale accordingly and do the final encoding.
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