Re: [MLUG] MP3 problem using sox & lame

[yoda <EMAIL:PROTECTED>]

Well, one way it would seem to me that it would work (though I'm just guessing)
is to convert it into a raw PCM format (either truly raw data, or .wav, etc.),
and then import it back into sox, forcing an option that will ccounteract it.
For example, the sample is playing twice as fast as it should, right? If it's
stereo, force it to mono (thus, what previously were right and left simultaneous
samples become mono samples in sequence, which would give you twice as many
samples per byte). Similarly, if it's sixteen bit, force it to eight.

I'm not sure what effects this will have since it's been converted to a lossy
format (mp3), but I think it's on the right track.

------------------------------------------------------
It's GNU/Linux damnit.
http://www.gnu.org/gnu/linux-and-gnu.html

On Wed, 3 Sep 2003, Walt Joyce wrote:

> I recording a conference call using sox & lame but made
> a mistake in the parameters. I want to know if I can
> convert the file (which plays at double the normal
> speed) into something useful (that one can listen to).
>
> The mistake I made was to specify 1 channel ("-c 1")
> instead of 2 channels ("-c 2"). Here is the command
> line I used:
>
> sox -t ossdsp -w -c -r 44100 -c 1 /dev/dsp -t raw -|lame -x -m s - ./testfile.mp3
>
> Is there any way to convert the result to normal
> speed, as if I typed:
>
> sox -t ossdsp -w -c -r 44100 -c 2 /dev/dsp -t raw -|lame -x -m s - ./testfile.mp3
>
> ??
>
>
> TIA.
>
> - Walt Joyce
> EMAIL:PROTECTED
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