[linux-audio-user] Pulling off old 2-track reels and trying to slow them down - SOLVED

Alan Taylor alan-linuxaudiouser at lowb.org
Sat Apr 30 18:03:26 EDT 2005


On Mon, 2005-04-25 at 17:38 -0400, Jesse Chappell wrote:
> If you can use ardour or alsaplayer to estimate the exact amount
> of resampling you need, you can then use:
> 
>      sndfile-resample -by <amount_ratio> infile.wav  outfile.wav
> 
> to do a high quality resampling.  You'll want to use the inverse
> of the varispeed you find to get the correct resampling.  The
> resulting file will have a weird sample rate, but if played back
> at the original sample rate of your capture will sound correct.
> 
> You can replace the samplerate field in the output wav file with
> the attached python script.  Run it as follows (example assumes
> a 48000 original capture):
> 
>    python wavsrmod.py 48000 outfile.wav


The above method resulted in a file that still sounded way too fast.
I'm not sure what I did wrong.  Knowing that I wanted to slow down my
recording to 25% of it's orginal speed, I tried using your suggestion of
the inverse "-by .750", I tried the normal representation of 25%
"-by .250", and I tried going directly to 25% speed with "-to 12000".
All resulted in a wav file that was too fast, and got even faster after
running your python script.

After messing around quite a bit, I tried doing the steps in reverse
order.  I ran your python script directly on my Ardour-exported wav file
"python wavsrmod.py 12000 reel1.wav".  That resulted in a 12KHz file
that was playable by mplayer at the correct sounding speed (but XMMS
doesn't like to play 12KHz files it seems).  Then, I ran
sndfile-resample to change it back to 48KHz.  The end result is the best
quality conversion thus far.

Thanks for all your help!



Alan





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