30 Apr
2005
30 Apr
'05
8:03 p.m.
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