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
