On Sun, 24 Apr 2005, Alan Taylor wrote:
On Sun, 2005-04-24 at 19:16 +0200, Wolfgang Woehl
wrote:
Alsaplayer sounds crap in varispeed. I'd use
Ardour for slowing things
down as it uses libsamplerate for that purpose. In the Mixer strip,
You are correct. While the results are still not ideal (due to the
original sample rate of 48KHz), the overall quality surpasses what I got
out of alsaplayer. Thanks.
Actually, for the real-time varispeed, ardour does *not* use
libsamplerate, but a handcrafted linear interpolation from Steve
Harris. This is also unsuitable for high quality archival work.
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
As mentioned by others, the higher capture rate you can achieve
the better quality you'll get, but depending on the source, 48k
might be good enough. Hope this helps.
jlc