>> I would mix the project at 48k or 96k > > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best > sound quality. if that would only be true ... a) any non-linearity introduces harmonics, some non-linearities introduce an infinite amount of harmonics, which will cause foldover distortion. the large the sampling-rate, the lower the foldover. b) delay-lines have a higher precision at higher sampling-rates c) the tuning of digital filters is more precise at higher sampling-rates due to the frequency warping in the blt iir filters may have a higher quantization noise, but that is the reason, why a good filter implementation is done in double-precision. frankly, 48k may be a good enough for distribution, but it is sub-optimal not for production ... and it is horrible for digital synthesis. fwiw, for digital synthesis (non-standard or distortion synthesis) i ended up rendering my compositions at 3mhz ... which was a good compromise between computation time and sound quality. best, tim note on a: if your signal processor introduces the Nth harmonic, you have to upsample your signal by a factor of N. or apply a pre-filter on your signal by nyquist/N. question for the reader: in order to completely prevent foldover distortion, how much do you have to upsample for a tanh waveshaper (a processor that introduces infinite harmonics)? _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
