[Guy Sherman] > Would the approach to use a sample-rate converter to essentially interpolate > samples, then do the processing, and then sample back down?
The principle is indeed the same, and you could use a converter library for this purpose. However, those converters are designed to work over a continuous range of samplerate ratios whereas the ratio chosen for oversampling is usually a fixed integer because this presents ample opportunity for optimisation. The interpolation filters in both cases are usually windowed sinc FIR (much like the Lanczos kernel in image resampling). > How does that work for live streams of data? As you intuit: you sample up, process, then sample back down, ending up with one output sample for every input sample. IIrc, http://quitte.de/dsp/caps.html contains at least two oversampled plugins and comes with source code. Cheers, Tim _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-dev