>Assume you have a Nyquist frequency square wave: 1, -1, 1, -1, 1, -1, 1, -1...
The sampling theorem requires that all frequencies be *below* the Nyquist frequency. Sampling signals at exactly the Nyquist frequency is an edge case that sort-of works in some limited special cases, but there is no expectation that digital processing of such a signal is going to work properly in general. But even given that, the interpolator outputting the zero signal in that case is exactly correct. That's what you would have gotten if you'd sampled the same sine wave (*not* square wave - that would imply frequencies above Nyquist) with a half-sample offset from the 1, -1, 1, -1, ... case. The incorrect behavior arises when you try to go in the other direction (i.e., apply a second half-sample delay), and you still get only DC. But, again, that doesn't really say anything about interpolation. It just says that you sampled the signal improperly in the first place, and so digital processing can't be relied upon to work appropriately. E On Tue, Aug 18, 2015 at 1:40 AM, Peter S <peter.schoffhau...@gmail.com> wrote: > On 18/08/2015, Nigel Redmon <earle...@earlevel.com> wrote: > >> > >> well, if it's linear interpolation and your fractional delay slowly > sweeps > >> from 0 to 1/2 sample, i think you may very well hear a LPF start to kick > >> in. something like -7.8 dB at Nyquist. no, that's not right. it's > -inf > >> dB at Nyquist. pretty serious LPF to just slide into. > > > > Right the first time, -7.8 dB at the Nyquist frequency, -inf at the > sampling > > frequency. No? > > -Inf at Nyquist when you're halfway between two samples. > > Assume you have a Nyquist frequency square wave: 1, -1, 1, -1, 1, -1, 1, > -1... > After interpolating with fraction=0.5, it becomes a constant signal > 0,0,0,0,0,0,0... > (because (-1+1)/2 = 0) > _______________________________________________ > music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp >
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