Is this Robin Whittle of Devilfish fame? I bought a Devilfish from you back in the mid-1990s. Best mod ever!
> On Aug 16, 2015, at 8:07 PM, Robin Whittle <r...@firstpr.com.au> wrote: > > Hi Shannon, > > If the number of reads from the delay line per sample cycle is high > enough, as a less expensive alternative to the most obvious solution > (higher order "interpolation" based on multiple samples before and > after, with some fancy set of coefficients calculated on the spot, or > looked up from a table of sufficiently high resolution, depending on the > fraction of a sample delay involved) you might like to consider > upsampling the input signal to twice the normal rate, and then doing > simple linear interpolation. > > This would not be mathematically perfect, since the high frequency > response would be slightly reduced if the delay fraction was 0.25 or > 0.75, whereas it would be flat for 0, and as flat as the upsampling > algorithm for 0.5 (I recall that the upsampling algorithm produces the > odd numbered samples in the final output, with the even ones being the > input samples). However, assuming the sampling rate is 44.1kHz, 48kHz > or higher, I think this slight variation is unlikely to be perceptible > to human ears. > > Robin > > > On 2015-08-16 6:09 PM, Sham Beam wrote: >> Hi, >> >> Is it possible to use a filter to compensate for high frequency signal >> loss due to interpolation? For example linear or hermite interpolation. >> >> Are there any papers that detail what such a filter might look like? >> >> >> Thanks >> Shannon >> _______________________________________________ >> music-dsp mailing list >> music-dsp@music.columbia.edu >> https://lists.columbia.edu/mailman/listinfo/music-dsp > _______________________________________________ > music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp
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