Maybe a side remark, interesting nevertheless: the filtering in digital domain, as compared with the analog good ol' electronics filters isn't the same in any of the important interpretations of sampled signals being put on any regular digital to analog converter, by and large regardless of the sampled data and it's known properties offered to the digital filter.
So, reconstructing the digital simulation of an analog filter into a electronic signal through either a (theoretically , or near-) perfect reconstruction DAC or an ordinary DAC with any of the widely used limited-time interval over sampled FIR or IIR simplified "reconstruction" filters, isn't going to yield a perfect equivalent of a normal, phase shift based electronics (or mechanics based) filter. Maybe unfortunately, but it's only an approximation, and no theoretically pleasing sounding mathematical derivation of filter properties is going to change that. It is possible to construct digital signals, where givens are hard-known about the signal which given a certain DAC will 'reconstruct' or simply result in an output signal which approaches a certain engineered ideal to any degree of accuracy. In general though, the signal between samples can only be known through perfect reconstruction filtering (taking infinite time and resources), and DACs that are used in studio and consumer equipment should be thoroughly signal prepared by pre-conditioning the digital signal feeding it such that it's very limited reconstruction filtering is used such that certain output signal ideals are approximated to the required degree of accuracy. Including even a modest filter in that picture isn't easy! Theo V. _______________________________________________ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
