... >> - *All* filtering you can do, either analog or digital, will >> inevitably have phase shifting as a consequence, no matter what people >> will try to tell you about correcting networks (check out the theory and >> preferably do your homework: ALWAYS is ALWAYS. > >wire? (or does that not count as a filter?)
:) Sure. There are things like wavelets with some suitable windowing (various kinds, complicated, in most cases theory prefers infinite sums/integrals) which can even do some sort of "orthogonal" filtering, probably also adding up (within some given accuracy limits) to "pass through" when done right. It is always a question of when you change a little thing (like compression of one or more bands) whether the result, which then doesn't anymore add up to a "wire", or simple N-sample delay, is something sensible. Also, many FIR filters sound not very natural because they don't implement the equivalent to reasonable accuracy of actual poles and zeros, causing in that interpretation signal distortion. And they may well be abused in the signal path for other purposes than the main filter function. Of course all kinds of compromises and accuracy fixes may be possible and useful.
Also, like with some per sample batch or averages based FFT-based filtering, very soon when not simply "reverse FFT-ing the ceptrums", the "straight" delay idea is left very far behind. Meaning that the actual filter convolution does a whole lot of stuff besides for instance an intended frequency/amplitude equalization.
Also, I meant by distortion, that the power sequence generated by for instance "analog equivalent" z-function function in the digital domain (with the same form as the jW or "s" transfer function in the Fourier domain), when reconstructed into an analog signal (after convolution with an input signal, say a test impulse) by either a close to perfect reconstruction, a "normal" DA converter oversampling filter, or just simple anit-aliasing, isn't going to be a perfect match in relevant cases.
Also, some filters would require the equivalent of "upsampling", which theoretically is an infinitely long filter, and to be done accurate requires in practice a pretty long (sinc-like) filter. Of course simple solutions can work a little magic, but distortion in the sense of harmonic, inter modulation and transient distortion is inevitable and quite audible, just like "imperfect reconstruction" in all normal DA (not AD) convertors.
I have the strong impression that in machines like the Kurzweil there's a compelling logic starting from the sample preparation, and ending with an overall machinery acting on the output signal which when the proper DSP and effects are used can make for pretty perfect output signals, but I have the impression it's complicated and works good only when all the steps are done right. Interesting though.
Also the DSP machinery in audio equipment can take into consideration that the human-perceived or electronically measured audio power resulting from using all kinds of DSP is kept limited, and made human-friendly, and that there is a well working warning mechanism when the limits of blasting audio waves into a listening space are reached. Many modernistic approaches appear to search for the opposite, unfortunately.
Theo V. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
