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![]() Ashhar Farhan wrote: you don't really have to know a lot of DSP to play around with this particular beast. very simply, you collect the audio samples in a first-in first-out buffer of about 250 slots. Each time a new sample is added at one end, a sample is retired at the other end. each of the 90 degree phase shift-ed samples is generated by simpy multiplying all the samples in the pipe with a individual 'magic' constants and adding them all up together. pretty basic stuff as far as programming goes. the magic constants are themselves quite complex to calculated, but that work has alread been done for you. The CD accompanying EMRFD has those constants in a text file under the DSP folder. Does this approximate the Hilbert Transform? Alan |
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