Alan Peake wrote in message . ..
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

It does if the "magic constants" are right. Consider
the complex frequency response of a Hilbert Transformer
and work out the impulse response. Unlike linear-phase
FIR filters, the impulse response of a Hilbert
Transformer isn't symmetric.

There are whole books on the subject. I have a copy
of _Hilbert Transforms in Signal Processing_ by Hahn,
which goes in to all of this in more than a little
detail. The challenge is turning the theoretical
impulse response (which is infinite) in to something
you can realize on finite hardware.

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