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- From: mek@guinan.psu.edu (Mark E. Kotanchek)
- Subject: Re: Imaginary signals
- Message-ID: <0931Hfqw_b@atlantis.psu.edu>
- Sender: news@atlantis.psu.edu (Usenet)
- Organization: Penn State Center for Academic Computing
- References: <TIM.93Jan10124435@ear-ache.mit.edu>
- Date: Tue, 12 Jan 93 15:59:28 GMT
- Lines: 66
-
- In article <TIM.93Jan10124435@ear-ache.mit.edu> tim@ear-ache.mit.edu (Tim
- Wilson) writes:
- > 880840m@axe.acadiau.ca (MICHAEL ALEXANDER MCKAY) writes:
- >
- > Is there a time-domain filter which, given a real signal as input,
- will
- > output the corresponding imaginary signal?
- > I've tried to make a filter with unity gain and pi/2 phase shift for
- > all frequencies, but with no luck.
- >
- > See Oppenheim and Shafer, _Digital Signal Processing_ (1st edition),
- > Section 7.4: "Hilbert Transform Relations for Complex Sequences." In
- > particular, see Section 7.4.1, "Design of Hiblert Transformers."
- > I quote: "...the ideal Hibert transformer or 90-degree phase shifter
- > takes its place alongside the ideal lowpass filter and ideal
- > bandlimited differentiator as valuable theoretical concepts which
- > correspond to noncausal systems and for which the system function
- > exists only in a restricted sense."
- >
- > In the same section, O&S discuss finite-duration approximations of the
- > ideal Hilbert transformer using frequency-sampling, windowing, and
- > equiripple approximations. Finally, they reference B. Gold, A. V.
- > Oppenheim, and C. M. Rader, "Theory and Implementation of the Discrete
- > Hibert Transform," _Proc. Symp. Computer Processing in
- > Communications_, Vol. 19, Polytechnic Press, 1970, New York for a
- > recursive implementation of a 90-degree phase-splitting system.
- >
- > Good luck.
- > --
- > Tim Wilson
- > Internet: tim@ear-ache.mit.edu
- > UUCP: mit-eddie!mit-athena!tim
-
- The Hilbert filter approach is the "correct" way to derive an analytic
- equivalent to a real-valued signal. However, if you can assume that your
- PASSBAND signal is also narrowband, there is an alternative. Assume that
- we have a passband centered at a frequency fc having a bandwidth BW and
- that the signal has been sampled at a frequency fS. Furthermore, the
- narrowband assumption implies that fc >> BW. Then, given the data stream
- s(k), we would:
-
- 1) Rotate the upper sideband to DC via
-
- y(k) = s(k)*exp(-j 2 pi fc k/fS)
-
- 2) Apply a low-pass filter to chop out the lower sideband (which is now
- centered on -2fc) and you have an basebanded analytic equivalent of your
- input signal. Make that low-pass filter of the decimating FIR type and you
- are probably picking up significant computational efficiencies.
-
- This techniques is approximately a digital equivalent of the quadrature
- sampling approach and is valid whenever quadrature sampling would be
- appropriate. (However, these days a digital implementation takes up less
- space than the analog so for embedded systems, the digital is generally
- preferable.)
-
- Hope this is of some use,
-
- Mark.
- --
- Mark Kotanchek
- Guidance & Control Dept - 363 ASB
- Applied Research Lab/Penn State
- P.O. Box 30
- State College, PA 16804
-
-