Difference between revisions of "Contributions:HilbertFilter"
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Revision as of 20:19, 20 March 2012
This filter computes the envelope or the phase of a signal using Hilbert transform. The discrete input signal x(n) is first transformed to its analytic representation (i.e., analytic signal), which is composed of real and imaginary parts.
The real part is the same input signal, and the imaginary part is the Hilbert transform of the input signal. The Hilbert transform is implemented as the convolution of the input signal with the filter h(n).
To get an ideal Hilbert transform, n must be infinitely long . However, for real time implementations, h(n) must be truncated and delayed to guarantee a causal filter. Thus, the FIR filter is defined as
for , where N must be an odd number representing the length of the filter. The resulting Hilbert transform is delayed by implying that the real part of the analytic signal must be delayed by the same amount too in order to estimate the envelope and phase of the input signal.
Not yet released: will be released at http://www.bci2000.org/svn/trunk/src/contrib/SignalProcessing/HilbertSignalProcessing/HilbertFilter.cpp
Cristhian Potes, Jeremy Hill
Source Code Revisions
- Initial development:
- Tested under:
- Known to compile under:
- Broken since:
This parameter may be one of
- 0 - Copy input signal
- no processing,
- 1 - Magnitude
- Hilbert envelope amplitude,
- 2 - Phase
- Hilbert phase,
- 3 - Real part
- original input signal, but with a delay to match its timing to the imaginary part.
- 4 - Imaginary part
- original signal filtered with an FIR-Hilbert transformer.
The delay of the Hilbert transform can be calculated as , where N is an odd number representing the length of the filter.