Impulse invariance

{{More footnotes|date=April 2009}}

Impulse invariance is a technique for designing discrete-time infinite-impulse-response (IIR) filters from continuous-time filters in which the impulse response of the continuous-time system is sampled to produce the impulse response of the discrete-time system. The frequency response of the discrete-time system will be a sum of shifted copies of the frequency response of the continuous-time system; if the continuous-time system is approximately band-limited to a frequency less than the Nyquist frequency of the sampling, then the frequency response of the discrete-time system will be approximately equal to it for frequencies below the Nyquist frequency.

Discussion

The continuous-time system's impulse response, h_c(t), is sampled with sampling period T to produce the discrete-time system's impulse response, h[n].

:h[n]=Th_c(nT)\,

Thus, the frequency responses of the two systems are related by

:H(e^{j\omega}) = \frac{1}{T} \sum_{k=-\infty}^\infty{ TH_c\left(j\frac{\omega}{T} + j\frac{2{\pi}}{T}k\right)}\,

If the continuous time filter is approximately band-limited (i.e. H_c(j\Omega) < \delta when |\Omega| \ge \pi/T), then the frequency response of the discrete-time system will be approximately the continuous-time system's frequency response for frequencies below π radians per sample (below the Nyquist frequency 1/(2T) Hz):

:H(e^{j\omega}) = H_c(j\omega/T)\, for |\omega| \le \pi\,

=Comparison to the bilinear transform=

Note that aliasing will occur, including aliasing below the Nyquist frequency to the extent that the continuous-time filter's response is nonzero above that frequency. The bilinear transform is an alternative to impulse invariance that uses a different mapping that maps the continuous-time system's frequency response, out to infinite frequency, into the range of frequencies up to the Nyquist frequency in the discrete-time case, as opposed to mapping frequencies linearly with circular overlap as impulse invariance does.

=Effect on poles in system function=

If the continuous poles at s = s_k, the system function can be written in partial fraction expansion as

:H_c(s) = \sum_{k=1}^N{\frac{A_k}{s-s_k}}\,

Thus, using the inverse Laplace transform, the impulse response is

:h_c(t) = \begin{cases}

\sum_{k=1}^N{A_ke^{s_kt}}, & t \ge 0 \\

0, & \mbox{otherwise}

\end{cases}

The corresponding discrete-time system's impulse response is then defined as the following

:h[n] = Th_c(nT)\,

:h[n] = T \sum_{k=1}^N{A_ke^{s_knT}u[n]}\,

Performing a z-transform on the discrete-time impulse response produces the following discrete-time system function

:H(z) = T \sum_{k=1}^N{\frac{A_k}{1-e^{s_kT}z^{-1}}}\,

Thus the poles from the continuous-time system function are translated to poles at z = eskT. The zeros, if any, are not so simply mapped.{{clarify|date=February 2013}}

=Poles and zeros=

If the system function has zeros as well as poles, they can be mapped the same way, but the result is no longer an impulse invariance result: the discrete-time impulse response is not equal simply to samples of the continuous-time impulse response. This method is known as the matched Z-transform method, or pole–zero mapping.

=Stability and causality=

Since poles in the continuous-time system at s = sk transform to poles in the discrete-time system at z = exp(skT), poles in the left half of the s-plane map to inside the unit circle in the z-plane; so if the continuous-time filter is causal and stable, then the discrete-time filter will be causal and stable as well.

=Corrected formula=

When a causal continuous-time impulse response has a discontinuity at t=0, the expressions above are not consistent.{{Cite journal|title = A correction to impulse invariance|journal = IEEE Signal Processing Letters|date = 2000-10-01|issn = 1070-9908|pages = 273–275|volume = 7|issue = 10|doi = 10.1109/97.870677|first = L.B.|last = Jackson| bibcode=2000ISPL....7..273J }}

This is because h_c (0) has different right and left limits, and should really only contribute their average, half its right value h_c (0_+), to h[0].

Making this correction gives

:h[n] = T \left( h_c(nT) - \frac{1}{2} h_c(0_+)\delta [n] \right) \,

:h[n] = T \sum_{k=1}^N{A_ke^{s_knT}} \left( u[n] - \frac{1}{2} \delta[n] \right) \,

Performing a z-transform on the discrete-time impulse response produces the following discrete-time system function

:H(z) = T \sum_{k=1}^N{\frac{A_k}{1-e^{s_kT}z^{-1}} - \frac{T}{2} \sum_{k=1}^N A_k}.

The second sum is zero for filters without a discontinuity, which is why ignoring it is often safe.

See also

References

{{Reflist}}

=Other sources=

{{Refbegin}}

  • Oppenheim, Alan V. and Schafer, Ronald W. with Buck, John R. Discrete-Time Signal Processing. Second Edition. Upper Saddle River, New Jersey: Prentice-Hall, 1999.
  • Sahai, Anant. Course Lecture. Electrical Engineering 123: Digital Signal Processing. University of California, Berkeley. 5 April 2007.
  • Eitelberg, Ed. "Convolution Invariance and Corrected Impulse Invariance." Signal Processing, Vol. 86, Issue 5, pp. 1116–1120. 2006