Impulse invariance

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 $|\Omega |\geq \pi /T$ ), 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 |\leq \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_{k}e^{s_{k}t}},&t\geq 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_{k}e^{s_{k}nT}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_{k}T}z^{-1}}}\,

Thus the poles from the continuous-time system function are translated to poles at z = e^s_kT. The zeros, if any, are not so simply mapped.^{[clarification needed]}

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 = s_k transform to poles in the discrete-time system at z = exp(s_kT), 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.^[1] 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_{k}e^{s_{k}nT}}\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_{k}T}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.

References

^ Jackson, L.B. (1 October 2000). "A correction to impulse invariance". IEEE Signal Processing Letters. 7 (10): 273–275. Bibcode:2000ISPL....7..273J. doi:10.1109/97.870677. ISSN 1070-9908.

Other sources

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

External links

Impulse Invariant Transform at CircuitDesign.info Brief explanation, an example, and application to Continuous Time Sigma Delta ADC's.

[1] Jackson, L.B. (1 October 2000). "A correction to impulse invariance". IEEE Signal Processing Letters. 7 (10): 273–275. Bibcode:2000ISPL....7..273J. doi:10.1109/97.870677. ISSN 1070-9908.

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v t e Digital signal processing
Theory	Detection theory Discrete signal Estimation theory Nyquist–Shannon sampling theorem
Sub-fields	Audio signal processing Digital image processing Speech processing Statistical signal processing
Techniques	Z-transform Advanced z-transform Matched Z-transform method Bilinear transform Constant-Q transform Discrete cosine transform (DCT) Discrete Fourier transform (DFT) Discrete-time Fourier transform (DTFT) Impulse invariance Integral transform Laplace transform Post's inversion formula Starred transform Zak transform
Sampling	Aliasing Anti-aliasing filter Downsampling Nyquist rate / frequency Oversampling Quantization Sampling rate Undersampling Upsampling