| Signal Processing Toolbox™ | ![]() |
Impulse invariance method for analog-to-digital filter conversion
[bz,az] = impinvar(b,a,fs)
[bz,az] = impinvar(b,a,fs,tol)
[bz,az] = impinvar(b,a,fs) creates a digital filter with numerator and denominator coefficients bz and az, respectively, whose impulse response is equal to the impulse response of the analog filter with coefficients b and a, scaled by 1/fs. If you leave out the argument fs, or specify fs as the empty vector [], it takes the default value of 1 Hz.
[bz,az] = impinvar(b,a,fs,tol) uses the tolerance specified by tol to determine whether poles are repeated. A larger tolerance increases the likelihood that impinvar interprets closely located poles as multiplicities (repeated ones). The default is 0.001, or 0.1% of a pole's magnitude. Note that the accuracy of the pole values is still limited to the accuracy obtainable by the roots function.
Convert an analog lowpass filter to a digital filter using impinvar with a sampling frequency of 10 Hz:
[b,a] = butter(4,0.3,'s');
[bz,az] = impinvar(b,a,10)
bz =
1.0e-006 *
-0.0000 0.1324 0.5192 0.1273 0
az =
1.0000 -3.9216 5.7679 -3.7709 0.9246
Illustrate the relationship between analog and digital impulse responses [2].
Note This example requires the impulse function from Control System Toolbox™ software. |
The steps used in this example are:
Use impinvar with a sampling frequency Fs of 10 Hz to scale the coefficients by 1/Fs. This compensates for the gain that will be introduced in Step 4 below.
Use Control System Toolbox impulse function to plot the continuous-time unit impulse response of an LTI system.
Plot the digital impulse response, multiplying the numerator by a constant (Fs) to compensate for the 1/Fs gain introduced in the impulse response of the derived digital filter.
[b,a] = butter(4,0.3,'s'); [bz,az] = impinvar(b,a,10); sys = tf(b,a); impulse(sys); hold on; impz(10*bz,az,[],10);
Zooming the resulting plot shows that the analog and digital impulse responses are the same.

impinvar performs the impulse-invariant method of analog-to-digital transfer function conversion discussed in reference [1]:
It finds the partial fraction expansion of the system represented by b and a.
It finds the transfer function coefficients of the system from the residues from step 1 and the poles from step 2.
[1] Parks, T.W., and C.S. Burrus, Digital Filter Design, John Wiley & Sons, 1987, pp.206-209.
[2] Antoniou, Andreas, Digital Filters, McGraw Hill, Inc, 1993, pp.221-224.
bilinear, lp2bp, lp2bs, lp2hp, lp2lp
![]() | ifft2 | impz | ![]() |
| © 1984-2008- The MathWorks, Inc. - Site Help - Patents - Trademarks - Privacy Policy - Preventing Piracy - RSS |