Transfer Function Models
Sampling the Impulse Response
Impulse Invariant MethodSearch Physical Audio Signal Processing
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The impulse-invariant method converts analog filter transfer
functions to digital filter transfer functions in such a way that the
impulse response is the same (invariant) at the sampling
instants [353], [371, pp.
216-219]. Thus, if
denotes the
impulse-response of an analog (continuous-time) filter, then the
digital (discrete-time) filter given by the impulse-invariant method
will have impulse response
, where
denotes the
sampling interval in seconds. Moreover, the order of the filter is
preserved, and IIR analog filters map to IIR digital filters.
However, the digital filter's frequency response is an aliased
version of the analog filter's frequency
response.R.3
To derive the impulse-invariant method, we begin with the analog transfer function