This tutorial describes the use of filtering techniques in Praat. It assumes you are familiar with the Intro.
Modern computer techniques make possible an especially simple batch filtering method: multiplying the complex spectrum in the frequency domain by any real-valued filter function. This leads to a zero phase shift for each frequency component. The impulse response is symmetric in the time domain, which also means that the filter is acausal: the filtered signal will show components before they start in the original.
Some very fast Infinite Impulse Response (IIR) filters can be defined in the time domain. These include recursive all-pole filters and pre-emphasis. These filters are causal but have non-zero phase shifts. There are versions that create new Sound objects:
And there are in-line versions, which modify the existing Sound objects:
A Finite Impulse Response (FIR) filter can be described as a sampled sound. Filtering with such a filter amounts to a convolution of the original sound and the filter:
Described in the Source-filter synthesis tutorial:
© ppgb, March 24, 2010