Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320 samples). I would like to convert this to a signal sampled with a varying sampling rate, for example, starting at 16 kHz, going down to 4 kHz at the end. This would involve a time-variant anti-alias-filter and non-uniform sub-sampling. Is there a standard method for doing this? Best regards, Andre
non-uniform-sampling
Started by ●September 10, 2007
Reply by ●September 10, 20072007-09-10
The most straightforward approach is IMO to reconstruct the continuous-time waveform through a lowpass filter, and evaluate it at the desired sampling instants: Every input sample represents a sin(x)/x pulse, centered around the input sample. That can be evaluated at the desired location of the output sample. In a practical implementation it comes down to design a suitable lowpass filter with a shorter impulse response. Search for "polyphase filter" or "Farrow interpolation". There was a similar thread recently on comp.dsp. Cheers Markus PS: If it is possible to process the whole signal at once via FFT, it's easy to evaluate the resulting sin() and cos() terms at any point in time.