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Think about I have an array of time and fundamental frequency of a given audio file. So what I need to do is extract the fundamental frequency from the audio. And what I am thinking is to apply a bandpass filter, for instance 100hz above and below to the given fundamental frequency corresponding to time.

Is it possible to do it? I know somehow it would be, but is there a known method that I can define the band pass filter parameters varying in time?

Thanks,

Baggio

katip_çelebi
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  • There is a class of adaptive filters, which self-adjusts their transfer function according to an optimization algorithm driven by an error signal. – divanov Mar 01 '14 at 06:21
  • Can you give me an example to how to implement? Think of I have a Time and F0 vector and apply a bandpass filter on those time intervals and around those F0's. – katip_çelebi Mar 01 '14 at 15:57
  • I mean I have a vector of time and fundamental frequency of a given audio file. SO I want to define a dynamic bandpass filter varying over time with the corresponding fundamental frequency. Is it possible to do it? By the way I have the signal processing package for matlab. – katip_çelebi Mar 01 '14 at 23:47
  • Adaptive filters are part of [DSP System Toolbox](http://www.mathworks.se/help/dsp/ref/adaptfilt.html) – divanov Mar 02 '14 at 10:54

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