I read bunch of materials for extracting feature from audio signal and they all tell me to break signal into segments, why don't we analyze all the audio signal? I don't know what are the advantages of doing that and how wide a segment should be? I only see 256 samples per frame or 512 samples per frame... what about 1028 per frame?
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Analyzing signals per segments, with proper windowing, is a way to cope with non-stationary in audio samples. With full-size analysis, features can get mixed. Segment-splitting is thus at play in many algorithms (mp3, shazam).
The length of window is often a matter of trade-offs, between data information and computing advantages:
- signal sampling (window length is quite meaningless without sampling rate), with respect to the following:
- analyzing or extracting informational content from the signal: various ranges of stationarity may exist in the data, or generally useful processing features,
- easiness in computing: the power-of-two length you mention can be beneficial (faster algorithms like in the FFT), parallel computing, dedicated hardware, closer to real-time analysis.
answered Jul 5, 2020 at 15:40
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$\begingroup$ thank you a lot. Is that all the advantages? $\endgroup$– MrnobodyCommented Jul 5, 2020 at 15:52
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1$\begingroup$ This covers a lot, and you can unfold the three items. For instance, in computing, you can build dedicated hardware, use parallel processing, etc. $\endgroup$ Commented Jul 5, 2020 at 16:03
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2$\begingroup$ @user51307: And of course there's the issue of quasi-real-time processing. If you have to wait for the signal to be complete before you start processing you can't do real-time processing. Think of speech enhancement: in many applications you want to do that in quasi-real-time, and not in an offline mode after you've recorded the complete signal. $\endgroup$– Matt L.Commented Jul 5, 2020 at 16:15
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1$\begingroup$ Matt L. did emphasizes a real computing issue. I've heard that what made Skype much better adopted (several times ago) was it's reduced lag, with respect to the bandwidth available at that time $\endgroup$ Commented Jul 5, 2020 at 16:22
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1$\begingroup$ Yep, and we mustn't forget that these short frames (20 ms) that we find even in less latency-bound standards like MPEG-1 Audio Layer II as used e.g. by the Digital Audio Broadcasting (DAB) standard is also owed to a time where a couple milliseconds of audio posed a serious memory buffer requirement for low-cost decoder equipment. $\endgroup$ Commented Jul 5, 2020 at 16:42