I want to compress voice data for transfer over
network,who could tell me about algorithm?
I want to compress voice data for transfer over
network,who could tell me about algorithm?
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The simplest approach would be to use an existing compression library (one that does MP3, for example). A decent "home-grown" method is to do an FFT on the data and filter out non-essential frequencies (outside of the range of voice, that is). I did that a while back and got pretty good results.Originally Posted by leetow2003
Could you tell me in detail?or some codes?Thank you.
Basically, what I did was take the FFT of a 16-bit audio stream and discarded non-voice frequencies, as well as frequencies with small amplitudes. Then I mapped the three components (frequency, amplitude, and phase) to 8-bit representations (so 3 bytes per spectral component). Finally, I compressed the result using arithmetic coding. The last step helped, but wasn't domain-specific enough to produce really spectacular compression; the correlation between each frame within the sample wasn't really exploited properly. For example, consider the ZIP archive format. It works really well with text because there is a direct correlation between each byte of data. But if you apply it to a buffer of 32-bit data, the efficiency typically goes down because the relationship between the data on a byte level is somewhat artificial. The key to achieving really good compression, then, is to model those inter-relationships as faithfully as possible. That said, the result was definitely satisfactory. Compression levels averaged out to roughly 10% - 33% of the uncompressed size, and the voice quality was decent. Had the second step been better modeled, I imagine those figures would have been half as much.
Last edited by gardhr; 12-23-2011 at 08:59 PM.
If I use 8-bit or 16-bit, what is the range about non-voice frequencies?
