rap@dana.UUCP (Rob Peck) (06/03/87)
I am interested in finding some kind of data compression algorithm that is suitable for compressing speech data. As I understand it, human speech has a great deal of redundancy to it, i.e. repetitions of virtually the same waveforms over a period of time, as well as slow changes in many cases from one waveform to the next. However, if one takes a set of audio samples of a spoken word, the samples will not fall in the right spots to show up any such redundancy. Thus, for a simplistic compression algorithm that looks for repeated sequences, no opportunity to compress would be noticed. Could someone point me to the appropriate literature? Or is there some public domain source code that is already available for this? The code needn't be fast on the analysis and compression. On playback, it should be pretty easy to expand, though. That is, play so many repetitions of this waveform at this sampling rate, then do this next one (or better still, adjust the current waveform until it looks like this new one, as a slewing to the new output... that'd be neat). I've read a little about FFT's, but once calculated, I have no idea how to use it or if it gives me remotely what I am looking for here. Please EMAIL directly to me. I will summarize any interesting responses to the Net. Thanks very much. Rob Peck ...ihnp4!hplabs!dana!rap