Abstract: A method of speech processing includes receiving at least two separate, but temporally overlapping speech waveforms in real time; extracting a pitch waveform from each of the speech waveforms by segmenting the speech waveform pitch synchronously, fixing an analysis window size, and analyzing the speech waveform in real time; concatenating each pitch waveform by interpolating the pitch waveform at each pitch epoch, synthesizing a synthesis window size according to a desired speech playback speed, and generating a synthesized output pitch waveform; queuing each of the output waveforms so as to sequence each speech waveform serially one after the other such that the waveforms are mutually separated upon playback; and outputting each of the queued output waveforms to a selected playback device.

Abstract: A system that synchronously segments a speech waveform using pitch period and a center of the pitch waveform. The pitch waveform center is determined by finding a local minimum of a centroid histogram waveform of the low-pass filtered speech waveform for one pitch period. The speech waveform can then be represented by one or more of such pitch waveforms or segments during speech compression, reconstruction or synthesis. The pitch waveform can be modified by frequency enhancement/filtering, waveform stretching/shrinking in speech synthesis or speech disguise. The utterance rate can also be controlled to speed up or slow down the speech.

Abstract: A voice communication processing system and method for processing a speech waveform as a digital bit stream having a reduced number of bits representing speech parameters. The bit representation of amplitude parameters is reduced by storing only probable amplitude parameter transitions corresponding to amplitude parameter indices in an amplitude table and by joint encoding the amplitude parameter indices over multiple frames. The bit representation of the pitch period is reduced by storing a range of pitch periods in a pitch table and by joint encoding pitch period indices corresponding to an average pitch period over two frames. The bit representation of the vocal tract filter coefficients is reduced by storing only probable filter coefficient transitions corresponding to filter coefficient indices in a filter coefficient table and by joint encoding the filter coefficient indices over two frames.

Abstract: Bandwidth compression of speech signals is done by aliasing the fricativeund signals (4-8 KHz) into the voiced-sound signal band for transmission (0-4 KHz). To minimize distortion, aliasing (foldover) is done only when the fricative band power is more than twice the voice band power.

Abstract: To provide alternative operation at 2.4 or 9.6 Kb/s bit rates, a multirate igital processor generates a data frame of 216 bits representing sync and voice encoded at both rates, with the 2.4 Kb/s embedded as a subset within the frame. Rate conversion is made by inserting or deleting 162 bits per frame. Speech signal is transformed by linear prediction into coefficients and residual signals. The coefficients are transmitted similarly at both 2.4 and 9.6 Kb/s rates. However, the residual is processed into a simple pitch/hiss excitation signal for 2.4 Kb/s, but is processed into a Fourier transform of its baseband (250-1500 Hz) for 9.6 Kb/s excitation signal transmission.