A Study of Time-Domain Speech Compression by Means of a New Analog Speech Processor
Time-domain speech compression using the SDA (sample, discard, abut) procedure at compression ratios of 0.25 to 0.75 is studied by means of a new analog speech processor and minicomputer algorithms. Fourier transform methods have been used to establish a correspondence between the quality of the reconstructed compressed speech waveforms and the subjective recognition of compressed speech. The result of two psychoacoustic experiments indicate that 1) the interruption frequency should be equal to the pitch frequency of the voice waveform for optimum recognition of the compressed speech, and 2) smoothing of the discontinuities with electronic techniques significantly improves the recognition of the compressed speech. The optimum smoothing parameters, window width and characteristic function, are also obtained from this study.
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