Showing papers on "Linear predictive coding published in 1971"

Predictive coding in a homomorphic vocoder

Abstract: Application of a type of predictive coding to the channel signals of a homomorphic vocoder has produced sizable bit rate reductions. With only slight degradation in speech quality, reduction (for the spectral envelope information) from 7800 to 4000 bits/s was achieved. A technique for obtaining the formant frequencies from the predictive coding parameters is described; this approach promises further bit rate reductions. As a by-product of this study of predictive coding, direct and cascade form speech synthesizers are compared on the basis of differing quantization effects.

Some Applications of a Small Digital Computer in Speech Processing

Abstract: An increase in the rate and the intelligibility of sound is highly desirable in speech communication. Also, it is useful to have an accurate and efficient method of obtaining desired segments of a speech sample. In this paper, the use of a small digital computer in processing the speech signal to achieve the above purposes is described with illustrations. On‐line simulation of the method of Fairbanks et al. [G. Fairbanks et al., IRE Trans. Audio 2, 7–12, (1954)] of increasing the speech rate has been achieved with flexible speed‐up ratios and sampling intervals. Increase of intelligibility in speech signals by converting them into dichotic signals with an interaural time delay is discussed. These dichotic signals have been obtained from the computer for time delays between 0 and 1 sec. To obtain different segments of a speech sample, the computer is programmed to store the speech sample and display its waveform on an oscilloscope, so that various segments of the speech sample can be extracted and also joi...

Low‐Bit‐Rate Speech Transmission by Linear Prediction of Speech Signals

Abstract: A method of encoding speech for transmission at low bit rates is described. At the transmitter, the current sample of the speech wave digitized at 10 kHz is predicted as a linear combination of the 12 previous samples. The optimum linear combination is determined by minimizing the mean‐squared error between the actual and the predicted values of the speech samples. The pitch period and a binary voiced‐unvoiced parameter are determined by performing a short‐time autocorrelation analysis of the speech wave. Fifteen parameters, namely, the 12 predictor coefficients, the pitch period, the rms value of the speech signal, and the binary voiced‐unvoiced parameter are encoded into 72‐bit frames and transmitted to the receiver at uniform intervals. Different transmission rates are obtained by varying the interval between adjacent frames. At the receiver, the decoded transmission parameters are used to control a speech synthesizer consisting of a linear recursive filter excited by a suitable combination of quasiper...

Quality of Speech Reconstituted from Reduced Spectral Data

Abstract: Characterizing speech spectra by means of statistical procedures reduces the amount of information needed to specify the data. When eigenvectors of the covariance matrix of short‐time digital spectral data are chosen to represent the speech spectra at an average accuracy of 90%, continuous speech can be characterized by seven dimensions. However, a proper test of the adequacy of such a data reduction scheme involves perceptual studies of the reconstituted speech signals. In order to establish the relationship between statistical accuracy and perceptual intelligibility, a computer program to reconstruct the speech signal from spectral data has been prepared. Preliminary listening tests to judge the qualities of speech signals reconstituted from reduced data have been made. The experiments and results are demonstrated and discussed.