scispace - formally typeset
Search or ask a question

Showing papers on "Speech coding published in 1972"



Journal ArticleDOI
TL;DR: An audio response unit used in data communication services that is able to respond to more than 7000 speech segments of 1-s duration in terms of partial autocorrelation (PARCOR) coefficient.
Abstract: This paper describes an audio response unit used in data communication services. The speech segments necessary to respond are stored in a large capacity magnetic drum in terms of partial autocorrelation (PARCOR) coefficients and excitation source information. PARCOR coefficient is a new parameter to express accurately the spectrum envelope of speech signals. Multiple speech signals can be synthesized simultaneously by means of a timemultiplexed digital filter composed of a high-speed arithmetic unit. The unit is able to respond to more than 7000 speech segments of 1-s duration.

49 citations


Journal ArticleDOI
R. W. Schafer1
TL;DR: Some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer are discussed.
Abstract: Digital signal processing techniques are becoming increasingly important in speech analysis and synthesis. These techniques can be implemented using a general purpose computer facility (often not in real time), or special purpose hardware realizations can be constructed. This paper discusses some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer. The survey concentrates on those speech processing techniques relevant to the development of sensory aids for the deaf.

18 citations


Journal ArticleDOI
TL;DR: Use of optimum channel encoding was shown to reduce considerably the channel capacity required to obtain a given speech quality, and the effect of the overload level of the quantizer on the system signal-to-noise ratio was determined.
Abstract: Isopreference tests were used to study the subjective effects of channel transmission errors on computer-simulated PCM and previous-sample feedback DPCM voice systems. Numerically scaled isopreference contours were obtained and plotted on planes having as axes the number of quantization bits N and the bit error probability p of the binary symmetric transmission channel. For any N and p considered, the quality of DPCM speech was found to exceed that of PCM speech. With N fixed and p decreasing, the improvement of DPCM relative to PCM increased to the point where quantization noise limited the performance of both systems. Two nonredundant codes, natural binary and folded binary, were used to encode the quantizer output. The two codes yielded comparable speech quality. The maximum subjective quality obtainable for white Gaussian and Rayleigh fading channels was determined as a function of channel capacity. Use of optimum channel encoding was shown to reduce considerably the channel capacity required to obtain a given speech quality. The subjective ratings were compared with measured system signal-to-noise ratios. Finally, the effect of the overload level of the quantizer on the system signal-to-noise ratio was determined.

14 citations


08 Mar 1972
TL;DR: A detailed study of various voice differential PCM systems applicable for digitally encoding 3, 4, 5 and 6 voice channels into a 153.6 kilobit per second digital signal which can be used in a switched telecommunication network.
Abstract: : The report contains a detailed study of various voice differential PCM systems applicable for digitally encoding 3, 4, 5 and 6 voice channels into a 153.6 kilobit per second digital signal which can be used in a switched telecommunication network. Transmission of both speech and data signals over these voice channels is considered. Included in the report are: Calculations of the performance of differential PCM systems with various data signals; An analysis of the performance of differential PCM systems with speech signals; A description of a differential PCM system constructed using standard T1 technology. The research reported here consists of theoretical studies, computer simulation, subjective testing and hardware construction. (Author)

2 citations


01 Apr 1972
TL;DR: In this article, the authors classified source encoding techniques into sampling and analog to digital conversion, codebook techniques, predictive subtractive coding and delta modulation, along with aperature and partitioning techniques.
Abstract: : Data compression, in the paper, is used to denote the reducing of an input data set prior to transmission, as opposed to data reduction, the analytical processing of the set upon reception. Source encoding techniques are classified into sampling and analog to digital conversion, codebook techniques, predictive subtractive coding and delta modulation, along with aperature and partitioning techniques. Some of the most recent work discussed includes adaptive predictive coding, picture bandwidth compression-source coding, and a techniques for subjective performance measures. The paper concludes with an information theoretic approach to data compression. (Author)

1 citations


01 Jan 1972
TL;DR: Some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer are discussed.
Abstract: Digital signal processing techniques are becoming increasingly important in speech analysis and synthesis. These techniques can be implemented using a general purpose computer facility (often not in real time), or special purpose hardware realizations can be constructed. This paper discusses some recent work in speech processing including design of digital filter bank spectrum analyzers, homorphic analyzers of speech, predictive coding, and hardware realization of a digital formant synthesizer. The survey concentrates on those speech processing techniques relevant to the development of sensory aids for the deaf.

1 citations



04 Apr 1972
TL;DR: Much of the redundancy in a speech or television signal is eliminated when that signal is encoded into digital form by a differential PCM encoder, so the performance of a DPCM system without entropy coding with one using entropy coding is compared.
Abstract: : Much of the redundancy in a speech or television signal is eliminated when that signal is encoded into digital form by a differential PCM encoder. Further coding of the differential PCM output using entropy coding techniques (Huffman or Shannon-Fano coding) can result in a further increase in the signal to quantizing noise ratio of 5.6 dB without increasing the transmission rate. This conclusion is reached by comparing the performance of a DPCM system without entropy coding with one using entropy coding. The DPCM without entropy coding uses a minimum mean square error quantizer (Max quantizer) while the system with entropy coding uses an equilevel density function. This is consistent with previous experiments with speech and television signals. Gaussian quantizer inputs result in only a 2.8 dB improvement for entropy coding.

1 citations