A High Performance Processor design is described which is employed in the study of digital signal processing techniques. The processor is seen to enhance algorithm development as a high speed peripheral to a general p...
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A High Performance Processor design is described which is employed in the study of digital signal processing techniques. The processor is seen to enhance algorithm development as a high speed peripheral to a general purpose minicomputer facility. In addition, it can operate in a stand-alone mode and has been used to implement a real-time linearpredictive vocoder system.
Selection of the appropriate orders of the numerators and denominators for specified lowpass infinite impulse response (IIR) digital filter design criteria can be difficult. REcent work by H. L. Gray, et. al. suggests...
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Selection of the appropriate orders of the numerators and denominators for specified lowpass infinite impulse response (IIR) digital filter design criteria can be difficult. REcent work by H. L. Gray, et. al. suggests a general method by which orders may be selected prior to filter design. In this note, infinite impulse response lowpass digital filters are designed from finite impulse response (FIR) lowpass equivalents, wherein the Gray technique is used to select the order before design. On determination of the order, the IIR denominator is determined from the autocorrelation of the FIR and the numerator is obtained using the minimum phase Hilbert transform relationship.
With rare exception, all presently available narrowband speech coding systems implement scalar quantization (independent quantization) of the transmission parameters (such as reflection coefficients or transformed ref...
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With rare exception, all presently available narrowband speech coding systems implement scalar quantization (independent quantization) of the transmission parameters (such as reflection coefficients or transformed reflection coefficients in LPC systems). In this paper a new approach called Vector Quantizatlon is presented. For very low data rates, realistic experiments have shown that vector quantization can achieve a given level of average distortion with fifteen to twenty fewer bits per frame than that required for optimized scalar quantizing approachs presently in use.
In this study, predictivecoding of area functions for data rate reduction of LPC coded speech is examined. The properties of several predictors are described, and the benefit obtained from use of the mutual informati...
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In this study, predictivecoding of area functions for data rate reduction of LPC coded speech is examined. The properties of several predictors are described, and the benefit obtained from use of the mutual information among the areas is assessed. coding methods examined include differential coding and two-dinmensional prediction. This experimental study uses an objective speech quality measure to aid in optimal bit allocation among the parameters.
The paper emphasizes the importance of normalization of parameters in identification of linear models as now commonly applied to digital signal processing. Classical LPC, Pisarenko, Prony methods are unified and compa...
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The paper emphasizes the importance of normalization of parameters in identification of linear models as now commonly applied to digital signal processing. Classical LPC, Pisarenko, Prony methods are unified and compared. The factorial approach plays a central role when additive noise is considered. The computational requirement is the determination of eigen vectors of correlation and covariance matrices. Various algorithms are then given including sequential estimation procedures in the covariance case. The methods are compared on close sinewaves merged in noise in terms of resolution, windowing, signal-to-noise ratio.
This paper describes in general the three main approaches to speech synthesis. The details of Waveform digitization with unique compression algorithms are described. A system for implementing this technique has been d...
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This paper describes in general the three main approaches to speech synthesis. The details of Waveform digitization with unique compression algorithms are described. A system for implementing this technique has been developed and realized in software and silicon. The procedure for generating ROM code and key circuit elements of the control chip are presented.
A connected digit recognizer is proposed in which a set of isolated word templates is used as reference patterns and an unconstrained dynamic time warping algorithm is used to literally "spot" the digits in ...
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A connected digit recognizer is proposed in which a set of isolated word templates is used as reference patterns and an unconstrained dynamic time warping algorithm is used to literally "spot" the digits in the string. Segmentation boundaries between digits are obtained as the termination point of the dynamic path from the previous time warp. A region around the boundary is searched for the optimum starting point for the succeeding digit. At each stage the recognizer keeps track of a set of candidate digit strings for each test string. The string with the smallest accumulated distance is used as the preliminary string estimate. For variable length digit strings of from 2 to 5 digits (where the recognizer was not told the length of the string), word error rates of about 2-3% and string error rates on the order of 8% were obtained for both speaker dependent and speaker independent systems.
We present an improved 4800 bps LPC vocoder system that virtually eliminates the buzzy effect from synthetic speech. Excitation signal in the new system is formed by adding high-pass filtered pitch pulses or random no...
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We present an improved 4800 bps LPC vocoder system that virtually eliminates the buzzy effect from synthetic speech. Excitation signal in the new system is formed by adding high-pass filtered pitch pulses or random noise to a baseband residual signal (0 - 600 Hz) that has been coded by pitch predictive PCM. Since the baseband residual is used as a part of excitation, the system is also robust to V/UV and pitch errors. According to our informal listening tests, the synthetic speech of the new system does not have the buzzy effect. As a result the vocoder speech quality is more natural than that of a conventional LPC vocoder.
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