Showing papers on "Speech coding published in 1973"

Transmission system for audio and coding signals in educational tv

Abstract: An educational TV system features a transmitter receiver for a plurality of audio and coding signals along with multiple pictures. Bursts of audio signals are modulated in pairs onto 3.6 megahertz quadrature phase subcarriers in the same way as conventional I and Q video signals are modulated onto a subcarrier in conventional TV systems. Modulated onto each quadrature phase of the 3.6 megahertz subcarrier is an audio pulse signal. The resulting burst of phase and amplitude modulated subcarriers are produced during a blanked guard-band interval. This process is repeated with another pair of audio signals whereby two time divided bursts of modulated subcarriers are produced during the blanked guard-band interval which is located midway within the horizontal scan line time period used to transmit modulated video signals on a subcarrier. Coding signals are introduced onto the carrier signal during the time of the back porch of the video signal. In the receiver, after detecting and demodulating the bursts of audio signals, switching and logic circuitry perform both audio and picture switching operations to select an audio signal for driving a loud-speaker or headphone. The conventional audio FM channel associated with TV signal transmissions is also provided and operates in the conventional manner.

Results of a Study of the Linear Prediction Vocoder

Abstract: The Linear Prediction Vocoder (LIPREDER) System is an analysis-synthesis speech processing system which efficiently converts speech into digital form at voiceband data rates. The objective of our study was to optimize the LIPREDER system in terms of performance and complexity for a wide selection of talkers and sentences at data rates of 3600 and 7200 b/s. The optimized LIPREDER system produces synthetic speech quality which represents a substantial improvement over previous voice encoders operating at voiceband data rates. Evaluation results indicate that for most talkers and text there is little or no perceptible quality difference between the synthetic speech produced by LIPREDER systems at 3600 and 7200 b/s. Additionally, implementation of the LIPREDER system appears to be technically feasible.

Speech Data Rate Reduction Part I: Applicability of Modern Estimation Theory

Abstract: Efficient coding of continuous speech signals for digital representation has attracted much interest in recent years. The underlying aim of efficient coding methods is to reduce the channel capacity required to represent a signal to meet a specific reconstruction fidelity criterion. To achieve this objective, modern speech data compression techniques rely on two very similar procedures. One procedure uses predictive deconvolution which subtracts from the current signal value that portion which can be predicted from its past and thus removes redundancy in the speech by removing sequential correlation. The signal thus requires fewer bits for equivalent quantization error. The second procedure involves identification of a complete mathematical model of the speech producing mechanism. This involves determination of the characteristics of the source that drives this transfer function. Data reduction is again achieved since the rate of change of the parameters of the speech model is much smaller than the rate of change of the speech waveform. This paper develops these data reduction procedures in terms of modern estimation theory, specifically a Kalman filter model, and illustrates the utility of this model as an analysis tool by means of an example based on a uniform tube which provides a qualitative assessment of the potential of the technique for application to real speech signals.

A rate-distortion bound on channel requirements for transmission of parameters of adaptive predictive coders, with speech applications

Abstract: A nonstationary signal source can be decomposed into subsources if it exhibits certain characteristics. Adaptive predictive coding is considered as a method of extracting these subsources from the composite source in such a way that the overall communication problem can then be viewed as two different, but connected, communication problems: 1) transmission of predictor parameters, and 2) transmission of the subsource using difference signals. An intermediate fidelity criterion is defined that describes the effects of predictor parameter distortion on the reconstruction of the difference signal. Then a rate-distortion bound on the channel requirements for the transmission of the predictor parameters is found, subject to a dual fidelity criterion. The approach is applicable to a wide range of nonstationary signal classes. The speech process is chosen here because of the wide interests in the information content of speech. The signal class consists specifically of selected voiced phrases.

An audio response unit for telephone needs

Abstract: An audio response unit has been built that synthetizes messages composed of a fixed sentence and any number from 0-999 999. The method used is synthesis by concatenation of words, and automatic corrections on pitch and rhythm are used to improve naturalness and intelligibility. The synthesizer is a part of a channel vocoder. This audio response unit is to be used in a telephone exchange to answer cost inquiries from the subscribers. Larger applications using touch-tone telephone as data input device are under consideration.

The Role of Acoustic Processing in Speech Understanding Systems

Abstract: : The author discusses methods and problems of acoustic signal processing for systems to enable machines to understand spoken communication Emphasis is on research outside of the ARPA-sponsored SUR (Speech Understanding Research) study This acoustic level processing includes three steps, not necessarily distinct: (1) preprocessing the original analog signal or its digitized form by basic techniques such as amplitude compression; (2) analysis of the preprocessed signals using fast Fourier transformations, digital filtering, etc; and (3) parameterizing the results in phoneme-sized chunks by formats, autocorrelation techniques, etc Problems include (1) environmental noise, (2) transducer limitations, (3) determining an appropriate parameterization technique, and (4) coping with wide phonetic, syntactic, and semantic variability of speech

Speech and Picture Processing

Abstract: : During the second half of the contract year the program continued the following studies: speech analysis by linear prediction, reconstruction of multidimensional signals from projections, development of a high speed digital processor for speech synthesis, and the design of two-dimensional recursive digital filters. These projects are summarized, and reprints of available publications are appended.