In this paper, we examine a number of concepts and issues concerning variable-rate coding of speech. We formulate the problem as a multistate coder (i.e., a coder that can operate at several bit rates) coupled with a time buffer. We first analyze the theoretical aspects of the problem by examining it in the context of a block processing formulation. We then suggest practical methods for implementing a variable rate coder based on a dynamic buffering approach. We also allude to a multiple user configuration of variable-rate coding for tasi-type applications. A practical example of a variable rate adpcm coder is presented and applied to speech coding. It is shown that by careful design the algorithm can be made to be as robust to channel errors as that of a fixed rate adpcm coder.

Variable rate coding of speech

Digital techniques for implementing voice response systems are discussed. In particular, attention is focused on practical methods for computer voice response that are likely to be important in the near future. The general method of concatenating isolated words and phrases to form a message is discussed, along with the various possibilities for digital representations of speech signals. The techniques used to implement an all-digital voice response system with multiline capability are presented. A wide variety of experimental applications of this particular system are also included.

Digital techniques for computer voice response: Implementations and applications

This paper presents the basic architecture and performance of a mobile radio multiaccess voice/data system. Natural pauses in conversational speech allow bandwidth saving through interleaving of data packets and talkspurts from different voice sources. A speech detector designed specifically for the mobile environment is presented. Blocking and delay performance of the multiaccess uplink is analyzed for voice traffic, assuming no traffic effects from the low priority data packets. Performance results from simulation are then presented for two downlink strategies in a two-hop virtual circuit in which a base station acts as a relay. The results verify also that the uplink analysis is valid for low voice traffic. For the data traffic, simulation results are presented in terms of data packet transmission delay and probability of collision with talkspurts. The results indicate that data flow may be limited by the collision factor. This work concludes that relative to conventional radio telephoning in which two channels are dedicated to each transmitter/receiver pair, a bandwidth reduction of 30-35 percent can be achieved.

An Integrated Voice/Data System for VHF/UHF Mobile Radio

In this paper we examine a number of concepts and issues concerning variable rate coding of speech. We formulate the problem as a multistate coder (i.e. a coder that can operate at several bit rates) coupled with a time buffer. We first analyze the theoretical aspects of the problem by examining it in the context of a block processing formulation. We also allude to a multiple user configuration of variable rate coding for TASI type applications. A practical example of a variable rate ADPCM coder is presented and applied to speech coding. It is shown that by careful design the algorithm can be made to be as robust to channel errors as that of a fixed rate ADPCM coder.

Variable rate coding

Speech detection is accomplished in conjunction with two-band subband encoding. A detection statistic T (i To ), used to estimate the short-term speech energy, is developed from energy estimates made in each subband. A speech presence energy threshold λ ON a speech silence energy threshold λ OFF and λ OFF are computed which adapt to the long-term speech level. The detection statistic is compared to the thresholds to make a decision concerning the presence or absence of speech. Also disclosed are considerations for extrapolating the detection to result in an arrangement with more than two subbands.

Speech-silence detection with subband coding

When speech is coded using a differential pulse-code modulation system with an adaptive quantizer, the digital code words exhibit considerable variation among all quantization levels during both voiced and unvoiced speech intervals. However, because of limits on the range of step sizes, during silent intervals the code words vary only slightly among the smallest quantization steps. Based on this principle, a simple algorithm for locating the beginning and end of a speech utterance has been developed. This algorithm has been tested in computer simulations and has been constructed with standard integrated circuit technology.

K. Jackson

Papers

Detecting the Presence of Speech Using ADPCM Coding