Showing papers on "Adaptive Multi-Rate audio codec published in 1982"

Predictive Coding of Speech at Low Bit Rates

Abstract: Predictive coding is a promising approach for speech coding. In this paper, we review the recent work on adaptive predictive coding of speech signals, with particular emphasis on achieving high speech quality at low bit rates (less than 10 kbits/s). Efficient prediction of the redundant structure in speech signals is obviously important for proper functioning of a predictive coder. It is equally important to ensure that the distortion in the coded speech signal be perceptually small. The subjective loudness of quantization noise depends both on the short-time spectrum of the noise and its relation to the short-time spectrum of the Speech signal. The noise in the formant regions is partially masked by the speech signal itself. This masking of quantization noise by speech signal allows one to use low bit rates while maintaining high speech quality. This paper will present generalizations of predictive coding for minimizing subjective distortion in the reconstructed speech signal at the receiver. The quantizer in predictive coders quantizes its input on a sample-by-sample basis. Such sample-by-sample (instantaneous) quantization creates difficulty in realizing an arbitrary noise spectrum, particularly at low bit rates. We will describe a new class of speech coders in this paper which could be considered to be a generalization of the predictive coder. These new coders not only allow one to realize the precise optimum noise spectrum which is crucial to achieving very low bit rates, but also represent the important first step in bridging the gap between waveform coders and vocoders without suffering from their limitations.

A 32 kb/s toll quality ADPCM codec using a single chip signal processor

Abstract: An ADPCM codec, that can provide toll quality speech at a 32 kb/s transmission rate, has been implemented on a single chip signal processor. Maximum effort has been paid to design a robust adaptation scheme for a quantizer and a predictor to withstand transmission bit errors. The codec employs a simplified robust quantizer and also employs a new backward adaptive predictor. The decoder, including the new adaptive predictor, has a structure having fixed poles and adaptive zeros, attaining both high prediction capability and robustness. The performance of a developed codec, which has analog interface capability through a PCM codec chip, satisfies the standard 64 kb/s PCM performance specification in CCITT recommendation G.712.

A high quality ADM LSI codec at 32 kbit/s for digital speech communications

Abstract: This paper describes a 32kbit/s adaptive delta modulation (ADM) CODEC which can be applied to digital portable telephones. This ADM CODEC employs a new adaptation algorithm and parameters adjusted for speech signals. The encoding algorithm makes it possible to achieve higher quality with the 32 kbit/s ADM CODEC than with 6 bit µ-law PCM coding. The developed ADM CODEC including a coder, a decoder and switched-capacitor filters on a chip can operate on a single power supply(3.6∼ 5V). The power dissipation is only 30 mW during normal operation and less than 10 mW in the power-down mode. The CMOS chip is an a 16-pin package, 7 × 4.3mm.

A.D.P.C.M Algorithms applied to wideband speech encoding (64 kbit/s, 0-7 kHz)

Abstract: The application of ADPCM coding technique with adaptive predictors (A.D.P.C.M (V)) and its performances, to transmit 7 kHz bandwith speech signals at 64 kbit/s is presented in this paper. Subjective performances of one ADPCM(V) with two different sets of parameters are evaluated by means of direct comparison tests, with respect to the classic 64 kbit/s P.C.M coder, a 128 kbit/s P.C.M coder and a 64 kbit/s subband coder. The results of the tests show that the ADPCM coding technique can be an efficient mean to transmit wideband speech signals. The quality of the A.D.P.C.M(V) coded speech compares well with that obtained with subbands coding technique. The quality of the transmitted speech, though slightly worse than "high fidelity speech", could be sufficient for communications systems such as audio or video conferencing systems, commentary links for broadcasting.