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Speech Signal Processing
Published in Richard C. Dorf, Circuits, Signals, and Speech and Image Processing, 2018
Jerry D. Gibson, Bo Wei, Hui Dong, Yariv Ephraim, Israel Cohen, Jesse W. Fussell, Lynn D. Wilcox, Marcia A. Bush
ADPCM operates at 32 kbits/sec or lower, and it achieves performance comparable to log-PCM by using a linear predictor to remove short-term redundancy in the speech signal before quantization. The most common form of ADPCM uses what is called backward adaptation of the predictors and quantizers to follow the waveform closely. Backward adaptation means that the predictor and quantizer are adapted based upon past reproduced values of the signal that are available at the encoder and decoder. No predictor or quantizer parameters are sent along with the quantized waveform values (called forward adaptation). By subtracting a predicted value from each input sample, the dynamic range of the signal to be quantized is reduced, and hence, good reproduction of the signal is possible with fewer bits [1].
Speech Coding
Published in Sadaoki Furui, Digital Speech Processing, Synthesis, and Recognition, 2018
ADPCM is a type of DPCM which includes backward adaptive quantization and/or backward adaptive prediction. This method is advantageous in that only the residual signals must be transmitted (Cummiskey et al., 1973). The method using backward (feedback) adaptive quantization and fixed prediction produces high quality in spite of its simple structure. In this method, the quantization step for the first-order difference using a fixed coefficient is controlled to adapt to input speech. This method produces an SNR of roughly 22 dB at 32 kbps (8-kHz sampling and 4-bit quantization), which is roughly 8 dB higher than that of log PCM at the same data rate. Subjective evaluation by the preference method indicates that the quality of 4-bit ADPCM is between that of 6-bit and 7-bit log PCM. This means that ADPCM can achieve an improvement of roughly 2.5 bits.
Speech Coding for Wireless Communications
Published in Jerry D. Gibson, Mobile Communications Handbook, 2017
ADPCM operates at 32 kbps or lower, and it achieves performance comparable to log-PCM by using an adaptive linear predictor to remove short-term redundancy in the speech signal before quantization. The most common form of ADPCM uses what is called backward adaptation of the predictors and quantizers to follow the waveform closely. Backward adaptation means that the predictor and quantizer are adapted based upon past reproduced values of the signal that are available at the encoder and decoder. No predictor or quantizer parameters are sent along with the quantized waveform values. By subtracting a predicted value from each input sample, the dynamic range of the signal to be quantized is reduced, and hence, good reproduction of the signal is possible with fewer bits.
Detection techniques for mitigating the nonlinear distortion of ADPCM link
Published in Australian Journal of Multi-Disciplinary Engineering, 2018
A recommended definition of the 32 kb/s Adaptive Differential Pulse Code Modulation (ADPCM) algorithm was published by International Telephone and Telegraph Consultative Committee (CCITT, the recent name is International Telecommunication Union, ITU) as recommendation G.721 (CCITT Recommendation G.721, 1984). This algorithm has the great advantage of being a proven international standard optimised for the operating environment of the telecommunication networks.