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This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance. The coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ) which selects the optimum ADPCM coder frame by frame and switches to it dynamically. To implement this coding algorithm, we used two Fujitsu DSPs (MB8764), 1.5 chips for the encoder and 0.5 chip for the decoder. Reconstructed speech with a 21 dB segmental SNR was obtained. With error correction, this codec provides good speech quality even with a bit-error rate of 10^-2 to 10^-3. To improve the subjective quality of the reconstructed speech, adaptive postfiltering was also applied. Since the processing delay of this codec is less than 10 ms, no echo-canceller is needed. Moreover, 2400 bps voice band data (CCITT Rec.V. 26) could be transmitted with a data error rate from 10^-7 to 5×10^-6, and G.III facsimiles were successfully transmitted using this codec. |