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Proceedings ArticleDOI

A 16 kbps ADPCM with multi-quantizer (ADPCM-MQ) codec and its implementation by digital signal processor

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TLDR
This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance, and the coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ).
Abstract
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.

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Citations
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Journal ArticleDOI

A low-delay CELP coder for the CCITT 16 kb/s speech coding standard

TL;DR: The official CCITT laboratory tests revealed that the speech quality of this 16 kb/s LD-CELP coder is either equivalent to or better than that of the CCITT G.721 standard 32-kb/s ADPCM coder for almost all conditions tested.
Journal ArticleDOI

Enhancement of ADPCM speech coding with backward-adaptive algorithms for postfiltering and noise feedback

TL;DR: It is shown that postfilters based on higher order LPC (linear predictive coding) models can provide very low distortion in terms of special tilt and can provide better speech enhancement than circuits based on the backward-adaptive pole-zero predictor in ADPCM (adaptive digital pulse code modulation).
Book ChapterDOI

A robust low-delay CELP speech coder at 16 kbits/s

TL;DR: A candidate algorithm for the new CCITT 16-kb/s speech coding standard is presented, based on backward-adaptive CELP (code-excited linear prediction) where the predictor and the excitation gain are updated by analyzing previously quantized signals.
Proceedings ArticleDOI

An 8 kb/s low-delay CELP speech coder

J.-H. Chen, +1 more
TL;DR: The authors have found a new way of CELP speech coding at or below 8 kb/s which achieves about the same speech quality with about the the same complexity but a much lower coding delay.
Proceedings ArticleDOI

An 8 kbps TC-MQ (time domain compression ADPCM-MQ) speech codec

TL;DR: A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed, based on time scale modification and sub-band coding with the aid ofADPCM with a multiquantizer that was confirmed by computer simulations.
References
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Journal ArticleDOI

Enhancement of ADPCM speech by adaptive postfiltering

TL;DR: An adaptive postfiltering procedure that can significantly enhance the quality of lower bit rate ADPCM is proposed and in a subjective test involving 18 listeners and two sentence-length test inputs, the enhanced 24- kb/s speech with an optimized postfilter design ranks very close to conventional 32-kb/sspeech.
Journal ArticleDOI

A High Performance LSI Digital Signal Processor for Communication

TL;DR: A newly developed CMOS LSI DSP (FDSP3) is described, which has a powerful multiplier, which operates at a rate of 10 M-operations/s, almost twice as fast as the fastest existing L SI DSP's.
Proceedings ArticleDOI

A high-efficiency speech coding algorithm based on ADPCM with multi-quantizer

TL;DR: A new 8 to 9.6 kbps ADPCM-MQ coding algorithm that includes tree coding to improve the per-sample quantizing characteristic, and sub-band coding with high frequency band reconstruction to realize a lower bit rate coding is described.