This paper presents an efficient LSP quantizer implementation for low bit-rate coders. The major feature of the quantizer is that it uses a truncated cepstral distance criterion for the code selection procedure. This approach has generally been considered too computationally costly. We utilized the quantizer with a moving-average predictor, two-stage-split vector quantizer and delayed decision. We have investigated the optimal parameter settings in this case and incorporated the quantizer thus obtained into an ITU-T 4-kbit/s speech coding candidate algorithm with a bit budget of 21 bits. The objective performance is better than that with a conventional weighted mean-square criterion, while the complexity is still kept to a reasonable level. The paper also describes the codebook design and techniques that were employed to achieve robustness in noisy channel conditions.

This paper presents a high quality 4-kbit/s speech coding algorithm based on a CELP algorithm. The coder operates on speech frames of 20 ms. The algorithm has following four main features: multiple sub-codebooks, backward adaptive mode switching, dispersed-pulse structure, and noise post-processing. The multiple sub-codebooks consist of a pulse-codebook and a random-codebook so that they can handle both signals, noise-like (e.g. unvoiced, stationary noise) and pulse-like (e.g. voiced). The backward adaptive mode switching is performed using decoded parameters; therefore, no additional mode bit is transmitted. The random-codebook size is switched with the backward adaptively selected mode. The subjective quality of unvoiced speech or noise-like signal can be improved by this switching operation because the random-codebook size is greatly increased in such signal mode. The dispersed-pulse structure provides better performance of sparse pulse excitation using dispersed pulses instead of simple unit pulses. The noise post-processing employs a stationary background noise generator for producing stationary noise signal. It significantly improves subjective quality of decoded signal under various background noise conditions. Subjective listening tests are conducted in accordance with ACR and DCR tests. The ACR test results indicate that the fundamental performance of the MDP-CELP is equivalent to that of 32-kbit/s adaptive differential pulse code modulation (ADPCM). The DCR test results show that the performance of the MDP-CELP is equivalent to or better than that of 8-kbit/s conjugate-structure algebraic code excited linear prediction (CS-ACELP) under several background noise conditions.