This paper presents a way to calculate the perceived importance of speech segments as a single value criterion, using a linear regression model. Unlike the commonly used voice activity detection (VAD) algorithms, this method allows us to obtain a finer priority granularity of speech segments. This can be used in conjunction with frequency scalable speech coding techniques and IP QoS techniques to achieve efficient and qualitycontrolled voice transmission. A simple linear regression model is used to calculate the estimated mean opinion score (MOS) of the various cases of missing speech segments.

Low-cost highly-efficient speech coding is important for personal multimedia communications. In this paper we propose a new low-complexity speech coding method called "Dual-Pulse CS-CELP (DP-CS-CELP)" at 7.8 kbit/s. This method is based on ITU-T G.729. To reduce the complexity, we applied a new excitation model to the random code vectors and simplified the LPC coding, adaptive codebook search, perceptual weighting, and the other structures. The number of operations for this encoder is 3.80 MOPS, which achieves real-time speech coding and decoding on personal computers. Although the efficiency of this coder is a little lower than that of G.729, the MOS listening test showed that the subjective quality was equivalent to or a little better than that of G.726 (32-kbit/s ADPCM).

A high-quality 8-kbit/s speech coder based on conjugate structure CELP (CS-CELP) is proposed that uses a trained sparse conjugate codebook. The trained sparse conjugate codebook improves speech quality for noisy speech. This codebook consists of two sub-codebooks and each sub-codebook consists of a random component and a trained component. Each component has excitation vectors consisting of a few pulses. In the random component, pulse position and amplitude are determined randomly. The trained component is determined by training. Subjective tests (differential mean opinion score, DMOS and mean opinion score, MOS) indicated that this codebook improves speech quality compared with the conventional trained codebook for noisy speech. The MOS showed that the quality of improved CS-CELP is equivalent to that of the 32-kbit/s ADPCM for clean speech.

This paper describes the design of a speech coder called pitch synchronous innovation CELP (PSI-CELP) for low hit-rate mobile communications. PSI-CELP is based on CELP, but has more adaptive excitation structures. In voiced frames, instead of conventional random excitation vectors, PSI-CELP converts even the random excitation vectors to have pitch periodicity by repeating stored random vectors as well as by using an adaptive codebook, in silent, unvoiced, and transient frames, the coder stops using the adaptive codebook and switches to fixed random codebooks. The PSI-CELP coder also implements novel structures and techniques: an FIR-type perceptual weighting filter using unquantized LPC parameters, a random codebook with a conjugate structure trained to be robust against channel errors, codebook search with delayed decision, a gain quantization with sloped amplitude, and a moving average prediction coding of LSP parameters, Our speech coder is implemented by DSP chips. Its coded speech quality at 3.6 kb/s with 2.0 kb/s redundancy is comparable to that of the Japanese full-rate VSELP coder at 6.7 kb/s with 4.5 kb/s redundancy. The basic structure of this PSI-CELP coder has been chosen as the Japanese half-rate speech codec for digital cellular telecommunications.>