This paper proposes a fast method of calculating high-order backward Linear Prediction (LP) coefficients for wideband Code Excited LP (CELP) coders operating at around 16 kbit/s. The fast calculation is achieved by a recursive calculation for the high-order autocorrelation of the decoded signal. The recursive calculation can be employed thanks to a novel method of converting the autocorrelation of the decoded signal to that of the residual signal. High-order backward LP coefficients are computed from the autocorrelation of the residual signal using the Levinson-Durbin (LD) procedure. The conversion approximately performs inverse-filtering using LP coefficients representing a corresponding envelope spectrum. Due to the recursive calculation, the proposed fast calculation method achieves 30% to 45% reduction in computations to calculate the high-order backward LP coefficients compared to the conventional method. Subjective tests show that a wideband Multi-Pulse based CELP (MP-CELP) coder at 16 kbit/s with the proposed method achieves comparable coding quality to that with the conventional one with 35% reduction in computations needed for calculation of the backward LP coefficients.

This paper proposes a method of low complexity speech mixing with speech codecs based on predictive coding for multimedia conferences. The proposed method applies a filter state management (FSM) technique to a partial mixing method in order to avoid inconsistency of the filter states of encoders. The inconsistency is created by switching of the encoders when the speakers to be mixed are switched. The results of subjective evaluations of speech quality show that the proposed method avoids the inconsistency, and achieves significantly higher speech quality than the conventional partial mixing method without the FSM and almost the same speech quality as the full mixing method. The complexity evaluation results show that the proposed method achieves much lower complexity than the full mixing method.

This paper presents the M-LCELP (Multi-mode Learned Code Excited LPC) speech coder, which has been developed for the next generation half-rate digital cellular telephone systems. M-LCELP develops the following techniques to achieve high-quality synthetic speech at 4kb/s with practically reasonable computation and memory requirements: (1) Multi-mode and multi-codebook coding to improve coding efficiency, (2) Pitch lag differential coding with pitch tracking to reduce lag transmission rate, (3) A two-stage joint design regular-pulse codebook with common phase structure in voiced frames, to drastically reduce computation and memory requirements, (4) An efficient vector quantization for LSP parameters, (5) An adaptive MA type comb filter to suppress excitation signal inter-harmonic noise. The MOS subjective test results demonstrate that 4.075kb/s M-LCELP synthetic speech quality is mostly equivalent to that for a North American full-rate standard VSELP coder. M-LCELP codec requires 18 MOPS computation amount. The codec has been implemented using 2 floating-point dsp chips.

This paper proposes a new pitch prediction method for 4 kbps CELP (Code Excited LPC) speech coding with 20 msec frame, for the future ITU-T 4 kbps speech coding standardization. In the conventional CELP speech coding, synthetic speech quality deteriorates rapidly at 4 kbps, especially for female and children's speech with short pitch period. The pitch prediction performance is significantly degraded for such speech. The important reason is that when the pitch period is shorter than the subframe length, the simple repetition of the past excitation signal based on the estimated lag, not the pitch prediction, is usually carried out in the adaptive codebook operation. The proposed pitch prediction method can carry out the pitch prediction without the above approximation by utilizing the current subframe excitation codevector signal, when the pitch prediction parameters are determined. To further improve the performance, a split vector synthesis and perceptually spectral weighting method, and a low-complexity perceptually harmonic and spectral weighting method have also been developed. The informal listening test result shows that the 4 kbps speech coder with 20 msec frame, utilizing all of the proposed improvements, achieves 0.2 MOS higher results than the coder without them.