This paper presents a high quality enhanced waveform interpolative (EWI) speech coder at low bit-rate. The system incorporates novel features such as optimization of the slowly evolving waveform (SEW) for interpolation, analysis-by-synthesis (AbS) vector quantization (VQ) of the SEW dispersion phase, dual-predictive AbS quantization of the SEW, efficient parameterization of the rapidly-evolving waveform (REW) magnitude, and VQ of the REW parameter, a special pitch search for transitions, and switched-predictive analysis-by-synthesis gain VQ. Subjective tests indicate that the 2.8 kb/s EWI coder's quality exceeds that of G.723.1 at 5.3 kb/s, and it is slightly better than that of G.723.1 at 6.3 kb/s.

This paper presents a high quality enhanced waveform interpolative (EWI) speech coder at 2.8 kbps. The system incorporates novel features such as: dual-predictive analysis-by-synthesis (AbS) quantization of the slowly evolving waveform (SEW), efficient parametrization of the rapidly evolving waveform (REW) magnitude, and AbS vector quantization (VQ) of the REW parameter. Subjective tests indicate that its quality exceeds that of G.723.1 at 5.3 kbps, and it is slightly better than that of G.723.1 at 6.3 kbps.

This paper presents an efficient analysis-by-synthesis vector quantizer for the dispersion phase of the excitation signal which was used to enhance a waveform-interpolative coder. The scheme can be used to enhance other harmonic coders, such as the sinusoidal-transform coder and the multiband-excitation coder. The scheme incorporates perceptual weighting, and does not require any phase unwarping. The proposed quantizer achieves a segmental signal-to-noise ratio of up to 14 dB for as low as 6-bit quantization. Subjective testing shows improvement in synthesized speech quality using the quantized phase over a male speaker extracted phase. The improvement was larger for female speakers.

This paper presents work based on an analysis-by-synthesis approach which aims to develop a reversible coding system for prosody, capable of deriving a `linguistic-like' surface phonological representation directly from acoustic data that is sufficient to reproduce a synthetic version of the original utterance without significant loss of linguistic information. With such a coding system, capable of representing any significant prosodic distinctions, the task of predicting such representations would be greatly simplified, becoming one of mapping between sets of symbolic representations. This approach has already been applied to the stylisation and symbolic coding of fundamental frequency curves by means of the INTSINT transcription system. An automatic version has also been proposed. This paper presents a preliminary proposal for an extension to the INTSINT system to cover segmental duration and the relative alignment of phonematic and tonal segments. 1. INTRODUCTION: INTSINT TRANSCRIP...

The Wideband-Audio Low-Delay CELP (LD-CELP) coder produces speech with quality as high as the CCITT 64 kb/s standard (G.722) at half the bitrate. The computational load of the encoder is almost 900% processor time of the 12.5 MIPS DSP32c. This makes a real-time implementation impractical. We investigated the Gain-Shape Vector-Quantization (GSVQ) in order to reduce the computational load of the encoder. This paper describes a real-time implementation of the LD-CELP encoder based on the AT&T SURFboard using two DSP32c operating in parallel. A computational load of 180% processor time has been achieved. The respective decoder requires 42% processor time. The implementation of a full-duplexed coding system requires three 12.5 MIPS Digital-Signal-Processors (DSPs) and has oneway coding delay of less than 1ms. The coder also performs well for non-speech wideband audio signals such as music.