This paper describes the scalable coder - G.729.1 - which has been recently standardized by ITU-T for wideband telephony and voice over IP (VoIP) applications. G.729.1 can operate at 12 different bit rates from 32 down to 8 kbit/s with wideband quality starting at 14 kbit/s. This coder is a bitstream interoperable extension of ITU-T G.729 based on three embedded stages: narrowband cascaded CELP coding at 8 and 12 kbit/s, time-domain bandwidth extension (TDBWE) at 14 kbit/s, and split-band MDCT coding with spherical vector quantization (VQ) and pre-echo reduction from 16 to 32 kbit/s. Side information - consisting of signal class, phase, and energy - is transmitted at 12, 14 and 16 kbit/s to improve the resilience and recovery of the decoder in case of frame erasures. The quality, delay, and complexity of G.729.1 are summarized based on ITU-T results.

This paper presents the core technology of novel enhancements to achieve toll quality at 4 kbps, our experiments and test results traditional CELP coding, coined eXtended CELP (eX-CELP). It is showed that this technology is also successful and suitable for centered on a combined and selective usage of closed-loop/open-both high and medium bit rates. Fig. I and Fig.2 illustrate the basic loop approach, and variant algorithm structure concept. The above structure of the eX-CELP encoder and decoder. two concepts are complemented by new features and refined One of the main themes of the eX-CELP technology is the existing technologies. The eX-CELP paradigm was used in judicious combination of the closed-loop approach and the open-several speech coding systems. It is the core technology of the loop approach, together with a careful selective usage of them. recently chosen candidate for the 3G-CDMA speech codec This mechanism is coined COLA, and its main objective is to standard. It was the best candidate for ITU-T 4 kbps codec intelligently employ the most appropriate approach for different qualification test, and became the basis technology for a types of input signals in order to preserve the perceptually consortium candidate to the ITU-T 4 kbps speech coding important contents. Another important feature in the eX-CELP competition.

During the years 1999 and 2000, the Telecommunication Industry Association (TIA) and the 3rd Generation Partnership Project 2 (3GPP2), managed a competition and a selection process for a new speech coding standard for CDMA applications. The new speech coding standard, which is coined Selectable Mode Vocoder (SMV), will become a service option in CDMA systems such as IS-95 and cdma2000, providing a higher quality, flexibility, and capacity over the existing speech coding service options, IS-96C, IS-127, and IS-733. Eight companies submitted candidates to selection phase. For all of the 36 test conditions, the Conexant SMV candidate was ranked at the top, or was statistically equivalent to other top-ranking candidates. The Conexant SMV candidate was chosen as the core speech coding technology for the SMV system. This paper describes the SMV algorithm developed by Conexant.

This paper presents the 4 kbit/s speech coding candidate submitted by AT&T, Conexant, Deutsche Telekom, France Telecom, Matsushita, and NTT for the ITU-T 4 kbit/s selection phase. The algorithm was developed jointly based on the qualification version of Conexant. This paper focuses on the development carried out during the collaboration in order to narrow the gap to the requirements in an attempt to provide toll quality at 4 kbit/s. This objective is currently being verified in independent subjective tests coordinated by ITU-T and carried out in multiple languages. Subjective tests carried out during the development indicate that the collaboration work has been successful in improving the quality, and that meeting a majority of the requirements in the extensive selection phase test is a realistic goal.

This paper addresses the problem associated with variable-dimension vector quantization, and presents a new quantization technique that combines a variable-size nonsquare transform (NST) with a fixed-dimension vector quantizer. We show that all linear dimension conversion methods can be treated as special cases of a general approach for linear dimension conversion formulated as NST. By incorporating the speech perceptual properties, we introduce a technique called weighted nonsquare transform vector quantization (WNSTVQ) for the quantization of speech spectral vectors. We show that the total perceptual weighted distortion can be separated into the weighted modeling distortion, which is solely determined by the choice of transforms in WNSTVQ and the weighted quantizer distortion. We discuss the factors that influence the performance of the WNSTVQ system and provide a complexity analysis for two WNSTVQ implementations. Finally, experimental results are presented to show that the WNSTVQ system has the ability to trade performance for computational complexity and memory storage by selecting suitable transforms and/or the length of fixed-dimension vectors.

A new hybrid speech coding technique is presented in this paper, which combines a frequency-domain parametric coder (for stationary voiced and stationary unvoiced speech) with a time-domain waveform coder (for transition speech). Our hybrid coder uses a parametric representation for the excitation of a linear-prediction filter. The excitation of stationary voiced speech is a sum of harmonic cosines with interpolated magnitudes and a synthetic phase model, the excitation for stationary unvoiced speech is a spectrally shaped noise, and the excitation for transition speech is a set of signed pulses. Signal alignment when switching between the harmonic excitation of stationary voiced speech and the pulse model used for transition speech is required, and achieved by special alignment procedures. A 4 kb/s hybrid coder, which achieves high-quality reconstructed speech, is described. The 4 kb/s hybrid coder employs a neural network classifier, and a novel pitch detection and harmonic bandwidth estimation algorithm. The locations of excitation pulses for coding transitions are determined by analysis-by-synthesis. A simple and efficient dimension conversion and quantization of the harmonic. Spectral magnitudes of voiced speech was devised, combining the general nonsquare transform (NST) or dimension conversion and a weighted vector quantization (VQ) approach. Subjective listening tests demonstrate that the 4 kb/s hybrid coding scheme competes favorably with CELP coders at low bit-rates.

In this paper we present a new approach for speech coding, which combines frequency-domain harmonic coding for periodic and "noise like" unvoiced segments of speech with a time-domain waveform coder for transition signals. This hybrid coder requires special handling of the boundary between voiced and transition segments. We outline the details of a 4 kbps hybrid coder and present subjective quality test results of this coder.