Jean-Pierre Adoul

Design and description of CS-ACELP: a toll quality 8 kb/s speech coder

This paper describes the 8 kb/s speech coding algorithm G.729 which has been standardized by ITU-T. The algorithm is based on a conjugate-structure algebraic CELP (CS-ACELP) coding technique and uses 10 ms speech frames. The codec delivers toll-quality speech (equivalent to 32 kb/s ADPCM) for most operating conditions. This paper describes the coder structure in detail and discusses the reasons behind certain design choices. A 16-b fixed-point version has been developed as part of Recommendation G.729 and a summary of the subjective test results based on a real-time implementation of this version are presented.

High quality coding of wideband audio signals using transform coded excitation (TCX)

This paper describes the application of transform coded excitation (TCX) coding to encoding wideband speech and audio signals in the bit rate range of 16 kbits/s to 32 kbits/s. The approach uses a combination of time domain (linear prediction; pitch prediction) and frequency domain (transform coding; dynamic bit allocation) techniques, and utilizes a synthesis model similar to that of linear prediction coders such as CELP. However, at the encoder, the high complexity analysis-by-synthesis technique is bypassed by directly quantizing the so-called target signal in the frequency domain. The innovative excitation is derived at the decoder by inverse filtering the quantized target signal. The algorithm is intended for applications whereby a large number of bits is available for the innovative excitation. The TCX algorithm is utilized to encode wideband speech and audio signals with a 50-7000 Hz bandwidth. Novel quantization procedures including inter-frame prediction in the frequency domain are proposed to encode the target signal. The proposed algorithm achieves very high quality for speech at 16 kbits/s, and for music at 24 kbits/s.<<ETX>>

GSM enhanced full rate speech codec

This paper describes the GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system. The GSM EFR codec has been jointly developed by Nokia and University of Sherbrooke. It provides speech quality at least equivalent to that of a wireline telephony reference (32 kbit/s ADPCM). The EFR codec uses 12.2 kbit/s for speech coding and 10.6 kbit/s for error protection. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec provides substantial quality improvement compared to the existing GSM full rate and half rate codecs. The old GSM codecs lack wireline quality even in error-free channel conditions, while the EFR codec provides wireline quality not only for error-free conditions but also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile to mobile calls).

The 'stretched'-Golay and other codes for high-SNR finite-delay quantization of the Gaussian source at 1/2 bit per sample

Error-correcting codes are considered as codebooks for high-performance vector quantization (VQ) of the IID Gaussian source at fractional bit rate. A family of good rate-one-half codes is introduced: the stretched Golay codes. The performance of these codes is compared to other good block codes, trellis-coded quantization, and other techniques. The stretched Golay codes are shown to outperform previously published results for block lengths 32, 40, 56, and 64. The good performance, together with fast decoding make these techniques attractive for applications such as low-bit-rate coding of speech. >

Embedded algebraic vector quantizers (EAVQ) with application to wideband speech coding

A new lattice vector quantization scheme, namely embedded algebraic vector quantizers (EAVQ), is proposed. This scheme makes use of spherical subsets of the rotated Gosset lattice RE/sub 8/ to constitute a vector-quantizer codebook. The codebook consists of several sub-codebooks and has an embedded structure. The codewords can be generated using an algebraic method and do not have to be stored. In combination with transform coded excitation (TCX) coding, this quantization technique is applied to 16 kbps wideband speech coding in order to quantize the so-called target signal in the frequency domain. Compared to a stochastic complex vector quantization scheme, EAVQ can achieve better performance and lead to significant savings of memory requirements.

Error intervals and cluster density in channel modeling (Corresp.)

This correspondence is concerned with binary processes and presents results with immediate applications in the modeling of digital channels for the purpose of evaluating code performance. It is demonstrated that cluster density can be analytically described from the distributions of intervals between errors. These relations and derived clustering properties hold for any stationary process. Analyses of real error data exemplify the use of these results in regard to channels having dependent inter-error intervals.

A critical statistic for channels with memory

We present a new descriptive statistic for channels with memory and show its utility a) in evaluating and comparing existing models for such channels and b) as a theoretical tool in defining the error-gap distribution characteristics of real channels. We demonstrate that certain kinds of real channel behavior cannot be adequately described by previously proposed models and offer an example of a better model that includes many of the earlier models as special cases.

Enhanced full rate speech codec for IS-136 digital cellular system

In this paper, we describe the enhanced full rate (EFR) speech codec that has recently been standardised for the North American TDMA digital cellular system (IS-136). The EFR codec, specified in the IS-641 standard, has been jointly developed by Nokia and University of Sherbrooke. The codec consists of 7.4 kbit/s speech (source) coding and 5.6 kbit/s channel coding (error protection) resulting in a 13.0 kbit/s gross bit-rate in the channel. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec offers speech quality close to that of wireline telephony (G.726 32 kbit/s ADPCM used as a wireline reference) and provides a substantial improvement over the quality of the current speech channel. The improved speech quality is not only achieved in error-free conditions, but also in typical cellular operating conditions including transmission errors, environmental noise, and tandeming of speech codecs.

8 kbit/s ACELP coding of speech with 10 ms speech-frame: a candidate for CCITT standardization

A toll quality speech codec at 8 kbit/s with a 10 ms speech-frame currently under standardization by the CCITT is presented. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP). Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Initial subjective tests showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kbit/s in error-free conditions and it outperforms G.726 under error conditions. The codec can support a frame erasure rate up to 3% with slight degradation and performs adequately under tandeming conditions. The algorithm has been implemented on a single fixed-point DSP for the CCITT qualification test. It requires about 24 MIPS.<<ETX>>

Nearest neighbor algorithm for spherical codes from the Leech lattice

The Leech lattice is a regular arrangement of points in 24-dimensional Euclidean space that yields an extremely dense packing when equal spheres are centered at these points. A subset of the Leech lattice can be used as a signal set for the Gaussian channel or as representative vectors for a vector quantizer. Of particular interest are the spherical codes (or code books) that consist of the points of the Leech lattice which lie on a sphere centered at the origin. The code points do not have to be stored because they can be obtained from a very small set of basic vectors using permutations of the components in a manner dictated by the words of the extended Golay code. A nearest-neighbor algorithm that works on this is developed to determine the point in the code closet to some arbitrary vector in R/sup 24/. The performance of this approach when quantizing independent identically distributed Gaussian samples is reported. >