This paper presents the two new ITU-T Recommendations G.722 Annex D and G.711.1 Annex F, which are stereo extensions of the wideband codecs ITU-T G.722 and G.711.1 and their superwideband extensions (G.722 Annex B and G.711.1 Annex D). An embedded scalable structure is used to add stereo extension layers on top of the wideband or superwideband core coding. Wideband stereo modes are supported at the bit rates of 64/80 and 96/128 kbit/s for G.722 and G.711.1 (respectively), while superwideband stereo modes are supported at 80/96/112/128 and 112/128/144/160 kbit/s. The parametric stereo coding model is based on a frequency domain downmix, wideband inter-channel differences estimation, quantization and synthesis, low complexity coherence analysis and synthesis, stereo transient detection and stereo post-processing. An overview of formal ITU-T characterization listening tests illustrates the performance of these codecs.

This paper presents the lowest bitrate mode (64 kbit/s) of the new superwideband (SWB, 50-14000 Hz) coder, recently standardized as ITU-T G.722 Annex B. This mode provides a superwideband extension of G.722 at 56 kbit/s with one 8 kbit/s enhancement layer divided in two sub-layers. The resulting bitstream is compatible with ITU-T G.722 at 64 kbit/s and can be viewed as watermarking G.722 least significant bits (LSBs). The novel technologies in this mode include G.722 enhancements (noise feedback coding, scalable quantization in G.722 higher band), as well as multimode bandwidth extension (BWE). Selected ITU-T characterization test results and additional informal test results show that the 64 kbit/s mode of G.722 Annex B gives high SWB quality with low delay and complexity.

This paper presents high quality monaural superwideband extensions to G.711.1 and G.722, recently standardized as Recommendations ITU-T G.711.1 Annex D and G.722 Annex B. The superwideband (50-14000 Hz) functionality is achieved using embedded scalable structure that adds extension layers on top of the wideband core codecs. The bit rates are extended to 96/112/128 and 64/80/96 kbit/s for G.711.1 and G.722, respectively. The main technologies include lower and higher band (0-4 kHz and 4-8 kHz) enhancements, 8-14 kHz bandwidth extension and transform coding based on algebraic vector quantization. The codecs' performance is illustrated with listening test results extracted from formal ITU-T Characterization tests.

This communication presents the recently standardized superwideband (SWB) extensions of ITU-T G.718 and G.729.1. These extensions were standardized as G.718 annex B and G.729.1 annex E. The SWB functionality is implemented using embedded scalable layers on top of the wideband (WB) core codecs, and it extends the bit rate of the codecs to 48 and 64 kbit/s for the G.718 and G.729.1, respectively. The main technology is a two-mode SWB coding method of the high frequencies. In addition, the G.729.1 SWB extension enhances the lower frequency range. The codec performance is illustrated with some listening test results extracted from the ITU-T Characterization phase.

This paper describes a 12.2–32 kbps scalable wideband speech and audio coder interoperable with GSM enhanced full-rate (EFR). This coder, referred to as EFR-EV, is designed using the ITU-T G.729.1 multi-stage coding structure. Specifically, EFR-EV consists of three stages: a code-excited linear prediction (CELP) stage derived from EFR, time-domain bandwidth extension (TDBWE), and time-domain aliasing cancellation (TDAC). In this paper, we show that the G.729.1 extension layers (i.e. TDBWE and TDAC) are quite generic for scalable codec design in the sense that they can be applied to EFR with limited adjustments. In addition, we propose a minor modification of the bit allocation procedure in TDAC stage, exploiting spectral masking only for higher frequency bands. The performance of EFR-EV and G.729.1 are evaluated in terms of objective/subjective quality, algorithmic delay, and complexity.

ITU-T G.729.1 is a scalable coder recently standardized in ITU-T for wideband telephony and voice over IP (VoIP) applications. Composed of three stages, this codec provides a scalable bitstream between 8 and 32 kbit/s both in narrowband and wideband. This paper describes the first stage which is a narrowband embedded CELP coder at 8 and 12 kbit/s. The 8 kbit/s layer ensures interoperability with ITU-T G.729 standard with a reduced complexity, and with a quality better than G.729 Annex A. At 12 kbit/s, G.729.1 reaches the quality level of the 11.8 kbit/s G.729 Annex E in spite of the embedded structure. The modifications brought to the original G.729 scheme to achieve this performance are explained and formal test results provided.

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 describes a combined adaptive transform codec (ATC) and code-excited linear prediction (CELP) algorithm, called ATCELP, for the compression of wideband (7 kHz) signals. The CELP algorithm applies mainly to speech, whereas the ATC mode is selected for music and noise signals. We propose a switching scheme between CELP and ATC mode and describe a frame erasure concealment technique. Subjective listening tests have shown that the ATCELP codec at bit rates of 16, 24 and 32 kbit/s achieved performances close to those of the CCITT G.722 at 48, 56 and 64 kbit/s, respectively, at most operating conditions.

This paper emphasizes the importance of the question of word-order variation in connectionist language modeling. More precisely, it investigates whether recurrent networks can integrate various linguistic constraints to process variable word-order languages. This paper reports three experiments on the understanding of spoken French that suggest that recurrent architectures could apply to the understanding of variable spoken languages.

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.

We describe a systematic procedure to replace vector quantization (VQ) with trellis-coded vector quantization (TCVQ) in existing CELP coders. Following this procedure, we design an 8/16 kbit/s TCVQ CELP coder. We analyze the performance of this method in terms of quality and complexity. Our results show that a CELP coder using TCVQ produces significantly better quality than the same coder using VQ, with reasonable complexity. By modifying the TCVQ CELP coder parameters one can favorably trade coding quality against complexity and/or delay.

We analyze the performance of a CELP coder where the vector quantization (VQ) of the excitation is replaced with trellis-coded vector quantization (TCVQ). Our results show that TCVQ performs significantly better than VQ, with reasonable complexity. This makes TCVQ a fair choice for trading quality against complexity and/or delay. We describe a systematic procedure to replace VQ with TCVQ for existing CELP coders. We propose an optimization algorithm to appropriately populate the trellis. We show how pseudo-Gray coding can be applied to the TCVQ codebook to improve intrinsic coder robustness to channel errors. Finally, we evaluate the complexity and performance of the method.

A novel approach for minimizing the CELP (code excited linear prediction) error criterion in the Z-transform domain is presented. A covariance method and an autocorrelation method of CELP coding are derived using the Z-transform formulation. The intermediate signals in the excitation selection are interpreted in terms of digital filtering, (inter)correlations, spectra, and inner products. For each method, specific algebraic codebooks permit fast algorithms in the selection of the excitation signal, leading to low complexity CELP coders for high-quality narrowband (0-4 kHz) and wideband (0-7 kHz) speech coding at reduced bit rates.>

In several speech coding applications the constraint of fixed bit rate and frame size is not imposed. For such a situation, a speech coding system performing speech segmentation and segment coding, both with reasonable complexity, is proposed. Speech is segmented according to its statistical properties. Each class of segment is encoded by a specific algorithm using algebraic excitation codebooks. An average bit rate between 5 and 6 kb/s yields a small degradation over a 9.6-kb/s fixed rate coder.>

The authors present an algebraic code-excited linear prediction (CELP) speech coder where the innovation codebook comes from the first spherical code of the Barnes-Wall lattice in 16 dimensions. Novel fast optimal algorithms for finding the best sequence in this Barnes-Wall shell innovation codebook are described. This algebraic codebook makes it possible to design a CELP coder at 9.6 kb/s with good quality and still implementable on a current digital-signal-processing chip.>

The paper discusses how binary error-correcting codes can be used to provide (+1/-1)-waveform codebooks that speed up search in CELP vocoders. Four coding techniques operating at half bit per sample with respectively 8, 16, 24 and 32 samples are compared in terms of complexity and SNR performance, Recent results on spherical codes from regular point lattices are also reported.

This paper describes a robust vector adaptive predictive coder which features a novel arrangement of a decoder to provide a finite memory synthesis and a new scheme for generating the excitation signal using an analysis by synthesis procedure. For bit error rates up to 10-2, our 16 kbit/s coder including error protection is shown to work efficiently on speech signals representative of a two-way mobile-network communication.

This paper concerns a new Quantization Scheme for efficient encoding of waveforms below or about one bit per sample. This technique is then applied to the encoding of baseband residual signals to demonstrate the feasibility of (baseband) Residual Excited LPC at 2400 hit/sec. The technique called "Spherical Vector Quantization" is described in which a block of n consecutive samples is quantized as a vector. The magnitude of the vector is transmitted independently of the vector's orientation. This vector's orientation is vector quantized using a codebook which can be seen as representing a set of N points on a unit hypersphere. The cases for blocks of n = 8 and 24 are discussed which make use of results by Conway and Sloane on regular point lattices. For n = 8, the algorithm is detailed which solves both the problem of finding the closest point on the hypersphere and the problem of determining the index of that point. Experiments involving SVQ in coding the speech baseband residual are described which show in particular that subband coding does not contribute any quality improvement when SVQ is used.