Hervé Taddei

ITU-T G.729.1: AN 8-32 Kbit/S Scalable Coder Interoperable with G.729 for Wideband Telephony and Voice Over IP

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.

Mode adaptive Unequal Error Protection for transform predictive speech and audio coders

In this paper we investigate the application of Unequal Error Protection (UEP) channel coding schemes to the Transform Predictive Coding (TPC) audio coding paradigm. The TPC paradigm has natural perceptual ordering of the bitstream that provides a good match with the UEP concept. Also introduced is the general idea of Mode Adaptive UEP schemes that adapt to the differing bit-error sensitivity profiles of signal adaptive multimode source-coders. By doing this the decoded signal quality is optimized jointly over the input signal, the channel conditions and the bit-error sensitivity profile of the source coder.

Noise reduction on speech codec parameters

The transmission of speech in mobile or packet networks requires the use of a speech codec. In order to improve the quality of speech in a noisy environment, a noise reduction algorithm is used. This noise reduction can either be done as pre-processing before speech encoding or in the network by decoding the bitstream, performing the speech enhancement in the time and/or frequency domain and re-encoding the speech. Both methods are computationally expensive. In this paper a new approach to reduce environmental background noise by modifying the codec parameters is discussed.

A Qualified ITU-T G.729EV Codec Candidate for Hierarchical Speech and Audio Coding

We present an embedded and hierarchical 8-32 kbit/s speech and audio coding algorithm that has been successfully submitted to the ITU-T as a candidate for ITU-T Rec. G.729.1 (ex G.729EV). The submitting consortium consisted of Siemens AG, Matsushita Electric Industrial Co. Ltd., and Mindspeed Technologies Inc. This contribution gives a comprehensive overview of the proposed codec, describes the implemented algorithms, and states a detailed characterization as well as results of the official G.729EV qualification tests

An Enhancement Layer for ACELP Coder

This paper presents an enhancement layer coming on top of G.729 Annex A codec. This enhancement layer consists of an optimization of the algebraic fixed codebook pulses that were selected by the Annex A codec during the fixed codebook search. The pulses selected by the ACELP codec have all the same absolute amplitude. Our method optimizes each of these amplitudes in order to increase the quality. This is a low-complexity method that allows the introduction of an intermediate bitrate in ACELP-based scalable coder. On top of this pulse optimization method that requires 1.6 kbit/s we introduce a second fixed codebook search with a limited number of pulses (2) and a bitrate of 2.4 kbit/s

Efficient coding of transitional speech segments in CELP

Speech coding at low and medium bit rates benefits from the assumption that the speech signal is stationary inside a subframe or even inside consecutive subframes. Speech signals however comprise a considerable amount of transient segments, such as onsets, which are usually difficult to encode with good quality. This paper proposes a method to improve the encoding of transient (sub-)frames by disregarding the adaptive codebook contribution and strengthening the fixed codebook contribution. This can principally be applied to any CELP-like standard speech coder while keeping the same bit rate. The proposed technique has been successfully employed in the ITU-T 4 kbps speech coder candidate commonly proposed by AT&T, Conexant currently Mindspeed, Deutsche Telekom, France Telecom, Matsushita, NTT, and Siemens in November 2001. Simulation results show the improvements achieved by this approach.