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IEEE Communications Magazine | 1997

ITU-T Recommendation G.729 Annex B: a silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications

Adit Benyassine; Eyal Shlomot; Huan-Yu Su; Dominique Massaloux; Claude Lamblin; Jean-Pierre Petit

This article describes Annex B to ITU-T Recommendation G.729. Annex B defines a low-bit-rate silence compression scheme designed and optimized to work in conjunction with both the full version of G.729 and its low-complexity Annex A. To achieve good quality low-bit-rate silence compression, a robust frame-based voice activity detector module is essential to detect inactive voice frames, also called silence or background noise frames. For these detected inactive voice frames, a discontinuous transmission module measures the changes over time of the inactive voice signal characteristics and decides whether a new silence information descriptor frame should be sent to maintain the reproduction quality of the background noise at the receiving end. If such a frame is needed, the spectrum and energy parameters describing the perceptual characteristics of the background noise are efficiently coded and transmitted using 15 b/frame. At the receiving end, the comfort noise generation module regenerates the output background noise using transmitted updates or previously available parameters. The synthesized background noise is obtained by linear predictive filtering of a locally generated pseudo-white excitation signal of a controlled level. This method of coding the background noise enables the achievement of bit-rate savings for coded speech at average rates as low as 4 kb/s during normal speech conversation while maintaining reproduction quality.


international conference on acoustics, speech, and signal processing | 2007

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

Stéphane Ragot; Balazs Kovesi; Romain Trilling; David Virette; Nicolas Duc; Dominique Massaloux; Stéphane Proust; Bernd Geiser; Martin Gartner; Stefan Schandl; Hervé Taddei; Yang Gao; Eyal Shlomot; Hiroyuki Ehara; Koji Yoshida; Tommy Vaillancourt; Redwan Salami; Mi Suk Lee; Do Young Kim

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.


Journal of the Acoustical Society of America | 2003

Speech manipulation for continuous speech playback over a packet network

Eyal Shlomot

In a speech communications network, continuous play of audio packets is achieved using a jitter buffer in a receiver. Audio packets are stored in the jitter buffer before decoding the audio packets into an audible output. When the level of stored audio packets approaches the full capacity of the jitter buffer, the rate at which the audio packets are played out of the jitter buffer is increased signaling a compression operation in the decoder. When the level of stored audio packets approaches an empty level of the jitter buffer, the rate which the audio packets are played out of the jitter buffer is reduced signaling an expansion operation in the decoder. Audio packets are not modified when the level of stored audio packets is within a predetermined range. A speed controller is provided to instruct the decoder to decode the audio packets according to either a compressed, expanded or normal audio packet status.


Archive | 2001

Speech communication system and method for handling lost frames

Adil Benyassine; Eyal Shlomot; Huan-Yu Su


Archive | 2003

Encoding and decoding speech signals variably based on signal classification

Yang Gao; Adil Benyassine; Jes Thyssen; Eyal Shlomot; Huan-Yu Su


Archive | 1995

Timing recovery scheme for packet speech in multiplexing environment of voice with data applications

Eyal Shlomot; Michael J. Simpson; Qiang Ye


Archive | 2008

Embedded silence and background noise compression

Eyal Shlomot; Yang Gao; Adil Benyassine


Archive | 2000

Fixed rate speech compression system and method

Yang Gao; Adil Benyassine; Jes Thyssen; Eyal Shlomot; Huan-Yu Su


Archive | 2003

Multi-mode bitstream transmission protocol of encoded voice signals with embeded characteristics

Yang Gao; Adil Benyassine; Jes Thyssen; Eyal Shlomot; Huan-Yu Su


Archive | 2007

Speech compression system and method

Yang Gao; Adil Benyassine; Jes Thyssen; Eyal Shlomot; Huan-Yu Su

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Yang Gao

Mindspeed Technologies

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Huan-Yu Su

Rockwell International

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Jes Thyssen

Mindspeed Technologies

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Adil c

Mindspeed Technologies

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