Mi Suk Lee

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.

An 8-12 Kbit/S Embedded CELP Coder Interoperable with ITU-T G.729 CIDER: First Stage of the New G.729.1 Standard

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.

A main speaker decision for a distributed telepresence system

In this paper, we propose a method to detect a main speaker and automatically change into ones high definition (HD) video for a distributed telepresence system, so that the users feel immersive and convenient. In contrast to centralized systems, user equipment (UE) performs the main speaker decision (MSD) with a time synchronization using network time protocol (NTP). The MSD method includes a voice activity detection (VAD) and post-corrections to remove unwanted voice detections and share the same main speaker. We emphasize an audio signal of the main speaker to become more immersive. The proposed approach is applied to the telepresence system developed by ETRI and shows good performances.

A forward-backward voice packet loss concealment algorithm for multimedia over IP network services

In this paper, we propose a voice packet loss concealment algorithm in order to improve voice quality for both multimedia over IP and voice over IP services. The proposed algorithm estimates the coding parameters of lost frames by combining forward and backward prediction from the good frames before and after the lost frames. The performance of the proposed algorithm is evaluated on the ITU-T G.729 coder, and it is compared with the performance of the conventional algorithms in terms of objective and subjective quality measures. From the PESQ score comparison and the listening test, it is found that the proposed algorithm provides better voice quality than the conventional ones.

Bandwidth extension of a narrowband speech coder for music delivery over IP

In this paper, we propose a bandwidth extension (BWE) algorithm of a narrowband speech coder for music delivery services over IP networks. The proposed BWE algorithm is based on an embedded structure of using a baseline coder followed by an enhancement layer. To minimize the bit-rate increase by the enhancement layer, the proposed algorithm shares spectral envelope and excitation parameters between the baseline coder and the enhancement layer. In this paper, we choose the iLBC as the baseline coder and mel-frequency cepstral coefficients (MFCCs) are used to reconstruct higher frequency components at the enhancement layer. By doing this, the bit-rate of the proposed BWE coder is 15.45 kbit/s which is just 0.25 kbit/s higher than the iLBC. We compare the quality of the proposed BWE coder with that of the iLBC, and it is shown from an informal listening test that the proposed BWE coder provides significantly better quality than the iLBC for all four different kinds of music genres such as pop, classical, jazz and rock.

An immersive telepresence platform based on distributed architecture

A novel telepresence platform is proposed in this paper to provide immersive video conferencing based on distributed architecture and enhanced QoS/QoE technology. MCU(Multi-point Control Unit)-based telepresence platform has been widely used as a popular architecture due to its easy implementation of multiple video-audio mixing and conference controls from a central point. MCU-based platform usually inputs all participants media and outputs mixed media for conference participants one by one through network. Proposed platform has a concept of main speaker by decision algorithm during conference and determines a main speaker who sends a high-resolution media to simple nodes. Simple nodes manage routing tables, and they copy the media packets and forward them to conference participants according to the tables. For the immersive service, a robust packet loss recovery algorithm newly developed covers upto 10% bursty loss to prevent interruption of conference often observed in commercial Internet. Proposed platform with distributed media control protocol with enhanced QoS/QoE technology based on distributed architecture has been designed and implemented by software. It shows meaningful reduction of traffic over network with the shorter end-to-end media delay than observed in MCU-based platform under same conditions.