Herve Marcel Taddei

Bandwidth Extension for Hierarchical Speech and Audio Coding in ITU-T Rec. G.729.1

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

Emerging ITU-T standard G.711.0 — lossless compression of G.711 pulse code modulation

The ITU-T Recommendation G.711 is the benchmark standard for narrowband telephony. It has been successful for many decades because of its proven voice quality, ubiquity and utility. A new ITU-T recommendation, denoted G.711.0, has been recently established defining a lossless compression for G.711 packet payloads typically found in IP networks. This paper presents a brief overview of technologies employed within the G.711.0 standard and summarizes the compression and complexity results. It is shown that G.711.0 provides greater than 50% average compression in typical service provider environments while keeping low computational complexity for the encoder/decoder pair (1.0 WMOPS average, <;1.7 WMOPS worst case) and low memory footprint (about 5k octets RAM, 5.7k octets ROM, and 3.6k program memory measured in number of basic operators).

ITU-T G.729.1 scalable codec for new wideband services

G.729.1 is a scalable codec for narrowband and wideband conversational applications standardized by ITU-T Study Group 16. The motivation for the standardization work was to meet the new challenges of VoIP in terms of quality of service and efficiency in networks, in particular regarding the strategic rollout of wideband service. G.729.1 was designed to allow smooth transition from narrowband (300-3400 Hz) PSTN to high-quality wideband (50-7000 Hz) telephony by preserving backward compatibility with the widely deployed G.729 codec. The scalable structure allows gradual quality increase with bit rate. A low-delay mode makes the coder especially suitable for high-quality speech communication. The article presents the standardization goals and process, an overview of the coding algorithm, and the codec performance in various conditions.

Applicability of UDP-lite for voice over IP in UMTS networks

This paper examines the application of UDP-lite for unequal error detection in packet-switched speech transmission via Internet protocols (voice-over-IP) over UMTS radio channels. Traditionally, UDP is used as transport layer protocol, which contains a checksum that covers the complete packet. Thus, any packet with residual bit errors is discarded. Speech codecs like AMR, however, can tolerate bit errors in less sensitive parts of the bitstream. A more recent development, UDP-lite, provides unequal error detection with a partial checksum that covers only the sensitive parts of a packet. Thus, only packets with errors in important bits are discarded. We compare the use of UDP-lite for UMTS channels with convolutional and channels with turbo coding. The results show that the achievable quality improvement by applying UDP-lite depends on the residual bit error distribution of the chosen UMTS channel coding method. While we determined a quality improvement for channels with convolutional coding, we did not get an improvement for turbo coding. Furthermore, when combined with header compression, the convolutional coder with use of UDP-lite can reach the performance of the turbo coder with use of UDP

Acoustic echo cancellation embedded in smart transcoding algorithm between 3GPP AMR-NB modes

Acoustic Echo Cancellation (AEC) is a necessary feature for mobile devices when the acoustic coupling between the microphone and the loudspeaker affects the communication quality and intelligibility. When implemented inside the network, decoding is required to access the corrupted signal. The AEC performance is strongly degraded by nonlinearity introduced by speech codecs. The Echo Return Loss Enhancement (ERLE) can be less than 10 dB for low bit rate speech codecs. We propose in this paper a coded domain AEC integrated in a smart transcoding strategy which directly modifies the Code Excited Linear Prediction (CELP) parameters. The proposed system addresses simultaneously problems due to network interoperability and network voice quality enhancement. The ERLE performance of this new approach during transcoding between Adaptive Multirate-NarrowBand (AMRNB) modes is above 45 dB as required in Global System for Mobile Communications (GSM) specifications.