Abstract
Multichannel audio is increasingly ubiquitous in consumer audio applications such as satellite radio broadcast systems; surround sound playback systems, multichannel audio streaming and other emerging applications. These applications often present challenging bandwidth constraints making parametric multichannel coding schemes attractive. Several techniques have been proposed recently to address this problem. Here we present a novel low bit rate five channel encoding system that has shown promising results. This technique called the Immersive Soundfield Rendition (ISR) System emphasizes accurate reproduction of multi-band temporal envelope. The ISR system also incorporates a very low over-head (blind upmixing) mode. The proposed multichannel coding system has yielded promising results for multi-channel coding in 0-12 kbps range. More information and audio demos are available at http://ww.atc-labs.com/isr.

Abstract
Bandwidth Extension has emerged as an important tool for the satisfactory performance of low bit rate audio and speech codecs. In this paper we describe the components of a novel integrated audio bandwidth extension toolkit (ABET). The ABET toolkit is a combination of two bandwidth extension tools: (i) The Fractal Self-Similarity Model (FSSM) for signal spectrum; and, (ii) Accurate Spectral Replacement (ASR). Combination of these two tools, which are applied directly to high frequency resolution representation of the signal such as the Modified Cosine Transform (MDCT), has several benefits for increased accuracy and coding efficiency of the high frequency signal components. At the same time the combination of the two tools entails a number of important algorithmic and perceptual considerations. In this paper we describe the components of the ABET bandwidth extension toolkit in detail. Algorithmic details, audio demonstrations, and, ABET configuration details are presented. Additional information and audio samples are available at http://www.atc-labs.com/abet/.

Abstract
This paper presents a new method performing real-time adaptive equalization of room acoustics in the frequency domain. The developed method obtains the frequency response of the room by means of the transparent insertion of a certain number of acoustic probe signals into the main audio spectrum. The opportunities for the insertion of tones are identified by means of a spectral analysis of the audio signal and using a psychoacoustic model of frequency masking. This enhanced version of the adaptive equalizer will be explained as well as its real time implementation on a TMS320C6713 DSP based platform. Finally the results of the acoustic tests and conclusions about its performance will be presented.

Abstract
3G mobile and wireless communication networks elicit new ways of multimedia human interaction and communication, notably two-way high-quality audio communication. This is inline with both the consumer expectation of new audio experiences and functionalities, and with the motivation of Telecom Operators to offer consumers new services and communication modalities. In this paper we describe the design and optimization of a nioriophonic audio coder (Audio Communication Coder -ACC) that features low-delay coding (< 50 ms) and intrinsic error robustness, while minimizing complexity and achieving competitive coding gains and audio quality at bit rates around 32 kbit/s and higher. ACC source, perceptual and bandwidth extension tools are described and an emphasis is placed on ACC structural and operational features making it suitable for real-time, two-way audio communication. A few performance results are also presented. Audio demos are available at http://www.atc-labs.corn/acc/ .

Abstract
In the Next Generation Networks (NGNs) users will carry multiple devices forming cooperative networks known as Personal Area Networks (PANs). Some existing technologies enable this type of networks, such as Bluetooth or IEEE 802.15.4, but a unified framework capable of self-organizing them dynamically in a full heterogeneous environment populated by these and other technologies still has to be defined. Also, these networks are envisioned to be connecting dynamically to the Internet, and may use two IP versions and their autoconfiguration mechanisms. In this paper we propose a new framework, the Autoconfiguration and Self-management of Personal Area Networks (ASPAN), which enables the automatic and dynamic deployment of PANs in the heterogeneous environments envisioned for NGNs and handles the automatic and dynamic connection of a PAN to the global Internet. Copyright 2006 ACM.

Abstract
Ad-hoc networks that are connected with the infrastructure Internet are named hybrid ad-hoc networks. In 4G communications scenarios, hybrid ad-hoc networks seem to be valuable since they may increase the coverage of wireless networks with minor costs. Using them, terminals out of range of an access point or a base station, or not having adequate network interfaces, may reach the operator's infrastructure via other terminals. This paper presents a hybrid ad-hoc network solution and a testbed implementation.