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
Consumer audio applications such as satellite radio broadcasts, multi-channel audio streaming and playback systems coupled with the need to meet stringent bandwidth requirements are eliciting newer challenges in parametric multichannel audio coding schemes. This paper describes the continuation of our research concerning the Immersive Soundfield Rendition (ISR) system. In particular we present detailed subjective result data benchmarking the ISR system in comparison to MPEG Surround and also characterizing the audio quality level at different sub-modes of the system. We also describe enhancements to various algorithmic components in particular the blind 2-to-5 channel upmixing algorithm and describe a novel scheme for providing enhanced stereo downmix at the receiver for improved decoding by conventional matrix decoding systems.

Abstract
Low bit rate audio coding often results in the loss of a number of key audio attributes such as audio bandwidth and stereo separation. Additionally, there is also typically a loss in the level of details and intelligibility and/or warmth in the signal. Due to the proliferation, e.g. on Internet, of low bit rate audio coded using a variety of coding schemes and bit rates over which the listener has no control, it is becoming increasingly attractive to incorporate processing tools in the player which can ensure a consistent listener experience. We describe a novel post-processing toolkit which incorporates tools for (i) Stereo Enhancement, (ii) Blind Bandwidth Extension, (iii) Automatic Noise Removal and Audio Enhancement, and, (iv) Blind 2-to-5 channel upmixing. Algorithmic details, listening results, and audio demonstrations are presented.

Abstract
In this paper, an evaluation of several methods allowing the estimation of the Harmonic-to-Noise Ratio (HNR) of sustained vowels was conducted. The HNR estimation methods are mainly based on time, spectral, and cepstral signal representations. An algorithm was implemented for each method and was tested with synthesized voice sounds in order to evaluate their accuracy. Tests were also conducted with real pathological voice sounds in order to evaluate the behaviour of the different methods under real conditions. © 2008 Division of Signal Processin.