Journal of the Audio Engineering Society

2008 October - Volume 56 Number 10


A computer simulation using finite-difference time-domain techniques has been used to model the low-frequency transfer function between a loudspeaker and virtual microphones in a rectangular room. The model for the analysis was validated with measurements in an IEC standard listening space. Using the mean sound field deviation, the cumulative spectral energy decay, and the clarity metric, the properties of a sample room can be examined mathematically and visually. At frequencies between 30 and 100 Hz, sound pressure varies as much as 30 dB from one position to another.

Adaptive RD Optimized Hybrid Sound Coding

Authors: Schijndel, Nicolle H. van; Bensa, Julien; Christensen, Mads G.; Colomes, Catherine; Edler, Bernd; Heusdens, Richard; Jensen, Jesper; Jensen, Søren Holdt; Kleijn, W. Bastiaan; Kot, Valery; Kövesi, Balázs; Lindblom, Jonas; Massaloux, Dominique; Niamut, Omar A.; Nordén, Fredrik; Plasberg, Jan H.; Vafin, Renat; Par, Steven van de; Virette, David; Wübbolt, Oliver

Rather than select a single type of codec optimized for a particular application, the authors propose a generic sound-coding technology that combines a range of different coding algorithms based on the current properties of the audio and the channel constraints. Rate-distortion optimization is a promising method for such a framework. Subjective listening tests show the advantages of this kind of flexibility in such applications as Internet radio, portable storage, and mobile communications. Bit allocation is not fixed but flexible, which allows for adaptation to time-varying inputs and changing channel properties.

By means of time–frequency signal processing, the effective directional response of stereo coincident microphones can be altered. This allows the spaciousness of the recording to be controlled without the use of artificial reverberation and with good localization. Effectively, the algorithm reduces the overlapping region of the microphone responses. Subjective testing shows that audio quality is very high and cardioid coincident stereo recordings are improved. Rear sound is rejected and the two signals remain in phase.

This paper presents a new paradigm for spatial audio teleconferencing that exploits the corpuscular nature of general audio parametric coding for low bit-rate audio compression. The paradigm consists of several processing tools. The spatial aspect is enacted by a dynamic rendition tool that optimizes placement of speech sources within the soundscape; the temporal aspects of a phoneme protection tool misalign corruptive and vulnerable speech elements in a multitalking environment. A novel means of echo suppression eliminates the need for adaptive filtering and real-time echo cancellation, which is difficult for multichannel audio.

In the above paper,1 in footnote 2 (p. 453) exponent b in pmon is missing. It should have appeared as follows...

Let the Games Be Mobile

Authors: Rumsey, Francis

[Feature] The rapid rise of interest in high-quality audio on mobile devices has led to a number of new initiatives in audio research. Broadband network access makes a range of new services possible, including downloadable games and even remixable music. So what are the challenges and possible solutions in this fast-evolving field? We review some recent workshops and papers from AES meetings dealing with the topic.

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