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AES Amsterdam 2008
Signal Processing, Sound Quality Design
Paper Session P15
Monday, May 19, 09:00 — 12:00
Chair: Jan Abildgaard Pedersen, Lyngdorf Audio - Skive, Denmark
P15-1 Characterization of the Multidimensional Perceptive Space for Current Speech and Sound Codecs—Thierry Etame, France Télécom R&D - Lannion Cedex, France, and University of Rennes, Rennes, France; Laetitia Gros, Catherine Quinquis, France Télécom R&D - Lannion Cedex, France; Gérard Faucon, Régine Le Bouquin Jeannes, INSERM - Rennes, France, and University of Rennes, Rennes, France
The purpose of our work is to produce a reference system that can simulate and calibrate degradations of speech and audio codecs which are currently used on telecommunications networks, for subjective assessment tests of voice quality. At first, 20 wideband codecs are evaluated through subjective tests with the general goal of producing the multidimensional perceptive space underlying the perception of current degradations. Then, from a verbalization task, it appears that the identified attributes are clear/muffle, high-frequency noise, noise on speech, and hiss. Finally, these dimensions are characterized with correlates such as spectral centroid, spectral flatness measure, Mean Opinion Score, and correlation coefficient.
Convention Paper 7410 (Purchase now)
P15-2 An Automatic Maximum Gain Normalization Technique with Applications to Audio Mixing—Enrique Perez Gonzalez, Joshua D. Reiss, Queen Mary, University of London - London, UK
A method for real-time magnitude gain normalization of a changing linear system has been developed and tested with a parametric filter design. The method is useful in situations where the maximum gain before feedback is needed. The method automatically calculates the appropriate gain that should be applied in order to maintain maximum unitary gain. The method uses an impulse measurement of a mathematical model of the system to be normalized. This is particularly useful for mixing engineers, who have to continually revise their gain structure in order to maximize gain before feedback. The system is also useful in many other situations where solving the analytical solution from the mathematical model is not possible.
Convention Paper 7411 (Purchase now)
P15-3 An Alternative Approach for the Convolution in Time-Domain: The Taches-Algorithm—Laurent Millot, Gérard Pelé, ENS Louis-Lumière - Noisy-le-Grand cedex France
We present an alternative temporal approach for convolution, providing a new algorithm, called the taches-algorithm. Based on interferences between the successive delayed and amplified output signals associated respectively with the impulses constituting the input signal, the taches-algorithm can give access immediately to the new output sample and have a low latency response using vector-based optimization of the calculation. With the taches-algorithm it is easy to change (even in real time) the impulse response while running the calculation, simply by updating the impulse response to use it for next samples, a task rather difficult to achieve using FFT convolution. Real time audio demonstrations using Pure Data and simple explanations of the taches-algorithm will be given.
Convention Paper 7412 (Purchase now)
P15-4 Performance of Independent Component Analysis when Used to Separate Competing Acoustic Sources in Anechoic and Reverberant Conditions—Ben Shirley, Paul Kendrick, University of Salford - Salford, Greater Manchester, UK
A review of existing methods for independent component analysis was carried out and a series of experiments conducted assessing the use of existing independent component analysis (ICA) methods to separate microphone sources in varied acoustic environments. Specifically the research looked at how effectively ICA could perform in a broadcast context using standard microphone techniques such as spaced omni and coincident crossed cardioid pairs. Experiments were carried out in an anechoic chamber and also in a listening room conforming to the ITU-R BS.1116-2 standard. Results clearly indicate the limitations of ICA when performed on audio material recorded in a reverberant environment; however it was still shown possible to achieve separation of signals of up to 12 dB even in these conditions.
Convention Paper 7413 (Purchase now)
P15-5 A Cross-Platform Audio Signal Processing Environment for Real-Time Audio Algorithm Development—Mika Ristimäki, Nokia Research Center - Helsinki, Finland; Matti Hämäläinen, Nokia Research Center - Tampere, Finland; Julia Turku, Nokia Research Center - Helsinki, Finland; Riitta Väänänen, Nokia Research Center - Tampere, Finland
This paper presents a real-time audio algorithm development environment for experimental audio system research. The backbone of the system is Pure Data audio signal processing platform, which enables flexible implementation of real-time audio systems. With the proposed development environment the user can concentrate on real-time audio algorithm development and performance evaluation in the workstation environment. We present the proposed algorithm design method and environment, and its application to an experimental Voice over Internet Protocol (VoIP) system development.
Convention Paper 7414 (Purchase now)
P15-6 New Enhancements to the Automatic Noise Removal (ANR) System Utilizing Improved Noise Statistics and Multi-Band Processing—Shamail Saeed, Harinarayanan E. V., ATC Labs - Noida, India; Deepen Sinha, ATC Labs - Chatham, NJ, USA; Anibal Ferreira, University of Porto - Porto, Portugal, and ATC Labs, Chatham, NJ, USA
We recently introduced a novel Automatic Noise Reduction (ANR) algorithm for the removal of wideband stationary/nonstationary noise from audio. Current noise reduction techniques exhibit certain undesirable characteristics. Distortion and/or alteration of the audio characteristics is a common problem. User intervention in identifying the noise profile is sometimes necessary. ANR uses a novel framework employing dominant component subtraction and restoration and performs better than conventional techniques in subjective tests. Here we describe three enhancements to ANR. The first of these increases the level of noise removal for the special case of stationary background noise. The second is a new tool for improving the temporal envelope coherence and yields additional noise removal. The third is a multi-band processing tool for conditioning time-frequency envelope for reduced listener fatigue.
Convention Paper 7415 (Purchase now)
Last Updated: 20080612, tendeloo