AES Munich 2009
Paper Session P1
P1 - Audio for Telecommunications
Thursday, May 7, 09:00 — 11:30
Chair: Damian Murphy
P1-1 20 Things You Should Know Before Migrating Your Audio Network to IP—Simon Daniels, APT - Belfast, Northern Ireland, UK
For many years, synchronous networks have been considered the industry standard for audio transport worldwide. Balanced analog copper circuits, microwave, and synchronous based systems such as V.35/X.21 or T1/E1 have been the traditional choice for studio transmitter and inter-studio links in professional audio broadcast networks. Readily available from all major service providers, the popularity of synchronous links has been largely due to the fact that they offer dedicated, reliable, point-to-point and bi-directional communication at guaranteed data and error rates. However, the reign of synchronous links as the preferred choice for STLs is currently coming under threat from a new challenger, in the form of IP-based network technology.
Convention Paper 7651 (Purchase now)
P1-2 Deploying Large Scale Audio IP Networks—Kevin Campbell, APT - Belfast, Northern Ireland, UK
This paper will examine the key considerations for those interested in deploying large-scale ip audio networks. It will include an overview of the main challenges and draw on the experience of national public broadcasters who have already migrated to IP. We will provide an overview of the key concerns such as jitter, delay, and link reliability that are valid for an IP network of any size. However, this paper will focus mainly on the issues arising from the greater complexity and
scale of large national and country-wide deployments. The paper will use illustrations and network applications from real-world deployments to illustrate the points.
Paper presented by Hartmut Foerster
Convention Paper 7652 (Purchase now)
P1-3 A Spatial Filtering Approach for Directional Audio Coding—Markus Kallinger, Henning Ochsenfeld, Giovanni Del Galdo, Fabian Kuech, Dirk Mahne, Richard Schultz-Amling, Oliver Thiergart, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
In hands-free telephony, spatial filtering techniques are employed to enhance intelligibility of speech. More precisely, these techniques aim at reducing the reverberation of the desired speech signal and attenuating interferences. Additionally, it is well-known that the spatially separate reproduction of desired and interfering sources enhance intelligibility of speech. For the latter task, Directional Audio Coding (DirAC) has proven to be an efficient method to capture and reproduce spatial sound. In this paper we propose a spatial filtering processing block, which works in the parameter domain of DirAC. Simulation results show that compared to a standard beamformer the novel technique offers significantly higher interference attenuation, while introducing comparably low distortion of the desired signal. Additional subjective tests of speech intelligibility confirm the instrumentally obtained results.
Convention Paper 7653 (Purchase now)
P1-4 A New Bandwidth Extension for Audio Signals without Using Side-Information—Kha Le Dinh, Chon Tam Le Dinh, Roch Lefebvre, Université de Sherbrooke - Sherbrooke, Quebec, Canada
The use of narrow bandwidth (300 – 3400 Hz) in the current telephone network limits the perceptual quality of telephone conversations. Changing to wideband network is a solution that can help to improve quality, but it will need a long time to upgrade. Thus, bandwidth extension can be seen as an alternative solution during the transition time. A new bandwidth extension method is presented in this paper. Without using any side-information, the proposed method can be applied as a post-processing step at the terminal devices, maintaining the compatibility to the current telephone network, and thus, no modification is needed in the network nodes. Experimental results show that the proposed solution can help to improve significantly the perceptual quality of narrowband telephone signal.
Convention Paper 7654 (Purchase now)
P1-5 Feature Selection vs. Feature Space Transformation in Music Genre Classification Framework—Hanna Lukashevich, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany
Automatic classification of music genres is an important task in music information retrieval research. Nearly all state-of-the-art music genre recognition systems start from the feature extraction block. The extracted acoustical features often could tend to be correlated or/and redundant, which can cause various difficulties in the classification stage. In this paper we present a comparative analysis on applying supervised Feature Selection (FS) and Feature Space Transformation (FST) algorithms to reduce the feature dimensionality. We discuss pros and cons of the methods and weigh the benefits of each one against the others.
Convention Paper 7655 (Purchase now)