120th AES Convention - Paris, France - Dates: Saturday May 20 - Tuesday May 23, 2006 - Porte de Versailles

4 Day Planner
Paper Sessions
Exhibitor Seminars
Application Seminars
Student Program
Special Events
Technical Tours
Heyser Lecture
Tech Comm Mtgs
Standards Mtgs
Hotel Reservation

AES Paris 2006

Home | Technical Program | Exhibition | Visitors | Students | Press

Last Updated: 20060403, mei

P4 - Audio in Computers & Audio Networking

Saturday, May 20, 14:00 — 17:20

Chair: Christophe Musialik, Algorithmix, Germany - Waldshut-Tiengen, Germany

P4-1 Newly Established IEC Standard on Audio Quality Measurement of Personal ComputersKenji Kurakata, National Institute of Advanced Industrial Science and Technology (AIST) - Tsukuba, Ibaraki, Japan; Masamichi Furukawa, Kenwood Corporation - Hachioji, Tokyo, Japan; JEITA Project Group - Chiyoda, Tokyo, Japan
A new IEC standard on audio quality measurement of personal computers (PCs) was published in December 2005, entitled IEC 61606-4 “Audio and audiovisual equipment—Digital audio parts—Basic measurement methods of audio characteristics —Part 4: Personal computer.” This standard prescribes methods for measuring PC audio quality, taking into account requirements of measuring conditions of PCs. Furthermore, a new measure of audio signal quality, short-term distortion, was introduced to describe PC-specific noise problems. This paper presents an outline of this standard.

Presentation is scheduled to begin at 14:00
Convention Paper 6659 (Purchase now)

P4-2 Scene Description Model and Rendering Engine for Interactive Virtual AcousticsJean-Marc Jot, Jean-Michel Trivi, Creative Advanced Technology Center - Scotts Valley, CA, USA
Interactive environmental audio spatialization technology has become commonplace in personal computers, where its primary current application is video game soundtrack rendering. The most advanced PC audio platforms available can spatialize 100 or more sound sources simultaneously over headphones or multichannel home theater systems, and employ multiple reverberation engines to simulate complex acoustical environments. This paper reviews the main features of the EAX environmental audio programming interface and its relation to the I3DL2 and MPEG-4 standards. A statistical reverberation model is introduced for simulating per-source distance and directivity effects. An efficient spatial reverberation and mixing architecture is described for the spatialization of multiple sound sources around a virtual listener navigating across multiple connected virtual rooms including acoustic obstacles.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at 14:20
Convention Paper 6660 (Purchase now)

P4-3 Intelligent Audio for GamesCol Walder, Revolution Recording - Sheffield, UK
Providing interactive audio for computer games has traditionally been seen as a challenge, particularly given the technological limitations of games consoles. With current advances in technology, however, there is the potential to take advantage of the benefits of interactivity. This paper proposes the use of Artificial Intelligence (AI) routines to control in-game audio with a focus on implementing techniques used in film sound for drama-based games. Soar architecture is presented as a good candidate for developing audio AI for games.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at 14:40
Convention Paper 6661 (Purchase now)

P4-4 A Frame Loss Concealment Technique for MPEG-AACSang-Uk Ryu, Kenneth Rose, University of California at Santa Barbara - Santa Barbara, CA, USA
An efficient method is proposed for frame loss concealment within the advanced audio coding (AAC) decoder, which can effectively mitigate the adverse impact of frame loss on reconstruction quality. The spectral information of the lost frame is first estimated in the modified discrete cosine transform (MDCT) domain via the known frame interpolation approach. The interpolated MDCT coefficients are then further refined by magnitude scaling and sign correction, which are differently designed for tonal and noise components of the source signal. In noise-like spectral bins, a shaped-noise insertion technique is employed to adjust the interpolated coefficients, while coefficients in tone-dominant bins are refined by magnitude scaling and novel sign correction techniques so as to optimize the fit of the corresponding time reconstruction with available partial signal information from neighboring frames. Subjective quality evaluations demonstrate that the proposed method achieves significant quality improvement over the shaped-noise insertion method adopted in commercial AAC decoders.

Presentation is scheduled to begin at 15:00
Convention Paper 6662 (Purchase now)

P4-5 Multiple Description Error Mitigation Techniques for Streaming Compressed Audio over a 802.11 Wireless NetworkCorey Cheng, Wenyu Jiang, Dolby Laboratories - San Francisco, CA, USA
This paper presents several multiple description (MD) coding techniques for error mitigation of compressed audio streamed over an 802.11b/g wireless network. Loosely speaking, an MD encoder generates several descriptions of the same source, and an MD decoder recreates the best estimate of the source from the descriptions it successfully receives. We propose a design for an MD architecture and simulate its integration into the AAC codec. We use packet loss traces gathered from an actual 802.11 b/g network to simulate the proposed codec’s error mitigation properties for various network traffic conditions. We examine how tuning several of the proposed codec’s parameters would affect the sound quality and overall bit rate of the proposed codec. Specifically, we show how interleaving, renormalization, and low-frequency variance estimation techniques can be used in conjunction with hierarchical correlating transforms to improve the sound quality of multiple description codecs.

Presentation is scheduled to begin at 15:20
Convention Paper 6663 (Purchase now)

P4-6 Single Frequency Networks for FM RadioPierre Soelberg, Selberg Broadcast & IT Consult - Købehavn S, Denmark
Single Frequency Networks (SFN) and Near Single Frequency Networks (NSFN) are usually not considered suitable for FM radio. Some countries are now replanning their FM bands for the use of (N)SFN, in order to make space for more stations. Even though some stations use it, like a station covering a highway, replanning the FM-band with the use of SFN for a whole country, is a different thing. The first country to do this was the Netherlands, and the first experiences with it are not as good as expected. The requirements for synchronization of FM transmitters used for (N)SFN are explained, and SFN networks are tested from real transmitter sites. The result is a proposed correction for the Dutch norm.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at 15:40
Convention Paper 6664 (Purchase now)

P4-7 A Paradigm for Wireless Digital Audio Home EntertainmentNikos Kokkos, University of Patras - Patras, Greece; Andreas Floros, Ionian University - Corfu, Greece; Nicolas-Alexander Tatlas, John Mourjopoulos, University of Patras - Patras, Greece
Despite recent advances in wireless networking technology, real-time streaming of CD-quality digital audio remains a challenging topic. In this paper a set of applications following the server-client model was developed, facilitating the transmission and playback of PCM-coded audio over wireless links. The implementation is based on typical personal computer (PC) platforms interconnected with off-the-shelf wireless networking hardware. Performance evaluation tests are presented under different networking parameters and link conditions, leading to an optimal set of parameters for high-quality wireless digital audio delivery.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at 16:00
Convention Paper 6665 (Purchase now)

P4-8 Online Acoustic Measurements in a Networked Audio SystemAki Härmä, Philips Research - Eindhoven, The Netherlands
A networked audio system consists of audio devices that are in the same physical environment and are connected by a network. The network connection makes it possible to perform continuous acoustic measurements between the devices. Such measurement data can be used, for example, to control the playback by the properties of the actual sound field produced. Continuous acoustic measurement involves transmission of audio data over the network. The bit-rate of the audio data should be low because the measurement is not a primary function of the networked system. In this paper we introduce a robust system for the networked audio measurements where the bit rate sent over the network is small.

Presentation is scheduled to begin at 16:20
Convention Paper 6666 (Purchase now)

P4-9 Design and Installation of Recording Studios for Vocational TrainingChris Bradley, James Watt College of Further & Higher Education - Greenock, Inverclyde, Scotland, UK; Billy Law, Mediaspec - Glasgow, Scotland, UK
This paper describes the design and installation of new recording studios for training of music and sound production allowing unparalleled direct student hands-on tuition. The design allows simultaneous recording from the live rooms to all twelve control rooms via digital distribution, enabling individual set up for a recording session, multitrack recording and subsequent mixdown. All recording sessions are saved to a centralized server that allows back-up and uploading to and from any other control room. Students can therefore import their work into any of the other control rooms at any time. Networking is through Gigabit Ethernet so transfer of work is fast, and students have their own password protected space, learning the importance of file management.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at16:40
Convention Paper 6667 (Purchase now)

P4-10 Flexible, High Speed Audio Networking for Hotels and Convention CentersRichard Foss, Rhodes University - Grahamstown, South Africa; Jun-ichi Fujimori, Yamaha Corporation - Hamamatsu, Japan; Nyasha Chigwamba, Rhodes University - Grahamstown, South Africa; Brad Klinkradt, Harold Okai-Tettey, Networked Audio Solutions - Grahamstown, South Africa
This paper describes the use of mLAN (music Local Area Network) to solve the problem of audio routing within hotels and convention centers. mLAN is a FireWire-based digital network interface technology that allows professional audio equipment, PCs, and electronic instruments to be easily and efficiently interconnected using a single cable. In order to solve this problem, an existing mLAN Connection Management Server, augmented with additional functionality, has been utilized. A graphical client application has been created that displays the various locations within a hotel/convention center and sends out appropriate routing messages in Extensible Mark-up Language (XML) to an mLAN connection management server. The connection management server, in turn, controls a number of mLAN audio distribution boxes on the FireWire network.

[Associated Poster Presentation in Session P10, Sunday, May 21, at 11:00]

Presentation is scheduled to begin at 17:00
Convention Paper 6668 (Purchase now)

  (C) 2006, Audio Engineering Society, Inc.