AES New York 2009
Paper Session P17

P17 - Audio Networks

Monday, October 12, 9:00 am — 12:00 pm
Chair: Richard Foss, Rhodes University - Grahamstown, South Africa

P17-1 Performance Metrics for Network Audio Systems: Methodology and Comparison—Nicolas Bouillot, Mathieu Brulé, Jeremy R. Cooperstock, McGill University - Montreal, Quebec, Canada
Network audio transmission is becoming increasingly popular within the broadcast community, with applications to Voice over IP (VoIP) communications, audio content distribution, and radio broadcast. Issues of end-to-end latency, jitter, and overall quality, including glitches of the delivered signal, all impact on the value of the technology. Although considerable literature exists comparing audio codecs, little has been published to compare systems in terms of their real-word performance. In response, we describe methods for accurately assessing the quality of audio streams transmitted over networks. These methods are then applied to an empirical evaluation of several audio compression formats supported by different streaming engines.
Convention Paper 7940 (Purchase now)

P17-2 An Integrated Connection Management and Control Protocol for Audio Networks—Richard Foss, Rhodes University - Grahamstown, South Africa; Robby Gurdan, Bradley Klinkradt, Nyasha Chigwamba, Universal Media Access Networks (UMAN) - Dusseldorf, Germany
With the advent of digital networks that link audio devices, there is a need for a protocol that integrates control and connection management, allows for streaming of all media content such as audio and video between devices from different manufacturers, and that provides a common approach to the control of these devices. This paper proposes such a protocol, named XFN (currently being standardized as part of the AES X170 project). XFN is an IP-based peer to peer network protocol in which any device on the network may send or receive connection management, control, and monitoring messages. Essential to the XFN protocol is the fact that each parameter in a device can be addressed via a hierarchical structure that reflects the natural layout of the device.
Convention Paper 7941 (Purchase now)

P17-3 Mixing Console Design Considerations for Telematic Music Applications—Jonas Braasch, Rensselaer Polytechnic Institute - Troy, NY, USA; Chris Chafe, Stanford University - Stanford, CA, USA; Pauline Oliveros, Doug Van Nort, Rensselaer Polytechnic Institute - Troy, NY, USA
This paper describes the architecture for a new mixing console that was especially designed for telematic live music collaborations. The prototype mixer is software-based and programmed in Pure Data. It has many traditional features but also a number of extra modules that are important for telematic projects: transmission test unit, latency meter, remote data link, auralization unit, remote sound level calibration unit, remote monitoring, and a synchronized remote audio recording unit.
Convention Paper 7942 (Purchase now)

P17-4 Comparison of Receiver-Based Concealment and Multiple Description Coding in an 802.11-Based Wireless Multicast Audio Distribution Network—Marcus Purat, Tom Ritter, Beuth Hochschule für Technik Berlin - Berlin, Germany
This paper presents aspects of a study of different methods to mitigate the impact of packet loss in a wireless distribution network on the subjective quality of compressed high fidelity audio. The system was simulated in Matlab based on parameters of an 802.11a WLAN in multicast-mode and the Vorbis codec. To aid the selection of the most appropriate packet loss concealment strategy not only the additional bandwidth, the processing requirements or the latency need to be considered. One important differentiating factor is the perceived subjective audio quality. Therefore an accurate estimate of the subsequent audio quality is required. Several simulation-based methods using psychoacoustic models of the human hearing system to quantify the subjective audio quality are compared.
Convention Paper 7943 (Purchase now)

P17-5 Audio-Over-IP Acceptance Test Strategy—Matthew O'Donnell, BSkyB (British Sky Broadcasting) - London, UK
Ensuring the integrity of an application that delivers audio-over-IP through Ethernet demands thorough acceptance testing during the development cycle, due to the effect of the potentially volatile “Best Effort” nature of IP transport upon performance of the application. This paper investigates attributes of protocols used on top of IP that must be taken into account during development and their impact on an audio-over-IP's Quality of Experience to the end user.
Convention Paper 7944 (Purchase now)

P17-6 Long-Distance Uncompressed Audio Transmission over IP for Postproduction—Nathan Brock, Michelle Daniels, University of California, San Diego - La Jolla, CA, USA; Steve Morris, Skywalker Sound - Marin County, CA, USA; Peter Otto, University of California, San Diego - La Jolla, CA, USA
The highly distributed nature of contemporary cinema postproduction has led many to believe that high-speed networking of uncompressed audio could significantly improve workflow efficiency. This paper will provide an overview of several significant issues with long-distance networking, including synchronization, latency, bandwidth limitations, and control protocols. We will present a recent networked postproduction demonstration, in which audio assets in Seattle, San Francisco, and San Diego along with local video assets were streamed to and controlled from a single DAW. These results are expected to lead to persistent wide-area networked postproduction environments to remotely access and control audiovisual assets.
Convention Paper 7945 (Purchase now)