v7.0, 20040922, me
Friday, October 29, 9:00 am 11:30 am
Session E AUDIO (INCLUDING TELEPHONY) OVER NETWORKS
Chair: Karlheinz Brandenburg, Fraunhofer Institute for Media Technology, Ilmenau, Germany
E-1 An RTP Payload Format for MIDIJohn Lazzaro, John Wawrzynek, University of California, Berkeley, CA, USA
The Real-Time Protocol (RTP) is an extensible transport for sending media streams over Internet Protocol packet networks. We describe a new payload format that extends RTP to transport MIDI (the Musical Instrument Digital Interface command language). The payload format encodes all commands that may legally appear on a MIDI 1.0 DIN cable. The format is suitable for interactive applications (such as the remote operation of musical instruments) and content-delivery applications (such as file streaming). The format may be used over lossy unicast and multicast networks, and defines tools for graceful recovery from packet loss to support use over lossy unicast and multicast networks (including wireless networks). Stream behavior, including the MIDI rendering method, may be specified during session setup. Rendering methods are specified using the extensible Multipurpose Internet Mail Extensions (MIME) registry.
Convention Paper 6207
E-2 Network Time Delay and Ensemble Accuracy: Effects of Latency, AsymmetryChris Chafe, Michael Gurevich, Stanford University, Stanford, CA, USA
Pairs of musicians were placed apart in isolated rooms and asked to clap a rhythm together. Each person monitored the others sound via headphones and microphone pickup, which was as close as possible. Time delay from source to listener was manipulated across trials. Trials were recorded and clap onset times were measured with an event detection algorithm. Longer delays produced increasingly severe tempo deceleration, and shorter delays (< 11.5 ms) produced a modest, but surprising, acceleration. The papers goal is to characterize effects of delay on rhythmic accuracy and identify the region most conducive to ensemble playing. The results have implication for networked musical performance. Network delay is a function of transmission distance and/or internetworking (routing) delays. The findings suggest that sensitive ensemble performance can be supported over rather long paths (e.g., San Francisco to Denver at about 20 ms, one-way). The finding that moderate amounts of delay are beneficial to tempo stability seems, at first glance, counterintuitive. We discuss the observed effect.
Convention Paper 6208
E-3 Practical Issues in Objective Speech Quality Assessment with ITU-T P.862Ville-Veikko Mattila, Nokia Research Center, Tampere, Finland; Antti Kurittu, Nokia Networks, Helsinki, Finland
The ITU-T P.862 Recommendation specifies the Perceptual Evaluation of Speech Quality (PESQ) algorithm that is the current industrial standard for the objective, intrusive assessment of the one-way speech quality of narrowband networks and speech codecs. The practical use of P.862, however, has raised several questions about the robustness, applicability, and accuracy of the algorithm. The current paper presents results from an investigation of these issues. The characteristics of test signals and the interferences of signal interfaces are shown to have a significant effect on the quality assessment with P.862. A measurement procedure is proposed to define the accuracy of P.862 in the comparison of different or unknown technologies. It is concluded that various test factors should be carefully defined so as different P.862 measurements to be comparable.
Convention Paper 6209
E-4 Integrated High-Performance Multichannel Audio InterconnectionMichael Page, Gary Cook, Peter Eastty, Richard Marshall, Sony Pro-Audio Lab, Oxford, UK
This paper describes a family of related technologies that provide a very high-performance audio interconnection system for professional applications. The first element is an advanced point-to-point audio interconnection based on a 100 Megabit Ethernet physical layer, which is currently undergoing AES standardization. The second element is a complementary gigabit-based high-capacity interconnection for applications such as backbone links. These are all linked together with a router technology that provides both low-latency audio channel routing and packet-switched control data routing. These technologies together provide a flexible, high-bandwidth digital audio infrastructure, which is ideally suited for applications requiring low, deterministic latency and high reliability.
Convention Paper 6210
E-5 An Internet Protocol (IP) Sound SystemTom Blank, Bob Atkinson, Michael Isard, James D. Johnston, Kirk Olynyk, Microsoft Corporation, Redmond, WA, USA
We describe a system that applies Internet concepts and software techniques to deliver audio from source to speakers using common computing hardware. The techniques overcome clocking and jitter problems. Microphones built into each transducer to locate loudspeakers allow the system to identify speaker placement, automatically compensate for off-center listening locations, adjust for inter-channel gain, delay, and do frequency response matching. A research prototype demonstrates the concepts and measures the resulting quality.
Convention Paper 6211