AES London 2010 Sunday, May 23, 13:30 — 18:30
Paper Session P11
P11 - Network, Internet, and Broadcast Audio
Chair: Bob Walker, Consultant - UK
P11-1 A New Technology for the Assisted Mixing of Sport Events: Application to Live Football Broadcasting—Giulio Cengarle, Toni Mateos; Natanael Olaiz; Pau Arumí, Barcelona Media Innovation Centre - Barcelona, Spain
This paper presents a novel application for capturing the sound of the action during a football match by automatically mixing the signals of several microphones placed around the pitch and selecting only those microphones that are close to, or aiming at, the action. The sound engineer is presented with a user interface where he or she can define and move dynamically the point of interest on a screen representing the pitch, while the application controls the faders of the broadcast console. The technology has been applied in the context of a three-dimensional surround sound playback of a Spanish first-division match.
Convention Paper 8037 (Purchase now)
P11-2 Recovery Time of Redundant Ethernet-Based Networked Audio Systems—Maciej Janiszewski, Piotr Z. Kozlowski, Wroclaw University of Technology - Lower Silesia, Poland
Ethernet-based networked audio systems has become more popular among audio system designers. One of the most important issues that is available from networked audio system is the redundancy. The system can recover after different types of failures—cable or even device failure. Redundancy protocols implemented by audio developers are different than protocols known from computer networks, but both may be used in an Ethernet-based audio system. The most important attribute of redundancy in the audio system is a recovery time. This paper is a summary of research that was done at Wroclaw University of Technology. It shows the recovery time after different types of failures, with different network protocols implemented, for all possible network topologies in CobraNet and EtherSound systems.
Convention Paper 8038 (Purchase now)
P11-3 Upping the Auntie: A Broadcaster’s Take on Ambisonics—Chris Baume, Anthony Churnside, British Broadcasting Corporation Research & Development - UK
This paper considers Ambisonics from a broadcaster’s point of view: to identify barriers preventing its adoption within the broadcast industry and explore the potential advantages were it to be adopted. This paper considers Ambisonics as a potential production and broadcast technology and attempts to assess the impact that the adoption of Ambisonics might have on both production workflows and the audience experience. This is done using two case studies: a large-scale music production of “The Last Night of the Proms” and a smaller scale radio drama production of “The Wonderful Wizard of Oz.” These examples are then used for two subjective listening tests: the first to assess the benefit of representing height allowed by Ambisonics and the second to compare the audience’s enjoyment of first order Ambisonics to stereo and 5.0 mixes.
Convention Paper 8039 (Purchase now)
P11-4 Audio-Video Synchronization for Post-Production over Managed Wide-Area Networks—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
A persistent challenge with enabling remote collaboration for cinema post-production is synchronizing audio and video assets. This paper details efforts to guarantee that the sound quality and audio-video synchronization over networked collaborative systems will be measurably the same as that experienced in a traditional facility. This includes establishing a common word-clock source for all digital audio devices on the network, extending transport control and time code to all audio and video assets, adjusting latencies to ensure sample-accurate mixing between remote audio sources, and locking audio and video playback to within quarter-frame accuracy. We will detail our instantiation of these techniques at a demonstration given in December 2009 involving collaboration between a film editor in San Diego and a sound designer in Marin County, California.
Convention Paper 8040 (Purchase now)
P11-5 A Proxy Approach for Interoperability and Common Control of Networked Digital Audio Devices—Osedum P. Igumbor, Richard J. Foss, Rhodes University - Grahamstown, South Africa
This paper highlights the challenge that results from the availability of a large number of control protocols within the context of digital audio networks. Devices that conform to different protocols are unable to communicate with one another, even though they might be utilizing the same networking technology (Ethernet, IEEE 1394 serial bus, USB). This paper describes the use of a proxy that allows for high-level device interaction (by sending protocol messages) between networked devices. Furthermore, the proxy allows for a common controller to control the disparate networked devices.
Convention Paper 8041 (Purchase now)
P11-6 Network Neutral Control over Quality of Service Networks—Philip Foulkes, Richard Foss, Rhodes University - Grahamstown, South Africa
IEEE 1394 (FireWire) and Ethernet Audio/Video Bridging are two networking technologies that allow for the transportation of synchronized, low-latency, real-time audio and video data. Each networking technology has its own methods and techniques for establishing stream connections between the devices that reside on the networks. This paper discusses the interoperability of these two networking technologies via an audio gateway and the use of a common control protocol, AES-X170, to allow for the control of the parameters of these disparate networks. This control is provided by a software patchbay application.
Convention Paper 8042 (Purchase now)
P11-7 Relative Importance of Speech and Non-Speech Components in Program Loudness Assessment—Ian Dash, Australian Broadcasting Corporation - Sydney, NSW, Australia; Mark Bassett, Densil Cabrera, The University of Sydney - Sydney, NSW, Australia
It is commonly assumed in broadcasting and film production that audiences determine soundtrack loudness mainly from the speech component. While intelligibility considerations support this idea indirectly, the literature is very short on direct supporting evidence. A listening test was therefore conducted to test this hypothesis. Results suggest that listeners judge loudness from overall levels rather than speech levels. A secondary trend is that listeners tend to compare like with like. Thus, listeners will compare speech loudness with other speech content rather than with non-speech content and will compare loudness of non-speech content with other non-speech content more than with speech content. A recommendation is made on applying this result for informed program loudness control.
Convention Paper 8043 (Purchase now)
P11-8 Loudness Normalization in the Age of Portable Media Players—Martin Wolters, Harald Mundt, Dolby Germany GmbH - Nuremberg, Germany; Jeffrey Riedmiller, Dolby Laboratories Inc. - San Francisco, CA, USA
In recent years, the increasing popularity of portable media devices among consumers has created new and unique audio challenges for content creators, distributors as well as device manufacturers. Many of the latest devices are capable of supporting a broad range of content types and media formats including those often associated with high quality (wider dynamic-range) experiences such as HDTV, Blu-ray or DVD. However, portable media devices are generally challenged in terms of maintaining consistent loudness and intelligibility across varying media and content types on either their internal speaker(s) and/or headphone outputs. This paper proposes a nondestructive method to control playback loudness and dynamic range on portable devices based on a worldwide standard for loudness measurement as defined by the ITU. In addition the proposed method is compatible to existing playback software and audio content following the Replay Gain (www.replaygain.org) proposal. In the course of the paper the current landscape of loudness levels across varying media and content types is described and new and nondestructive concepts targeted at addressing consistent loudness and intelligibility for portable media players are introduced.
Convention Paper 8044 (Purchase now)
P11-9 Determining an Optimal Gated Loudness Measurement for TV Sound Normalization—Eelco Grimm, Grimm-Audio; Esben Skovenborg, tc-electronic; Gerhard Spikofski, Institute of Broadcast Technology - Berlin, Germany
Undesirable loudness jumps are a notorious problem in television broadcast. The solution consists in switching to loudness-based metering and program normalization. In Europe this development has been led by the EBU P/LOUD group, working toward a single target level for loudness normalization applying to all genres of programs. P/LOUD found that loudness normalization as specified by ITU-R BS.1770-1 works fairly well for the majority of broadcast programs. However, it was realized that wide loudness-range programs were not well-aligned with other programs when using ITU-R BS.1770-1 directly, but that adding a measurement-gate provided a simple yet effective solution. P/LOUD therefore conducted a formal listening experiment to perform a subjective evaluation of different gate parameters. This paper specifies the method of the subjective evaluation and presents the results in term of preferred gating parameters.
Convention Paper 8154 (Purchase now)
P11-10 Analog or Digital? A Case-Study to Examine Pedagogical Approaches to Recording Studio Practice—Andrew King, University of Hull - Scarborough, North Yorkshire, UK
This paper explores the use of digital and analog mixing consoles in the recording studio over a single drum kit recording session. Previous research has examined contingent learning, problem-solving, and collaborative learning within this environment. However, while there have been empirical investigations into the use of computer-based software and interaction around and within computers, this has not taken into consideration the use of complex recording apparatus. A qualitative case study approach was used in this investigation. Thirty hours of video data was captured and transcribed. A preliminary analysis of the data shows that there are differences between the types of problems encountered by learners when using either an analog or digital mixing console.
Convention Paper 8045 (Purchase now)