In This Section
- 137th Audio Engineering Society Convention Breaks Records and Draws Acclaim from Attendees, Exhibitors and Presenters Alike
- Convention reminds West-Coast audio community, “If It’s About Audio, It’s At AES!”
- AES 2014 Election Results
- The results are in!
- Time to Vote: 2014 AES Elections
- Deadline was Friday, July 11th
- AES Showcases Latest Audio Innovations and Best Practices at 136th International Convention in Berlin
- Successful Technical Program and Exhibition draw high attendance and positive response from attendees and exhibitors alike
Paper Session 4
Paper Session 4: Broadcasters, Teleconferencing, and Wireless
Sonja Langhans (IRT), session moderator
Sunday, Nov. 20, 9:30am, CalIT2 Auditorium
9:30. Lars Jonsson and Mathias Coinchon: EBU Standardisation of Audio over IP for Contribution Systems
Audio over IP end units are increasingly being used in radio operations for streaming of radio programmes to and from studio sites. The IP networks used are often well-managed private networks with controlled Quality of Service. The open Internet is also increasingly being used for radio contribution. Radio correspondents now have a choice to use either ISDN, or the Internet to deliver their reports. ISDN services will be closed down in some countries. The European Broadcasting Union project ACIP has created an interoperability standard, which has been developed jointly by members of the EBU and manufacturers. The document, TECH 3326 , has since its publication quickly been implemented and used by more than 20 manufacturers worldwide for end units for streaming contribution over IP. Several plug tests have been made between manufacturers to enhance and confirm the standard. The requirements for interoperability are based on the use of RTP over UDP for the audio session and SIP for signaling. IETF RFC documents for commonly used audio formats in radio contribution define the packet payload audio structure. This document gives an overview of the standard and its applications in broadcasting.
10:00. Christian Hoene, Michael Haun, and Mansoor Hyder: Measuring Complexity and Latency of a Spatial Audio Teleconferencing System
Spatial audio increases the quality of teleconferencing. Multiple talkers can be separated by taking advantage of the Cocktail-Party Effect and participants can be identified by their locations in a virtual acoustic environment. In this paper, we present performance measurements of our spatial teleconferencing application 3DTel. We use ours implementation to see to what extend spatial audio increases the latency and computational complexity of teleconferencing sessions.
10:30. Jeffrey Berryman: Technical Criteria for Professional Media Networks
Recent standards in digital networking have rekindled industry hope for universal media networking standards. In the professional realm, these standards promise to enable fully interoperable multivendor audio/video system solutions that offer unprecedented levels of functionality and flexibility at relatively low cost. However, in order to realize this potential, the standards must meet certain specific requirements of professional media system applications. In this report, those requirements are summarized.
11:00. Karen Collins, Peter Taillon, and Bill Kapralos: Experimenting With a Framework for Networked Mobile Audio Arrays
This project investigates the use of networked smartphones for distributed audio. The core application is to convert smartphones to loudspeakers, to which audio events can be transmitted over a wireless network in an array. The system has applications in business, education, gaming, and social environments. We report on initial experiments with the system, where investigation revolves around gauging the end-user experience. We also provide design suggestions for future social-audio games based on mobile technology.
11:30. Seppo Nikkilä: Introducing wireless organic digital audio: a multichannel streaming audio network based on IEEE 802.11 standards
Until recently the multichannel audio market has been dominated by systems with synthetic or blended audio where new channels are created from the two original stereo channels or mixed from a few independent signals. The majority of these systems only have the CD, or worse yet, MP3 sound quality. This may be acceptable for computer games and adventure movies but is certainly not good enough for serious music work. To opposite this unsatisfactory trend we have introduced our organic sound format with eight fully independent original channels each with 24-bit samples at the sampling rate of 192 kHz. Further we introduce the wireless organic digital audio distribution network to transmit this group of signals to a dynamic group of loudspeakers with intelligent, high quality receivers. Our system features isochronous transfer with very low system delay, excellent channel-to-channel synchronization, clocks drift compensation, and complete system configuration flexibility.