Technical Program
Papers
• Sessions by Industry
Category
• Detailed Calendar
• Convention Planner
• Paper Sessions
• Workshops
• Tutorials
• Master Classes
• Live Sound Seminars
• Exhibitor Seminars
• Special Events
• Student Program
• Historical
• Technical Tours
• Technical Council
• Standards Committee
• Heyser Lecture
|
AES Amsterdam 2008
Poster Session P12
P12 - Audio Archiving, Storage, Restoration, and Content Management & Audio Networking
Sunday, May 18, 14:00 — 15:30
P12-1 Drift, Wow, and Flutter Measurement and Reduction in Shrunken Movie Soundtracks—Przemek Maziewski, Adam Kupryjanow, Andrzej Czyzewski, Gdansk University of Technology - Gdansk, Poland
The paper presents the method and algorithms used to determine and reduce drift, wow, and flutter in shrunken movie tapes. The idea behind the algorithms is to use image processing for calculating the local tape shrinkage, which is one of the reasons for drift, wow, and flutter. The shrinkage can be calculated via analyzing the image height of: a movie frame, sprocket hole, pitch, or another standardized movie tape element; and then it can be expressed as the drift, wow, and flutter characteristic. After the characteristic determination both the soundtrack and movie frames can be corrected. The paper presents the description of the image based drift, wow, and flutter determination method and the experiments confirming the theoretical findings.
Convention Paper 7392 (Purchase now)
P12-2 The Norwegian Institute of Recorded Sound: From Collection to Archive to Public Private Partnership—Mark Drews, University of Stavanger - Stavanger, Norway; Jacqueline von Arb, Norwegian Institute of Recorded Sound - Stavanger, Norway
In 2006, the Norwegian Institute of Recorded Sound (NIRS) entered into a partnership with Memnon Audio Archiving Services to form MemNor, a commercial audio archiving service based in Stavanger, Norway. This paper traces the evolution of the Norwegian Institute of Recorded Sound from a private collection of music recordings to a municipally funded audio archive to a public private partnership and discusses the past, the current, and the future challenges involved. Details of ongoing activities is included.
Convention Paper 7393 (Purchase now)
P12-3 Cable-free Audio Delivery for Home Theater Entertainment Systems—Andreas Floros, Ionian University - Corfu, Greece; Nicolas-Alexander Tatlas, John Mourjopoulos, Dimitris Grimanis, University of Patras - Patras, Greece
Real time, multichannel audio content delivery over the air is expected to significantly simplify the interconnection complexity required for setting up typical home theater applications. However, despite the technological advantages of wireless networking standards related to high transmission rates and Quality-of-Service support, a number of issues has to be additionally addressed, such as multiple loudspeaker synchronization and packet delay/losses containing compressed quality and multiplexed audio data. In this paper further developments in the area of wireless audio delivery are presented by considering in detail multichannel reproduction for wireless home theater applications. Using both subjective and objective performance evaluation criteria, it is shown that cable-free multichannel audio playback is feasible under specific networking and audio coding conditions.
Convention Paper 7394 (Purchase now)
P12-4 Adaptive Playout for VoIP Based on the Enhanced Low Delay AAC Audio Codec—Jochen Issing, Nikolaus Färber, Manfred Lutzky, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) codec extends the application area of the Advanced Audio Coding (AAC) family toward high quality conversational services. Through the support of the full audio bandwidth at low delay and low bit rate, it offers excellent support for enhanced VoIP applications. In this paper we provide a brief overview of the AAC-ELD codec and describe how its codec structure can be exploited for IP transport. The overlapping frames and excellent error concealment make it possible to use frame insertion/deletion in order to adjust the playout time to varying network delay. A playout algorithm is proposed that estimates the jitter on the network and adapts the size of the de-jitter buffer in order to minimizes buffering delay and late loss. Considering typical network conditions and the same average delay, it is shown that the playout algorithm can reduce the loss rate by more than one magnitude compared to fixed playout.
Convention Paper 7395 (Purchase now)
Last Updated: 20080612, tendeloo
|
