Digital recording of performing arts: formats and conversion

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Operating-system support for distributed multimedia

Multimedia applications place new demands upon processors, networks and operating systems. While some network designers, through ATM for example, have considered revolutionary approaches to supporting multimedia, the same cannot be said for operating systems designers. Most work is evolutionary in nature, attempting to identify additional features that can be added to existing systems to support multimedia. Here we describe the Pegasus project's attempt to build an integrated hardware and operating system environment from\ud the ground up specifically targeted towards multimedia

A continuous media transport protocol

A protocol for the transmission of digitized audio and video data must schedule data transmission such that each audio or video sample is available in the receiver\u27s buffer by its play-out deadline without overflowing the allotted buffer space at the receiver. While traditional data communication protocols provide error-free data transmission through the use of time-outs and retransmissions, the unbounded number of retransmissions that a packet may suffer results in an unboundable network transmission delay. Thus, such protocols are not appropriate for the real-time transmission of digitized audio and video data;This dissertation examines the network environments likely to be used for digitized audio and video data transmission and develops an audio/video transport-layer communication protocol which for such network environments. A unique feature of this protocol is the calculation of a transmission schedule which can minimize the buffer requirements at the receiver while still meeting the real-time play-out deadlines of the individual audio and video streams

A framework for realistic 3D tele-immersion

Meeting, socializing and conversing online with a group of people using teleconferencing systems is still quite differ- ent from the experience of meeting face to face. We are abruptly aware that we are online and that the people we are engaging with are not in close proximity. Analogous to how talking on the telephone does not replicate the experi- ence of talking in person. Several causes for these differences have been identified and we propose inspiring and innova- tive solutions to these hurdles in attempt to provide a more realistic, believable and engaging online conversational expe- rience. We present the distributed and scalable framework REVERIE that provides a balanced mix of these solutions. Applications build on top of the REVERIE framework will be able to provide interactive, immersive, photo-realistic ex- periences to a multitude of users that for them will feel much more similar to having face to face meetings than the expe- rience offered by conventional teleconferencing systems

Multimedia player implementation on embedded systems

Thesis (Master)--Izmir Institute of Technology, Electronics and Communication Engineering, Izmir, 2008Includes bibliographical references (leaves: 85-88)Text in English; Abstract: Turkish and Englishxi, 90 leavesThere has been a surge in the number of digital audio and video content in recent years. Advances in the compression and storage technologies and improvements in the speed of internet connection have enabled widespread use of multimedia content. A wide variety of devices have been introduced to decode and play these media contents.Initially designed as a mere voice communication device, the mobile phones nowadays come equipped with a variety of multimedia capabilities including media players despite their limited system resources.Nowadays, huge servers host dramatically increased audio and video contents Users prefer to watch these contents while streaming rather than downloading them first. So, streaming media players are responsible to present multimedia contents without annoying interrupts.This thesis firstly introduces challenges in design and implementation of a streaming media player and then proposes solutions. Main challenges are keeping audio-video synchronization and server-client synchronization and detecting stream type, handling of multithreaded operations and buffer management. Audio-video synchronization problem is solved by using audio as master stream. Server-client synchronization problem is solved by designing a playback mechanism that keeps synchronization with the server by tuning the playback rate of a streaming media without losing lip-sync between audio and video. The proposed streaming player also has a feature of identifying the type of a media stream very rapidly without using a discrete stream inspector module. The presented design is heavily multithreaded which is implemented on Linux platform, moreover it is also convenient for and implementable on any multithreaded platform

Binaural Spatialization for 3D immersive audio communication in a virtual world

Realistic 3D audio can greatly enhance the sense of presence in a virtual environment. We introduce a framework for capturing, transmitting and rendering of 3D audio in presence of other bandwidth savvy streams in a 3D Tele-immersion based virtual environment. This framework presents an efficient implementation for 3D Binaural Spatialization based on the positions of current objects in the scene, including animated avatars and on the fly reconstructed humans. We present a general overview of the framework, how audio is integrated in the system and how it can exploit the positions of the objects and room geometry to render realistic reverberations using head related transfer functions. The network streaming modules used to achieve lip-synchronization, high-quality audio frame reception, and accurate localization for binaural rendering are also presented. We highlight how large computational and networking challenges can be addressed efficiently. This represents a first step in adequate networking support for Binaural 3D Audio, useful for telepresence. The subsystem is successfully integrated with a larger 3D immersive system, with state of art capturing and rendering modules for visual data

Error resilient video transcoding for robust inter-network communications using GPRS

A novel fully comprehensive mobile video communications system is proposed in this paper. This system exploits the useful rate management features of the video transcoders and combines them with error resilience for transmissions of coded video streams over general packet radio service (GPRS) mobileaccess networks. The error-resilient video transcoding operation takes place at a centralized point, referred to as a video proxy, which provides the necessary output transmission rates with the required amount of robustness. With the use of this proposed algorithm, error resilience can be added to an already compressed video stream at an intermediate stage at the edge of two or more different networks through two resilience schemes, namely the adaptive intra refresh (AIR) and feedback control signaling (FCS) methods. Both resilience tools impose an output rate increase which can also be prevented with the proposed novel technique in this paper. Thus, an error-resilient video transcoding scheme is presented to give robust video outputs at near target transmission rates that only require the same number of GPRS timeslots as the nonresilient schemes. Moreover, an ultimate robustness is also accomplished with the combination of the two resilience algorithms at the video proxy. Extensive computer simulations demonstrate the effectiveness of the proposed system

Database of audio records

Diplomka a prakticky castDiplome with partical part

Network streaming and compression for mixed reality tele-immersion

Bulterman, D.C.A. [Promotor]Cesar, P.S. [Copromotor