ITC/USA 2010 Conference Proceedings / The Forty-Sixth Annual International Telemetering Conference and Technical Exhibition / October 25-28, 2010 / Town and Country Resort & Convention Center, San Diego, California / Synchronized video is crucial for data acquisition and telecommunication applications. For real-time applications, out-of-sync video may cause jitter, choppiness and latency. For data analysis, it is important to synchronize multiple video channels and data that are acquired from PCM, MIL-STD-1553 and other sources. Nowadays, video codecs can be easily obtained to play most types of video. However, a great deal of effort is still required to develop the synchronization methods that are used in a data acquisition system. This paper will describe several methods that TTC has adopted in our system to improve the synchronization of multiple data sources.

Dans le cadre de la nouvelle société de l'information multimédia et communicante, cette thèse propose des contributions méthodologiques et techniques relatives à la représentation, l'animation et la transmission des objets virtuels.<br /><br />Les méthodes existantes sont analysées de façon comparée et les performances des standards multimédias actuels évaluées en termes de réalisme d'animation et de débit de transmission. Pour surmonter les limitations mises en évidence, un nouveau cadre de modélisation et d'animation de personnages virtuels est proposé. Le modèle SMS (Skeleton, Muscle and Skin), fondé sur le concept de contrôleur de déformation d'un maillage, est introduit et sa formulation mathématique développée. Le graphe de scène 3D et le flux de compression associés à SMS sont décrits. L'approche SMS est évaluée dans le cadre d'un nouveau service de transmission télévisuelle d'un signeur virtuel destinés aux déficients auditifs. Le modèle SMS a été promu dans le standard MPEG-4 version 5.

In the immediate future, multimedia product distribution through the Internet will become main stream. However, it can also have the side effect of unauthorized duplication and distribution of multimedia products. That effect could be a critical challenge to the legal ownership of copyright and intellectual property. Many schemes have been proposed to address these issues; one is digital watermarking which is appropriate for image and video copyright protection. Videos distributed via the Internet must be processed by compression for low bit rate, due to bandwidth limitations. The most widely adapted video compression standard is MPEG-4. Discrete cosine transform (DCT) domain watermarking is a secure algorithm which could survive video compression procedures and, most importantly, attacks attempting to remove the watermark, with a visibly degraded video quality result after the watermark attacks. For a commercial broadcasting video system, real-time response is always required. For this reason, an FPGA hardware implementation is studied in this work. This thesis deals with video compression, watermarking algorithms and their hardware implementation with FPGAs. A prototyping VLSI architecture will implement video compression and watermarking algorithms with the FPGA. The prototype is evaluated with video and watermarking quality metrics. Finally, it is seen that the video qualities of the watermarking at the uncompressed vs. the compressed domain are only 1dB of PSNR lower. However, the cost of compressed domain watermarking is the complexity of drift compensation for canceling the drifting effect.

Pour pallier un manque d'outils spécialisés que les enseignants puissent s'approprier et utiliser en classe, l'objectif de cette étude, à l'intersection entre informatique et éducation, vise la réalisation d'un logiciel permettant de créer des supports de cours numériques. Basé sur la technologie MPEG-4 (compression, composition multimédia, interactivité) l'outil est cohérent avec une architecture de mutualisation de contenus. Trois axes de recherche en découlent. Le premier décrit la réalisation d'un prototype d'outil dont l'IHM basée sur celle des « assistants d'installation » apporte une simplicité d'utilisation pour les enseignants « novices », et qui dispose de modules destinés à enrichir les différents médias par des fonctionnalités. Le deuxième axe présente le « module vidéo » qui permet de créer des zones sensibles et des marques temporelles. Le dernier axe décrit un espace 3D de représentation des contenus multimédias dont les niveaux de structuration définis sont conservés.

<p>In this thesis work a part of the video coding standard H.264 has been implemented. The part of the video coder that is used to code the I-frames has been implemented to see how well suited it is for regular image coding. The big difference versus other image coding standards, such as JPEG and JPEG2000, is that this video coder uses both a predictor and a transform to compress the I-frames, while JPEG and JPEG2000 only use a transform. Since the prediction error is sent instead of the actual pixel values, a lot of the values are zero or close to zero before the transformation and quantization. The method is much like a video encoder but the difference is that blocks of an image are predicted instead of frames in a video sequence.</p> / <p>I det här examensarbetet har en del av videokodningsstandarden H.264 implementerats. Den del av videokodaren som används för att koda s.k. I-bilder har implementerats för att testa hur bra den fungerar för ren stillbildskodning. Den stora skillnaden mot andra stillbildskodningsmetoder, såsom JPEG och JPEG2000, är att denna videokodaren använder både en prediktor och en transform för att komprimera stillbilderna, till skillnad från JPEG och JPEG2000 som bara använder en transform. Eftersom prediktionsfelen skickas istället för själva pixelvärdena så är många värden lika med noll eller nära noll redan innan transformationen och kvantiseringen. Metoden liknar alltså till mycket en ren videokodare, med skillnaden att man predikterar block i en bild istället för bilder i en videosekvens.</p>

The purpose of this thesis is to present an attractive service for the third generation mobile network that includes streaming video. A prototype application for this service is to be built. The technique behind streaming video is to be presented so that it comes clear what kind of problems and solutions that are associated with streaming. Finally, a platform for streaming video is to be tested and evaluated through different channels. The attractive service presented in this thesis is MMS - Multimedia Messaging Service. Today's popular SMS is evolving beyond text to multimedia. Multimedia will be part of the next generation messaging service called MMS, which will, in an advanced shape, include streaming video. MMS is expected to be a successful service for the next generation?s cell phones. The WAP Forum and the 3GPP industry groups are responsible for standardizing MMS. The standard defines an MMS architecture, which has a number of key elements that interact with each other. The prototype application that was built is called mVideo Messaging and is an MMS that is built on the basis of the MMS standard. The kernel of the prototype is a platform from PacketVideo that makes it possible to stream video over wireless networks. Theories and tests makes it clear that the parameters affecting the video quality can be found at the source/target while coding and compressing, as well as at the streaming-channel. At the channel there are above all three network problems - packet loss, end-to-end delay and delay jitter. To deal with these matters, new protocols have been developed. At the source/target it is important use an efficient compression scheme. MPEG-4 is a new compression scheme that suits very well for streaming video through wireless channels. MPEG- 4 make use of scalability, is object oriented, and is optimized for streaming between 9,6 kbps and 4 Mbps. The service proposed in this thesis as a future service for 3G is practicable. It is also shown that the service can be built using the technology of today.

<p>The purpose of this thesis is to present an attractive service for the third generation mobile network that includes streaming video. A prototype application for this service is to be built. The technique behind streaming video is to be presented so that it comes clear what kind of problems and solutions that are associated with streaming. Finally, a platform for streaming video is to be tested and evaluated through different channels. </p><p>The attractive service presented in this thesis is MMS - Multimedia Messaging Service. Today's popular SMS is evolving beyond text to multimedia. Multimedia will be part of the next generation messaging service called MMS, which will, in an advanced shape, include streaming video. MMS is expected to be a successful service for the next generation?s cell phones. </p><p>The WAP Forum and the 3GPP industry groups are responsible for standardizing MMS. The standard defines an MMS architecture, which has a number of key elements that interact with each other. The prototype application that was built is called mVideo Messaging and is an MMS that is built on the basis of the MMS standard. The kernel of the prototype is a platform from PacketVideo that makes it possible to stream video over wireless networks. </p><p>Theories and tests makes it clear that the parameters affecting the video quality can be found at the source/target while coding and compressing, as well as at the streaming-channel. At the channel there are above all three network problems - packet loss, end-to-end delay and delay jitter. To deal with these matters, new protocols have been developed. At the source/target it is important use an efficient compression scheme. MPEG-4 is a new compression scheme that suits very well for streaming video through wireless channels. MPEG- 4 make use of scalability, is object oriented, and is optimized for streaming between 9,6 kbps and 4 Mbps. </p><p>The service proposed in this thesis as a future service for 3G is practicable. It is also shown that the service can be built using the technology of today.</p>

Thesis (M. S.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2008. / Committee Chair: Bing Benny; Committee Co-Chair: Fred B-H. Juang; Committee Member: Gee-Kung Chang. Part of the SMARTech Electronic Thesis and Dissertation Collection.

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.