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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.

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.

[pt] As tarefas de segmentação e classificação automáticas de áudio vêm se tornando cada vez mais importantes com o crescimento da produção e armazenamento de mídia digital. Este trabalho se baseia em características do padrão MPEG, que é considerado o padrão para acervos digitais, para gerir algoritmos de grande eficiência para realizar essas arefas. Ao passo que há muitos estudos trabalhando a partir do vídeo, o áudio ainda é pouco utilizado de forma eficiente para auxiliar nessas tarefas. Os algoritmos sugeridos partem da leitura apenas dos fatores de escala presentes no Layer 2 do áudio MPEG para ambas as tarefas. Com isso, é necessária a leitura da menor quantidade possível de informações, o que diminui significativamente o volume de dados manipulado durante a análise e torna seu desempenho excelente em termos de tempo de processamento. O algoritmo proposto para a classificação divide o áudio em quatro possíveis tipos: silêncio, fala, música e aplausos. Já o algoritmo de segmentação encontra as mudanças ignificativas de áudio, que são indícios de segmentos e mudanças de cena. Foram realizados testes com diferentes tipos de vídeos, e ambos os algoritmos mostraram bons resultados. / [en] With the growth of production and storing of digital media, audio segmentation and classification are becoming increasingly important. This work is based on characteristics of the MPEG standard, considered to be the standard for digital media storage and retrieval, to propose efficient algorithms to perform these tasks. While there are many studies based on video analysis, the audio information is still not widely used in an efficient way. The suggested algorithms for both tasks are based only on the scale factors present on layer 2 MPEG audio. That allows them to read the smallest amount of information possible, significantly diminishing the amount of data manipulated during the analysis and making their performance excellent in terms of processing time. The algorithm proposed for audio classification divides audio in four possible types: silent, speech, music and applause. The segmentation algorithm finds significant changes on the audio signal that represent clues of audio segments and scene changes. Tests were made with a wide range of types of video, and both algorithms show good results.

Streaming multimedia quality is impacted by two main factors: capacity constraint and packet loss. To match the capacity constraint while preserving real-time playout, media scaling can be used to discard the encoded multimedia content that has the least impact on perceived video quality. To limit the impact of lost packets, repair techniques, e.g. forward error correction (FEC), can be used to repair frames damaged by packet loss. However, adding data to facilitate repair requires further reduction of the original multimedia data, making the decision of how much repair data to use of critical importance. Assuming a limited network capacity and the availability of an estimate of the current packet loss rate along a flow path, selecting the best distribution of FEC packets for video frames with inherent interframe encoding dependencies can be cast as a constraint optimization problem that attempts to optimize the quality of the video stream. This thesis presents an Adjusting Repair and Media scaling with Operations Research (ARMOR) system. An analytical model is derived for streaming video with FEC and media scaling. Given parameters to represent network loss as well as video frame types and sizes, if the number of FEC packets per video frame type and media scaling pattern is specified, the model can estimate the video quality at the receiver side. The model is then used in an operations research algorithm to adjust the FEC strength and media scaling level to yield the best quality under the capacity constraint. Four different combinations of FEC type and media scaling method are studied: Media Independent FEC with Temporal Scaling (MITS), Media Independent FEC with Quality Scaling (MIQS), Media Independent FEC with Temporal and Quality Scaling (MITQS), and Media Dependent FEC with Quality Scaling (MDQS). The analytical experiments show: 1) adjusting FEC always achieves a higher video quality than streaming video without FEC or with a fixed amount of FEC; 2) Quality Scaling usually works better than Temporal Scaling; and 3) Media Dependent FEC (MDFEC) is typically less effective than Media Independent FEC (MIFEC). A user study is presented with results from 74 participants analysis shows that the ARMOR model can accurately estimate users¡¯perceptual quality. Well-designed simulations and a realistic system implementation suggests the ARMOR system can practically improve the quality of streaming video.

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.

Error Correction and Concealment of Block Based, Motion-Compensated Temporal Prediction, Transform Coded Video David L. Robie 133 Pages Directed by Dr. Russell M. Mersereau The use of the Internet and wireless networks to bring multimedia to the consumer continues to expand. The transmission of these products is always subject to corruption due to errors such as bit errors or lost and ill-timed packets; however, in many cases, such as real time video transmission, retransmission request (ARQ) is not practical. Therefore receivers must be capable of recovering from corrupted data. Errors can be mitigated using forward error correction in the encoder or error concealment techniques in the decoder. This thesis investigates the use of forward error correction (FEC) techniques in the encoder and error concealment in the decoder in block-based, motion-compensated, temporal prediction, transform codecs. It will show improvement over standard FEC applications and improvements in error concealment relative to the Motion Picture Experts Group (MPEG) standard. To this end, this dissertation will describe the following contributions and proofs-of-concept in the area of error concealment and correction in block-based video transmission. A temporal error concealment algorithm which uses motion-compensated macroblocks from previous frames. A spatial error concealment algorithm which uses the Hough transform to detect edges in both foreground and background colors and using directional interpolation or directional filtering to provide improved edge reproduction. A codec which uses data hiding to transmit error correction information. An enhanced codec which builds upon the last by improving the performance of the codec in the error-free environment while maintaining excellent error recovery capabilities. A method to allocate Reed-Solomon (R-S) packet-based forward error correction that will decrease distortion (using a PSNR metric) at the receiver compared to standard FEC techniques. Finally, under the constraints of a constant bit rate, the tradeoff between traditional R-S FEC and alternate forward concealment information (FCI) is evaluated. Each of these developments is compared and contrasted to state of the art techniques and are able to show improvements using widely accepted metrics. The dissertation concludes with a discussion of future work.

<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>

Made available in DSpace on 2016-06-02T19:05:17Z (GMT). No. of bitstreams: 1 DissMML.pdf: 1887422 bytes, checksum: 213fe8c28a477776b42e8c6ef861a171 (MD5) Previous issue date: 2003-08-21 / This work describes the implementation of a support structure to 3D virtual networked games, based on the emergent standard multiuser MPEG-4 in a Gnutella hybrid peer-to-peer network. This solution minimizes the disadvantages of the existent hybrid solutions, that they are based on proxies, which have to be re-configured whenever a new application appears in the net. For that, the code that implements the Gnutella network it was modified from way to include a service of search of games and of active sessions. Two defined components and specified by the emergent standard MPEG-4 MU were implemented and integrated into the Gnutella network for games session control and updating of the scenes. When a node designated as controller leaves, another should assume in a fast and continuous way. Those, among other challenges in the implementation of multiuser games as peer-to-peer applications, they will be discussed in this work, together with the integration of the technologies Gnutella and MPEG-4 MU. The evaluation of this implementation allowed to conclude the some topics about the adaptation of those networks in the support to applications that demand continuous collaboration, as it is the case of a 3D game in that multiples participant constantly alter the scene and also the viability of implementing a session controller in one of the nodes of the network. / Este trabalho descreve a implementação de uma estrutura de suporte a jogos virtuais 3D em rede, baseada no padrão emergente MPEG-4 multiusuário em uma rede Gnutella peer-to-peer híbrida. Esta solução minimiza as desvantagens das soluções híbridas existentes, que são baseadas em proxies, as quais têm que ser re-configuradas sempre que uma nova aplicação surge na rede. Para isso, o código que implementa a rede Gnutella foi modificado de modo a incluir um serviço de busca de jogos e de sessões ativas. Dois componentes definidos e especificados pelo padrão emergente MPEG-4 MU foram implementados e integrados à rede Gnutella para controle de sessão de jogos e atualização das cenas. Quando um nodo designado como controlador sai, outro deve assumir de forma rápida e contínua. Esses, entre outros desafios na implementação de jogos multiusuário como aplicações peer-to-peer, serão discutidos neste trabalho, juntamente com a integração das tecnologias Gnutella e MPEG-4 MU. A avaliação desta implementação nos permitiu chegar a algumas conclusões sobre a adequação dessas redes no suporte a aplicações que exigem colaboração contínua, como é o caso de um jogo 3D em que múltiplos participantes alteram a cena constantemente e também a viabilidade de se implementar um controlador de sessão em um dos nodos da rede.

Pour avoir une idée sur la qualité du réseau, la majorité des acteurs concernés (opérateurs réseau, fournisseurs de service) se basent sur la Qualité de Service (Quality of Service). Cette mesure a montré des limites et beaucoup d’efforts ont été déployés pour mettre en place une nouvelle métrique qui reflète, de façon plus précise, la qualité du service offert. Cette mesure s’appelle la qualité d’expérience (Quality of Experience). La qualité d’expérience reflète la satisfaction de l’utilisateur par rapport au service qu’il utilise. L’évaluation de la qualité d’expérience est devenue primordiale pour les fournisseurs de services et les fournisseurs de contenus. Cette nécessité nous a poussés à innover et mettre en place des nouvelles méthodes pour estimer la QoE. Dans cette thèse, nous travaillons sur l’estimation de la QoE dans le cas des communications Voix sur IP et dans le cas de la vidéo sur IP. Nous étudions les performances et la qualité des codecs iLBC, Speex et Silk pour la VoIP et les codecs MPEG-2 et H.264/SVC pour la vidéo sur IP. Nous étudions l’impact que peut avoir la majorité des paramètres réseaux, des paramètres sources (au niveau du codage) et destinations (au niveau du décodage) sur la qualité finale. Afin de mettre en place des outils précis d’estimation de la QoE en temps réel, nous nous basons sur la méthodologie Pseudo-Subjective Quality Assessment. La méthodologie PSQA est basée sur un modèle mathématique appelé les réseaux de neurones artificiels. En plus des réseaux de neurones, nous utilisons la régression polynomiale pour l’estimation de la QoE dans le cas de la VoIP. / Quality of Experience (QoE) is the key criteria for evaluating the Media Services. Unlike objective Quality of Service (QoS) metrics, QoE is more accurate to reflect the user experience. The Future of Internet is definitely going to be Media oriented. Towards this, there is a profound need for an efficient measure of the Quality of Experience (QoE). QoE will become the prominent metric to consider when deploying Networked Media services. In this thesis, we provide several methods to estimate the QoE of different media services: Voice and Video over IP. We study the performance and the quality of several VoIP codecs like iLBC, Speex and Silk. Based on this study, we proposed two methods to estimate the QoE in real-time context, without any need of information of the original voice sequence. The first method is based on polynomial regression, and the second one is based on an hybrid methodology (objective and subjective) called Pseudo-Subjective Quality Assessment. PSQA is based on the artificial neural network mathematical model. As for the VoIP, we propose also a tool to estimate video quality encoded with MPEG-2 and with H.264/SVC. We studied also the impact of several network parameters on the quality, and the impact of some encoding parameters on the SVC video quality. We tested also the performance of several SVC encoders and proposed some SVC encoding recommendations.