10 research outputs found

    Edge-guided image gap interpolation using multi-scale transformation

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    This paper presents improvements in image gap restoration through the incorporation of edge-based directional interpolation within multi-scale pyramid transforms. Two types of image edges are reconstructed: 1) the local edges or textures, inferred from the gradients of the neighboring pixels and 2) the global edges between image objects or segments, inferred using a Canny detector. Through a process of pyramid transformation and downsampling, the image is progressively transformed into a series of reduced size layers until at the pyramid apex the gap size is one sample. At each layer, an edge skeleton image is extracted for edge-guided interpolation. The process is then reversed; from the apex, at each layer, the missing samples are estimated (an iterative method is used in the last stage of upsampling), up-sampled, and combined with the available samples of the next layer. Discrete cosine transform and a family of discrete wavelet transforms are utilized as alternatives for pyramid construction. Evaluations over a range of images, in regular and random loss pattern, at loss rates of up to 40%, demonstrate that the proposed method improves peak-signal-to-noise-ratio by 1–5 dB compared with a range of best-published works

    Edge-Guided Image Gap Interpolation Using Multi-Scale Transformation

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    Adaptive delivery of real-time streaming video

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    Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 87-92).While there is an increasing demand for streaming video applications on the Internet, various network characteristics make the deployment of these applications more challenging than traditional Internet applications like email and the Web. The applications that transmit data over the Internet must cope with the time-varying bandwidth and delay characteristics of the Internet and must be resilient to packet loss. This thesis examines these challenges and presents a system design and implementation that ameliorates some of the important problems with video streaming over the Internet. Video sequences are typically compressed in a format such as MPEG-4 to achieve bandwidth efficiency. Video compression exploits redundancy between frames to achieve higher compression. However, packet loss can be detrimental to compressed video with interdependent frames because errors potentially propagate across many frames. While the need for low latency prevents the retransmission of all lost data, we leverage the characteristics of MPEG-4 to selectively retransmit only the most important data in order to limit the propagation of errors. We quantify the effects of packet loss on the quality of MPEG-4 video, develop an analytical model to explain these effects, and present an RTP-compatible protocol-which we call SR-RTP--to adaptively deliver higher quality video in the face of packet loss. The Internet's variable bandwidth and delay make it difficult to achieve high utilization, Tcp friendliness, and a high-quality constant playout rate; a video streaming system should adapt to these changing conditions and tailor the quality of the transmitted bitstream to available bandwidth. Traditional congestion avoidance schemes such as TCP's additive-increase/multiplicative/decrease (AIMD) cause large oscillations in transmission rates that degrade the perceptual quality of the video stream. To combat bandwidth variation, we design a scheme for performing quality adaptation of layered video for a general family of congestion control algorithms called binomial congestion control and show that a combination of smooth congestion control and clever receiver-buffered quality adaptation can reduce oscillations, increase interactivity, and deliver higher quality video for a given amount of buffering. We have integrated this selective reliability and quality adaptation into a publicly available software library. Using this system as a testbed, we show that the use of selective reliability can greatly increase the quality of received video, and that the use of binomial congestion control and receiver quality adaptation allow for increased user interactivity and better video quality.by Nicholas G. Feamster.M.Eng

    Robust mode selection for block-motion-compensated video encoding

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    Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.Includes bibliographical references (p. 129-132).by Raynard O. Hinds.Ph.D

    Détection et dissimulation de la détérioration visuelle issue du décodage de séquences H.264 corrompues

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    Compte tenu de leur nature, les réseaux mobiles sont plus fortement enclins à la corruption de données que leurs contreparties filaires. Même si les données se rendent à destination, les bits endommagés entrainent le rejet des paquets qui les encapsulent. Ces pertes ont un impact important sur la qualité de l’expérience de l’utilisateur lors de la consultation de flux vidéos ou de la vidéophonie, et ce, surtout lorsque la retransmission n’est pas envisageable. On restreint l’usage des approches conventionnelles de résilience aux erreurs, tels la retransmission de trames ou l’ajout de trames redondantes, car elles imposent un fardeau considérable aux réseaux mobiles, ceux-ci étant déjà fortement achalandés. Dans cet ouvrage, nous proposons la réutilisation sélective des données corrompues afin d’augmenter la qualité visuelle de séquences endommagées. Cette sélection est guidée par une nouvelle approche de détection de la détérioration visuelle dans le domaine des pixels. Elle combine la mesure des effets de bloc (discontinuités spatiales en bordure de blocs) à l’estimation du mouvement. Notre méthode a été testée sur un ensemble de 17 séquences QCIF codées en H.264 avec des QP de 16 à 28 et soumis à des taux d’erreurs de 0.0004 à 0.0032. Nos résultats de simulation démontrent qu’il est possible de décoder des trames corrompues. La probabilité d’un décodage réussi varie de 20 % à 70 % selon les paramètres d’encodage et le taux d’erreurs subies lors du transport. De plus, notre algorithme, développé en fonction de la norme H.264, réussit à effectuer le bon choix de 81 % à 86 % et 88 % à 91 % des cas (selon les conditions). Lorsque notre algorithme est combiné au décodeur de référence H.264, nous observons un gain moyen 0.65 dB à 0.86 dB de PSNR par rapport au calque de trame et calque de tranche respectivement pour nos conditions de test
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