Scalable Video Coding

The Discrete Cosine Transform is statistically optimal transform for first order Markov signals, which is widely used in image and video coding. However, in intra frame coding of H.264/AVC, it is knowm that after directional intra prediction, there is still anisotropic features left in residue signals. And the features are related to the intra prediction mode. In order to represent the corresponding features, mode dependent directional transform (MDDT), which is based on Karhunen Loeve transform (KLT), were derived and adopted into JMKTA software for High Efficiency Video Coding (HEVC). However, within the MDDT scheme, each prediction mode has one correspondent transform. In some cases, the transform basis functions can not represent the residue signal very well. Therefore, in this paper, we extend the transform candidates for each prediction mode. By introducing such a transform competition scheme, the proposed algorithm can capture some irregular residue pattern, therefore achieving a better energy compaction. Experimental results indicate that these transforms can improve the compression efficiency of the intra residue. 10-0106320635©2010 APSIPA. All rights reserved.

This paper examines a multiple-access (MAC) channel where users are
powered by batteries with significant internal resistance. Due to this
resistance, only a portion of the power drawn from the battery reaches the
load, while the remaining power is dissipated as heat, depending on the
discharge current. Users also consume continuous circuit power during
transmission, with no power consumption when the device is off. We
derive the maximum cumulative rates and define the feasible rate regions
under different transmission scenarios. In conventional time-division
multiple access (TDMA), where users transmit in separate time slots, the
optimal aggregate rate is not achieved when internal resistance is present,
unlike in ideal battery systems. To address this, we propose a hybrid
NOMA-TDMA scheme that combines non-orthogonal multiple access
(NOMA) with TDMA, allocating slots for both orthogonal single-user and
non-orthogonal joint transmission. Through convex optimization and
numerical analysis, we demonstrate that the maximum achievable rate
regions of pure NOMA and TDMA are strictly contained within those of
the hybrid scheme. The framework is extended from single-user and two
user cases to general multi-user MAC(U≥2), highlighting the trade-off
between circuit energy losses and resistive losses. The hybrid NOMA
TDMA approach consistently provides performance gains by balancing
these losses and achieving larger feasible rate regions.

We propose a robust and scalable transmission scheme based on Multiple Description (MD) coding. Each description is encoded using a standard video codec such as MPEG-2 or H.264. This scheme provides equal or better robustness with respect to other solutions such as Forward Error Correction (FEC), at lower system complexity; also, it provides scalability by selectively dropping descriptions. 1

A new video coding paradigm, distributed video coding, has been the focus of many recent studies. In this paper we present a simple video coding framework, based on the principles of distributed coding, that can be applied to any video coding standards with minor modifications. The framework allows spatial scalability of the non-reference frames, and does not need any feedback channel between the encoder and decoder. The complexity in the encoder is reduced since the non-reference frames are coded at lower spatial resolution. At the decoder, side-information is generated using the reference frames through motion estimation and compensation. Results using the H.263+ standard are shown.

Distributed Video Coding is a new video coding paradigm that enables video codecs with reversed complexity. It is well-known that the performance of any distributed video coder is heavily dependent on the quality of the side information generated by motion-estimation at the decoder. As a continuation of previous works, in this paper we propose to use a super-resolution method to up-sample the non-key frames using the key frames as reference, in order to improve the side information generation in a distributed video coding scheme with spatial scalability. The framework, proposed in previous works, enables reduced encoding complexity by reduced spatial-resolution encoding of the non-reference frames, followed by Wyner-Ziv coding of the Laplacian residue. The side information generation is iterative, in the first iteration the high-frequency data of matching blocks from the key frames are added to the low-resolution blocks of the interpolated downsampled frames. Results are presented using the state-of-the-art video coding standard H.264/AVC.

A motion-based side-information generation scheme with semi super-resolution for a scalable Wyner-Ziv coder framework is introduced. It is known that the performance of any Wyner-Ziv coder is heavily dependent on the efficiency of the side-information generation. We propose an iterative block based scheme to generate a semi super-resolution frame using the past and future reference frames which should be coded at full-resolution. To enable this side-information generation the framework should allow for low encoding complexity, reducing the spatial resolution only in the non-reference frames. The enhancement layer is produced using a residual frame of the reduced resolution encoded frame. The decoder first decodes the low resolution base layer and iteratively generates the side-information, along with channel decoding, to obtain a higher quality version of the decoded frame. Results of the implementation of the framework using the motion-based side-information in the H.263+ and H.264 standards are presented.

We propose a mixed resolution framework based on full resolution key frames and spatial-reduction-based Wyner-Ziv coding of intermediate nonreference frames. Improved ratedistortion performance is achieved by enabling better sideinformation generation at the decoder side and better rateallocation at the encoder side. The framework enables reduced encoding complexity by low resolution encoding of the nonreference frames, followed by Wyner-Ziv coding of the Laplacian residue. The quantized transform coefficients of the residual frame are mapped to cosets without the use of a feedback channel. A study to select optimal coding parameters in the creation of the memoryless cosets is made. Furthermore, a correlation estimation mechanism that guides the parameter choice process is proposed. The decoder first decodes the low resolution base layer and then generates a super-resolved side-information frame at full resolution using past and future key frames. Coset decoding is carried using side-information to obtain a higher quality version of the decoded frame. Implementation results are presented for the H.264/AVC codec.

The paper proposes a research on the process of replacing the classical radio broadcast (analog)
with the digital one, with a focus on Norway, the first country in the world to make this change in 2017. We
propose to find out what were the steps, the costs, the technical or human needs, but also what were the
possible changes in radio production that occurred after digitization. Also, we will try to we see the
advantages of the digital radio, to the detriment of the analogue one.

In this paper a simple and robust method for estimation of distorted video quality, which is perceived by human observer in mobile video streaming applications, is proposed and assessed. Increasing bandwidth of mobile communication systems expand the variety of offered multimedia services such as video streaming. However, the quality of these services is very dependent on rapidly varying mobile communication conditions. Most widely used video quality estimation methods, such as Peak Signal to Noise Ratio (PSNR), Structural Similarity (SSIM), and Video Quality Metric (VQM) are based on the presence of full or reduced reference video. These methods could be used to assess video quality of video transmission system only during test stage and in the limited number of scenarios. In order to assess user experienced video quality in real conditions, methods with no reference must be employed. Such existing methods as video quality metric use bit-error rate that has low correlation with human perceived video quality. More precise methods usually are too complex and require too much processing power that cannot be tolerated in handheld mobile devices. In this paper it is shown that developed no reference low complexity video quality estimation method based on H.264/AVC video stream packet structure delivers estimate of received video quality comparable with results of subjective MOS tests.

Video monitoring applications for underground rely on wireless train-to-wayside communication systems which require high data rate as well as high Quality of Service (QoS) level. In order to satisfy both constraints we propose a combined source and channel coding approach in the context of MIMO (Multiple Input Multiple Output) video transmission. In the present case, MIMO transmission is based on the PHY layer of IEEE 802.11n Wi-Fi standard currently deployed in a railway tunnels. Two different strategies are studied: first, the association between Multiple Description Coding (MDC) and a STBC (Space Time Block Code) MIMO scheme is considered when no channel information is available at transmitter side. In the case when perfect channel information is available at transmitter side (CSIT), a Singular Value Decomposition of the MIMO channel is possible. This transmission scheme is then associated with scalable video coding, which consists here in the separation of the scene into different Regions Of Interest (ROI). The creation of the regions of interest is based on the Flexible Macroblock Ordering (FMO) technique introduced in the new H.264/AVC compression standard. The stream associated to the area with the maximal perceptual relevance is transmitted on the eigen-channel with the higher gain. Consequently, this strategy which provides unequal protection against channel errors, allows guaranteeing better robustness and acceptable reconstructed video quality at the control-centre. The two different strategies of transmission have been evaluated thanks to realistic simulations. Two antenna configurations representative of real cases encountered in railway tunnels are considered. The channel model is generated by using the correlation based Kronecker model obtained by computing the channel matrix with a 3D ray tracing tool. Simulation results show that the two proposed solutions allow enhancing the reconstructed video quality

When communicating at a very-low bit-rate, video coders are unable to preserve high visual quality for all images. A selection of key regions according to human viewing and understanding may therefore be useful: it allows extracting essential features and to code them with a high quality, while the remainder of the image is coarsely transmitted. Such an approach requires a signal decomposition allowing an adaptive spatial variant bit allocation. Matching Pursuits (MP) is an iterative procedure leading to such a signal representation. Incorporated within a motion compensated video codec, MP allows for adaptive representation of the Displaced Frame Difference. Bit allocation procedures for Matching Pursuits differ from the ones used for DCT-or other transform-based signal representations. For transformbased methods, the signal is first projected on a complete set of orthogonal base functions. Coefficients are then quantized. The quantization selects which information is transmitted. So, the selection is constrained by the chosen transform. On the contrary, MP explicitly select the information to be transmitted among a large and overcomplete set of functions and quantize it according to a fixed and constant step. For MP, bit allocation is equivalent to function selection. This makes Matching Pursuits well-suited for spatial variant bit allocation, since functions matched to important objects can be chosen. In this paper, we first show the abilities of Matching Pursuits to allocate an available bit-budget according to a semantic understanding of the scene. This semantic information can be provided either automatically or interactively. Second, the possibility to incorporate perceptive criteria within the MP coding algorithm is investigated.

In this paper, Reduced Resolution Depth Compression (RRDC) is proposed for Scalable Video Coding (SVC) to improve the 3D video rate distortion performance. RRDC is applied by using Down-Sampling and Up-Sampling (DSUS) of the depth data of the stereoscopic 3D video. The depth data is down-sampled before SVC encoding and up-sampled after SVC decoding operation. The proposed DSUS method reduces the overall bit rates and consequently: 1) improves SVC rate distortion for 3D video, particularly at lower bit rates in error free channels; and 2) improves 3D SVC performance for 3D transmission in error prone channels. The objective quality evaluation of the stereoscopic 3D video yields higher PSNR values at low bit rates for SVCDSUS compared to the original SVC (SVC-Org), which makes it advantageous in terms of reduced storage and bandwidth requirements. Moreover, the subjective quality evaluation of the stereoscopic 3D video further confirmed that the perceived stereoscopic 3D video quality of the SVC-DSUS is very similar to the stereoscopic 3D video of the SVC-Org by up to 98.2%.

No início da década passada, observou-se um crescimento significativo do uso da Internet, assim como do número de computadores interligados. Aplicações de rede, tal como FTP, WWW e Telnet, são tradicionalmente baseadas em protocolos unicast (um-para-um). Com o surgimento de IP Multicast, criou-se o potencial para uma nova classe de aplicações, as multiparticipantes. Certas aplicações necessitam a transmissão confiável de um conjunto de pacotes para um grupo de receptores; esta pesquisa investiga protocolos que satisfazem esse requisito de maneira escalável na Internet. Em particular, são comparados através de análise e simulação um conjunto de modelos de protocolos multicast pertencentes à classe de protocolos baseados em polling (mecanismo de controle de implosão baseado no controle sobre respostas de receptores) em oposição a outras técnicas de obtenção de feedback. Estes protocolos (ou modelos gerais) são extraídos da literatura científica. As métricas utilizadas na comparação são taxa de transferência (throughput) e custo de rede (largura de banda).

Scalable multimedia data transmission are subject to specific constraints such as the Quality of Service (QoS) of sensitivity classes and the transmission rate (yielding a maximum size of each frame to send). Many scalable source decoders are used to discarding data than processing an erroneous stream. This featuring class structure is helpful to define a strategy that determines the maximum number of classes to send and delivers the suitable protection and transmission scheme (coding rates and modulation) to apply in accordance with the transmission constraints. It leads to the possible truncation of frame parts transmitted with an Unequal Error Protection (UEP) scheme for severe channel conditions. In a MPEG-4 speech frames context, we compare our approach to other methods using equal error and existing UEP schemes. It results in a significant improvement of the Peak SNR (PSNR) quality in poor channel transmission conditions.

The Data Distribution Service (DDS) middleware is enjoying a rapid adoption in high-performance, mission-critical networks. At the same time, the H.264 Scalable Video Coding (SVC) has been recently standardized and it is deemed to be an effective solution for video streaming over a channel with timevarying bandwidth, like the wireless one. In these conditions, it is critical to adapt the video bit-rate to the actual wireless capacity, and bit-rate adaptation is extremely simple for a H.264 SVC video. In this paper we devise, evaluate and demonstrate a technique for streaming H.264 SVC video over a DDS middleware. The contribution is threefold: i) we design a structure of the DDS data-unit able to carry H.264 SVC video-units; ii) we devise a receiver-driven rate-control mechanism based on our DDS dataunit and exploiting specific DDS functionality; iii) we implement and show the effectiveness of our mechanism in an 802.11 wireless scenario, comparing our proposal with other solutions.

This paper is among the first works to document experimental results for application-aware H.264 Scalable Video Coding (SVC) support over Wireless LANs. Application-aware support is achieved by introducing a bandwidth throttling device, called Virtual BottleNeck (VBN), before the WLAN Access Point. Throttling is set to a bandwidth slightly smaller than the actual WLAN capacity (either known or estimated), so that all packet/frame losses occur inside the VBN. Here, loss events are controlled by a scheduling mechanism devised to operate with information taken from the H.264 Network Abstraction Layer Units (NALUs). Despite its relative simplicity, the implemented scheduler exhibits effective video adaptation performance and close to optimal bandwidth efficiency. Setting up the trial was not trivial due to the lack of suitable publicly available tools. We filled this gap by implementing and integrating several separate software modules, e.g., streaming server, NALU dependency filtering, video frame concealment, etc. As a final result, the experimental trial supports the full delivery chain for an SVC stream with the only limitation of an off-line stream conversion for play-out and Peak Signal-to-Noise Ratio (PSNR) measurement purposes, due to the unavailability of real time SVC players. 1 The source code of the software as well as all the videos produced and discussed throughout the remainder of this paper can be found at the URL -for download purposes note that videos are 300-400 MB each.

Although Peer-to-Peer live video streaming over wireless mesh networks (WMNs) is considered a promising technology, some important challenges such as interference, mobility and limited available resources in gadgets (e.g. Smartphones and Tablets) may significantly reduce the perceived video quality. GREENIE and MATIN, in our previous studies, provided an efficient routing protocol in WMNs and a video streaming method based on random network coding (RNC), respectively. Therefore, their integration in the form of an enhanced framework, named GAZELLE, can considerably increase the video quality on these gadgets by decreasing the video distortion, dependency distortion, initial start-up delay and end-to-end delay. Findings using a precise simulation in OMNET?? show that GAZELLE noticeably outperforms other frameworks. GAZELLE not only decreases the imposed computational complexity and transmission overhead due to using RNC considerably, but it also efficiently routes video packets through those gadgets which does not require neither high battery energy sources nor high CPU power.

Transmission of multimedia, especially transmission of image/video information, requires high reliability and high digital transmission speed. But, in case of mobile communication systems, since transmission reliability is degraded by multipath fading also the frequency bandwidth is severely limited, it is difficult to realize the multimedia transmission without applying techniques to overcome these limitations. For achieving a high quality and high speed in digital image transmission in a band-limited fading channel, Hierarchical modulation is a better method. This method is also constellation is used to give different degrees of error protection for the most important information of the image content. A higher order i.e. 256-HQAM where non-uniform signal constellation with hierarchical oriented OFDM modulator has been proposed. Proposed method is used to provide lesser BER and Bandwidth efficiency in the compressed image data at lower channel Signal to Noise Ratio (SNR). An information theoretical approach is applied to analyze the performance of proposed technique. Using the approach, it is quantitatively evaluated how much the proposed technique outperforms. BER and PSNR of reconstructed image of proposed methodology, which include uniform and non-uniform constellations, over AWGN environments, have been calculated to analyze the performance. The performance of proposed and existing technique has been evaluated by doing simulation using MATLAB 7.6 (R2008a).

HTTP Adaptive Streaming of video content is becoming an integral part of the Internet and accounts for the majority of today's traffic. Although Internet bandwidth is constantly increasing, video compression technology plays an important role and the major challenge is to select and set up multiple video codecs, each with hundreds of transcoding parameters. Additionally, the transcoding speed depends directly on the selected transcoding parameters and the infrastructure used. Predicting transcoding time for multiple transcoding parameters with different codecs and processing units is a challenging task, as it depends on many factors. This paper provides a novel and considerably fast method for transcoding time prediction using video content classification and neural network prediction. Our artificial neural network (ANN) model predicts the transcoding times of video segments for state of the art video codecs based on transcoding parameters and content complexity. We evaluated our method for two video codecs/implementations (AVC/x264 and HEVC/x265) as part of large-scale HTTP Adaptive Streaming services. The ANN model of our method is able to predict the transcoding time by minimizing the mean absolute error (MAE) to 1.37 and 2.67 for x264 and x265 codecs, respectively. For x264, this is an improvement of 22% compared to the state of the art.

HTTP adaptive video streaming is a widespread and sought-after technology on the Internet that allows clients to dynamically switch between different stream qualities presented in the bitrate ladder to optimize overall received video quality. Currently, there exist several approaches of different complexity for building such a ladder. The simplest method is to use a static bitrate ladder, and the more complex one is to compute a per-title encoding ladder. The main drawback of these approaches is that they do not provide bitrate ladders for scenes with different visual complexity within the video. Moreover, most modern methods require additional computationally-intensive test encodings of the entire video to construct the convex hull, used to calculate the bitrate ladder. This paper proposes a new fast per-scene encoding approach called FAUST based on 1) quick entropy-based scene detection and 2) prediction of optimized bitrate ladder for each scene using an artificial neural network. The results show that our model reduces the mean absolute error to 0.15, the mean square error to 0.08, and the bitrate to 13.5% while increasing the difference in video multimethod assessment fusion to 5.6 points.

HTTP adaptive streaming of video content becomes an integrated part of the Internet and dominates other streaming protocols and solutions. The duration of creating video content for adaptive streaming ranges from seconds or up to several hours or days, due to the plethora of video transcoding parameters and video source types. Although, the computing resources of different transcoding platforms and services constantly increase, accurate and fast transcoding time prediction and scheduling is still crucial. We propose in this paper a novel method called fast video transcoding time prediction and scheduling (FastTTPS) of x264 encoded videos based on three phases: (i) transcoding data engineering, (ii) transcoding time prediction, and (iii) transcoding scheduling. The first phase is responsible for video sequence selection, segmentation and feature data collection required for predicting the transcoding time. The second phase develops an artificial neural network (ANN) model for segment...

HTTP Adaptive Streaming of video content is becoming an integral part of the Internet and accounts for the majority of today's traffic. Although Internet bandwidth is constantly increasing, video compression technology plays an important role and the major challenge is to select and set up multiple video codecs, each with hundreds of transcoding parameters. Additionally, the transcoding speed depends directly on the selected transcoding parameters and the infrastructure used. Predicting transcoding time for multiple transcoding parameters with different codecs and processing units is a challenging task, as it depends on many factors. This paper provides a novel and considerably fast method for transcoding time prediction using video content classification and neural network prediction. Our artificial neural network (ANN) model predicts the transcoding times of video segments for state of the art video codecs based on transcoding parameters and content complexity. We evaluated our method for two video codecs/implementations (AVC/x264 and HEVC/x265) as part of large-scale HTTP Adaptive Streaming services. The ANN model of our method is able to predict the transcoding time by minimizing the mean absolute error (MAE) to 1.37 and 2.67 for x264 and x265 codecs, respectively. For x264, this is an improvement of 22% compared to the state of the art.

We use a new method based on discrete fuzzy transforms for coding/decoding frames of color videos in which we determine dynamically the GOP sequences. Frames can be differentiated into intraframes, predictive frames, and bidirectional frames, and we consider particular frames, called Δ-frames (resp., R-frames), for coding P-frames (resp., B-frames) by using two similarity measures based on Lukasiewicz -norm; moreover, a preprocessing phase is proposed to determine similarity thresholds for classifying the above types of frame. The proposed method provides acceptable results in terms of quality of the reconstructed videos to a certain extent if compared with classical-based F-transforms method and the standard MPEG-4.

The IEEE 802.11 MAC l a yer uses RTS/CTS/ACK signaling to improve the reliability of unicast transmissions, but in current products there is no special signaling support for multicast transmissions. Therefore, the packet loss rate of multicast communication is generally higher than that of unicast communication. This paper evaluates a new MAC-layer multicasting technique called Leader-Driven Multicast (LDM). Instead of multicasting to a group address, the source node sends a unicast stream to the address of a leader node among the receivers. All other receivers passively receive this unicast stream, including retransmissions. This paper investigates the e ects of the LDM protocol on interactive video multicasting across WLANs. Results are presented from a simulation study using di erent packet loss models and di erent loss correlations among receivers. The results indicate that LDM is potentially bene cial even when there is no loss correlation among receivers. When losses do exhibit some positive correlation LDM can provide a very high quality multicast stream to WLAN nodes, especially when combined with application-level error control techniques.

Mobile phones have become essential part of modern life and continue to change the way people communicate with each other. The camera phones have become ubiquitous in the recent past and video communication services such as videophone are getting ...

Rate control is a complicated problem in the H.264/AVC coding standard, extra computation is usually needed for the existing rate control schemes to estimate the complexity of frames or macroblocks (MBs). However, during transcoding, information from precoded video could be used to simplify the rate control. In this paper, we propose a low-complexity rate control scheme for transcoding from H.263 to H.264/AVC. The relationship between the rate of the precoded video and both the rate and distortion of the transcoded video are studied. By using only the rate information from the precoded video, we introduce a row-layer bit allocation and perform average rate shaping across a row of MBs. Estimation error diffusion is also introduced. The proposed scheme has sufficiently lower computational complexity than other methods as there is no explicit complexity measurement of MBs and complicated parameters updating. Experimental results show the effectiveness of the proposed scheme.

Rate control is a complicated problem in the H.264/AVC coding standard, extra computation is usually needed for the existing rate control schemes to estimate the complexity of frames or macroblocks (MBs). However, during transcoding, information from precoded video could be used to simplify the rate control. In this paper, we propose a low-complexity rate control scheme for transcoding from H.263 to H.264/AVC. The relationship between the rate of the precoded video and both the rate and distortion of the transcoded video are studied. By using only the rate information from the precoded video, we introduce a row-layer bit allocation and perform average rate shaping across a row of MBs. Estimation error diffusion is also introduced. The proposed scheme has sufficiently lower computational complexity than other methods as there is no explicit complexity measurement of MBs and complicated parameters updating. Experimental results show the effectiveness of the proposed scheme.

Mode baseline encoder video H.263 menerapkan teknik estimasi dan kompensasi gerak dengan satu vector gerak untuk setiap macroblock. Prosedur area pencarian menggunakan pencarian penuh dengan akurasi setengah pixel pada bidang [16,15.5] membuat prediksi di tepian frame tidak dapat diprediksi dengan baik. Peningkatan unjuk kerja pengkodean prediksi interframe encoder video H.263 dengan optimalisasi teknik estimasi dan kompensasi gerak diimplementasikan dengan penambahan area pencarian [31.5,31.5] (unrestricted motion vector, Annex D) dan 4 motion vector (advanced prediction mode, Annex F). Hasil penelitian menunjukkan bahwa advanced mode mampu meningkatkan nilai SNR sebesar 0.03 dB untuk sequence video claire, 0.2 dB untuk sequence video foreman, 0.041 dB untuk sequence video Glasgow, dan juga mampu menurunkan bit rate pengkodean sebesar 2.3 % untuk video Claire, 15.63 % untuk video Foreman, dan 9.8% untuk video Glasgow dibandingkan dengan implementasi 1 motion vector pada pengkodean baseline mode.

En el presente trabajo se describe un método para transmitir video codificado en H.264/AVC. La transferencia de datos se realiza adaptando el envío de datos al desempeño de la red para garantizar una transmisión continua. La transferencia de datos entre los terminales servidor y cliente se realiza mediante sockets TCP .El video original se codifica en H.264/AVC para diferentes niveles de bitrate usando la librería GStreamer. Luego cada video codificado en H.264/AVC es segmentado con la herramienta VideoSplitter, desarrollada como parte del proyecto. La segmentación se realiza a nivel de GOPs. El objetivo de segmentar el video es facilitar la conmutación entre fuentes de video de diferente bitrate para adaptar las transferencia al desempeño de la red. Segmentar el video codificado tiene como ventajas el escalado del servicio de entrega de video digital y el aprovechamiento máximo de los recursos de hardware en el lado del servidor.

Monografia (graduação)—Universidade de Brasília, Faculdade UnB Gama, Curso de Engenharia Eletrônica, 2014.O objetivo deste trabalho é reduzir o gasto de energia para a decodificação de imagens em receptores de rádio digital. Ao reduzir a resolução de imagens, sem perda significativa de qualidade, após o processo de codificação reduz-se a taxa de transmissão, proporcionando menor ocupação espectral e maior velocidade de recepção, o que leva a um processamento mais rápido, com menor gasto de energia. Para tratar a imagem, utilizou-se o método proposto por Carlos Salazar, que utiliza a Transformada Discreta dos Cossenos para separar as componentes de baixa e de alta frequência. Assim, desenvolveu-se um algoritmo que realiza esta operação, para ser aplicado em sistemas de rádio digital, como por exemplo, na transmissão de páginas em HTML na faixa de ondas curtas. Entretanto, nada impede que as imagens resultantes das técnicas sejam transmitidas utilizando outros tipos de modulações digi...

This paper presents a new selective and scalable encryption (SSE) method for intra dyadic scalable coding framework based on wavelet/subband (DWTSB) for H.264/AVC. It has been achieved through the scrambling of quantized transform coefficients (QTCs) in all the subbands of DWTSB. To make the encryption scalable, it takes advantage of the prior knowledge of the frequencies which are dominant in different high frequency (HF) subbands, as traditional zigzag scan is not that efficient for them. Thus, by scrambling the scan order of QTCs in the intra scalable coding framework of H.264/AVC, we were able to get encryption and compression for enhancement layers (ELs) simultaneously. Watermarking has been integrated in the proposed architecture to avoid the requirement of separate keys for each spatial layer. The algorithm is better suited for multimedia streaming as the bitrate of the encrypted bitstream is lower than the original bitrate. Besides offering SSE, the proposed algorithm, when applied to different benchmark video sequences, outperformed the standard zigzag scan in terms of bitrate.

This paper develops a new adaptive scanning methodology for intra frame scalable coding framework based on a subband/wavelet(DWTSB) coding approach for MPEG-4 AVC/H.264 scalable video coding (SVC). It attempts to take advantage of the prior knowledge of the frequencies which are present in different higher frequency subbands. We propose dyadic intra frame coding method with adaptive scan (DWTSB-AS) for each subband as traditional zigzag scan is not suitable for high frequency subbands. Thus, by just modification of the scan order of the intra frame scalable coding framework of H.264, we can get better compression. The proposed algorithm has been theoretically justified and is thoroughly evaluated against the current SVC test model JSVM and DWTSB through extensive coding experiments for scalable coding of intra frame. The simulation results show the proposed scanning algorithm consistently outperforms JSVM and DWTSB in PSNR performance. This results in extra compression for intra frames, along with spatial scalability. Thus Image and video coding applications, traditionally serviced by separate coders, can be efficiently provided by an integrated coding system.

This paper presents a novel method for the protection of copyrighted multimedia content. The problem of selective encryption (SE) is being addressed along with the compression for the state of the art video codec H.264/AVC. SE is performed in the context-based adaptive binary arithmetic coding (CABAC) module of video codec. For this purpose, CABAC is converted to an encryption cipher. It has been achieved through scrambling of equal length bin strings. In our scheme, CABAC engine serves the purpose of encryption cipher without affecting the coding efficiency of H.264/AVC by keeping exactly the same bit rate, generating completely compliant bitstream and requires negligible computational power. Owing to no escalation in bit rate, our encryption algorithm is better suited for real-time multimedia streaming. it is perfect for playback on handheld devices because of negligible increase in processing power. Nine different benchmark video sequences containing different combinations of motion, texture and objects are used for experimental evaluation of the proposed algorithm.

In this paper, we present a novel approach for active fingerprinting of state of the art video codec H.264/AVC. Tardos probabilistic fingerprinting code is embedded in H.264/AVC video signals using spread spectrum watermarking technique in both luma and chroma. Tardos code is embedded in intra as well as inter frames. The embedding has been performed in the non-zero quantized transformed coefficients (QTCs), which are above a certain threshold while taking into account the reconstruction loop, to avoid the uncontrollable increase in bitrate of video bitstream. A comprehensive analysis of payload and PSNR trade-off is presented for benchmark video sequences. Different linear and non-linear collusion attacks have been performed in the pixel domain to show the robustness of the proposed approach.

The paper reports research on original techniques for compression of chrominance data in video sequences transmitted at very low bit rates. Two chrominance components are converted into one scalar chrominance signal using vector quantization in chrominance plane. Three methods for scalar chrominance compression based on transform and predictive coding are described. The important issue is that presented algorithms exploit mutual correlation between vector quantized chrominance and the luminance component of video data. In transform coding techniques, the activity of reconstructed luminance estimated for its transform coecients controls quantization steps of respective transform coecients of scalar chrominance. In the approach based on predictive coding, detection of active areas in the luminance component allows for ecient coding of chrominance prediction error, where unnecessary transmission of zero{valued errors is avoided. The paper presents experimental results obtained for the proposed methods.

The paper describes a multi-layer video coder based on spatiotemporal scalability and data partitioning. The base layer represents video sequences with reduced spatial and temporal resolution. Decoding of a middle layer gives full resolution images but with lower quality as compared to those obtained from the enhancement layer also. The bitrate overhead measured relative to the single layer MPEG-2 bitstream varies about 10% -25% for progressive television test sequences.. The coder is able to produce three layer-bitstreams with similar bitrates. The base layer is fully MPEG-2 compatible and the whole structure exhibits high level of compatibility with individual building blocks of MPEG-2 coders.

HEVC (MPEG-H Part 2 and H.265) is a new coding technology which is expected to be deployed on the market along with new video services in the near future. HEVC is a successor of currently widely used AVC (MPEG-4 Part 10 and H.264). In this paper, the quality coding gains obtained for the Cascaded Pixel Domain Transcoder of AVC-coded material to HEVC standard are reported. Extensive experiments showed that transcoding with bitrate reduction allows the achievement of better rate-distortion performance than by compressing an original video sequence with the use of AVC at the same (reduced) bitrate.

The paper describes video codecs that provide the functionality of spatio-temporal scalability that enables construction of highly scalable video codecs. The coder exploits wavelet decomposition for intraframe coding and combined temporal prediction and spatial interpolation for P-and B-frames. In a two-layer system, the bitrate overhead measured relative to the single layer MPEG-2 bitstream varies about 10% for progressive television test sequences. The base layer bitstream constitutes about 40% of the overall bitstream. In multilayer systems, the bitrate ratio of consecutive layers is about 1:2.5. The base layer encoder is fully compatible with the MPEG-2 video coding standard. The paper comprises experimental results obtained for a two-layer system for progressive video sequence compression.

The paper describes a three-layer video coder based on spatiotemporal scalability and data partitioning. The base layer represents video sequences with reduced spatial and temporal resolution. Decoding of a middle layer gives full resolution images but with lower quality as compared to those obtained from the enhancement layer also. The bitrate overhead measured relative to the single layer MPEG-2 bitstream varies about 5% -25% for progressive television test sequences. The base layer is fully MPEG-2 compatible and the whole structure exhibits high level of compatibility with individual building blocks of MPEG-2 coders. The paper reports experimental results that prove useful properties of the coder proposed.

The goal of the MASCOT project was to develop new video coding schemes and tools that provide both an increased coding efficiency as well as extended scalability features compared to technology that was available at the beginning of the project. Towards that goal the following tools would be used: • metadata-based coding tools; • new spatiotemporal decompositions; • new prediction schemes. Although the initial goal was to develop one single codec architecture that was able to combine all new coding tools that were foreseen when the project was formulated, it became clear that this would limit the selection of the new tools. Therefore the consortium decided to develop two codec frameworks within the project, a standard hybrid DCT-based codec and a 3D wavelet-based codec, which together are able to accommodate all tools developed during the course of the project.

Motion information scalability is an important requirement for a fully scalable video codec, especially in low bit rate or small resolution decoding scenarios, for which the fully scalable motion model (SMM) has been proposed. SMM can collaborate flawlessly with other scalabilities, such as spatial, temporal and quality, in a scalable video codec and it performs better than nonscalable motion model. To further improve the SMM, this paper further extends the algorithm to support hierarchical B frame structure and bidirectional or multidirectional motion estimation and, moreover, provides the corresponding rate distortion optimized estimation. Several simulation results are presented to verify the improvements of this extension.