Video Transmission

The popularity of interactive video streaming applications have pushed researchers to propose mechanisms for supporting these applications over the Internet. Several studies showed that interactive operations are well supported if the end-to-end delay, experienced by the application traffic, is kept lower than a pre-defined threshold. Recently, it has been proposed a new approach that acts on the video QoS (by dropping frame) in order to provide interactive features. The mechanism has been tested with Motion JPEG videos. The contribution of this paper is to evaluate the mechanism with MPEG videos, in order to investigate the behavior of the mechanism when the video is encoded with one of the more popular inter-frame techniques. Further, since the mechanism acts on the video QoS, we also present a QoS evaluation of the perceived video play out quality.

This work presents a visual telepresence system that captures and projects panoramic images into immersive environments in real-time. A number of images are captured with a ring of cameras (a special mounting of multiple digital cameras that captures synchronized images of a 360 degrees scene). These images are processed by a PC cluster before their transmission. A panoramic image is locally rendered by the proposed optimized stitching method that computes a positioning map for all pixels. Then, a complete cylindrical image is sent over the Internet to a multi-projection virtual environment called CAVERNA Digital®. Since the panoramic view is projected in natural size onto the screens of the virtual environment, the user has the sensation of being present at the remote scene.

The error-resilient for video transmission over the Internet in which regarded as the packet erasure channel is always a tough task and has gained lots of attentions. The main contradictory problem lies between error-resilient and bandwidth usage. Additional redundant data has to be added to achieve robust transmission which leads to huge bandwidth usage. In this paper, an error-resilient scheme called Wyner-Ziv Error-Resilient (WZER) based on a receiver driven layered Wyner-Ziv (WZ) coding framework is proposed. The WZER purposely emphasizes on the protection of the Region of Interest (ROI) area in the frame thus to achieve the better tradeoff between the bandwidth usage and error-resilience. WZER is designed to work for the scenario of wavelet based video coding over packet erasure channel, where several techniques including automatic ROI detection, ROI mask generation, Rate distortion optimization (RDO) quantization, WZ coding with layer design, and packet level Low Density Parity Check (LDPC) code are used. The performances of the proposed WZER are simulated based on average PSNR of luminance, perceptual reconstruction and bandwidth usage and compared with normal Forward Error Correction (FEC) full protection scheme and no protection scheme. The results show the advantages of the proposed WZER over traditional FEC protection, especially in the aspects of the recovery of the subject area and bandwidth efficiency.

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.

In this paper, we are interested in resource allocation strategies for wireless time-hopping ultra-wide band (TH-UWB) communications with multiple rate capabilities between users. Multiple rates are achieved by assigning different processing gains, i.e. N f , to users. For this purpose, the multiple-access interference (MAI) variance accounting for multi-rate is needed. It is a challenging task due to the lack of a suitable closedform expression for the MAI variance in a multi-rate context. We further study the multi-rate resource allocation problem in uplink TH-UWB systems for which an optimal search cannot be envisaged due to the exponential complexity induced. Our contribution lies in three-fold: i) A new intercode correlation expression accounting for multi-rate communications is derived, and the variance of the MAI averaging over the codes is obtained. ii) The multi-rate resource allocation problem is tackled by relaxing the integer constraint on the processing gains and modeled via a signomial programming problem. iii) Based on this, a branch and bound (BB) algorithm is derived for the allocation of the processing gains in TH-UWB systems. We also propose a really simple heuristic with linear complexity for the N f allocation. We show that the algorithm proposed outperforms the BB algorithm in average throughput and average starvation rate 1 .

The region of image compression since it is applicable to various fields of image processing. The key goal of compression is to cut back the space for storage and decrease the transmission cost and maintain the visual quality of image. This paper presents fundamental approach for analysis and evaluating the image compression techniques. This paper describes the many techniques which can be used for image compression and describes about artifacts which can be consequence of compression and their reduction techniques. This paper presents a new Canny edge based restoration method to remove artefacts from compressed image. Firstly standard JPEG compression is used to compress the first image. Compressed image contains various visible artefacts like blurring artifacts, blocking artifacts and ringing artifacts. Then to remove ringing artifacts proposed method i.e. image restoration as a post processing method is used. Experimental effects of proposed method by use of various performance parameters are good.

This research work studies the performance of the internet services of institution of higher learning in Nigeria. Data was collated from Lagos State University of Science and Technology (LASUSTECH) as case study of this research work. The problem of Internet Bandwidth optimization in the institution of higher learning in Nigeria was extensively addressed in this paper. The operation of the Link-Load balancer which provides an efficient costeffective and easy-to-use solution to maximize utilization and availability of Internet access is discussed. In this research work, the Lagrange's method of interpolation was used to predict effective internet data bandwidth for significantly increasing number of internet users. The linear Lagrange's interpolation model (LILAGRINT model) was proposed for LASUSTECH. The predictions allow us to view the effective internet data bandwidth with respect to the corresponding acceptable number of internet users as the number of user's increases. The integrity of the model was examined, verified and validated at the ICT department of the institution. The LILAGRINT model was integrated into the management of ICT and tested. The result showed that the proposed LILAGRINT model proved to be highly effective and innovative in the area of internet data bandwidth predictability.

As random network coding (RNC) considerably increases the network throughput, it has been of great interest for video streaming over wireless mesh networks (WMNs). However, mobile video users suffer from high transmission overhead due to the transmission of large coefficient vectors as headers and an excessive imposed decoding computational complexity due to using the Gauss-Jordan elimination method in RNC. This complexity cannot be supported by the embedded mobile processors. To overcome these limitations, this study analyses the impact of applying a method that simplifies RNC requirements on WMNs. This method is based on the generation of a full rank coefficients matrix without any linear dependency among its vectors. Nodes encapsulate one instead of n coefficients entries into a packet which leads to very low transmission overhead. Receivers can obtain the inverted coefficients matrix by performing very few arithmetic operations. Consequently, wireless nodes experience very low decoding computational complexity eliminating the need for powerful processors and high battery energy sources. The wireless medium is also less occupied and the transmission processes are shorter. Simulation results in the OMNeT++ framework depict that the applied method provides high video quality on the nodes by addressing the mentioned challenges, even if high mobility rates exist in the WMN.

Migration of news gathering feeds from SD to HD introduce capacity and performance challenges for RF channels using existing picture coding and modulation technologies MPEG-2 picture coding 4.5Mbps -8Mbps provides acceptable SD contribution picture quality 14Mbps -18Mbps provides acceptable HD contribution picture quality DVB modulation DVB-S provides robust transmission using QPSK, FEC=3/4 ~6.0Mbps data rate for a 5.5MHz satellite channel DVB-T provides robust transmission using QPSK, FEC=1/2, GI=1/8 ~5.5Mbps data rate for a 8MHz terrestrial channel Both are sufficient for MPEG-2 SD but not for MPEG-2 HD! Furthermore, in the USA, BAS channel relocation has reduced bandwidth available for terrestrial news gathering 17MHz channel bandwidth reduced to 12MHz Dual 8MHz channels (pedestals) are no longer possible

The scientific and business communities are showing considerable interest in wireless sensor networks (WSN). The availability of low-cost, small-scale components like CPUs, radios, and sensors, which are often combined into a single chip, is crucial. Parallel to the evolution of WSNs, the concepts of the IoT have been evolving in recent years. Wireless communication technologies may play a significant role in the implementation of IoT, despite the fact that IoT does not need or require any particular technology for communication. WSN assisted IoT networks can drive several applications in many industries. The proposed research explores the possibility of enhancing energy efficiency in WSN-assisted IoTN by balancing various challenging sensor network performance metrics. The base station's current placement inside the sensing field is predetermined by the preexisting routing algorithms. Our study examines the impact of base station placement outside and within the prescribed sens...

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.

Self-organizing wireless metropolitan area networks connect-ing portable GSM/UMTS base stations are a valuable asset to rescue forces in disaster recovery. These networks can be formed by routers em-ploying multiple commercially available IEEE 802.16-2004 interfaces. In this paper, we present and evaluate an algorithm for self-organized topol-ogy construction and channel allocation. The algorithm suggests (i) to estimate the capacity of potential wireless links from the measured signal-to-noise ratio, (ii) to run a modified Dijkstra algorithm to select links for topology construction in a way that maximizes network capacity and (iii) to greedily allocate channels to minimize interference. We evaluate our algorithm by simulations using QualNet. With the settings preferred by our industry partner, we find that 12 routers per square kilometer are sufficient to enable coverage. Trading path qualities and traffic load on gateways to the wired network significantly improves performance.

In this paper, we propose a temporal error concealment algorithm for video transmission in an error-prone environment. The error concealment algorithm employed an edge detection algorithm and progressive median motion vector concealment. First, the edges are detected and concealed portion by portion. Then, the corrupted MB is partitioned by the reconstructed edges and each partition is concealed by progressive median motion vector individually. The proposed algorithm shows better performance on both objective and subjective quality over existing temporal error concealment algorithm.

Since the quality of compressed video is vulnerable to errors, video transmission over unreliable Internet is very challenging today. Two-Hypothesis Motion-Compensated Prediction (THMCP) has been shown to have Error Resilience (ER) capability for video transmission, where each macroblock is predicted from its previous two frames. In this paper, we propose a novel ER approach named Alternate Motion-Compensated Prediction (AMCP). In addition to two-hypothesis prediction, one-hypothesis prediction is alternately used. We use some schemes to determine which kind of prediction should be used, so that in some cases of loss, the propagated error can be first decreased to some extent before it spreads to the subsequent frames. As a result, the expected converged error is less than that obtained from THMCP with fixed weights (THMCPF). Both analysis and simulation results are given to show that AMCP performs better than THMCPF, in terms of both compression efficiency and ER capability.

This paper presents an intelligent, rate-limited multicast video transmission optimization scheme for video distribution over 802.11 wireless networks based on packet retransmissions. We propose a problem formulation which involves the characteristics of the encoded stream together with the behaviour of the wireless channel. Using standard Stochastic Dynamic Programming techniques, optimal control policies are obtained off-line. These policies are optimal in the sense of minimizing the expected distortion at the terminal. In addition, the on-line complexity or our approach is very low since the optimization problem is solved off-line. The performance of our scheme has been evaluated in a real scenario and compared with that of a limited rate ARQ algorithm. The results for our proposed system show a higher packet recovery rate and a better protection of information with a higher priority.

In this paper we introduce the concept of multiple description video transcoding (MDVT). MDVT converts a single description encoded video into two or more descriptions at an intermediate node in the network. The objective of our MDVT approach is to adapt the video transmission to a multi-radio environment where two or more independent transmission paths exist between the intermediate node and the receiver. The sender does not have to be aware of the transcoding process and the multi-path transmission. MDVT can for instance be applied for multi-mode terminals that are simultaneously connected to two wireless access technologies, e.g., UMTS and WLAN. We compare MDVT with multiple description coding at the sender (MDC-S) as well as with MDC at the intermediate node (MDC-I) where the incoming single description video is decoded and re-encoded into multiple descriptions and the transmission is optimized separately for each path. We present a fast greedy method that can be used to perform multiple description video transcoding in real-time at low complexity. Our experimental results show that we can achieve performance similar to MDC-S where the sender has to be aware of the availability of multiple paths. Compared to MDC-I we observe more than 2 dB gain in reconstruction quality.

This paper describes a new error correction scheme for flexible and robust high-throughput real time video transmission systems able to cope with difficult high mobility channels. The scheme is based on combining OFDM transmission with Turbo Coding and deep time interleaving. Simulations of different channel models and field test has shown that the provided solution provides relevant improvement versus DVB-T based solutions.

A low complexity hierarchical quadrature amplitude modulation (QAM) symbol bits allocation algorithm for unequal error protection of video transmission over wireless channels is proposed in this paper. An unequal error protection (UEP) scheme using hierarchical QAM, which takes into consideration the non-uniformly distributed importance of intracoded frame (I-frame) and predictive coded frame (P-frame) in a group of pictures (GOP), is first proposed. In order to optimally allocate the hierarchical QAM's high priority (HP), medium priority (MP) and low priority (LP) symbol bits to the H.264/AVC video, a generic solution for optimal allocation of hierarchical QAM's high priority (HP), medium priority (MP) and low priority (LP) symbol bits to the H.264/AVC video is then proposed. Finally, a low complexity symbol bits allocation algorithm, namely ranking search algorithm, which reduces the computational complexity of the proposed optimal symbol bits allocation algorithm, is proposed. Simulation results show that our proposed UEP scheme outperforms the classical equal error protection (EEP) scheme and also the previous UEP scheme.

Compressed video sequences are very vulnerable to channel disturbances when transmitted through the burst-packet-loss wireless channels. Transmission errors not only corrupt the current frame, but may also propagate to succeeding frames. We find that a number of current proposed post-processing error concealment methods exploit the spatial/temporal redundancy in the video stream can not actually conceal the transmission errors effectively, because they ignored the fact that the neighboring blocks can be easily lost by the burst packet losses and the correlation was considered only in the current frame, which all limited the ability to obtain an optimal recovery. In this paper, a novel error concealment algorithm is proposed for video transmissions in a wireless environment. Simulation proves that the proposed algorithm can be chosen as good trade-off between video quality and computational load in wireless networks.

A new technique of spatio-temporal scalability coding is considered for video with bitrates of order of megabits per second. Both temporal and spatial resolutions are reduced for data transmitted in a base layer. The temporal resolution reduction is obtained ...

Layering techniques for video transmission are being actively considered as a way of satisfying receivers in a heterogeneous environment. A layering scheme based on a combination of network requirements and the perceptual quality of video is an attractive proposition. In this paper a feed-forward post-codec controller has been described that sends video sequence in an optimally layered way. For optimization the controller considers the perceptual quality of video, load on sender and load on network. For the controller the correlations have been developed experimentally and deductively. The quality of video can be judged either subjectively or objectively. For subjective test, in the absence of any other standard, the ITU-R and ITU-T, standards for digital TV may be used. There are a number of objective tests. Some of these, like peak signal to noise ratio (PSNR) and root mean square error (RMSE), are simple but do not correlate well with the subjective evaluation. Other more complex...

Video content represents a large portion of global network traffic, and it is expected that this trend will continue as 4K video streaming becomes more popular and more applications such as entertainment and surveillance become available. Wireless networks are becoming the most important technologies for delivering cost-effective content, including video. The unpredictability of wireless channels, as well as the sensitivity of video content to packet losses, errors, and delays makes video delivery over wireless network difficult. Furthermore, when compared to wire links, congestion and complex traffic patterns make wireless channels much worse. Packet losses and network errors can be mitigated with the help of modern codecs. This study looks at the modern video codec H.264/Advanced Video Coding (AVC) and how to use the built-in error resilience tools to improve end-to-end video quality over simulated wireless networks. Network Simulator 3 (NS-3) was used and a framework for video quality assessment based on EvalVid 2.7 tools was employed. We tested the performance of several coding structures in the presence of different intra coding techniques. Results for less active video sequences showed that the IBP coding structure with intra period of 18 achieved best performance at lower loss rates. The addition of an intra MB line or extra random intra MBs did not seem to offer extra protection against errors for these sequences. For more active video sequences, IBBBP coding structure without or with an intra MB line performed better than all other configurations at all loss rates.

Traffic smoothing for delivery of online VBR media streams is one of the most important problems in designing multimedia systems. Given an available client buffer, a window size and a window-sliding size, previous window-based online smoothing methods try to reduce peak bandwidth allocated in each window. However, as bandwidths allocated in different windows are minimized independently, these methods require a large peak bandwidth for transmitting entire stream. In this paper, a new window-based method is proposed. It introduces two new ideas, the dynamic window-sliding size and the aggressive work-ahead, for delivery of online VBR media streams. Our ADWS (aggressive and dynamic window-sliding) method can automatically decide the suitable window-sliding sizes for different windows. Thus, the allocated peak bandwidth can be further reduced. By examining various media streams, ADWS is shown to be effective and efficient. Considering the online transmission of movie Star Wars with a 90...

With the advent of new digital video systems that utilize compression to achieve a savings in transmission or storage bandwidth, the quality of the received output video can be dependent not only upon the inherent spatial and temporal information content of the input video but also upon the dynamic variability of the digital communications channel. Therefore, out-of-service quality measurements using video test signals or scenes may not relate at all to the resultant received quality of actual program material. Furthermore, traditional in-service quality measurements made by injecting test signals into the non-visible portion of the video signal (e.g., the vertical interval in the NTSC or PAL video standard) are not applicable to modern digital video systems. Thus, a new method is required to make in-service video quality measurements on actual program material. This paper describes a new test instrument for measuring the quality of a video transmission or storage system where the input and output of the system may be spatially separated, and when there is no a priori knowledge of the input video. The test instrument makes continuous quality measurements by ( ) extracting statistics from sequences of processed input and output video frames, (2) communicating these extracted statistics between the input and the output ends using an ancillary-data channel of arbitrary bandwidth, (3) computing individual video quality parameters from the communicated statistics that are indicative of the various perceptual aspects of video quality (e.g., spatial, temporal, color), and ( ) calculating a composite video quality metric by combining the individual video quality parameters. The test instrument makes coarser quality measurements (coarser in the sense that the extracted statistics come from larger spatial-temporal regions) when smaller capacity ancillary-data channels are available and finer quality measurements when larger capacity ancillary-data channels are available. The design goal for the test instrument is to make the most accurate in-service video quality measurements given the available ancillary-data-channel bandwidth (mobile telephone connections, modem connections over the Public Switched Telephone Network (PSTN), Internet connections, Local Area Network (LAN) connections, satellite connections, cable connections, etc.).

We propose a real-time video transmission scheme, referred to as SNP/VQR, which is capable of providing spatial and temporal scalabilities. SNP/VQR eliminates error propagation by combining bandwidth adaptability with error concealment. A software implementation of SNP/VQR is tested over a wireless network with 5% to 20% of simulated packet losses. SNP/VQR produces a relatively constant visual quality in our packet loss studies.

Transmitting video with portable devices (such as smartphone), over heterogeneous networks with satellite portions, to a Remote Monitoring Host (RMH) represents a hot scientific and technical topic. Unfortunately, heterogeneity often implies impairments such as packet losses, due to errors and congestion, which negatively affect the video quality. To guarantee a satisfactory video fruition at the RMH, an application layer joint coding algorithm for video transmission is presented. It adaptively applies both video compression and encoding to protect the sent video stream, at the application layer, on the basis of the overall network condition estimated in terms of both maximum allowable throughput of the network and quality (packet cancellations or lossiness). A preliminarily performance study, carried out with real implementation of the algorithm, compares the joint coding against fixed schemes.

In this paper we propose an adaptive Joint Source and Channel Coding (JSCC) associated with adequate modulation algorithm to provide a near channel capacity video transmission rate to achieve better system performance. In the proposed algorithm we apply MPEG-2 video coding scheme, and turbo channel coding technique as well as BPSK/QPSK for modulation method. Rates assigned to MPEG-2 source coding and turbo channel coding schemes are based on the feedback information from Performance Control Unit (PCU) based on channel capacity limitation. Therefore, an improved performance for the suggest system design is promised over Rayleigh fading channel conditions.

A video-on-demand system, dubbed NetMovie, based on MPEG-2 encoded video transmission over ATM networks has been designed and built. It uses a real-time MPEG-2 decoder board on a client personal computer to display compressed video delivered from a remote server over an FDDI/ATM network. Audio play-back is provided using off-the-shelf, digital audio boards. Video compression is done off-line using software encoders. Several transport layer techniques, including dual priority transmission and slice-based packetization, have been developed for error resilient, real-time transmission and audio-video synchronization. The system is integrated with the World Wide Web to provide real-time, high-quality video delivery to end points with high bandwidth connections to the Internet. It demonstrates feasibility of an economical, MPEG-2 video-on-demand system based on readily available hardware and software. NetMovie has been successfully used to demonstrate an MPEG-2 based, international video-on-demand application between NJ, USA and Kyoto, Japan during the 1994 plenipotentiary meeting of the ITU.

Abstract. This paper describes methods of extracting region boundaries from the frames of an image sequence by combining information from spatial or temporal cooccurrence matrices of the frames. It summarizes past work on the uses of cooccurrence matrices for image segmentation; qualitatively describes the peaks (clusters of high values) that can be expected to occur in cooccurrence matrices when the image(s) contain smooth regions separated by stationary or moving boundaries; and describes methods of extracting stationary or moving region boundaries from an image by combining information from spatial and temporal cooccurrence matrices. 1 Cooccurrence matrices and their uses Cooccurrence matrices, originally called gray-tone spatial dependency matrices, were introduced by Haralick et al. [1], who used them to define textural properties of images. Let I be an image whose pixel gray levels are in the range 0,...,255. Let δ = (u, v) be an integer-valued displacement vector; δ specifies...

Standardized wireless transmissions include fixed-size fragments per packet. Relying on this structure, two schemes for distributing redundancy across fragments are considered in this paper to enable blind identification without higher-order statistics of received signals. We prove that both schemes guarantee blind identifiability as well as symbol detectability and propose a subspace based blind identification method. We test their relative merits, and compare them with existing alternatives using simulations.

In dual-frame video coding, one long-term reference (LTR) and one short-term reference (STR) frames are used for motion estimation and compensation. In previous work, it was shown that the performance of video coding can be improved by pulsing the quality of LTR frames in dual-frame video coding, but this increases the encoder delay buffer size. Also, buffer constrained real-time video transmission requires an efficient rate control algorithm to meet the delay requirement. In this paper, we propose a rate control algorithm for dualframe video coding under a delay buffer constraint. With the proposed rate control algorithm and motion activity detection for determining the LTR quality, simulation results using H.264/AVC show a significant PSNR improvement over H.264 rate control and other rate control algorithms for dualframe video coding.

With the growth of broadband networks, Video-on-Demand (VoD) has become realistic. Many significant broadcasting schemes have been proposed to reduce the viewers' waiting time. However, clients are required to receive video segments altogether from all channels. This disregard not only needs a lot of client bandwidth, but also incurs more buffer requirements. To escape from these constraints, this work proposes the adjustable interleaving staircase-harmonic broadcasting scheme (AISHB), which offers a tradeoff between any two of three resources: server bandwidth, client buffer spaces, and client bandwidth. When client bandwidth is not limited, the scheme requires a client to buffer only 25% of a playing video and the waiting time is slightly higher than the optimal waiting time. In comparison with the fast broadcasting, recursive frequency-splitting, and harmonic broadcasting, AISHB saves the buffer requirements by 50%, 33%, and 33%. If client bandwidth is restricted, AISHB achi...

Video transmission is expected to be the next big thing in personal communication systems. While text messaging applications have already experienced an explosive growth, the exchange of multimedia content is still lacking architectural solutions which enable it to become the next killer application. The EU project MEDIEVAL (MultimEDia transport for mobIlE Video AppLications) [1] aims at filling this gap. In this paper we describe the approaches envisioned by the project for what concerns the data link layer, with special emphasis on wireless access and its related cross-layer issues. After presenting some general project aspects, we will review the state of the art and enumerate several open issues, concerning the identification of the most suitable radio technologies for video support, as well as their integration and required enhancements to enable efficient video transport.

The paper studies the impact of the introduction of pipeline forwarding on the network of a large Internet Service Provider trying to provide a detailed analysis of the trade-offs between costs and benefits. The network of Telecom Italia is considered as a case study.

This paper studies an application of Turbo codes to compressed image/video transmission and presents an approach to improving error control performance through joint channel and source decoding (JCSD). The proposed approach to JCSD includes error-free source information feedback, error-detected source information feedback, and the use of channel soft values (CSV) for source signal postprocessing. These feedback schemes are based on a modification of the extrinsic information passed between the constituent maximum a posteriori probability (MAP) decoders in a Turbo decoder. The modification is made according to the source information obtained from the source signal processor. The CSV's are considered as reliability information on the hard decisions and are further used for error recovery in the reconstructed signals. Applications of this joint decoding technique to different visual source coding schemes, such as spatial vector quantization, JPEG coding, and MPEG coding, are examined. Experimental results show that up to 0.6 dB of channel SNR reduction can be achieved by the joint decoder without increasing computational cost for various channel coding rates.

This paper proposes a set of performance enhancements techniques aiming to improve the perceived performance of video transmission across the IEEE 802.11e network. The proposed mechanism preserves the video Quality of Experience (QoE) by protecting the I-Frames transmitted as part of the Group of Pictures (GoP) during queue congestion. The method is evaluated using the NS-3 simulator with the Evalvid module and the results demonstrate the video flows will have better Peak Signal to Noise ratio (PSNR) and less video frame drops compared to the original IEEE 802.11e queueing. Keywords—IEEE 802.11e; EDCA; QoE; NS-3; Evalvid

Broadband Networks based on the Asynchronous Transfer Mode (ATM) will support, among others, traffic coming from variable bit rate video (VBR) codecs1, which are capable of maintaining a constant picture quality of the decoded image. The characterization of such VBR video sources becomes important in the analysis and design of Broadband Integrated Services Digital Networks (B-ISDN). The network architecture and its characteristics, such as cell-loss probabilities, transmission delay, high-speed statistical multiplexing gain, buffering, etc., are strongly related to the statistical properties of the sources and the coding schemes involved. On the other hand, a characterization of the traffic generated by a VBR source is necessary in order to allocate resources in ATM networks, as well as, to keep a satisfactory quality of service (QoS). In the call establishment phase, service requirements are negotiated between the user and the network to establish a Traffic Contract. The source tra...

The issues of green technology nowadays give an inspiration to the researcher to make all the future design to be energy efficient. Medium Access Control (MAC) layer is the most effective layer to provide energy efficient due to its ability to control the physical radio directly. One of the important applications in the future is a video transmission that can be transmitted with low-cost and low power consumption. MPEG-4 is one of the international standards for moving video. MPEG-4 provide better compression and primarily design at low bit rate communication. In order to achieve good quality for video application, the design at MAC layer must be strong. Therefore, to increase the performance of the MPEG-4 in IEEE 802.15.4, in this paper we propose a cross layer design between MAC layer and Application layer. A priority queue will be implemented at MAC scheduling depends on the level of frame important in MPEG-4 format frame. A distributed Time division Multiple Access (TDMA) will be used for MAC protocol to provide reliable data transmission for high priority frame.

With the increase in the bandwidth & the transmission speed over the internet, transmission of multimedia objects like video, audio, images has become an easier work. In this paper we provide an approach that can be useful for transmission of video objects over the internet without much fuzz. The approach provides a ontology based framework that is used to establish an automatic deployment of video transmission system. Further the video is compressed using the structural flow mechanism that uses the wavelet principle for compression of video frames. Finally the video transmission algorithm known as RRDBFSF algorithm is provided that makes use of the concept of restrictive flooding to avoid redundancy thereby increasing the efficiency.

The great end-to-end delays are the major factor to influence the visual quality of real-time video across the Internet using TCP as transport layer protocol. In this paper, we present a video frame schedule scheme for rate adaptive real-time video transmission over TCP. The scheme schedules video frames between the application layer sender-buffer, the TCP sender-buffer and TCP receiver-buffer and can regulate automaticlly the video frame rate and play out buffer delays according to the network congestion level. The sheme requires only an extra buffer of application layer and can significantly cut down the end-to-end delays of real-time video without any modification to the network infrastructure or TCP protocol stack. The performance of the proposed solution is evaluated through extensive simulations using the NS-2 simulator.

1. Summary MPEG-4 AVC/AAC has been accepted as an audio-visual encoding format by a number of standards bodies and is poised for wide adoption in IP-based video services over consumer broadband connections (IPTV). In spite of the agreement on the encoding ...

With the continuous increase of network capacities, video transmissions in medicine are becoming an effective tool for second opinion diagnosis, archiving and teaching environments. This study investigates video compression delay times and transmission impairments for different compression formats and describes the resulting picture qualities. For the evaluation endoscopic video sequences were produced with different compression formats and bandwidth requirements. In a second step an impairment tool was used to introduce error rates to the video material to simulate network behavior. A group of medical experts evaluated the video sequences rating visibility of errors, artifacts, sharpness, overall picture quality and suitability for a medical diagnosis. The results clearly establish lower boundaries for picture quality deteriorated by compression and network impairments, and introduce limits for medical assessments.

Visual observations constitute important input to different meteorological tasks. Previously, these observations were made by human observers and manually reported to the meteorologists. With the employment of video technology, this kind of observations can be automated. This paper presents one approach to visual weather observations. This involves software compression and transmission of digital video. The paper presents a modified version of the PVC algorithm and shows that this algorithm is well suited for software compression of color video for use as visual information in the field of meteorology.

This paper presents a new technique to extract objects from a real complex background so that a video sequence can be decomposed into a set of objects as required for object oriented video compression techniques. The proposed method is based on a background subtraction technique. However, instead of using a fixed background, the system relies on predicting one from a previously constructed virtual model of the environment. Thus, camera movements are allowed. These movements are estimated by means of a tracker device. We also present the virtual model construction technique for indoor environments. The method has been successfully tested for several different video sequences including capture errors, partially mapped virtual environments and camera positioning errors. Further work will focus on extending the virtual models not only to environment, but also to objects, and integrating the method in a MPEG4 standard compression system.

A technology that deals with the spectrum scarcity and underutilization is cognitive radio (CR), where by spectrum sensing is one of the most important aspects. Multiple sensors perform cooperative spectrum sensing to reduce shadowing and multipath fading in the network. Due to the limitations of energy in sensors, energy efficiency emanate as significant issue in sensor-aided CR networks. Scheduling of each group of sensor active time can definitely reduce energy consumption and boost network life time. The sensors are divided into groups depending on the geographical position, only one group of sensors is turned on at a time while maintaining the necessary detection and false alarm thresholds. Each group is activated independently and non-activated are set in a low energy sleep mode to boost the network lifetime. Also throughput optimization is achieved by increasing the bit rates of data received to the fusion center which decrease the reporting time of secondary users. Analysis and simulation are presented by considering the performance of energy detection which discovers spectrum holes or white spaces and cooperative spectrum sensing approaches by using AND, MAJORITY and OR rule.

This paper presents an investigation of issues related to The process of converting. the variable rate image source the support of real-time compressed digital video on shared access packet transmission systems. Packet transport mechanisms are motivated for compressed digital video because of their inherent ability to average data from many bursty variable rate sources, potentially elim-to a' fixed channel rate rnvolves rate buffer adaptive encoding [l2I9 which tends to increase video dec implementation complexity 9 while also resulting in inating the need for buffer controlled adaptation of the video coding algorithm. The feasibility of employing conventional link-and trans-undesirable scene-sequence-dependent image quality artifacts. port-level protocol services to transmit compressed video is examined by focusing on two specific practically important scenarios for compressed video transmission: 1) multipoint-to-multipoint video transmission using a 200 Mbit/s implicit token passing (ITP) fiber-optic LAN and 2) point-to-multipoint broadcast video distribution using a 90 Mbit/s packet-TDM direct-broadcast satellite channel. In order to evaluate the performance of such shared access broadband packet video systems, we develop accurate simulation models, driven by realistic "broadcast quality" AST-DPCM compressed video sources, for the example ITP-LAN and packet-TDM systems. The models are used to determine design tradeoffs between channel throughput, video quality (measured by clipping probability), and the transport-level and mediaaccess-level protocol features and parameters implemented in the packet video network interface unit (NIU).