In a surveillance system with a huge number of cameras, the number of videos to be transmitted and displayed is usually restricted by network bandwidth and the resource of display terminal. Given that the source video is captured at high quality, a network transcoder is used to send video with lower data rate as the default for ordinary scenes, while only extraordinary or unusual scenes are sent with higher quality. With such a scenario, it is necessary to switch from a low quality version of the video to a higher quality video with low latency and in a seamless manner. This paper presents a network transcoder that is able to change the content and the quality of videos seamlessly and with low latency. The novelty of the proposed scheme is possible to change the quality and camera in the same session. Moreover, this paper describes an RTSP enhancement that enables this dynamic transcoding function. Finally, an evaluation of the results is provided.

In this paper, we propose a high-speed UDP-based media transport with an adaptive error control. The proposed adaptive transport scheme controls the adaptive forward error correction (FEC) and automatic repeat request (ARQ) schemes by monitoring underlying networks so that it can effectively adapt to the fluctuation of given high-speed networks. The monitored feedbacks from the receiver enable the sender to aware of current reception status (i.e., rate/type of packet loss and packet delay). Based on this, the proposed adaptive media transport attempts to provide controlled reliability by adaptively controlling ARQ as well as adaptive FEC schemes. Experiments with high-speed network testbed have been conducted to verify the performance of the proposed transport that demonstrates the enhanced throughput at the speed of up to one Gbps.

As real-time media applications based on IP multicast networks spread widely, end-to-end QoS (quality of service) provisioning for these applications have become very important. To guarantee the end-to-end QoS of multi-party media applications, it is essential to monitor the time-varying status of both network metrics (i.e., delay, jitter and loss) and system metrics (i.e., CPU and memory utilization). In this paper, targeting the multicast-enabled AG (Access Grid) a next-generation group collaboration tool based on multi-party media services, the applicability of hybrid monitoring scheme that combines active and passive monitoring is investigated. The active monitoring measures network-layer metrics (i.e., network condition) with probe packets while the passive monitoring checks both application-layer metrics (i.e., user traffic condition by analyzing RTCP packets) and system metrics. By comparing these hybrid results, we attempt to pinpoint the causes of performance degradation and explore corresponding reactions to improve the end-to-end performance. The experimental results show that the proposed hybrid monitoring can provide useful information to coordinate the performance improvement of multi-party real-time media applications.

The GMF4iTV project (Generic Media Framework for Interactive Television) is an IST European project that consists of an end-to-end broadcasting platform providing interactivity on heterogeneous multimedia devices such as Set-Top-Boxes and PCs according to the Multimedia Home Platform (MHP) standard from DVB. This platform allows the content providers to create enhanced audiovisual contents with a degree of interactivity at moving object level or shot change from a video. The end user is then able to interact with moving objects from the video or individual shots allowing the enjoyment of additional contents associated to them (MHP applications, HTML pages, JPEG, MPEG4 files...). This paper focus the attention to the issues related to metadata and content transmission, synchronization, signaling and bitrate allocation of the GMF4iTV project.

Digital television is a reality providing new facilities for multimedia content delivery. Due to the large amount of available content from multiple broadcasters, the use of personalized recommendation techniques can help the user in the process of consuming A/V content. The combination of content description standards as MPEG7 and TV Anytime, and Personal Digital Recorders (PDRs) may allow to the user to organize the TV Program Guide of its interest whenever and however he wants, achieving a real "a la carte TV". In this paper we present an end-to-end digital TV system able to offer A/V contents according to the TV viewer preferences following DVB and TV Anytime standards.

A novel H.264 error resilient scheme aiming at stopping or reducing error propagation using the SP/SI coded macroblocks is proposed to maintain the quality of transmitted video through erroneous channels in this work. Specifically, for each encoded macroblock, we encode additional predicted versions using different reference frames (or different prediction methods) and save them as SP/SI macroblocks. During transmission, these SP/SI macroblocks are used to replace the originally coded macroblocks in the output video stream if they are affected by previous errors detected by the receiver. The way to encode these SP/SI macroblocks ensures that such a replacement will not cause any mismatch at the decoder side. It is confirmed by experimental results that the proposed scheme is effective in reducing error propagation so as to enhance the error resilient capability of H.264 video. This scheme introduces a small amount of overhead in the bit rate only when there are transmission errors, and does not have an overhead when no error occurs.

In the most recent H.264/AVC video codec, a loss-aware rate-distortion based coding mode selection method uses K copies of random variable channel behavior and decoder pairs to estimate the expected distortion at the encoder. The expected distortion can be estimated accurately if K is chosen large enough. However, the added complexity at the encoder is obviously at least K times the decoder complexity. In this paper, we modify another popular rate-distortion based mode selection method, the recursive optimal pre-sample estimate (ROPE) algorithm, by recursively calculating the standard deviation of pixel samples instead of the second moment values. Since the standard deviation values have the same order as the first moment of samples, the revised ROPE algorithm can be easily adopted to the H.264/AVC encoding process. Our experimental results show that the H.264/AVC coder with the revised ROPE algorithm has comparable error resilient capabilities, but it has only between one third to one half of the computation complexity of the current H.264/AVC codec with K=30.

Rate control is an important technique in Constant Bit Rate(CBR) coding. AVC standards contains several inter and intra prediction modes. The current AVC software is difficult to adopt the existing rate control techniques, because the Rate Distortion Optimization (RDO) based on prerequisite quantization parameters determines the optimal prediction of each macroblock. This paper proposes an efficient rate control algorithm at macroblock level for AVC standard by considering adaptive quantization parameter. Experimental results show that the proposed rate control algorithm can accurately achieve the target bit rate. Although the PSNR of object quality of video sequences is degraded a little degraded, but the coding efficiency is similar to or better than of the fixed quantization parameter video coding.

A multi-hypothesis motion compensated prediction (MHMCP) scheme, which predicts a block from a weighted superposition of more than one reference blocks in the frame buffer, is proposed and analyzed for error resilient visual communication in this research. By combining these reference blocks effectively, MHMCP can enhance the error resilient capability of compressed video as well as achieve a coding gain. In particular, we investigate the error propagation effect in the MHMCP coder and analyze the rate-distortion performance in terms of the hypothesis number and hypothesis coefficients. It is shown that MHMCP suppresses the short-term effect of error propagation more effectively than the intra refreshing scheme. Simulation results are given to confirm the analysis. Finally, several design principles for the MHMCP coder are derived based on the analytical and experimental results.

Multiple description coding (MDC) is a well-known robust data compression algorithm designed to minimize the distortion caused by data loss in packet-based communication systems. Several MDC schemes for transmitting wavelet compressed images have been developed. However, these MDC schemes cannot be adopted for digital mobile wireless applications where both packet loss and bit error are present, because individual description in these MDC schemes usually does not have adequate error resilience capability to combat the bit error in transmission. In this paper, we propose an algorithm to achieve robust communication over error prone transmission channels with both packet loss and bit error. We integrate the multiple description scalar quantization (MDSQ) with the multiple wavelet tree image coding method in order to provide an excellent error resilient capability. Two descriptions are generated independently by using index assignment of MDSQ. For each description, multiple sub-sampling is applied to split the wavelet coefficients of the source image into multiple sub-sources. Each sub-source is then entropy coded using the SPIHT algorithm and followed by a channel coding scheme that combines cyclic redundancy code (CRC) and rate compatible punctured convolutional (RCPC) code to offer unequal error protection to the entropy coded bits. The unequal error protection channel coding rate is designed based on the bit error sensitivity of different bit planes to achieve maximum end-to-end quality of service. Experimental results show that the proposed scheme not only exhibits an excellent error resilient performance but also demonstrates graceful degradation over error prone channels with changing rate of packet loss and bit error.

We examine the validity of TCP-friendliness and the utility of equation-based congestion control as a mean for the media streaming service in 802.11 multi-hop networks. Our investigation is motivated by the recent findings of the bandwidth-delay product of 802.11 multi-hop networks via intensive computer simulations. We show that TCP behavior in 802.11 multi-hop networks is generally inefficient and unstable because the standard TCP is too aggressive considering the low bandwidth-delay product of 802.11 networks. We introduce the concept of the `fractional window' mechanism (which resembles the stop-and-go protocol) for TCP, and show that the modified TCP is more stable and efficient while outperforming the legacy TCP in 802.11-based ad hoc networks. Finally, we show that many of the typical problems of TCP protocol (performance, network fairness, and TCP-friendliness) in 802.11 networks can be solved simply by increasing the bandwidth-delay product of 802.11 networks without any TCP modification. Based on this observation, we conclude that the assumption of the ideal steady-state TCP behavior is generally invalid in 802.11 multi-hop networking environment, and that congestion control based on TCP-friendly equation can hardly provide TCP-fair throughput and smoothness in 802.11 multi-hop networking environment.

In the mobile IP-enabled wireless LAN (WLAN), packet flows are corrupted due to the handoff of a mobile node at the link and network layers, which results in burst packet losses. This transient behavior hurts time-critical streaming media applications so much. Many solutions have been suggested to address this handoff-related problem. However, even with recent low-latency and smooth handoff options, seamless streaming (i.e., without any playback disruption due to the handoff) is not easy to achieve in the practical situation. Since no packet will be delivered to the mobile node during the handoff process, this can cause temporary packet underflows at the streaming client. Thus, to avoid packet underflows, one has to pre-buffer the streaming client sufficiently before the handoff. Note that the required level of pre-buffering depends on the efficiency of adopted link-/IP-layer handoff options. In this paper, we are targeting seamless MPEG-4 streaming over the mobile IP-enabled WLAN. For this, we are introducing a seamless media streaming framework by estimating the accurate pre-buffering level to compensate the handoff latency and proposing a combination of schemes as a preliminary version of the proposed framework. We utilize a Layer 2 (L2) trigger (event) about a change of an associated access point (AP) with a wireless network interface to reduce the agent discovery time in total handoff latency. A packet forwarding with buffering mechanism is also realized at the foreign agent (FA). With these two options, we can nearly eliminate packet losses during the handoff period. Then, we perform pre-buffering adjustment based on handoff latency experimentally measured and analytically obtained by the handoff transient time analysis. The experimental results show that the proposed approach can eliminate packet losses during the handoff period and provide the feasibility of seamless MPEG-4 video streaming over the mobile IP-enabled WLAN.

Multiple description coding (MDC) is a source coding technique that involves coding the source information into multiple descriptions. When these descriptions are transmitted over different channels in packet network or error-prone wireless environment, graceful degradation can be achieved even if part of descriptions are not received by the receiver. When MDC is applied to wavelet subband based image coding, it is possible to introduce correlation between the descriptions in each subband. In this paper, we consider using such a correlation as well as potentially error corrupted description as side information in the decoding to formulate the MDC decoding as a Wyner Ziv decoding problem. If only part of descriptions is lost, however, their correlation information is still available. Therefore, the proposed Wyner Ziv decoder can recover the description by using the correlation information and the error corrupted description as side information. High quality reconstruction can still be obtained by combining the decoded descriptions from Wyner Ziv decoder. The proposed scheme takes advantage of an efficient way to use the correlation information, thus makes the system more robust to channel error corruption. Experimental result shows that, comparing to conventional multiple description wavelet based image coding, the PSNR of the received and decoded image could be improved noticeably when coding at the same bit rate.

Joint source-channel coding schemes have been proven to be effective ways for reliable multimedia communications. In this paper, a joint source-channel decoding (JSCD) scheme that combines the hidden Markov source (HMS) estimation and low-density parity-check (LDPC) coding is proposed for the standard MPEG-2 video transmission. The LDPC code of the proposed scheme has a near-Shannon-limit error-correcting capability, while the HMS estimator may accurately extract the residual redundancy within the MPEG-2 video stream without any prior information. Furthermore, with a joint iterative decoding algorithm, the estimated source redundancy may be well exploited by the LDPC decoder, and the channel decoding feedback may refine the subsequence HMS estimation, thereby effectively improving the system performance. On the other hand, we also show that the proposed JSCD scheme has approximately the same computation complexity as that of the standard decoding scheme. Moreover, it is worth noting that the proposed scheme is based on separation encoding schemes, which is very convenient to be applied to existing multimedia transmission systems.

In this paper, we present a novel reliable MPEG-4 based packetisation scheme over UMTS (universal mobile telecommunications service) networks. Using this scheme, some lost crucial RTP packets can be delivered within a multiplex RTP packet. With the feedback of RTCP control packets, this paper presents a decision-making procedure by which we can calculate the number of the redundant packets in a multiplex packet during a RTCP interval, i.e., this procedure can precisely reflect current network congestion state dynamically. To support the proposed procedure, two conditions must be satisfied. The first one is the RTP packets loss constraint and the second one is the RTP packets delay limitation. Moreover, through investigating related RTP payload formats and FGS (fine granularity scalability) framework, we can deliver BL (MPEG-4 basic layer) data by the proposed scheme. EL (MPEG-4 enhancement layer) data will be delivered as the best-effort traffic. Finally, simulations over UMTS networks demonstrate the validity of this scheme. Further comparison between this scheme and the traditional schemes also is done. The results show that our scheme improves QoS of services significantly.

Over the last three years, mobile handsets have evolved from voice based services to multimedia terminals that is gradually leading towards a new era of consumer entertainment fostered by mobile communications and consumer networks. The need for Mobile Digital Rights Management (MDRM) solutions is thus intensified in order to safeguard mobile media content. Various types of content protection solutions have been proposed. In this paper, we investigate current status, the standardization effort, and the challenges in the development of MDRM technologies. Sample state-of-the-art media security technologies for MDRM are also discussed.

A new paradigm of one-way hash function, called the distance-preserving hash function (DP hash function), is proposed and a soft multimedia content authentication scheme is developed accordingly in this work. The DP hash function has a distinct characteristic from regular hash functions. That is, when it is equipped with a correct key, the difference between two outputs reflects the distance between the corresponding inputs. However, if a wrong key value is used, the DP hash function reduces to a regular one-way hash function. We examine the theoretical aspect of the DP hash function and propose a practical way for its construction. The DP hash value of the multimedia feature vector is used as the authenticator for the corresponding content. By comparing the DP hash of received data and the received DP hash (authenticator), users can obtain the error between the received data and the original data. This error information indicates a degree of authenticity and allows users to render a soft decision rather than an "authentic or non-authentic" hard decision. Such a soft decision is helpful in applications where a small amount of distortion in the target data is acceptable, such as digital audio and video.

In this paper, a stereo image watermarking scheme using discrete wavelet transform (DWT) and feature-based window maching algorithm(FMA) is proposed. A watermark data is embedded into the right image of a stereo image pair by using the DWT algorithm and disparity data is extracted from the left and watermarked right images. And then, disparity data and the left image are transmitted to the recipient through the communication channel. At the receiver, the watermarked right image is reconstructed from the received left image and disparity data by employing the FMA. From the difference between the watermarked and original right images, the embedded watermark image can be finally extracted. From experiments using the stereo image pair of 'Friends' and a watermark data of '3DRC', it is found that PSNR of the watermark image extracted from the reconstructed right image through the FMA and DWT algorithms can be increased up to 2.87dB, 2.58dB on average by comparing with those of the FMA and DCT algorithm when the quantizer scale(Q.S) is kept to be 16 and 20, respectively.

Security has become an important issue as information technology has become increasingly pervasive in our everyday lives. Security risks arise with a display that shows decrypted information. In this paper, we propose a secure information display technique by use of visual cryptography. Its decryption requires no special computing devices and is implemented using only human vision. To improve security of the key to decode the encrypted displayed image, multiple decoding masks have been constructed based on visual secret sharing scheme. The proposed display appears as a random pattern to anyone who looks at it unless the person views the displayed image through stacked multiple decoding masks. We have constructed code sets to share the secret image information with a displayed image and multiple decoding masks. The decoding process is a kind of optical processing based on spatial encoding and it needs no computer calculation, which means that no decrypted data exists in a computer system. Thus, the proposed display technique is secure against theft of the decrypted data and eavesdropping of the display signals, and provides a limited viewing zone. Multiple decoding masks increase security of the decoding masks.

In this paper, a novel shot boundary detection approach is presented, based on the popular region growing segmentation method - Watershed segmentation. In image processing, gray-scale pictures could be considered as topographic reliefs, in which the numerical value of each pixel of a given image represents the elevation at that point. Watershed method segments images by filling up basins with water starting at local minima, and at points where water coming from different basins meet, dams are built. In our method, each frame in the video sequences is first transformed from the feature space into the topographic space based on a density function. Low-level features are extracted from frame to frame. Each frame is then treated as a point in the feature space. The density of each point is defined as the sum of the influence functions of all neighboring data points. The height function that is originally used in Watershed segmentation is then replaced by inverting the density at the point. Thus, all the highest density values are transformed into local minima. Subsequently, Watershed segmentation is performed in the topographic space. The intuitive idea under our method is that frames within a shot are highly agglomerative in the feature space and have higher possibilities to be merged together, while those frames between shots representing the shot changes are not, hence they have less density values and are less likely to be clustered by carefully extracting the markers and choosing the stopping criterion.

Digital archiving and efficient retrieval of radiological scans have become critical steps in contemporary medical diagnostics. Since more and more images and image sequences (single scans or video) from various modalities (CT/MRI/PET/digital X-ray) are now available in digital formats (e.g., DICOM-3), hospitals and radiology clinics need to implement efficient protocols capable of managing the enormous amounts of data generated daily in a typical clinical routine. We present a method that appears to be a viable way to eliminate the tedious step of manually annotating image and video material for database indexing. MPEG-7 is a new framework that standardizes the way images are characterized in terms of color, shape, and other abstract, content-related criteria. A set of standardized descriptors that are automatically generated from an image is used to compare an image to other images in a database, and to compute the distance between two images for a given application domain. Text-based database queries can be replaced with image-based queries using MPEG-7. Consequently, image queries can be conducted without any prior knowledge of the keys that were used as indices in the database. Since the decoding and matching steps are not part of the MPEG-7 standard, this method also enables searches that were not planned by the time the keys were generated.

Online media server scheduling algorithms in distributed video-on-demand (VoD) systems are studied in this work. We first formulate a general server scheduling problem based on the VoD service model, where the failure rate and the server-side network bandwidth consumption are identified as two main cost factors in the system. The distributed server scheduler consists of two parts; namely, the request migration scheme and the dynamic content update strategy. By improving the random early migration (REM) scheme, we propose a cost-aware REM (CAREM) scheme to reduce the network bandwidth consumption in the migration process. Furthermore, to accommodate the video popularity and/or client population change, we use the server-video affinity to measure the importance of placing a specific video copy on that server. The dynamic content update strategy uses the server-video affinity metric to reconfigure video copies on media servers. We conduct extensive simulations to measure the performance of proposed algorithms. It can be shown that CAREM together with the dynamic content update strategy can improve the system performance by reducing the request failure rate as well as the server bandwidth consumption.

The noise-alike nature of astronomical images imposes a great challenge on compression. Due to the lack of correlation among adjacent pixels, it is very difficult to achieve good compression result using standard algorithms. To address the above challenge, a novel object-based compression method is proposed in this paper. Based on object analysis, the astronomical entities presented in the image are classified into two categories: clear and faint objects. For the former, a zerotree based wavelet compression algorithm is employed to achieve scalable coding; for the latter, a predictive coding method is used to preserve their location and intensity. The objective is to enhance the detection of faint object in astronomical images while providing a good overall visual effect. Experiment results demonstrate the superior performance of our proposed algorithm.

Intelligent video pre-processing and authoring techniques that facilitate people to create MTV-style music video clips are investigated in this research. First, we present an automatic approach to detect and remove bad shots often occurring in home video, such as video with poor lighting or motion blur. Then, we consider the generation of MTV-style video clips by performing video and music tempo analysis and seeking an effective way in matching these two tempos. Experiment results are given to demonstrate the feasibility and efficiency of the proposed techniques for home video editing.

With the availability of various wireless link-layer technologies, such as Bluetooth, WLAN and GPRS, in one wireless device, ubiquitous communications can be realized through managing vertical handoff in the environment of wireless overlay networks. In this paper, we propose a vertical handoff management system based on mobile IPv6, which can automatically manage the multiple network interfaces on the mobile device, and make decisions on network interface selection according to the current situation. Moreover, we apply our proposed vertical handoff management with JPEG-2000 codec to the wireless application of map image access. The developed system is able to provide seamless communications, as well as fast retrieve any interested map region with any block size, in different resolutions and different color representations directly from the compressed bitstream.

In this paper we propose the way to form a kind of medicine Image information sharing platform, which is based on PACS and broadband network. And we also discuss some key technologies used in building up the platform, such as sharing information between heterogeneous data sources based on HL7, storing and transmission the medical images based on DICOM. The study result shows that it can make full use of those heterogeneous data resources currently in different hospitals, and give them a good way to share the data.

This paper present video surveillance system based on double cameras that can make up the insufficiency of current video surveillance system. One camera with wide-angle lens monitors the full scene, another camera with zoom lens captures high distinguishable face image. The system detects face in the full scene image and makes sure the position of face, controls the rotatable camera moving. The system only save full scene images with moving object and face images that can occupy fewer memory space and be better suited for face retrieval and face recognition.

The power consumption for the battery-supplied DSP-embedded multimedia systems based on a test platform, i.e. TI C64x, is analyzed in this research. We focus on the behavior of some frequently used compress/decompress functional modules. In particular, a MPEG-4 simple-profile decoder consisting of these modules is evaluated at the highest compiler optimization level so as to understand power allocation in embedded multimedia systems. Two DCT schemes are examined to find out a better power behavior. The integer DCT can reach 47% power saving as compared with an implementation of the float DCT. Overall, our studies provide a better understanding of the system-level power modeling and consumption estimation for embedded multimedia applications, and suggest some optimization methods.

An image registration technique for compressed video such as motion JPEG or the I picture of MPEG based on the matching of DCT (Discrete Cosine Transform) coefficients is investigated in this research. Several simple features such as the DC value and a couple of low-frequency AC coefficients in the DCT domain are first extracted to indicate the edge strength and orientation inside each block for the image alignment purpose. Next, we conduct a coarse-level image segmentation task to filter out irrelevant regions. Then, for the regions of interest, we perform a more detail analysis to get the edge map. Finally, the alignment parameters are determined based on the information contained by the edge map. It is shown by experimental results that the proposed method reduces the computational cost of image registration dramatically as compared with the pixel domain registration technique while achieving certain quality of composition.

A progressive 3D mesh coding scheme using the octree-based space partitioning is proposed in this work, which achieves better coding efficiency than the state-of-the-art kd-tree-based codec. Given a 3D mesh, the quantized 3D vertices are first partitioned into an octree structure. The octree is then traversed from the root and gradually to the leaves. During the traversal, each 3D cell in the tree front is subdivided into eight child cells through three orthogonal cell bi-partitionings. For each cell subdivision, the information of nonempty child cells is encoded. To encode the information, two approaches (i.e. the bit-pattern coding approach and the nonempty-child-cell-tuple coding approach) are implemented and compared. In addition to the geometry coding, the local connectivity update associated with each cell subdivision is also encoded. Furthermore, selective cell subdivision is performed in the tree front to provide better rate-distortion performance, especially at low bitrates. It is shown in experimental results that the geometry coding cost is around 4.2 bits per vertex (bpv) for 8-bit coordinate quantization and 14.3 bpv for 12-bit coordinate quantization, and the connectivity coding cost is 3.3 bpv on the average.

In the case of high bit rate image transmission or having lots of packets, the FEC (forward error correction) encoding and decoding processes in the ULP (unequal loss protection) based schemes should be applied to individual packet groups instead of all the packets in order to avoid long processing delay. In this paper, we propose a layered ULP (L-ULP) scheme for fast and efficient FEC allocations among different packet groups and also within each packet group. The numerical results show that the proposed L-ULP scheme is quite promising for fast image transmission over packet loss networks.

With the increasing popularity of digital camera, organizing and managing the large collection of digital photos effectively are therefore required. In this paper, we study the photo album sorting, clustering and compression techniques in DCT frequency domain without having to decompress JPEG photos into spatial domain firstly. We utilize the first several non-zero DCT coefficients to build our feature set and calculate the energy histograms in frequency domain directly. We then calculate the similarity distance of every two photos, and perform photo album sorting and adaptive clustering algorithms to group the most similar photos together. We further compress those clustered photos by a MPEG-like algorithm with variable IBP frames and adaptive search windows. Our methods provide a compact and reasonable format for people to store and transmit their large number of digital photos. Experiments prove that our algorithm is efficient and effective for digital photo processing.

In this paper, we present techniques based on multiple wavelet-tree coding for robust image transmission. The algorithm of set partitioning in hierarchical trees (SPIHT) is a state-of-the-art technique for image compression. This variable length coding (VLC) technique, however, is extremely sensitive to channel errors. To improve the error resilience capability and in the meantime to keep the high source coding efficiency through VLC, we propose to encode each wavelet tree or a group of wavelet trees using SPIHT algorithm independently. Instead of encoding the entire image as one bitstream, multiple bitstreams are generated. Therefore, error propagation is limited within individual bitstream. Two methods based on subsampling and human visual sensitivity are proposed to group the wavelet trees. The multiple bitstreams are further protected by the rate compatible puncture convolutional (RCPC) codes. Unequal error protection are provided for both different bitstreams and different bit segments inside each bitstream. We also investigate the improvement of error resilience through error resilient entropy coding (EREC) and wavelet tree coding when channels are slightly corruptive. A simple post-processing technique is also proposed to alleviate the effect of residual errors. We demonstrate through simulations that systems with these techniques can achieve much better performance than systems transmitting a single bitstream in noisy environments.

In this paper, a video encryption algorithm based on MPEG4 codec is proposed, which combines encryption process with MPEG4 compression process. This algorithm is composed of two parts: VOP (Video Object Plane) encryption and VOL (Video Object Layer) encryption. In VOP encryption, the intra-encoded VOP is encrypted by inter-block permutation, the intra-encoded code blocks are encrypted by bit-plane permutation and coefficient sign encryption, and the other code blocks are encrypted by motion vector encryption. In VOL encryption, the basic VOLs are encrypted while others left unchanged. It can obtain high speed, for only parts of VOLs are encrypted. And it supports direct bit-rate control, which means that the encrypted images can also be decrypted correctly after recompression or bit-rate conversion. What's more, it is more robust to transmission errors than traditional ciphers since the errors causing in the encrypted bit-stream cannot spread out. These properties make it suitable for secure multimedia transmission over network, such as Video-on-Demand systems, multimedia communication networks, mobile multimedia or wireless multimedia networks, and so on.