The Internet has witnessed rapidly increasing streaming traffic to various mobile devices. In this paper, through analysis of a server-side workload and experiments in a controlled lab environment, we find that current practice has introduced a significant amount of redundant traffic. In particular, for the popular iOS based mobile devices, accessing popular Internet streaming services typically involves about 10%-70% redundant traffic. Such a practice not only over-utilizes and wastes resources on the server side and the network (cellular or Internet), but also consumes additional battery power on user's mobile devices and leads to possible monetary cost. To alleviate such a situation without changing the server side or the client side, we design and implement CStreamer that can transparently work between existing mobile clients and servers. We have implemented a prototype and installed on Amazon EC2. Experiments conducted based on this prototype show that CStreamer can completely eliminate the redundant traffic without degrading user's QoS.

The Internet has witnessed rapidly increasing streaming traffic to various mobile devices. In this paper, we find that for the popular iOS based mobile devices, accessing popular Internet streaming services typically involves about 10% - 70% unnecessary redundant traffic. Such a practice not only overutilizes and wastes resources on the server side and the network (cellular or Internet), but also consumes additional battery power on users' mobile devices and leads to possible monetary cost. To alleviate such a situation without changing the server side or the iOS, we design and implement a CStreamer prototype that can transparently work between existing iOS devices and media servers. We also build a CStreamer iOS App to enable end users to access Internet streaming services via CStreamer. Experiments conducted based on this prototype running on Amazon EC2 show that CStreamer can completely eliminate the redundant traffic without degrading user's QoS.

Receiving Internet streaming services on various mobile devices is getting increasingly popular, and cloud platforms have also been gradually employed for delivering streaming services to mobile devices. While a number of studies have been conducted at the client side to understand and characterize Internet mobile streaming delivery, little is known about the server side, particularly for the recent cloud-based Internet mobile streaming delivery. In this work, we aim to investigate the Internet mobile streaming service at the server side. For this purpose, we have collected a 4-month server-side log on the cloud (with 1,002 TB delivered video traffic) from a top Internet mobile streaming service provider serving worldwide mobile users. Through trace analysis, we find that 1) a major challenge for providing Internet mobile streaming services is rooted from the mobile device hardware and software heterogeneity. In this workload, we find over 3,400 different hardware models with more than 100 different screen resolutions running 14 different mobile OS and three audio codecs and four video codecs. 2) To deal with the device heterogeneity, CPU-intensive transcoding is used on the cloud to customize the video to the appropriate versions at runtime for different devices. A video clip could be transcoded into more than 40 different versions to serve requests from different devices. 3) Compared to videos in traditional Internet streaming, mobile streaming videos are typically of much smaller size (a median of 1.68 MBytes) and shorter duration (a median of 2.7 minutes). Furthermore, the daily mobile user accesses are more skewed following a Zipf-like distribution but users' interests also quickly shift. Considering the huge demand of CPU cycles for online transcoding, we further examine server-side caching to reduce the total CPU cycle demand from the cloud. We show that a policy considering different versions of a video altogether outperforms other intuitive ones when the cache size is limited.

Receiving Internet streaming services on various mobile devices is getting more and more popular. To understand and better support Internet streaming delivery to mobile devices, a number of studies have been conducted. However, existing studies have mainly focused on the client side resource consumption and streaming quality. So far, little is known about the server side, which is the key for providing successful mobile streaming services. In this work, we set to investigate the Internet mobile streaming service at the server side. For this purpose, we have collected a one-month server log (with 212 TB delivered video traffic) from a top Internet mobile streaming service provider serving worldwide mobile users. Through trace analysis, we find that (1) a major challenge for providing Internet mobile streaming services is rooted from the mobile device hardware and software heterogeneity. In this workload, we find over 2800 different hardware models with about 100 different screen resolutions running 14 different mobile OS and 3 audio codecs and 4 video codecs. (2) To deal with the device heterogeneity, transcoding is used to customize the video to the appropriate versions at runtime for different devices. A video clip could be transcoded into more than 40 different versions in order to serve requests from different devices. (3) Compared to videos in traditional Internet streaming, mobile streaming videos are typically of much smaller size (a median of 1.68 MBytes) and shorter duration (a median of 2.7 minutes). Furthermore, the daily mobile user accesses are more skewed following a Zipf-like distribution but users' interests also quickly shift. Considering the huge demand of CPU cycles for online transcoding, we further examine server-side caching in order to reduce CPU cycle demand. We show that a policy considering different versions of a video altogether outperforms other intuitive ones when the cache size is limited.

With the increasing deployment of Internet P2P/overlay streaming systems, more and more clients use mobile devices, such as smart phones and PDAs, to access these Internet streaming services. Compared to wired desktops, mobile devices normally have a smaller screen size, a less color depth, and lower bandwidth and thus cannot correctly and effectively render and display the data streamed to desktops. To address this problem, in this paper, we propose PAT (Peer-Assisted Transcoding) to enable effective online transcoding in P2P/overlay streaming. PAT has the following unique features. First, it leverages active peer cooperation without demanding infrastructure support such as transcoding servers. Second, as online transcoding is computationally intensive while the various devices used by participating clients may have limited computing power and related resources (e.g., battery, bandwidth), an additional overlay, called metadata overlay, is constructed to instantly share the intermediate transcoding result of a transcoding procedure with other transcoding nodes to minimize the total computing overhead in the system. The experimental results collected within a realistically simulated testbed show that by consuming 6% extra bandwidth, PAT could save up to 58% CPU cycles for online transcoding.

Community streaming is an enhanced form of joint content viewing where a sense of community is reinforced by the addition of interactive visual overlays, controlled in real-time by viewers, on top of a shared video stream. As a concrete example, we describe a community video system called ECHO, where personalized avatars are overlaid on top of a real-time encoded video stream of an Internet game for multicast consumption. Recognizing that only the visual overlays are generated live, we propose schemes that encode and schedule the live and non-live portions of the overlaid video separately in order to exploit the difference in delay sensitivity of the two, leading to video streams that contain two sub-streams with different delay constraints. We show that, in the known channel case, a low complexity ldquoearliest deadline firstrdquo packet scheduling algorithm minimizes receiver buffer delay. We also analyze the case where multiple streams are multiplexed, which allows us to quantify the potential gains of allowing different delay constraints for different sub-streams. We show that a ldquowater fillingrdquo strategy maximizes the total number of streams that can be supported. Simulation results show that the bandwidth necessary to maintain low-latency for visual overlays is reduced by about 40% when our proposed sub-stream approach is used. For multiplexing of multiple streams, our approach can increase the number of supported streams (e.g., a 30% increase when around ten streams are multiplexed).

This paper studies the temporal error propagation of small errors in a single frame for H.264 video. Such small errors can arise due to imperfect recovery from loss, e.g., through error concealment. The key contribution of this paper include demonstrating empirically that small errors tend to amplify over time, and is primarily caused by rounding errors in motion compensation and selective application of deblocking filter based on thresholding. Some methods of reducing error amplification is also presented.

The increase of aggregate Internet bandwidth and the rapid development of 3G wireless networks demand efficient delivery of multimedia objects to all types of wireless devices. To handle requests from wireless devices at runtime, the transcoding-enabled caching proxy has been proposed to save transcoded versions to reduce the intensive computing demanded by online transcoding. Constrained by available CPU and storage, existing transcoding-enabled caching schemes always selectively cache certain transcoded versions, expecting that many future requests can be served from the cache. But such schemes treat the transcoder as a black box, leaving no room for flexible control of joint resource management between CPU and storage. In this paper, we first introduce the idea of meta-caching by looking into a transcoding procedure. Instead of caching certain selected transcoded versions in full, meta-caching identifies intermediate transcoding steps from which certain intermediate results (calledmetadata) can be cached so that a fully transcoded version can be easily produced from the metadata with a small amount of CPU cycles. Achieving big saving in caching space with possibly small sacrifice on CPU load, the proposed meta-caching scheme provides a unique method to balance the utilization of CPU and storage resources at the proxy. We further construct a model to analyze the meta-caching scheme. Based on the analysis, we proposeAMTrac,AdaptiveMeta-caching forTranscoding, which adaptively applies meta-caching based on the client request patterns and available resources. Experimental results show that AMTrac can significantly improve the system throughput over existing approaches.

The five articles in this special section focus on multimedia applications in mobile/wireless context. Two papers focus on video transmission over 802.11a/b and WiMax, the next discusses a visual sensor network in wireless environments, and the last two propose supporting technologies for two mobile applications that rely on multimedia interactions.

Massive Internet media distribution demands prolonged continuous consumption of networking and disk bandwidths in large capacity. Many proxy-based Internet media distribution algorithms and systems have been proposed, implemented, and evaluated to address the scalability and performance issue. However, few of them have been used in practice, since two important issues are not satisfactorily addressed. First, existing proxy-based media distribution architectures lack an efficient media distribution control mechanism. Without copyright protection, content providers are hesitant to use proxy-based fast distribution techniques. Second, little has been done to protect client privacy during content accesses on the Internet. Straightforward solutions to address these two issues independently lead to conflicts. For example, to enforce distribution control, only legitimate users should be granted access rights. However, this normally discloses more information (such as which object the client is accessing) other than the client identity, which conflicts with the client's desire for privacy protection. In this article, we propose a unified proxy-based media distribution protocol to effectively address these two problems simultaneously. We further design a set of new algorithms in a cooperative proxy environment where our proposed scheme works efficiently and practically. Simulation-based experiments are conducted to extensively evaluate the proposed system. Preliminary results demonstrate the effectiveness of our proposed strategy.

Semi-fragile authentication schemes are found to be useful to maintain authentication in the event of certain allowed manipulation of source content. This paper focuses on a quantization based algorithm, and extends it to more general uniform quantizers. Earlier work is found to be a special case of our generalized solution which provides more useful guideline when selecting quantizer for this type of authentication. Experimental results verified our analytical conclusion.

Many algorithmic efforts have been made to address technical issues in designing a streaming media caching proxy. Typical of those are segment-based caching approaches that efficiently cache large media objects in segments which reduces the startup latency while ensuring continuous streaming. However, few systems have been practically implemented and deployed. The implementation and deployment efforts are hindered by several factors: 1) streaming of media content in complicated data formats is difficult; 2) typical streaming protocols such as RTP often run on UDP; in practice, UDP traffic is likely to be blocked by firewalls at the client side due to security considerations; and 3) coordination between caching discrete object segments and streaming continuous media data is challenging. To address these problems, we have designed and implemented a segment-based streaming media proxy, called SProxy. This proxy system has the following merits. First, SProxy leverages existing Internet infrastructure to address the flash crowd. The content server is now free of the streaming duty while hosting streaming content through a regular Web server. Thus, UDP based streaming traffic from SProxy suffers less dropping and no blocking. Second, SProxy streams and caches media objects in small segments determined by the object popularity, causing very low startup latency, and significantly reducing network traffic. Finally, prefetching techniques are used to pro-actively preload uncached segments that are likely to be used soon, thus providing continuous streaming. SProxy has been extensively tested and we show that it provides high quality streaming delivery in both local area networks and wide area networks (e.g., between Japan and the U.S.).

SP-frame is a new picture type supported by H.264, and supports functions such as rate-switching and random-access. In this paper, we investigate several complementary methods to improve the coding efficiency of SP-frames. We show that by appropriately choosing reference pictures, the size of secondary SP frames can be reduced by up to 40% and 2% for random-access and rate-switching, respectively. We also demonstrate that a simple rule exists that allows the joint selection of the two quantization parameters associated with SP frames to minimize "requantization" error. Results shows 0.1 dB PSNR improvement over comparable choices.

Hosting Internet streaming services has its unique challenges. Aiming at making Internet streaming services be widely and easily adopted in practice, in this paper, we have designed and implemented a system, called SProxy that can leverage existing Internet infrastructure to free the streaming content providers so that they only need to host streaming content through a regular Web server. SProxy has been extensively tested and evaluated and it provides high quality streaming delivery in both local area networks and wide area networks (e.g. between Japan and US)

Bit rate adaptation is one of the most important types of video transcoding. With H.264 becoming the predominant video codec of choice in video coding and streaming, prudent rate-adaptation techniques should be developed. In this paper, we investigate certain critical points in the spectrum of rate shaping requests. We show that the selection of quantization step sizes may not have monotonic effects on rate-distortion characteristics in the transcoding sense. In other words, distortion in the regular sense can be different from the distortion in the context of transcoding in which a requantization process is carried out. We show in a generic form that careful selections of the step size can lead to much improved performance. Experiments based on both simulation and real transcoding show the effectiveness of the proposed solution

Massive Internet media distribution demands pro longed continuous consumption of networking and disk band widths in large capacity. Many proxy-based Internet media distribution algorithms and systems have been proposed, implemented, and evaluated to address the scalability issue. However, few of them have been used in practice, since two important issues are not satisfactorily addressed. First, existing proxy-based media distribution architectures lack an efficient media distribution control mechanism. Without protection on the Internet, content providers are hesitant to use existing fast distribution techniques. Second, little has been done to protect client privacy during client accesses. Straightforward solutions to address these two issues independently lead to conflicts. For example, to enforce distribution control, only legitimate users should be granted access rights. However, this normally discloses more information (such as which object the client is accessing) other than the client identity, which conflicts with the client's desire for privacy protection. In this paper, we propose a unified proxy-based media distribution protocol to effectively address these two problems simultaneously. We further design a set of new algorithms for cooperative proxies where our proposed scheme works practically. Simulation results show that our proposed strategy is efficient

Streaming video to mobile devices like cellular phones is an important emerging application. However, mobile thin clients have limited capabilities and offer restricted interfaces to the end-user and the Telco infrastructure has significant technical barriers which limit the introduction of new protocols. This paper describes our design of an end-to-end interactive video service. We outline the technological and user-interaction challenges of providing one typical type of interactive controls-time-scale modification (TSM), when streaming compressed audio-video data within the Telco infrastructure in a way that maintains audio-video synchronization and respects the video frame and bit-rate contracts

The increase of aggregate Internet bandwidth and the rapid development of 3G wireless networks demand efficient delivery of multimedia objects to all types of wireless devices. To handle requests from wireless devices at runtime, the transcode-enabled caching proxy has been proposed and a lot of research has been conducted to study online transcoding. Since transcoding is a CPU-intensive task, the transcoded versions can be saved to reduce the CPU load for future requests. However, extensively caching all transcoded results can quickly exhaust cache space. Constrained by available CPU and storage, existing transcode-enabled caching schemes always selectively cache certain transcoded versions, expecting that many future requests can be served from the cache while leaving CPU cycles for online transcoding for other requests. But such schemes treat the transcoder as a black box, leaving little room for flexible control of joint resource management between CPU and storage. In this paper, we first introduce the idea of meta-caching by looking into a transcoding procedure. Instead of caching certain selected transcoded versions in full, meta-caching identifies intermediate transcoding steps from which certain intermediate results (called metadata) can be cached so that a fully transcoded version can be easily produced from the metadata with a small amount of CPU cycles. Achieving big saving in caching space with possibly small sacrifice on CPU load, the proposed meta-caching scheme provides a unique method to balance the utilization of CPU and storage resources at the proxy. We further construct a model to analyze the meta-caching scheme. Based on modeling results, we propose AMTrac, Adaptive Meta-caching for Transcoding, which adaptively applies meta-caching based on the client request pattern and available resources. Experimental results show that our proposed AMTrac can significantly improve the system throughput over existing approaches.

Researchers often use segment-based proxy caching strategies to deliver streaming media by partially caching media objects. The existing strategies mainly consider increasing the byte hit ratio and/or reducing the client perceived startup latency (denoted by the metric delayed startup ratio). However, these efforts do not guarantee continuous media delivery because the to-be-viewed object segments may not be cached in the proxy when they are demanded. The potential consequence is playback jitter at the client side due to proxy delay in fetching the uncached segments, which we call proxy jitter. Thus, for the best interests of clients, a correct model for streaming proxy system design should aim to minimize proxy jitter subject to reducing the delayed startup ratio and increasing the byte hit ratio. However, we have observed two major pairs of conflicting interests inherent in this model: (1) one between improving the byte hit ratio and reducing proxy jitter, and (2) the other between improving the byte hit ratio and reducing the delayed startup ratio. In this study, we first propose and analyze prefetching methods for in-time prefetching of uncached segments, which provides insights into the first pair of conflicting interests. Second, to address the second pair of the conflicting interests, we build a general model to analyze the performance tradeoff between the second pair of conflicting performance objectives. Finally, considering our main objective of minimizing proxy jitter and optimizing the two tradeoffs, we propose a new streaming proxy system called Hyper Proxy. Synthetic and real workloads are used to evaluate our system. The performance results show that Hyper Proxy generates minimum proxy jitter with a low delayed startup ratio and a small decrease of byte hit ratio compared with existing schemes.

Inter frames in compressed video make transcoding a complicated task since motion information is involved. This is especially true for downscale (spatial resolution) video transcoding in which input motion information cannot be precisely mapped to the output, which causes motion drift. Often carried out in full resolution, the drift compensation loop is computing intensive. In this paper, we model the entire process of downscale transcoding and propose a transcoding architecture that performs all computing-intensive loops in reduced resolution. We show that the computing load is significantly reduced while the quality is preserved.

Rate control via /spl rho/-domain rate-distortion analysis (both rate and distortion are considered as functions of /spl rho/, the percentage of zero transform coefficients) relies on a distortion model which can be erroneous in many situations, especially at the macroblock level (Z. He and S. K. Mita, 2002). This paper uses an accurate distortion measurement in the transform domain to drive a better macroblock level bit allocation. Specifically, we propose an algorithm based on an accurate distortion-to-/spl rho/ mapping in order to minimized the distortion while matching the target rate. Experimental results show that our proposed algorithm achieved 0.1 to 0.3 dB better PSNR compared to the pure /spl rho/-domain rate control.

Requantization has many applications in the processing of compressed multimedia content. However, no work has thorough analysis on its dynamics. A previous work has only numerically evaluated this process since it is found difficult to track analytically. We present a new approach in this paper to carry out an analytical study based on analysis of the intervals using number theory. This work proves some of the insights proposed before as propositions. The theoretical results also serve as a foundation in further algorithm development for requantization-related applications

For downscale (spatial resolution reduction) video transcoding, previous work has shown that motion vectors in the input video can be reused so that the costly motion estimation process can be avoided. Since the derived motion vectors may not be perfectly aligned with the input ones, a drift compensation loop consisting of inverse transform and motion compensation (or transform domain motion compensation) modules is required. However, the motion compensation is always performed based on macroblocks in the full resolution, which makes the full-resolution drift compensation inevitable. This fact renders the drift compensation loop as the next computational bottleneck. Through analytical modeling of the transcoding architectures, we propose a new architecture that employees a submacroblock motion compensation method. This method enables the drift compensation loop to be performed entirely in the reduced resolution. In addition, the memory space required for the architecture is significantly reduced since the full-resolution frame buffers are no longer required. Extensive analysis and experiments show that the proposed system achieves further reduction in computing load comparing to prior systems with minimum quality degradation.

With the falling price of memory, an increasing number of multimedia servers and proxies are now equipped with a large memory space. Caching media objects in the memory of a proxy helps to reduce the network traffic, the disk I/O bandwidth requirement, and the data delivery latency. The running buffer approach and its alternatives are representative techniques to caching streaming data in the memory. There are two limits in the existing techniques. First, although multiple running buffers for the same media object co-exist in a given processing period, data sharing among multiple buffers is not considered. Second, user access patterns are not insightfully considered in the buffer management. In this paper, we propose two techniques based on shared running buffers in the proxy to address these limits. Considering user access patterns and characteristics of the requested media objects, our techniques adaptively allocate memory buffers to fully utilize the currently buffered data of streaming sessions, with the aim to reduce both the server load and the network traffic. Experimentally comparing with several existing techniques, we show that the proposed techniques achieve significant performance improvement by effectively using the shared running buffers.

With the falling price of the memory, an increasing number of multimedia servers and proxies are now equipped with a large DRAM memory space. Caching media objects in the memory of a proxy helps to reduce network traffic, disk I/O bandwidth requirement, and data delivery latency. The running buffer approach and its alternatives are representative techniques to cache streaming data in the memory. However, there are two limits in the existing techniques. First, although multiple running buffers for the same media object co-exist in a given processing period, data sharing among the multiple buffers is not considered. Second, user access patterns are not insightfully considered in the buffer management. In this paper, we propose two techniques based on shared running buffers (SRB) in the proxy to address these limits. Considering user access patterns and characteristics of the requested media objects, our techniques adoptively allocate memory buffers to fully utilize the currently buffered data of streaming sessions, with the aim to reduce both the server load and the network traffic. Experimentally comparing with several existing techniques, we show that the proposed techniques have achieved significant performance improvement by effectively using the shared running buffers.

H.264 uses a 4-tap integer transform (DCT-like) to produce transform coefficients. This is different from the 8-tap DCT that is used in most prior video coding standards such as MPEG-x. The paper propose two integer algorithms for transcoding from 8-tap DCT data to 4-tap DCT-like data. The approximate nature of the integer transcoding algorithms is quantitively evaluated. The results show that the approximation is hidden in most cases when bit rate reduction is considered.

Recent years have witnessed significant efforts in deriving and embedding semantic information in content for improved content retrieval, adaptation, and distribution. Relatively little work considers leveraging this semantic information in the infrastructure to better serve the needs of clients and achieve better cost-effectiveness of infrastructure resources. In this paper, we identify semantics that can be derived and extracted from the various components in a content distribution infrastructure, namely content semantics (from content source), infrastructure semantics and client semantics (from content consumer). We develop a semantic-enhanced distribution and adaptation framework (SEDAN) that achieves superior efficiency and provides content access and adaptation features that were not previously possible

Recent years have witnessed significant efforts in deriving and embedding semantic information in content for improved content retrieval, adaptation, and distribution. Relatively little work considers leveraging this semantic information in the infrastructure to serve the needs of clients better and achieve better cost-effectiveness of infrastructure resources. We identify semantics that can be derived and extracted from the various components in a content distribution infrastructure, namely content semantics (from content source), infrastructure semantics and client semantics (from content consumer). We develop a semantic-enhanced distribution and adaptation framework (SEDAN) that achieves superior efficiency and provides content access and adaptation features that were not previously possible.

Researchers often use segment-based proxy caching strategies to deliver streaming media by partially caching media objects. The existing strategies mainly consider increasing the byte hit ratio and/or reducing the client perceived startup latency (denoted by the metric delayed startup ratio). However, these efforts do not guarantee continuous media delivery because the to-be-viewed object segments may not be cached in the proxy when they are demanded. The potential consequence is playback jitter at the client side due to proxy delay in fetching the uncached segments, which we call proxy jitter. Thus, for the best interests of clients, a correct model for streaming proxy system design should aim to minimize proxy jitter subject to reducing the delayed startup ratio and increasing the byte hit ratio. However, we have observed two major pairs of conflicting interests inherent in this model: (1) one between improving the byte hit ratio and reducing proxy jitter, and (2) the other between improving the byte hit ratio and reducing the delayed startup ratio. In this study, we first propose an active prefetching method for in-time prefetching of uncached segments, which provides insights into the first pair of conflicting interests. Second, we further improve our lazy-segmentation scheme which effectively addresses the second pair of the conflicting interests. Finally, considering our main objective of minimizing proxy jitter and optimizing the two trade-offs, we propose a new streaming proxy system called Hyper Proxy by effectively coordinating both prefetching and segmentation techniques. Synthetic and real workloads are used to systematically evaluate our system. The performance results show that the hyper proxy system generates minimum proxy jitter with a low delayed startup ratio and a small decrease of byte hit ratio compared with existing schemes.

With the wide availability of high-speed network access, we are experiencing high quality streaming media delivery over the Internet. The emergence of ubiquitous computing enables mobile users to access the Internet with their laptops, PDAs, or even cell phones. When nomadic users connect to the network via wireless links or phone lines, high quality video transfer can be problematic due to long delay or size mismatch between the application display and the screen. Our proposed solution to this problem is to enable network proxies with the transcoding capability, and hence provide different, appropriate video quality to different network environment. The proxies in our transcoding-enabled caching (TeC) system perform transcoding as well as caching for efficient rich media delivery to heterogeneous network users. This design choice allows us to perform content adaptation at the network edges. We propose three different TeC caching strategies. We describe each algorithm and discuss its merits and shortcomings. We also study how the user access pattern affects the performance of TeC caching algorithms and compare them with other approaches. We evaluate TeC performance by conducting two types of simulation. Our first experiment uses synthesized traces while the other uses real traces derived from an enterprise media server logs. The results indicate that compared with the traditional network caches, with marginal transcoding load, TeC improves the cache effectiveness, decreases the user-perceived latency, and reduces the traffic between the proxy and the content origin server.

In bit rate reduction video transcoding, blockness in the transcoded result can become a major problem. Previous works have shown that deblocking can be achieved by recomputing some of the DCT coefficients. However, in video transcoding, these DCT coefficients are subject to a requantization process, which may render the deblocking ineffective. We propose to consider deblocking in conjunction with the requantization process therefore accelerating the transcoding processes. In addition, the method also motivates a design of an optimized quantizer selection algorithm that boosts the deblocking capability of the video transcoder.

This paper proposes novel concepts of meta-caching and meta-transcoding which indicate that intermediate results (metadata) of service (e.g., video transcoding) sessions are cached and subsequent identical service requests are served through transcoding from the cached metadata. The proposed methods provide a foundation to achieving superior storage and computation resource balance at service proxies. It is primarily useful in reducing the computing load of service proxies at the server side. The paper further identifies the appropriate metadata and its characteristics for three types of video transcoding tasks. Simulations based on these characteristics validate that the proposed methods achieve superior computing load reduction on transcoding service proxies.

Streaming media objects are often cached in segments. Previous segment-based caching strategies cache segments with constant or exponentially increasing lengths and typically favor caching the beginning segments of media objects. However, these strategies typically do not consider the fact that most accesses are targeted toward a few popular objects. In this paper, we argue that neither the use of a predefined segment length nor the favorable caching of the beginning segments is the best caching strategy for reducing network traffic. We propose an adaptive and lazy segmentation based caching mechanism by delaying the segmentation as late as possible and determining the segment length based on the client access behaviors in real time. In addition, the admission and eviction of segments are carried out adaptively based on an accurate utility function. The proposed method is evaluated by simulations using traces including one from actual enterprise server logs. Simulation results indicate that our proposed method achieves a 30% reduction in network traffic. The utility functions of the replacement policy are also evaluated with different variations to show its accuracy.

With the introduction of the next generation wireless network, mobile devices access increasingly more media-rich content. However, a key factor that prevents a mobile device from accessing multimedia content is that it does not have enough display real estate to render the content that is traditionally created for the desktop web client. Moreover, the wireless network typically has lower bandwidth compared to a wired network of the same time frame. Therefore, a transcoder is needed somewhere in the network to transform multimedia content to the appropriate form factor and bandwidth requirement. This paper introduces an extremely fast transcoder of such kind. Comparing to the previous methods, additional computation is saved while the quality of transcoding is maintained. This saving in computing resources maps into the benefit that more concurrent sessions can be supported on one transcoding device. This scalability is crucial for the wireless network to handle user requests that might be very intensive at times.

Media transcoding can be used to enable mobile devices that have low resolution and low bit-rate capabilities to access content created for stationary, desktop clients with high bandwidth connections. By deploying transcoding servers near the edge of a large network, it becomes possible to support clients with multiple resolutions, while limiting the bandwidth requirements in the core network.

Digital video is becoming widely available in compressed form, such as a motion JPEG or MPEG coded bitstream. In applications such as video browsing or picture-in-picture, or in transcoding for a lower bit rate, there is a need to downscale the video prior to its transmission. In such instances, the conventional approach to generating a downscaled video bitstream at the video server would be to first decompress the video, perform the downscaling operation in the pixel domain, and then recompress it as, say, an MPEG, bitstream for efficient delivery. This process is computationally expensive due to the motion-estimation process needed during the recompression phase. We propose an alternative compressed domain-based approach that computes motion vectors for the downscaled (N/2xN/2) video sequence directly from the original motion vectors for the N/spl times/N video sequence. We further discover that the scheme produces better results by weighting the original motion vectors adaptively. The proposed approach can lead to significant computational savings compared to the conventional spatial (pixel) domain approach. The proposed approach is useful for video severs that provide quality of service in real time for heterogeneous clients.

This paper presents an algorithm for DC-only fade-out operation on MPEG compressed video. The operation has no restriction on picture coding type (I, P or B) or GOP structure. The fade-out process can start at any frame and end at any frame. The DC-only approach assures an independent processing of the frames regardless of the coding dependencies among them. We take advantage of the temporal dependencies inherited from MPEG sequence to process the MPEG stream sequentially. An error-correction scheme is proposed to cope with the accumulated fade-out error due to adaptive quantization employed by MPEG. We also discuss syntax manipulations required to perform the fade-out operation directly in the MPEG domain. Real time fade-out can be achieved.

For direct manipulation in the compressed domain, we develop a DCT domain convolution theorem which besides exploiting the sparseness of the DCT domain representation also exploits the orthogonality and symmetry in the DCT domain representation. These properties lead to efficient compressed domain based processing methods unlike their spatial domain counterparts, where such properties are not available. This theorem can be used in a variety of image and video editing functions when the image and video data are available only as a JPEG or MPEG bitstream. We illustrate the use of these DCT domain convolution theorems in a typical video editing application such as video bluescreen editing.

Conventional processing of JPEG or MPEG compressed image or video data involves first decompressing the data and applying the desired processing function, and then the processed data are recompressed for the purposes of transmission or storage. We propose an alternate processing pipeline which involves direct manipulation of the JPEG or MPEG compressed domain representation to achieve the desired spatial domain processing. For direct manipulation in the compressed domain, we develop a discrete cosine transform (DCT)-domain convolution theorem which besides exploiting the sparseness of the DCT-domain representation also exploits the orthogonality and symmetry in the DCT-domain representation. These properties lead to efficient compressed domain-based processing methods unlike their spatial domain counterparts, where such properties are not available. This theorem can be used in a variety of image and video editing functions when the image and video data are available only as a JPEG or MPEG bitstream. We illustrate the use of the DCT-domain convolution theorem in a typical video editing application such as video bluescreen editing.

Digital video is becoming widely available only in the compressed form. In applications such as video browsing or picture-in-picture or transcoding for a lower bit rate, there is a need to downscale the video before transmission. In such instances, the conventional approach of generating a downscaled video bitstream at the video server is to first decompress the video, perform the downscaling operation in the pixel domain and then recompress it for efficient delivery. This process is computationally expensive due to the motion-estimation process needed during the recompression phase. We propose an alternative compressed domain based approach that computes the motion vectors for the downscaled (N/2/spl times/N/2) video sequence directly from the original motion vectors residing in the N/spl times/N video sequence. Furthermore, the downscaling of the DCT information is also done in the DCT domain. The proposed approach can lead to significant computational savings compared to the conventional spatial domain approach.

No abstract available.