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This paper discusses the background, challenges, opportunities for parallelization of low-latency video transport of JPEG XS streams over the real-time network protocol (RTP). Transport of compressed video signals over a constant bitrate channel requires smoothing buffers at encoder and decoder side. To limit their sizes in partical implementations, part 2 of the JPEG XS standard defines a normative buffer model along a sequential hypothetical reference decoder. Due to its sequential nature, it cannot be directly applied to high-speed applications that necessarily depend on multithreading. In this paper, the JPEG XS Part 2 buffer model is introduced, then the current state of JPEG XS RTP transport standardization is discussed, and multiple strategies for multithreaded encoding under latency contraints within this application are reported and analyzed.

Mobile compute environments provide a unique set of user needs and expectations that designers must consider. With increased multimedia use in mobile environments, video encoding methods within the smart phone market segment are key factors that contribute to positive user experience. Currently available display resolutions and expected cellular bandwidth are major factors the designer must consider when determining which encoding methods should be supported. The desired goal is to maximize the consumer experience, reduce cost, and reduce time to market. This paper presents a comparative evaluation of the quality of user experience when HEVC and AVC/H.264 video coding standards were used. The goal of the study was to evaluate any improvements in user experience when using HEVC. Subjective comparisons were made between H.264/AVC and HEVC encoding standards in accordance with Doublestimulus impairment scale (DSIS) as defined by ITU-R BT.500-13. Test environments are based on smart phone LCD resolutions and expected cellular bit rates, such as 200kbps and 400kbps. Subjective feedback shows both encoding methods are adequate at 400kbps constant bit rate. However, a noticeable consumer experience gap was observed for 200 kbps. Significantly less H.264 subjective quality is noticed with video sequences that have multiple objects moving and no single point of visual attraction. Video sequences with single points of visual attraction or few moving objects tended to have higher H.264 subjective quality.

The effective usage of transmission resources is becoming very important problem as new coding systems are extremely efficient and data transmission channels have higher throughput. That force us to look at the rate control algorithms. This paper presents test results of using local bitrate estimation instead of basic rate control algorithms for video sequences with intra-frame prediction. Local bitrate estimation enables inter and extrapolation of rho-rate function in neighborhood of currently used QP. This approach simplifies rate control process by excluding complex error-based operations.

We describe a method for searching videos in large video databases based on the activity contents present in the videos. Being able to search videos based on the contents (such as human activities) has many applications such as security, surveillance, and other commercial applications such as on-line video search. Conventional video content-based retrieval (CBR) systems are either feature based or semantics based, with the former trying to model the dynamics video contents using the statistics of image features, and the latter relying on automated scene understanding of the video contents. Neither approach has been successful. Our approach is inspired by the success of visual vocabulary of "Video Google" by Sivic and Zisserman, and the work of Nister and Stewenius who showed that building a visual vocabulary tree can improve the performance in both scalability and retrieval accuracy for 2-D images. We apply visual vocabulary and vocabulary tree approach to spatio-temporal video descriptors for video indexing, and take advantage of the discrimination power of these descriptors as well as the scalability of vocabulary tree for indexing. Furthermore, this approach does not rely on any model-based activity recognition. In fact, training of the vocabulary tree is done off-line using unlabeled data with unsupervised learning. Therefore the approach is widely applicable. Experimental results using standard human activity recognition videos will be presented that demonstrate the feasibility of this approach.

Nowadays, most video material is coded using a non-scalable format. When transmitting these single-layer video bitstreams, there may be a problem for connection links with limited capacity. In order to solve this problem, requantization transcoding is often used. The requantization transcoder applies coarser quantization in order to reduce the amount of residual information in the compressed video bitstream. In this paper, we extend a requantization transcoder for H.264/AVC video bitstreams with a rate-control algorithm. A simple algorithm is proposed which limits the computational complexity. The bit allocation is based on the bit distribution in the original video bitstream. Using the bit budget and a linear model between rate and quantizer, the new quantizer is calculated. The target bit rate is attained with an average deviation lower than 6%, while the rate-distortion performance shows small improvements over transcoding without rate control.

In this paper, the conformance of hypothetical reference decoder (HRD) is addressed for H.264 when there is jitter among the transmission of packets via a channel without packet loss but variation among transmission delay. The sending rate is decoupled from the coding one. Both the jitter and the total size of coded bitstream are taken into consideration such that the values of buffer size and initial buffer delay are minimized, especially when the sending rate is greater than the coding one. Sufficient conditions are derived for the conformance of a coded bitstream to a HRD at the constant delay. These conditions are then used to design iterative algorithms to determine a minimal buffer size and a minimal initial buffer delay for the decoder. A novel interpolation method is also presented such that it is suitable for a wide range of sending rates.

For real-time video streaming applications over the constant bit rate channels, it is highly desired that video signals can be encoded in not only good average quality but also smooth video quality. However, in the case that the network resource is sufficiently large and the video quality has reached the target quality, the quality smoothing is not necessary and the rate smoothing is desired to avoid overusing the unnecessary network resource but also achieve a smoothed traffic rate. In this paper, we propose a novel real-time rate-smoothed encoding scheme by applying the low pass filtering idea. Both theoretical analysis and experimental results show that the proposed rate-smoothed encoding scheme can achieve a target average quality while significantly reducing the peak rate and the rate variance. We have further proposed a joint quality and rate smoothed encoding scheme, which can provide adaptive smoothing according to different situations. Experimental results show that the proposed joint smoothing scheme can make an optimal balance between the quality fluctuation and the rate fluctuation, and hence improve the overall system performance.

An operational rate control (RC) scheme based on two-stage encoding is studied in this research, where frame-layer rate control with a constant bit rate to achieve constant video quality is examined. In the first encoding stage, the R-D optimized mode decision and its associated motion estimation (RDO) as well as DCT/Q, IQ/IDCT and entropy coding are performed for all macroblocks (MBs) for a target frame using an initial quantization parameter (QP), which is the QP of its previous frame. In the second encoding stage, the residual signal from the first stage is encoded using several QP values around the initial QP. Given the target bits and distortion for the current frame, the residual signal is finally encoded using the QP determined by comparing target bits and distortion with actual bits and distortion. To reduce the additional coding complexity of the two-stage encoding, upper and lower bounds around target bits and distortion are employed to reduce the number of encoding required in the second stage. Experimental results are given to show the superior performance of two proposed rate control algorithms, where one targets at the constant bit rate while the other at constant quality.

Bit-plane coding techniques have been prop osed as an efficient way to achieve the goal of fine granularity scalability (FGS). Both in image and video coding standards such as JPEG 2000 and MPEG-4, bit-plane coding techniques have been utilized to code a universal bit-st ream for various bandwidth applications. W ith the bit-plane coding, a bit-stream can be truncated according to the available bandwidth. Hen ce, the video quality can be very sensitive to the network bandwidth variation especially while streaming a video over the Internet or a wireless link. For overall human perception, a consistent and smooth quality video is much better than to have high vi deo quality in just a short period but with highly quality variation in general. We classify th e quality variation into two categories: inter-frame and intra-frame quality variation. The inter-frame quality variation may cause by the constant bit rate coded (CBR) base layer video or the network bandwidth variation, while the intra -frame quality variation causes mainly by the various motion types. For example, moving objects are usually more difficult to be encoded. This paper focuses on the inter-frame quality variation because quality variation is more significant between frames than within a frame. An enhancement layer labeling algorithm is proposed to reduce the inter-frame quality variation, and a dynamic bit-plane truncation scheme is utilized to maintain the smooth streaming video quality. The proposed algorithm has no overhead in the decoder, and the experimental results portray that our prop osed algorithm can greatly el iminate the inter-frame quality variation. Keywords: Bit-plane, smooth quality streaming, MPEG-4, fine granularity scalability, FGS, scalable video coding.

Leaky prediction based FGS (Fine Granularity Scalability) can achieve better coding efficiency than the baseline FGS. However, for leaky prediction based FGS (L-FGS), constant quality constrained bit allocation, i.e., how to optimally allocate bits given the current channel bandwidth, is still an open problem. In this paper, based on the accurate R-D (Rate-Distortion) model developed in our previous work, we propose a constant quality constrained bit allocation scheme for L-FGS. The proposed scheme is a combination of offline and online processes. During the offline stage, we perform the L-FGS encoding and collect the necessary feature information. At the online stage, given the transmission bandwidth at that time, we quickly estimate the R-D curves of a sequence of consecutive video frames based on our previously developed R-D model and then perform the corresponding bit allocation using a sliding window technique. Experimental results show that our proposed bit allocation algorithm can achieve much more smooth video quality than the traditional uniform bit allocation under both CBR (constant bit rate) and VBR (variable bit rate) channels.

Most video rate-control research emphasizes constant bit-rate (CBR) applications. These aim to produce a CBR bitstream with the highest possible quality, within the bitrate constraint and with no consideration for quality variation. In this paper, two MPEG-4 Constant-Quality (CQ) CBR controls are proposed. These aim to produce a CBR bitstream that meets a target quality level whenever possible. The Frame-level Laplacian CQ (FLCQ) algorithm uses a distortion model based on a Laplacian model for DCT coefficients. In contrast, the MB-level Viterbi CQ (MVCQ) algorithm uses the Viterbi algorithm to determine the best combination of MB-QP’s. “CQ” is measured by the deviation of the mean quality from the target quality, and by quality variance over time. Simulation results suggest that the proposed algorithms perform better than Q2 and TM5 under these measures. In some cases, they produce bitstreams with fewer bits while having higher average PSNR, and smaller variance. The FLCQ algorithm has more variation in quality than the MVCQ algorithm. With extra computational complexity, the MVCQ algorithm gives the best performance over all algorithms tested. Often, it precisely meets the target PSNR with no variation. This is truly a CQ rate-control algorithm.

An optimal broadcasting scheme under the presence of secondary content (i.e. advertisements) is proposed. The proposed scheme works both for movies encoded in a Constant Bit Rate (CBR) or a Variable Bit Rate (VBR) format. It is shown experimentally that secondary content in movies can make Video-on-Demand (VoD) broadcasting systems more efficient. An efficient algorithm is given to compute the optimal broadcasting schedule with secondary content, which in particular significantly improves over the best previously known algorithm for computing the optimal broadcasting schedule without secondary content.

The introduction of new ATM service categories increases the benefits of ATM, making the technology suitable for a virtually unlimited range of applications. Connection Admission Control (CAC) is defined as the set of actions taken by the network during the call (virtual connection) set-up phase, or during call re-negotiation phase, to determine whether a connection request can be accepted or rejected. Network resources (port bandwidth and buffer space) are reserved to the incoming connection at each switching element traversed, if so required, by the service category. The major focus of this paper is call admission in the context of multi-service, multi-class ATM networks. Several strategies suggesting rules on bandwidth sharing are found in the litterature. This study investigates particularly the Complete Sharing approach. Two service categories are concerned, namely, Constant Bit rate/Deterministic Bit Rate (CBR/DBR) and Variable Bit Rate/Statistical Bit Rate (VBR/SBR). Each service category is represented by a set of call classes corresponding to different bandwidth needs. We propose two algorithms to solve the underlying Markovian system: Product-form and Recursive solutions. A performance study based on the latter algorithm is implemented. We analyze the results of this very sharing strategy and set the not-to-violate limits for a beneficial use of it.

We present a new approach to video coding that applies video analysis based on global motion features. A super-resolution mosaic is build for each frame to be encoded from a number of previously transmitted frames. This super-resolution mosaic is used to detect macroblocks that are only affected by global motion. For such macroblocks no prediction error is transmitted. They are purely reconstructed by prediction from the super-resolution mosaic which results in significant bitrate savings. Our results indicate total bitrate savings of 20% and more compared to a state of the art H.264/AVC codec at the same visual quality.

A flow control for streaming multimedia data over UDP on IP network is presented. The bitrate adaptation algorithm embedded in the protocol is considered to be an end-to-end mechanism in the application level. The flow control system constantly maintains the streaming buffer at a prescribed capacity even with bursty network losses by adapting the multimedia bitrate B from the streaming codec. A congestion control algorithm which is considered to be located in a lower level than that of the flow control mechanism is presented. It works together with the presented flow control to resolve network congestion problems while maintaining a degree of TCP-friendliness by changing the sending rate R at the server. Simulation results obtained from NS2 have shown better resource allocation can be obtained, and an overall incrase in the average sending rate and hence better quality of streaming media is observed.

There have been increasing market demands to develop more effective schemes for applying MPEG video coding standard to meet the challenges from the emerging applications. We present a real-time multi-pass MPEG video coding sys tem for constant bit rate (CBR) applications, such as Video-on-Demand (VoD) over ADSL. This scheme employs two MPEG encoders. Both encoders operate at the same CBR. The second encoder has a buffer to delay the input by an amount of time relative to the first encoder to create a look-ahead window. In encoding, the first encoder collects statistical and rate-quality characteristic information. An on-line processor then uses the collected information to derive the best coding strategy for the second encoder to encode the incoming frames in the look-ahead window. The second encoder uses the encoding parameters from the processor as the coding guide to execute the coding strategy and produce the compressed bitstream. Results on an IBM MPEG-2 encoder testing system with a 15 frame look-ahead window have shown that the scheme can obtain 0.6 to 1.6dB improvement in PSNR over the conventional single encoder on MPEG test video sources. Moreover, measured by the Tektronix Inc.'s Picture Quality Analysis (PQA) System, this scheme achieved 0.87 ~ 1.35 reduction in PQA index. The visual quality improvment is also very evident.

End-to-End (ETE) Quality of Service (QoS) is critical for next generation wireless multimedia communication systems. To meet the ETE QoS requirements, Universal Mobile Telecommunication System (UMTS) requires not only meeting the 3GPP QoS requirements [1-2] but also mapping external network QoS classes to UMTS QoS classes. There are four Quality of Services (QoS) classes in UMTS; they are Conversational, Streaming, Interactive and Background. There are eight QoS classes for LAN in IEEE 802.1 (one reserved). ATM has four QoS categories. They are Constant Bit Rate (CBR) - highest priority, short queue for strict Cell Delay Variation (CDV), Variable Bit Rate (VBR) - second highest priority, short queues for real time, longer queues for non-real time, Guaranteed Frame Rate (GFR)/ Unspecified Bit Rate (UBR) with Minimum Desired Cell Rate (MDCR) - intermediate priority, dependent on service provider UBR/ Available Bit Rate (ABR) - lowest priority, long queues, large delay variation. DiffServ (DS) has six-bit DS codepoint (DSCP) available to determine the datagram's priority relative to other datagrams and therefore, up to 64 QoS classes are available from the IPv4 and IPv6 DSCP. Different organisations have tried to solve the QoS issues from their own perspective. However, none of them has a full picture for end-to-end QoS classes and how to map them among all QoS classes. Therefore, a universal QoS needs to be created and a new set of QoS classes to enable end-to-end (ETE) QoS provisioning is required. In this paper, a new set of ETE QoS classes is proposed and a mappings algorithm for different QoS classes that are proposed by different organisations is given. With our proposal, ETE QoS mapping and control can be implemented.© (2002) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper describes a bit allocation algorithm that can achieve a constant bit rate when coding multiple video objects (MVOs), while improving the rate-distortion (R-D) performance over the reference method for MPEG-4 object-based rate control. In object-based coding, bit allocation is performed at the object level and temporal rates of different objects may vary. The proposed algorithm in this paper deals with these two issues. We pay particular attention to maintenance of buffer occupancy levels and propose a new method for spatio-temporal trades-offs for object-based coding. In order to improve the coding efficiency, we also consider several new R-D coding modes. These modes are available at the encoder to determine the best R-D performance under different coding conditions and are chosen automatically. Simulation results demonstrate moderate improvements at low and high bit rates. One important key aspect of the proposed algorithm is that the actual coded bits in the proposed algorithm are similar to the target bits over a wide range of bit rates. Consequently, the proposed algorithm has not experienced the buffer overflow/underflow over the entire range of bit rates.

An encoder-based rate smoothing technique is developed to reduce the rate burstiness in video coding. At first, we propose a new source model to accurately estimate the rate-distortion (R-D) function of each video frame with very low computational complexity. Based on the estimated R-D function, in the second step, we propose a rate smoothing algorithm to make a tradeoff between the constant bit rate and the constant picture quality. The tradeoff forces the output video stream to have both a smoothed rate shape and a consistent video presentation quality, which is highly desirable in video coding and transportation.

With the increasing importance of heterogeneous networks and time-varying communication channels, such as packet-switched networks and wireless communications, fine scalability has become a highly desirable feature in both image and video coders. A single highly scalable bitstream can provide precise rate control for constant bitrate (CBR) traffic and accurate quality control for variable bitrate (VBR) traffic. In this paper, we propose two methods that provide constant quality video under buffer constraints. These methods can be used with all scalable coders. Experimental results using the Motion JPEG2000 coder demonstrate substantial benefits.

MPEG family codecs generate variable-bit-rate (VBR) compressed video with significant multiple-time-scale bit rate variability. Smoothing techniques remove the periodic fluctuations generated by the codification modes. However, global efficiency concerning network resource allocation remains low due to scene-time-scale variability. RCBR techniques provide suitable means to achieving higher efficiency. Among all RCBR techniques described in literature, 2RCBR mechanism seems to be especially suitable for video-on demand. The method takes advantage of the knowledge of the stored video to calculate the renegotiation intervals and of the client buffer memory to perform work-ahead buffering techniques. 2RCBR achieves 100% bandwidth global efficiency with only two renegotiation levels. The algorithm is based on the study of the second derivative of the cumulative video sequence to find out sharp-sloped inflection points that point out changes in the scene complexity. Due to its nature, 2RCBR becomes very adequate to deliver MPEG2 scalable sequences into the network cause it can assure a constant bit rate to the base MPEG2 layer and use the higher rate intervals to deliver the enhanced MPEG2 layer. However, slight changes in the algorithm parameters must be introduced to attain an optimal behavior. This is verified by means of simulations on MPEG2 video patterns.© (2001) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Most existing H.263+ rate control algorithms, e.g. the one adopted in the test model of the near-term (TMN8), focus on the macroblock layer rate control and low latency under the assumptions of with a constant frame rate and through a constant bit rate (CBR) channel. These algorithms do not accommodate the transmission bandwidth fluctuation efficiently, and the resulting video quality can be degraded. In this work, we propose a new H.263+ rate control scheme which supports the variable bit rate (VBR) channel through the adjustment of the encoding frame rate and quantization parameter. A fast algorithm for the encoding frame rate control based on the inherent motion information within a sliding window in the underlying video is developed to efficiently pursue a good tradeoff between spatial and temporal quality. The proposed rate control algorithm also takes the time-varying bandwidth characteristic of the Internet into account and is able to accommodate the change accordingly. Experimental results are provided to demonstrate the superior performance of the proposed scheme.

An algorithm was developed with the goal of automatically recognizing isochronous traffic in a packet switched network. Designed with constant bit rate traffic in mid, this algorithm examines the interarrival times between packets in flows classifying as isochronous those flows that have both an average inter arrival times between packets in flows classifying is isochronous those flows that have both an average interarrival time consistent with an isochronous application and sufficiently low variance for interarrival times. The algorithm was analyzed based on a priori considerations and was then applied to actual traffic in an effort to characterize the parameters for appropriate use. Traffic flows from several common isochronous applications were examined focusing on common CODECs. From a priori considerations, the algorithm has the desirable characteristics of building intelligence into the network and being purely local to the individual switch or router on which it is installed while totally transparent to other equipment. While the approach appears sound in general principle, the isochronous applications examined show a much higher degree of variability in packet rates than expected. Moreover, most common applications seem to be shifting to CODECs that produce variable bit rate traffic. Thus, using the algorithm with such applications would require a st of threshold parameters that would likely result in a high misclassification rate.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The increasing demand on high quality digital video broadcasting systems has driven the developments of several coding standards, the motion picture experts group (MPEG) standard 4 being the most recent. Since these standards only define the bitstream syntax, such essential features of the coder as controlling the output bitrate have to be implemented individually. An important problem arising here is the optimization of the bitrate-distortion trade-off, since in general the resulting bitstream has to meet the requirements of a communication channel with limited bandwidth. Obviously the output bitrate can be influenced by adjusting the quantization of the coder, but however a general functional relation to predict bitrate and distortion at a given quantization for a frame does not exist, since these parameters depend strongly on the properties of the actual video sequence. With the help of a predictor adapting on a specific video sequence, it would be possible to run an algorithm, which calculates the optimal quantization parameters causing the coder to meet the bitrate requirements of the given channel. Main objective of this paper is to design az model of a MPEG-4 coder with reduced complexity that is capable to predict the functional relation between quantization, bitrate and distortion for the next frame to be coded of a given video sequence. This model will be implemented as a neuronal network forming a two-layer perceptron.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper addresses the problem of efficient traffic shaping while providing QoS guarantees for Constant Bit Rate (CBR) sources. A simulation model is developed to capture the performance of several mechanisms that schedule cells according to the Generic Cell Rate Algorithm which is used for policing of connections. Results show that efficient shaping with QoS guarantees can be only achieved by a multi- level priority scheme. The suggested shaper mechanism absorbs the Cell Delay Variation of CBR cells and minimizes the buffer required at the receiver end.© (1997) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The rate control problem can be greatly simplified by using a wavelet coder, since the desired bit rate for aparticular frame can be easily reached by using its embedding property. We convert the rate control problem to abit allocation problem for each frame, and solve it with two models. Frames are assumed to be independent in the basic model while their dependency is taken into account in the advanced model. Two computationally efficient rate control algorithms are then derived. Extensive experiments are performed to demonstrate the superior performanceof the wavelet coder with the proposed rate control schemes over the MPEG standard with test model 5 (TM5).Keywords: rate control, video coding, wavelet coding. 1 INTRODUCTION Rate control is essential to video transmission through constant bit rate (CBR) channels such as ISDN, Ti andbroadcasting channels. It is also important in other applications such as video editing and video storage with CD- ROM or digital video disk (DVD) . The

Broadening the ATM offer with a full set of service categoriesLivio Lambarelli, Matteo D'Ambrosio, Paolo CastelliCSELT, Cemtro Studi E Laboratori Telecomunicazioni S.p.A.Via Reiss Romoli, 274,10148 Torino, Italy)ABSTRACTThe paper aims at providing a comprehensive picture of the new set of ATM-layer services defined in parallel by the ITU-T and The ATM Forum, outlining the potential benefits that can derive, both to users and network service providers, from theavailability ofthese new functionalities in an ATM-based network.Terminological differences between the ITU-T and ATM Forum documents are compared and clarified and the approachesto the use of quality of service (QoS) parameters and classes are discussed. The so-called Service Categories (using the ATMForum nomenclature) are Constant Bit Rate (CBR), Variable Bit Rate (VBR), Available Bit Rate (ABR) and Unspecified BitRate (UBR).The ABR is the newest and the one raising major discussions, as specific congestion control mechanisms, which dorepresent a real innovation in the ATM framework, have been defmed for its support. The paper therefore introduces the ABRservice in some details and discusses some still open issues and utilisation perspective.

Variable bit rate (VBR) video is currently by far the most interesting and challenging real-time application for B- ISDN/ATM environments. To define bandwidth allocation schemes which provide an adequate quality of service (QoS) for VBR applications and minimize the waste of bandwidth, the effects of the video applications on the network must be investigated. While the modeling of VBR video sources has recently received significant attention, there presently exists no widely accepted model which lends itself to mathematical analysis. Currently MPEG is the reference standard for moving video compression. In the first part of the paper a VBR implementation of MPEG-2 coding algorithm is described and a comparison with the constant bit rate (CBR) algorithm is performed. To investigate the impact that the coding parameters have on the characteristics of the traffic delivered to the network a long sample of the movie 'The Sheltering Sky' has been coded with two different quantizer steps. The generated trace was used to obtain a detailed statistical analysis of the traffic generated by an MPEG-2 encoder. Starting from this statistical analysis an analytically tractable model is developed, analyzed and used to study bandwidth allocation problems.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper describes the design and implementation of a file server for variable bit rate continuous media. Most continuous media file servers have been designed for constant bit rate streams. We address the problem of building a server where each stream may have a different bit rate and, more importantly, where the bit rate within a single stream may vary considerably. Such servers will become increasingly more important because of Variable Bit Rate compression standards such as is possible with MPEG-2.

In this paper we compare techniques for storage and real-time retrieval of Variable Bit Rate (VBR) video data for multiple simultaneous users. The motivation for considering VBR is that video results in inherently time varying data, and as such, with the same average bit rate, higher quality can be achieved with VBR than with Constant Bit Rate (CBR). We propose and compare the following three classes of VBR data placement and retrieval techniques: Constant Time Length (CTL) places and retrieves data in blocks corresponding to equal playback durations, Constant Data Length (CDL) places and retrieves constant-sized data blocks, and a hybrid solution uses CDL placement but retrieves a variable number of blocks in each service round. We have found that CTL data placement has much lower buffer requirements than CDL but suffers from fragmentation during video editing. We show hybrid placement to have both advantages of high efficiency and low fragmentation. We also address the issue of admission control policies by comparing statistical and deterministic techniques. `Statistical' admission control uses statistics of the stored data to ensure that the probability of `overload' does not exceed a prespecified threshold. `Deterministic' control uses the actual stored video bit traces to regulate the number of admitted users. We consider two types of deterministic admission control: data-limit and ideal deterministic. Data-limit admission control admits users based on precomputing the total amount of data requested by all users in future service rounds. In contrast, ideal deterministic admission control not only precomputes the total amount of data requested, but also assumes we have control of data placement at the disk sector level in order to precompute the future seek and rotation times. We provide a cost/benefit analysis of the above placement/retrieval/admission control techniques and conclude that CTL and hybrid placement/retrieval techniques can reduce the total system cost by up to a factor of 3 in comparison with the strategy of padding the VBR video trace to achieve a constant data rate. For read-only systems, CTL has the lowest cost per user. For writable systems, the hybrid technique achieves a good compromise between low cost and low fragmentation. We find that all forms of deterministic admission control can outperform statistical, but the greatest gain comes from using ideal deterministic admission control. We note, however, that this admission control may be difficult to implement on standard disk controllers. Finally, we have implemented a full disk model simulator that operates 1000 times faster than the real-time disk. Results using the simulator are very close to those measured on the real disk, making the simulator useful for future experiments.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Variable bit rate transmission opens the way to constant quality video coding. However, this requires adifferent approach from the traditional constant bit rate coding techniques since a constant distortion does notyield a constant quality We introduce a method to determine the maximum distortion locally, and a techniqueto minimise the bit rate regarding this local maximum distortion. Also, it is shown that with bit stream shapingand peak bit rate control the bit rate will always be lower than in a CBR source with similar quality in the most difficult scenes. Keywords: VBR, Constant quality coding, MPEG, bit stream shaping, peak rate control 1 Introduction: CBR versus VBR In the field of video compression for transmission and storage, the digital signal processing world meets thetelecommunication world in a supply and demand kind of way. The telecommunication world delivers band-width, the digital signal processing world uses the bandwidth as efficient as possible. Most video compressiontechniques are designed for communication channels with a constant bandwidth. The compression performancefor digital video, however, strongly depends on the contents of the video scenes, leading to a varying quality of

DQRAP (distributed queuing random access protocol), is a multiple access protocol that offers a throughput equal to the offered traffic up to a load of one with a lesser delay than any other multiple access protocol. DQRAP utilizes three control minislots and two distributed queues: transmission and collision resolution, to provide this performance for all values of 'a', i.e., the size of the network is immaterial. This paper describes: (a) how DQRAP can act as the switching fabric for a distributed ATM switch serving an HFC (hybrid fiber-coax) network; and (b) how CBR (constant bit rate) traffic is supported on our distributed ATM switch, i.e., a DQRAP/ATM switch. Simulation results representing the operation of a DQRAP/ATM switch operating in two modes: (a) the control minislots in dataslots allocated to CBR traffic are utilized in the reservation process; and (b) the control minislot capacity in those slots is allocated to CBR data.© (1995) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper proposes a new buffer-control technique for MPEG Video. The goal is to minimize the distortion while keeping the change in distortion between consecutive frames small. We formulate this goal as a constrained optimization problem and find the optimal solution using an iterative gradient search method. A preliminary experiment on a sub- GOP (4 frames, IBBP) shows that the solution is very close to the global optimal point. Further experiments on short video sequences show that, at the same bit rate and buffer constraint, the technique generates output sequences with smaller and more stable mean square error than other techniques, and strictly constant bit rate for every group of pictures. This technique should be particularly useful for off-line encoding or for one encoder/many decoders situations, where the higher computational cost required at the encoder can be afforded pays off by allowing higher quality sequences or smaller decoder buffers.© (1995) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Specific mechanisms are required to carry MPEG coded information over an ATM network. After a brief description of the MPEG-2 video coding algorithm, this paper discusses the "isochronism" related requirements for the transmission of the coded data and the problems associated with the use of an anisochronous transmission mechanism. The main problems, concerning the choice of the ATM Adaptation Layer (AAL) for each level of interaction, are discussed. Even if the MPEG stream is inherently bursty, the technical literature considers both the Constant Bit Rate (CBR)and Variable Bit Rate (VBR) solutions. The paper reports on a study concerning the AAL choice, in relation with the degreeof interactivity required by the application and with the delay constraints.A further issue, is an overview of different scenarios for the provision of video services, by "MPEG over ATM". Differentpossible network and service architectures are analyzed, and various solutions for storing video information are proposed.Three different service architectures are discussed: the fully centralized, the quasi-centralized and the fully distributed.

The paper presents and analyzes two alternatives of interworking modes for interconnection of IEEE 802.6 based Metropolitan Area Networks (MANs) and ATM based B-ISDN. The two alternatives are the `cell-to-slot interworking' and the `frame interworking.' The cell-to-slot mode modifies ATM-cells to DQDB-slots and vice versa, whereas the frame interworking mode reassembles ATM-cells and converts the resulting AAL frames to DQDB IMPDU frames. In the paper, simulation results of a DQDB network are presented, where one of the DQDB stations represents an IWU getting aggregated traffic from B-ISDN. The aggregated traffic is modelled by a mixture of various constant bitrate and variable bitrate traffic sources representing different real-time sources and background traffic. It turns out that the cell-to-slot interworking mode is much more useful for keeping bandwidth requirements of real-time traffic than the frame interworking mode which causes additional burstiness due to reassembly. Based on these results one can conclude that the cell-to-slot interworking mode is much more appropriate for real-time applications than the frame interworking mode.© (1993) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

CRMA-II (Cyclic Reservation Multiple Access, Version II) is a media access control protocol for local and metropolitan area networks working in the Gbit/s range. The access scheme is based on direct access to a slotted medium combined with a reservation scheme guaranteeing both high utilization of network capacity and global fairness. The performance of CRMA-II is improved in comparison with the forerunner CRMA protocol by installing insertion buffers at each station. Network throughput is pushed to values far beyond the transmission capacity by spatial reuse of slots while buffer-insertion technique reduces media access time but increases round-trip delay. In this paper a new media access scheme for constant bitrate (CBR) traffic is proposed. For acknowledged CBR connections CBR slots have immediate access to the transmission media. While there is one insertion buffer for each priority of data traffic, CBR traffic is allowed to bypass insertion buffers to ensure minimum delay for constant bitrate services. In this paper a performance evaluation of the CRMA-II media access scheme and the new access mechanism for CBR traffic are presented. The evaluation was done using discrete event driven simulations.© (1993) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

This paper describes a procedure for coding video in the HDTV Interlaced format (HDI) at rates of 34 Mbit/s while satisfying constraints of an embedded specified constant bit rate stream, compatible to an Interlaced Enhanced Definition TV signal (EDI). The term compatibility is used here to imply both (a) resolution compatibility and (b) that a decoder conforming to a specific syntax, should be able to decode an easily extractable stream from the information intended for a higher quality decoder, a feature that is sometimes referred as backward compatibility. Resolution compatibility, while solved easily when dealing with progressively scanned signals is more intricate in the case of interlaced signals and requires careful considerations in the design of the procedure useful for generating the lower resolution (or compatible) signal. Instead of performing a temporal filtering which presents the drawback of blurring the contours in motion, the proposed approach consists of addressing these issues in a spatial hierarchical coding scheme, in which special care is given to the choice of filters, in order to obtain a compatible signal with a good tradeoff between vertical and temporal definition, and with an appropriate suppression of aliasing. The hierarchical decomposition is carried out on a field basis and two constrained filter banks are actually designed in order to deal with the phase shift inherent to field processing procedures.© (1992) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

The transmission of digital video signals to the end user, which is known as secondary distribution by CCITT, is one of the key point of Broadband ISDN (B-ISDN). Sufficient picture quality can be obtained at a bitrate of 17 Mbit/s (or even lower) by using a hybrid DCT scheme working on blocks of 8 pels by 8 lines of the same frame (instead of lines of the same field) associated to block matching motion estimation and compensation. The optimization of the block matching algorithm (BMA) and the adaption of the hybrid DCT scheme to blocks designed on a frame basis are presented hereafter.