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Multimedia, IEEE Transactions on

Issue 1 • Date Mar 2001

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Displaying Results 1 - 14 of 14
  • Multidescription video streaming with optimized reconstruction-based DCT and neural-network compensations

    Page(s): 123 - 131
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (172 KB) |  | HTML iconHTML  

    Packet and compression losses are two sources of quality losses when streaming compressed video over unreliable IP networks, such as the Internet. In this paper, we propose two new approaches for concealing such losses. First, we present a joint sender-receiver approach for designing transforms in multidescription coding (MDC). In the receiver, we use a simple interpolation-based reconstruction algorithm, as sophisticated concealment techniques cannot be employed in real time. In the sender we design an optimized reconstruction-based discrete cosine transform (ORB-DCT) with an objective of minimizing the mean squared error, assuming that some of the descriptions are lost and that the missing information is reconstructed by simple averaging at the destination. Second, we propose artificial neural network to compensate for compression losses introduced in MDC. Experimental results show that our proposed algorithms perform well in real internet tests View full abstract»

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  • Scalable services via egress admission control

    Page(s): 69 - 81
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (192 KB)  

    Allocating resources for multimedia traffic flows with real-time performance requirements is an important challenge for future packet networks. However, in large-scale networks, individually managing each traffic flow on each of its traversed routers has fundamental scalability limitations, in both the control plane's requirements for signaling, state management, and admission control, and the data plane's requirements for per-flow scheduling mechanisms. In this paper, we develop a scalable architecture and algorithm for quality-of-service management termed egress admission control. In our approach, resource management and admission control are performed only at egress routers, without any coordination among backbone nodes or per-flow management. Our key technique is to develop a framework for admission control under a general “black box” model, which allows for cross traffic that cannot be directly measured, and scheduling policies that may be ill-described across many network nodes. By monitoring and controlling egress routers' class-based arrival and service envelopes, we show how network services can be provisioned via scalable control at the network edge. We illustrate the performance of our approach with a set of simulation experiments using highly bursty traffic flows and find that despite our use of distributed admission control, our approach is able to accurately control the system's admissible region under a wide range of conditions View full abstract»

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  • A probabilistic framework for semantic video indexing, filtering, and retrieval

    Page(s): 141 - 151
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (256 KB)  

    Semantic filtering and retrieval of multimedia content is crucial for efficient use of the multimedia data repositories. Video query by semantic keywords is one of the most difficult problems in multimedia data retrieval. The difficulty lies in the mapping between low-level video representation and high-level semantics. We therefore formulate the multimedia content access problem as a multimedia pattern recognition problem. We propose a probabilistic framework for semantic video indexing, which call support filtering and retrieval and facilitate efficient content-based access. To map low-level features to high-level semantics we propose probabilistic multimedia objects (multijects). Examples of multijects in movies include explosion, mountain, beach, outdoor, music etc. Semantic concepts in videos interact and to model this interaction explicitly, we propose a network of multijects (multinet). Using probabilistic models for six site multijects, rocks, sky, snow, water-body forestry/greenery and outdoor and using a Bayesian belief network as the multinet we demonstrate the application of this framework to semantic indexing. We demonstrate how detection performance can be significantly improved using the multinet to take interconceptual relationships into account. We also show how the multinet can fuse heterogeneous features to support detection based on inference and reasoning View full abstract»

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  • A voicing-driven packet loss recovery algorithm for analysis-by-synthesis predictive speech coders over Internet

    Page(s): 98 - 107
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (172 KB) |  | HTML iconHTML  

    In this paper, a novel voice-driven adaptive packet loss recovery algorithm is proposed to lessen the possible voice degradation and error propagation for analysis-by-synthesis speech coders in Internet applications. After voicing classification, we adaptively adopt random noise generation, multiresolution excitation generation, or pulse tracking procedure to recover the lost packets, By applying the algorithm to the G.723.1 coder, simulation results show that the proposed algorithm is superior to the recovery algorithm embedded in the G.723.1 standard through the subjective evaluation View full abstract»

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  • RTP/I-toward a common application level protocol for distributed interactive media

    Page(s): 152 - 161
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (96 KB) |  | HTML iconHTML  

    Distributed interactive media are media that involve communication over a computer network as well as user interactions with the medium itself. Examples of this kind of media are shared whiteboard presentations and networked computer games. One key problem of this media class is that a large amount of common functionality is currently redesigned and redeveloped for each single medium. In order to solve this problem we present a media model and an application level protocol called RTP/I. Derived from the experience gained with audio and video transmission using RTP, RTP/I is defined as a new protocol framework which reuses many aspects of RTP while it is thoroughly adapted to meet the demands of distributed interactive media. By identifying and supporting the common aspects of distributed interactive media RTP/I allows the reuse of key functionality in form of generic services. Furthermore RTP/I makes it possible for applications of different vendors to interact with each other in a standardized way View full abstract»

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  • Real-time traffic transmission over the Internet

    Page(s): 33 - 40
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (132 KB)  

    Multimedia applications require the transmission of real-time streams over a network. These streams often exhibit variable bandwidth requirements, and require high bandwidths and guarantees from the network. This creates problems when such streams are delivered over the Internet. To solve these problems, recently, a small set of differentiated services has been introduced. Among these, Premium Service is suitable for transmitting real-time stored stream (full knowledge of the stream characteristics). It uses a bandwidth allocation mechanism (BAM) based on the stream peak rate. Due to the variable bandwidth requirement, the peak rate BAM can waste large amount of bandwidth. In this paper we propose a new BAM that uses less bandwidth than the peak rate BAM, while providing the same service. Our BAM does not affect the real-time stream quality of service (QoS) and does not require any modification to the Premium Service Architecture. We also introduce several frame dropping mechanisms that further reduce bandwidth consumption subject to a QoS constraint when coupled with the above BAM. The proposed BAM and the dropping mechanisms are evaluated using Motion JPEG and MPEG videos and are shown to be effective in reducing bandwidth requirements. Further, since VCR operations are very useful in video streaming, we propose a mechanism that introduces these operations in our BAM. Through simulations we show the effectiveness of this mechanism View full abstract»

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  • Error control for receiver-driven layered multicast of audio and video

    Page(s): 108 - 122
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (300 KB)  

    We consider the problem of error control for receiver-driven layered multicast of audio and video over the Internet. The sender injects into the network multiple source layers and multiple channel coding (parity) layers, some of which are delayed relative to the source, Each receiver subscribes to the number of source layers and the number of parity layers that optimizes the receiver's quality for its available bandwidth and packet loss probability. We augment this layered FEC system with layered pseudo-ARQ. Although feedback is normally problematic in broadcast situations, ARQ can be simulated by having the receivers subscribe and unsubscribe to the delayed parity layers to receive missing information. This pseudo-ARQ scheme avoids an implosion of repeat requests at the sender and is scalable to an unlimited number of receivers, We show gains of 4-18 dB on channels with 20% loss over systems without error control and additional gains of 1-13 dB when FEC is augmented by pseudo-ARQ in a hybrid system, Optimal error control in the hybrid system is achieved by an optimal policy for a Markov decision process View full abstract»

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  • Traffic specifications for the transmission of stored MPEG video on the Internet

    Page(s): 5 - 17
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (336 KB)  

    Guarantees of quality-of-service (QoS) in the real-time transmission of stored video on the Internet is a challenging task for the success of many video on demand (VoD) applications. Two QoS classes have been specified by the IETF Integrated Services (intserv) Working Group: Guaranteed Services and Controlled-Load Services. For both of them, it is necessary to provide traffic sources with the capability of calculating the traffic characteristics to be declared to the network, Tspec, on the basis of a limited set of parameters statistically characterizing the traffic and the required level of QoS. The target of this paper is to develop an algorithm for the evaluation of the Tspec parameters which characterize the video stream when a given QoS is required. To this end an analytical framework modeling an MPEG stored-video server and the access network node is introduced. The video emission process is modeled with a switched batch Bernoulli process (SBBP), and performance in the video-server smoother is analytically evaluated. Then the token bucket at the network access point, loaded by the output traffic of the video-server smoother, is modeled to calculate the probability of marking nonconforming data packets View full abstract»

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  • On packetization of embedded multimedia bitstreams

    Page(s): 132 - 140
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (180 KB) |  | HTML iconHTML  

    We study the problem of packetizing embedded multimedia bitstreams to improve the error resilience of source (compression) codes. This problem is important because of the increasing popularity of embedded compression methodology and its suitability for scalable streaming media over IP or/and mobile IP. We study various packetization schemes against packet erasure at both low and high bit rates. Maximizing packetization efficiency for embedded bitstreams is formulated as a discrete optimization problem and globally optimal packetization (OP) algorithms are proposed under different settings. Suboptimal packetization algorithms are also devised to reduce the complexity of the OP algorithms. In order to assess their effectiveness, the proposed packetization algorithms are used to packetize embedded image and video bitstreams with simulated packet loss. Experimental results show that our OP algorithms slightly outperforms suboptimal ones. In addition to confirming the superiority of the OP algorithms, these results also provide justification of heuristic packetization methods published in the literature View full abstract»

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  • Broadcast quality video over IP

    Page(s): 162 - 173
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB)  

    We consider the problem of designing systems for the transmission of video signals of the quality found in current television broadcasts, over high-speed segments of the public IP network. Our most important contribution is the definition of a network/coder interface for IP networks which gathers channel state information, and then sets parameters of the video coder to maximize the quality of the signal delivered to the receiver, while remaining fair to other data or video connections. This interface plays a role analogous to that of a Leaky Bucket controller, in that it specifies traffic shaping parameters which result in simultaneous good Quality-of-Service (QoS) for the source and good network performance. Since the network is not assumed to provide any form of QoS guarantee, fundamental to our construction is a hidden Markov model for the channel, based on which the interface solves a problem of optimal stochastic control, to decide how to configure the encoder. Other contributions are: a) modifications to the standard Internet transport protocol, to make it suitable for the transport of delay-constrained traffic and to gather channel state information, and b) the design of an error-resilient video coder. Experimental studies reveal that the proposed system is able to stream video signals of the quality of current TV-broadcasts, among hosts in wide-area networks connected to the experimental vBNS backbone View full abstract»

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  • An integrated source transcoding and congestion control paradigm for video streaming in the Internet

    Page(s): 18 - 32
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    In this work we present an end-to-end optimized video streaming system comprising of synergistic interaction between a source packetization strategy and an efficient and responsive, TCP-friendly congestion control protocol [Linear Increase Multiplicative Decrease with History (LIMD/H)]. The proposed source packetization scheme transforms a scalable/layered video bitstream so as to provide graceful resilience to network packet drops. The congestion control mechanism provides low variation in transmission rate in steady state and at the same time is reactive and provably TCP-friendly. While the two constituent algorithms identified above are novel in their own right, a key aspect of this work is the integration of these algorithms in a simple yet effective framework. This “application-transport” layer interaction approach is used to maximize the expected delivered video quality at the receiver. The integrated framework allows our system to gracefully tolerate and quickly react to sudden changes in the available connection capacity due to the onset of congestion, as verified in our simulations View full abstract»

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  • The MPEG-4 fine-grained scalable video coding method for multimedia streaming over IP

    Page(s): 53 - 68
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    Real-time streaming of audiovisual content over the Internet is emerging as an important technology area in multimedia communications. Due to the wide variation of available bandwidth over Internet sessions, there is a need for scalable video coding methods and (corresponding) flexible streaming approaches that are capable of adapting to changing network conditions in real time. In this paper, we describe a new scalable video-coding framework that has been adopted recently by the MPEG-4 video standard. This new MPEG-4 video approach, which is known as Fine-Granular-Scalability (FGS), consists of a rich set of video coding tools that support quality (i.e., SNR), temporal, and hybrid temporal-SNR scalabilities. Moreover, one of the desired features of the MPEG-J FGS method is its simplicity and flexibility in supporting unicast and multicast streaming applications over IF View full abstract»

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  • MQ: an integrated mechanism for multimedia multicasting

    Page(s): 82 - 97
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    This paper studies the integration of multimedia multicasting, with the consideration of multicast with end-to-end QoS guarantees by resource reservation, dynamic join and departure of participants, user heterogeneity, scalability, robustness, and loop-free control. A protocol called MQ, Multicast with QoS, is proposed to support multimedia group communications with QoS guarantees for heterogeneous recipients. With MQ, while resource reservation is de-coupled from QoS multicast routing, they are integrated in a way to avoid the problem of “sender-oriented” path determination, a problem that occurs when RSVP is used in conjunction with QoS routing for heterogeneous reservations. Being a truly receiver-oriented and integrated scheme for multimedia multicasting, MQ supports such integration in a robust, scalable and loop-free way. It also accommodates heterogeneous users with varied QoS, dynamically adjusts QoS trees to improve resource utilization, and guarantees end-to-end QoS requirements. We have conducted simulations to evaluate the performance of the proposed mechanism. MQ demonstrates its advantages over the conventional loosely coupled integration of TP multicasting, resource reservation and QoS routing, in terms of better accommodation of heterogeneous users, higher scalability, lower blocking probability for users to join groups with service guarantees, and more efficient resource utilization to enhance system performance View full abstract»

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  • Classification based mode decisions for video over network

    Page(s): 41 - 52
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    A video encoder has to make many mode decisions in order to achieve the goal of a low bit rate, high quality, and fast implementation. We propose a general classification based approach to making such mode decisions accurately and efficiently. We first illustrate the approach using the Intra-Inter coding mode decision. We then focus on the decision to skip or code a frame for rate control of video over networks. Using the classification used approach we show improvement in the rate-distortion sense. We then extend the work to scalable video coding in choosing between scalability modes and examine the performance of our approach over error prone networks, using simulated packet losses View full abstract»

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Aims & Scope

The scope of the Periodical is the various aspects of research in multimedia technology and applications of multimedia.

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Chang Wen Chen
State University of New York at Buffalo