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1
Lee, Il-Sung. "Traffic shaping for variable-bit-rate MPEG-2 video." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ37266.pdf.
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Sowden, Bradley Claude. "The performance of DS-CDMA cellular systems with variable-bit-rate traffic." Thesis, University of Auckland, 2009. http://hdl.handle.net/2292/5211.
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The deployment of third generation (3G) cellular systems is resulting in a transition from cellular systems that predominantly carry constant-bit-rate (CBR) voice traffic to multi-service packet based systems that predominantly carry variable-bit-rate (VBR) traffic. With 3G DS-CDMA cellular systems there is a direct relationship between user traffic and propagation dependent performance as additional traffic causes increased system interference. This thesis investigates the impact of VBR traffic on the propagation dependent performance of DS-CDMA cellular systems that utilise frame-by-frame dynamic resource allocation on the radio channel. A DS-CDMA cellular system model is developed and the downlink performance of both outdoor macro-cellular and indoor pico-cellular systems is evaluated with a variety of traffic types. Both traffic scheduling performance and propagation dependent performance are evaluated as the two are inter-linked. Scenarios are identified where propagation dependent performance is sensitive to the statistical properties of the user traffic streams and it is shown that a significant performance difference potentially exists between different traffic types when the number of users per cell is low. When a significant performance difference does exist, burstier more variable traffic generally results in superior propagation dependent performance. The base transceiver station (BTS) transmitter power mean and variance provides a good indication of the level of propagation dependent performance regardless of the specific traffic type. Traffic scheduling policies that deliberately reduce the variability of user traffic streams are considered and in terms of propagation dependent performance these are shown to have a minimal impact on the performance difference between different traffic types. The implications of VBR traffic on DS-CDMA cellular system design are outlined and it is shown that VBR traffic can be approximated as CBR traffic in many scenarios and this is a convenient approximation as it simplifies system design and detailed traffic models do not need to be developed.
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Voo, Charles. "Management of low and variable bit rate ATM Adaptation Layer Type 2 traffic." University of Western Australia. School of Electrical, Electronic and Computer Engineering, 2003. http://theses.library.uwa.edu.au/adt-WU2004.0030.
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Asynchronous Transfer Mode (ATM) Adaptation Layer Type 2 (AAL2) has been developed to carry low and variable bit rate traffic. It provides high bandwidth efficiency with low packing delay by allowing voice traffic from different AAL2 channels to be multiplexed onto a single ATM virtual channel connection. Examples of where AAL2 are used include the Code Division Multiple Access and the Third Generation mobile telephony networks. The main objective of this thesis is to study traditional and novel AAL2 multiplexing methods and to characterise their performance when carrying low and variable bit rate (VBR) voice traffic. This work develops a comprehensive QoS framework which is used as a basis to study the performance of the AAL2 multiplexer system. In this QoS framework the effects of packet delay, delay variation, subjective voice quality and bandwidth utilisation are all used to determine the overall performance of the end-to-end system for the support of real time voice communications. Extensions to existing AAL2 voice multiplexers are proposed and characterised. In the case where different types of voice applications are presented to the AAL2 multiplexer, it was observed that increased efficiency gains are possible when a priority queuing scheme is introduced into the traditional AAL2 multiplexer system. Studies of the voice traffic characteristics and their effects on the performance of the AAL2 multiplexer are also investigated. It is shown that particular source behaviours can have deleterious effect on the performance of the AAL2 multiplexer. Methods of isolating these voice sources are examined and the performance of the AAL2 multiplexer re-evaluated to show the beneficial effects of a particular source isolation technique. The extent to which statistical multiplexing is possible for real time variable VBR sources is theoretically examined. These calculations highlight the difficulties in multiplexing VBR real time traffic while maintaining guaranteed delay bounds for these sources. Based on these calculations, multiplexing schemes that incorporate data transfers within the real time traffic transfer are proposed as alternatives for utilising unused bandwidth caused by the VBR nature of the voice traffic.
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Zhang, Shu Carleton University Dissertation Engineering Systems and computer. "Congestion control in frame relay networks with variable bit rate compressed voice and data traffic." Ottawa, 1993.
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Chan, Henry C. B. "Reservation arbitrated access for variable bit rate isochronous traffic transport over dual bus metropolitan area networks." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq25031.pdf.
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Griffin, Wesley. "Quality Guided Variable Bit Rate Texture Compression." Thesis, University of Maryland, Baltimore County, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10159930.
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<p> The primary goal of computer graphics is to create images by rendering a scene under two constraints: quality, producing the image with as few artifacts as possible, and time, producing the image as fast as possible. Technology advances have both helped to satisfy these constraints, with Graphics Processing Unit (GPU) advances reducing image rendering times, and to exacerbate these constraints, with new HD and virtual reality displays increasing rendering resolutions. To meet both constraints, rendering uses texture mapping which maps 2D textures onto scene objects. Over time, the count and resolution of textures has increased, resulting in dramatic growth of data storage requirements. Compression can help to reduce these storage requirements. </p><p> I present a rigorous texture compression evaluation methodology using final rendered images. My method can account for masking effects introduced by the texture mapping process while leveraging the perceptual-rigor of current Image Quality Assessment metrics. Building on this evaluation methodology, I present a demonstration of guided texture compression optimization that minimizes the bitrate of compressed textures while maximizing the quality of final rendered images. Guided texture compression will help with the scalability problem for optimizing texture compression in real-world scenarios.</p>
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Biswas, Md Israfil. "Internet congestion control for variable-rate TCP traffic." Thesis, University of Aberdeen, 2011. http://digitool.abdn.ac.uk:80/webclient/DeliveryManager?pid=182264.
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The Transmission Control Protocol (TCP) has been designed for reliable data transport over the Internet. The performance of TCP is strongly influenced by its congestion control algorithms that limit the amount of traffic a sender can transmit based on end-to-end available capacity estimations. These algorithms proved successful in environments where applications rate requirements can be easily anticipated, as is the case for traditional bulk data transfer or interactive applications. However, an important new class of Internet applications has emerged that exhibit significant variations of transmission rate over time. Variable-rate traffic poses a new challenge for congestion control, especially for applications that need to share the limited capacity of a bottleneck over a long delay Internet path (e.g., paths that include satellite links). This thesis first analyses TCP performance of bursty applications that do not send data continuously, but generate data in bursts separated by periods in which little or no data is sent. Simulation analysis shows that standard TCP methods do not provide efficient support for bursty applications that produce variable-rate traffic, especially over long delay paths. Although alternative forms of congestion control like TCP-Friendly Rate Control and the Datagram Congestion Control Protocol have been proposed, they did not achieve widespread deployment. Therefore many current applications that rely upon User Datagram Protocol are not congestion controlled. The use of non-standard or proprietary methods decreases the effectiveness of Internet congestion control and poses a threat to the Internet stability. Solutions are therefore needed to allow bursty applications to use TCP. Chapter three evaluates Congestion Window Validation (CWV), an IETF experimental specification that was proposed to improve support for bursty applications over TCP. It concluded that CWV is too conservative to support many bursty applications and does not provide an incentive to encourage use by application designers. Instead, application designers often avoid generating variable-rate traffic by padding idle periods, which has been shown to waste network resources. CWV is therefore shown to not provide an acceptable solution for variable-rate traffic. In response to this shortfall, a new modification to TCP, TCP-JAGO, is proposed. This allows variable-rate traffic to restart quickly after an inactive (i.e., idle) period and to effectively utilise available network resources while sending at a lower rate than the available rate (i.e., during an application-limited period). The analysis in Chapter five shows that JAGO provides faster convergence to a steady-state rate and improves throughput by more efficiently utilising the network. TCP-JAGO is also shown to provide an appropriate response when congestion is experienced after restart. Variable-rate TCP traffic can also be impacted by the Initial Window algorithm at the start or during the restart of a session. Chapter six considers this problem, where TCP has no prior indication of the network state. A recent proposal for a larger initial window is analysed. Issues and advantages of using a large IW over a range of scenarios are discussed. The thesis concludes by presenting recommendations to improve TCP support for bursty applications. This also provides an incentive for application designers to choose TCP for variable-rate traffic.
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Chin, Hin-Soon. "Transmission of variable bit rate video over an Orwell ring." Thesis, Loughborough University, 1989. https://dspace.lboro.ac.uk/2134/32384.
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Asynchronous Transfer Mode (ATM) is fast emerging as the preferred information transfer technique for future Broadband Integrated Services Digital Networks (BISON), offering the advantages of both the simplicity of time division circuit switched techniques and the flexibility of packet switched techniques. ATM networks with their inherent rate flexibility offer new opportunities for the efficient transmission of real time Variable Bit Rate (VBR) services over such networks. Since most services are VBR in nature when efficiently coded, this could in turn lead to a more efficient utilisation of network resources through statistical multiplexing. Video communication is typical of such a service and could benefit significantly if supported with VBR video over ATM networks.
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Vogeleer, Karel De. "On the Quality of Delivery for Variable Bit Rate Video." Licentiate thesis, Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00519.
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Several factors affect the perceived quality of a video streamed over a network. Technical aspects related to the codec and the video play an important role besides the delivery of the video content in a timely and error-free manner. If the streamed video frames do not arrive in time, temporal artifacts may become observable during the video playback. This has potentially an adverse effect on the user experience. The study of temporal artifacts during the video playback is the area of concern for Quality of Delivery. The licentiate work investigates the Quality of Delivery for Variable Bit Rate video sent over wireless technologies. The ability of the network to deliver data to the video’s jitter buffer before its playout deadline is studied. Jitter buffer exhaustions are of particular interest as they tell something about the Quality of Delivery. A number of experiments are conducted where a set of videos with different bit-rates are streamed to a mobile device over a wireless LAN and a W-CDMA network. The video streams are recorded and analyzed based on specific Quality of Service parameters and are related to the states of the jitter buffer. The statistical tools Support Vector Machines and the Mahalanobis distance are applied to the parameters to obtain a model that can classify the jitter buffer states. The performance evaluation indicates that the Mahalanobis distance can classify the jitter buffer state marginally better than the Support Vector Machines. However, the Support Vector Machines can produce more reliable predictions compared to the Mahalanobis distance. It is also observed that the perceived Quality of Delivery is not only affected by the behavior of the wireless networks but also by the behavior of the streaming server. Finally, the ability of the Quality of Service parameters to describe the Quality of Delivery is studied. The results indicate that metrics based on the packet arrival-rate and packet inter-arrival time are most suitable in this particular case.
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Kim, Hyun Soo Electrical Engineering & Telecommunications Faculty of Engineering UNSW. "Speech analysis techniques useful for low or variable bit rate coding." Awarded by:University of New South Wales. School of Electrical Engineering and Telecommunications, 2005. http://handle.unsw.edu.au/1959.4/22050.
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We investigate, improve and develop speech analysis techniques which can be used to enhance various speech processing systems, especially low bit rate or variable bit rate coding of speech. The coding technique based on the sinusoidal representation of speech is investigated and implemented. Based on this study of the sinusoidal model of speech, improved analysis techniques to determine voicing, pitch and spectral estimation are developed, as well as noise reduction technique. We investigate the properties and limitations of the spectral envelope estimation vocoder (SEEVOC). We generalize, optimize and improve the SEEVOC and also compare it with LP in the presence of noise. The properties and applications of morphological filters for speech analysis are investigated. We introduce and investigate a novel nonlinear spectral envelope estimation method based on morphological operations, which is found to be very robust against noise. This method is also compared with the SEEVOC method. A simple method for the optimum selection of the structuring set size without using prior pitch information is proposed for many purposes. The morphological approach is then used for a new pitch estimation method and for the general sinusoidal analysis of speech or audio. Many of the new methods are based on a novel systematic analysis of the peak features of signals, including the study of higher order peaks. We propose a novel peak feature algorithm, which measure the peak characteristics of speech signal in time domain, to be used for end point detection and segmentation of speech. This nonparametric algorithm is flexible, efficient and very robust in noise. Several simple voicing measures are proposed and used in a new speech classifier. The harmonic-plus-noise decomposition technique is improved and extended to give an alternative to the methods used in the sinusoidal analysis method. Its applications to pitch estimation, speech classification and noise reduction are investigated.
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Yearwood, Mario A. (Mario Anton). "Variable bit rate voice over ATM using compression and silence removal." Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/42724.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1997.<br>Includes bibliographical references (leaves 45-46).<br>by Mario A. Yearwood.<br>M.Eng.
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Dhillon, Abinder. "Multimedia integration of variable bit rate sources over the FDDI network." Thesis, University of Ottawa (Canada), 1994. http://hdl.handle.net/10393/6666.
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Multimedia is the future of today's communication field. Multimedia communications involve the transmission of different sources, such as video, voice, data and graphics, on the same network. This thesis focuses on the performance study of the Fibre Distributed Data Interface (FDDI) network when multimedia sources are integrated. In the first part of the study, Variable Bit Rate (VBR) video and data traffic are studied on FDDI. The second part of the study is the integration of VBR video and voice sources. In order to improve the multiplexing gain, the bit dropping algorithm is used on voice sources. It is shown that due to the dynamic bandwidth transfer property of FDDI, multimedia sources can be integrated without affecting the quality of service required by various media. Performance measures such as delay, variance and efficiency are calculated for each component of multimedia. Also using the bit dropping algorithm on voice sources, the delay and the probability of loss of voice packets decreases while the quality of voice is much higher than required.
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Lee, Jeffrey C. "ARCHITECTURAL CONSIDERATIONS FOR A VARIABLE BIT RATE DATA ACQUISITION TELEMETRY ENCODER." International Foundation for Telemetering, 2007. http://hdl.handle.net/10150/604502.
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ITC/USA 2007 Conference Proceedings / The Forty-Third Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2007 / Riviera Hotel & Convention Center, Las Vegas, Nevada<br>Modern telemetry systems require flexible bit rate telemetry encoders in order to optimize mission formats for varying data rate requirements and/or signal to noise conditions given a fixed transmitter power. Implementing a variable bit rate telemetry encoder requires consideration of several possible architectural topologies that place different system requirements on data acquisition modules within the encoder in order to maintain adequate signal fidelity of sensor information. This paper focuses on the requirements, design considerations and tradeoffs associated with differing architectural topologies for implementing a variable bit rate encoder and the resulting implications on the encoder systems data acquisition units.
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Watson, Scott Douglas. "Low and variable bit-rate speech coding for asynchronous transfer mode networks." Thesis, University of Liverpool, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.367239.
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Makaroff, Dwight J. "Design, implementation and evaluation of a variable bit-rate continuous media file server." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp02/NQ34586.pdf.
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Kalyanaraman, Shivkumar. "Traffic management for the available bit rate (ABR) service in asynchronous transfer mode (ATM) networks /." The Ohio State University, 1997. http://rave.ohiolink.edu/etdc/view?acc_num=osu1487946776020645.
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Taylor, Elvin Lattis Jr. "Modeling and Simulation of a Video-on-Demand Network Implementing Adaptive Source-Level Control and Relative Rate Marking Flow Control for the Available Bit Rate Service." Thesis, Virginia Tech, 1997. http://hdl.handle.net/10919/31097.
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The Available Bit Rate (ABR) service class for the Asynchronous Transfer Mode (ATM) protocol was originally designed to manage data traffic. ABR flow control makes no guarantees concerning cell transfer delay or cell delay variation. A closed-loop feedback mechanism is used for traffic management. To use this class of service for video transport, the video source will accept feedback from the network and adapt its source rate based on this status information. The objective of this research is to assess the ability of the ATM ABR service class to deliver Moving Picture Experts Group version 1 (MPEG-1) video. Three approaches to source-level control are compared: (i) arbitrary loss or no control method, (ii) selective discard of MPEG B-pictures, and (iii) selective discard of MPEG B- and P-pictures. Performance is evaluated based on end-to-end delay, congested queue occupancy levels, network utilization, and jitter. A description of the investigation, assumptions, limitations, and results of the simulation study are included.<br>Master of Science
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Fahmy, Sonia. "Traffic management for point-to-point and multipoint available bit rate (ABR) service in Asynchronous Transfer Mode (ATM) Networks /." The Ohio State University, 1999. http://rave.ohiolink.edu/etdc/view?acc_num=osu1488191124569846.
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Vandalore, Bobby. "Traffic management to enhance Quality of Service (QoS) of multimedia over Available Bit Rate (ABR) in Asynchronous Transfer Mode (ATM) networks /." The Ohio State University, 2000. http://rave.ohiolink.edu/etdc/view?acc_num=osu1488196234911491.
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Mahlanyane, Nkululeko S. "Using a low-bit rate speech enhancement variable post-filter as a speech recognition system pre-filter to improve robustness to GSM speech." Master's thesis, University of Cape Town, 2003. http://hdl.handle.net/11427/5176.
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Includes bibliographical references.<br>Performance of speech recognition systems degrades when they are used to recognize speech that has been transmitted through GS1 (Global System for Mobile Communications) voice communication channels (GSM speech). This degradation is mainly due to GSM speech coding and GSM channel noise on speech signals transmitted through the network. This poor recognition of GSM channel speech limits the use of speech recognition applications over GSM networks. If speech recognition technology is to be used unlimitedly over GSM networks recognition accuracy of GSM channel speech has to be improved. Different channel normalization techniques have been developed in an attempt to improve recognition accuracy of voice channel modified speech in general (not specifically for GSM channel speech). These techniques can be classified into three broad categories, namely, model modification, signal pre-processing and feature processing techniques. In this work, as a contribution toward improving the robustness of speech recognition systems to GSM speech, the use of a low-bit speech enhancement post-filter as a speech recognition system pre-filter is proposed. This filter is to be used in recognition systems in combination with channel normalization techniques.
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Gatton, Tim. "BEST SOURCE SELECTORS AND MEASURING THE IMPROVEMENTS." International Foundation for Telemetering, 2005. http://hdl.handle.net/10150/604871.
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ITC/USA 2005 Conference Proceedings / The Forty-First Annual International Telemetering Conference and Technical Exhibition / October 24-27, 2005 / Riviera Hotel & Convention Center, Las Vegas, Nevada<br>After years of tracing the evolution and solutions to finding the best data, I learned that
it isn’t best source selection that we all want. What we need is best data selection.
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Halbach, Till. "Error-robust coding and transformation of compressed hybered hybrid video streams for packet-switched wireless networks." Doctoral thesis, Norwegian University of Science and Technology, Faculty of Information Technology, Mathematics and Electrical Engineering, 2004. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-136.
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<p>This dissertation considers packet-switched wireless networks for transmission of variable-rate layered hybrid video streams. Target applications are video streaming and broadcasting services. The work can be divided into two main parts.</p><p>In the first part, a novel quality-scalable scheme based on coefficient refinement and encoder quality constraints is developed as a possible extension to the video coding standard H.264. After a technical introduction to the coding tools of H.264 with the main focus on error resilience features, various quality scalability schemes in previous research are reviewed. Based on this discussion, an encoder decoder framework is designed for an arbitrary number of quality layers, hereby also enabling region-of-interest coding. After that, the performance of the new system is exhaustively tested, showing that the bit rate increase typically encountered with scalable hybrid coding schemes is, for certain coding parameters, only small to moderate. The double- and triple-layer constellations of the framework are shown to perform superior to other systems.</p><p>The second part considers layered code streams as generated by the scheme of the first part. Various error propagation issues in hybrid streams are discussed, which leads to the definition of a decoder quality constraint and a segmentation of the code stream to transmit. A packetization scheme based on successive source rate consumption is drafted, followed by the formulation of the channel code rate optimization problem for an optimum assignment of available codes to the channel packets. Proper MSE-based error metrics are derived, incorporating the properties of the source signal, a terminate-on-error decoding strategy, error concealment, inter-packet dependencies, and the channel conditions. The Viterbi algorithm is presented as a low-complexity solution to the optimization problem, showing a great adaptivity of the joint source channel coding scheme to the channel conditions. An almost constant image qualiity is achieved, also in mismatch situations, while the overall channel code rate decreases only as little as necessary as the channel quality deteriorates. It is further shown that the variance of code distributions is only small, and that the codes are assigned irregularly to all channel packets.</p><p>A double-layer constellation of the framework clearly outperforms other schemes with a substantial margin. </p><p>Keywords — Digital lossy video compression, visual communication, variable bit rate (VBR), SNR scalability, layered image processing, quality layer, hybrid code stream, predictive coding, progressive bit stream, joint source channel coding, fidelity constraint, channel error robustness, resilience, concealment, packet-switched, mobile and wireless ATM, noisy transmission, packet loss, binary symmetric channel, streaming, broadcasting, satellite and radio links, H.264, MPEG-4 AVC, Viterbi, trellis, unequal error protection</p>
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Shao, Wen-Hsuan, and 邵文炫. "Traffic Monitoring of Variable Bit-Rate Video in ATM Networks." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/08236268976905663123.
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碩士<br>國立交通大學<br>電信研究所<br>81<br>This thesis examines the problem of video transport over ATM
networks using MPEG-based video compression coder. The
constraints on the traffic of a MPEG-based codec in ATM
environment are investigated and the interactions between the
codec and the network are exploited. ATM networks, as proposed
by CCITT as the solution for the future B-ISDN, will provide
high flexibility of variable bit-rate (VBR) video service. And
the policing function has been introduced to monitor some
traffic characteristics of any connection in order to protect
the networks. By integrating network source-policing into the
VBR encoder, the available network bandwidth can be better
utilized. According to a commonly proposed ATM network-user
contract, the effect of encoder rate-control for the leaky
bucket channel is examined. With the considerations of ATM cell
loss, a 2-layered coding scheme is applied to keep picture
quality high.
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鍾昭德. "An admission control scheme for variable-bit-rate video traffic in atm environment." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/04904965713764302337.
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Yeh, Li-Bin, and 葉禮彬. "Variable Bit Rate (VBR) Video Traffic Control: A Wavelet Approach for Self-Similar Parameters Estimation." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/13141591230543421119.
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碩士<br>國立中興大學<br>電機工程學系<br>88<br>More recently, the variable-bit-rate (VBR) video over ATM networks is found to exhibit self-similar characteristics. Neglecting the self-similarity would lead to overly optimistic performance predictions and inadequate network resources allocation. In other words, the presence of self-similarity has serious implications on the performance of networks. A key parameter characterizing self-similar processes is the Hurst parameter H, which is designed to capture the degree of self-similarity. In order to determine if a given time series exhibits self-similarity, a method is needed to estimate H for a given time series. In this thesis we present an estimation tool by using wavelet transform. An important feature of the wavelet-based tool is the conceptual and practical simplicity, consisting essentially in measuring the slope. The other advantage of wavelet-based tool is that we can arbitrary choose vanishing moment to remove pre-selected trends. Moreover, the Hurst parameter can be accurately estimated from the power-law behavior of wavelet coefficients. A finite-duration impulse response (FIR) multilayer network is used to predict the number of incoming cells at the next period. Based on the information provided by the FIR multilayer network, we apply the wavelet-based tool to measure the Hurst parameters of predicted data. Then we use the Norros formula to estimate the bandwidth requirement for each predicted data. The results show that the lowest effective bandwidth requirement for MPEG1 video traffic over ATM networks can be obtained. Then we use the random early detection (RED) algorithm for traffic congestion control based on the predictive bandwidth. The simulation results indicate that the actual data has higher throughput by using RED.
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Huang, Yu Wen, and 黃郁文. "Gaussian Approximation Based Admission Control Algorithm for Variable Bit Rate Traffic in IEEE 802.11e WLANs." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/87320593503475720148.
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碩士<br>國立交通大學<br>電信工程系所<br>94<br>For Quality of Service (QoS) requirements of real-time traffic, IEEE 802.11 working group introduces a QoS-aware channel access mechanism, called Hybrid Coordination Function (HCF), which consists of contention-based Enhanced Distributed Channel Access (EDCA) and contention-free HCF Controlled Channel Access (HCCA). The TXOP allocation and the admission control units of the HCCA reference scheduler are only appropriate for constant bit rate (CBR) flows. It may result in serious packet loss for variable bit rate (VBR) flows.
In this thesis, we propose a simple admission control algorithm which adopts Gaussian distribution to approximate VBR traffic. Numerical results obtained from computer simulations show that our proposed algorithm can effectively and efficiently allocate Transmission Opportunity (TXOP) durations to QoS-enhanced stations (QSTAs) to guarantee a predefined packet loss probability. Moreover, our proposed scheme can easily handle multiple VBR flows of the same QSTA to get the advantage of multiplexing gain.
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Yen, Lee-Lung, and 顏立隆. "VARIABLE BIT RATE IMBE VOCODER." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/ajscxk.
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碩士<br>大同大學<br>通訊工程研究所<br>102<br>The improved multi-band excitation (IMBE) vocoder is developed from the multi-band excitation (MBE) speech model. It can provide good speech quality with low bit rate, primarily due to its special form of treating excitation signals. Specifically, for each speech frame, through spectral analysis, the IMBE vocoder divides the whole spectrum into sub-bands, each consisting three harmonics. Then each sub-band is discriminated as voiced or unvoiced. By doing this, the IMBE vocoder allows a speech frame having voiced and unvoiced sound at the same time, i.e., the nature of mixed sound. And this is why the IMBE vocoder can reconstruct synthetic speech with more nature quality.
In this thesis, for the IMBE vocoder, with decreasing upper limited number of sub-bands (the original setting is 12), we investigated the corresponding bit-rate reduction and the possible quality degradation of synthesized speech. Our experiments proved that, with the 4000 Hz system bandwidth, high-pitched female voice will be affected less due to the decreasing of sub-bands. For example, only 8 sub-bands are required for a voice with 170 Hz fundamental frequency. On the contrary, more than 12 sub-bands are required for voice with fundamental frequency lower than 121 Hz. Our experiments also showed that, within acceptable speech intelligibility, the bit-rate reduction can be achieved if a certain amount of speech quality degradation is tolerable.
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Chiang, Chun-Yi, and 江俊毅. "Variable Bit-Rate Color Image Quantization." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/16912230493175865794.
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碩士<br>靜宜大學<br>資訊管理學系研究所<br>98<br>As the images between neighboring pixels high similarity, so based on the characteristics, we proposed two methods of variable bit-rate about color image quantization in this paper. The first method focuses on the degree of similarity between the top and left pixel for encoding pixel to determine the state of the palette to use size. Using this approach we hope to find a better palette color in a short time.
The second method splits the given image into equilateral and fixed-size blocks and then calculates the degree of similarity between pixels in every split block. If the similar degree is high, this block only takes a common representative color in palette. On the contrary if the similar degree of block is widely different, the comparison that this block must do the similar intensity once again after splitting, until it is unable to split again.
Our method not only can solve the conventional color image quantization that different bit-rate would correspond to a different size of palette, but also under conditions similar to the bit-rate the quality of reconstructed images are better than traditional methods.
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Xu, Jian-Wei, and 許建偉. "A VARIABLE-BIT RATE VIDEO SOURCE MODEL." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/29549402298994900796.
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Lin, Tzung-Liang, and 林宗良. "A VARIABLE BIT RATE CELP SPEECH CODER." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/88623392182276404082.
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碩士<br>大同大學<br>電機工程研究所<br>90<br>In the era of mobile and network communication, speech is still the most natural and convenient manner for human to exchange information. Attempts are made continuously to pursuit speech coding techniques with lower bit rates and better synthetic speech quality. The use of variable bit rate (VBR) coders is undeniably an attractive approach for maintaining speech quality at lower average bit rate. The aim of this thesis is thus to design a VBR speech coder based on the Code Excited Linear Prediction (CELP) at 4.8 kbps, which was standardized as Federal Standard FS-1016 in 1982.
The basic idea of our system design is from the observation: the frame size of most speech coders is around 20-30 ms, while the speech signal is slowly time-varying, e.g., vowel sounds may last for 200-300 ms, during which the vocal tract remains nearly unchanged. This observation suggests that the speech parameters, such as the Linear Predictive Coding (LPC) parameters and the Line Spectral Frequencies (LSFs), may share high similarity between the current frame and some temporally closed previous frames. This means that it is not necessarily to transmit a set of new parameters for each frame. Instead, speech parameters of a previous frame may be used in the decoder to save the bit rate.
Based on the concept described above, we introduced an adaptive forward/backward quantization (AFBQ) [1] scheme to reduce the required for transmitting of LPC parameters. Specifically, the spectral distances between the current frame and some previous frames are calculated and an experimentally determined threshold is used to decide either the LPC parameters of the current frame should be transmitted or it is sufficient to transmit only a location index of a previous frame. The AFBQ scheme can reduce bit rate at a minimum cost of speech quality. To further reduce the bit rate, instead of using a 34-bit scalar quantizer, the proposed VBR coder utilizes a 10-bit vector quantizer (VQ) for the quantization of the LSF parameters. On the effort of speech quality improvement, we adopted a perceptual weighted distance measure in the LSF vector quantizer and incorporated an interpolation scheme for LSF parameters to smooth the spectral changes in the synthetic speech. The LSF interpolation scheme can improve the speech quality without the need of transmitting extra bits, but at the cost of 15-ms longer coding delay.
Our experimental results and informal listening test showed that, by using the AFBQ scheme and LSF vector quantizer, the proposed VBR speech coder could maintain speech quality at a lower average bit rate. For example, the VBR coder at 3.9 kbps can retain the average segmental signal-to-noise ratio (segSNR) with only 0.6 dB lower than that of the 4.8 kbps CELP coder, and their synthetic speech quality can hardly be differentiated. The experimental results also showed that the inclusion of the LSF interpolation scheme did improve the speech quality with a higher average segSNR of 0.8 dB.
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Lien, Kai-Cheng, and 廉凱成. "A VARIABLE BIT RATE MELP SPEECH CODER." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/48041242170593651845.
Full textAbstract:
碩士<br>大同大學<br>通訊工程研究所<br>90<br>In the era of third-generation (3G) wireless personal communications, though applications of multimedia such as video and data communication have become more and more popular, speech communication is still one of the most important mobile radio services. Consequently, speech coding techniques that can compress speech information into as few parameters as possible is increasingly important. To achieve the goal, the use of variable bit rate speech coders is certainly an attractive approach to retain overall high voice quality at low average bit rate. The aim of this thesis is thus to develop a variable-rate speech coder based on the Federal Standard 1017 (FS-1017), a 2.4 kbps mixed excitation linear prediction (MELP) coder originally developed by the Texas Instruments and then standardized by the U.S. Department of Defense.
In the FS-1017 standard 2.4 kbps MELP coder, sampling rate is 8 kHz with 16 bit resolution and the frame size is 22.5 ms. Each frame has a bit stream of 54 bits, wherein 25 bits are LPC coefficients, which account for 46% of the required bandwidth. Therefore effectively quantization of the LPC coefficients is essential to reduce overall bit rate for the MELP coder.
In the FS-1017 MELP coder, LPC parameters are transformed to line spectral frequencies (LSFs) and then quantized by a fixed four-stage vector quantizer. However, our experimental results showed that, with only one- to three-stage VQ, more than 20% quantized LSFs could satisfy the requirement of “transparent quantization,” i.e., having an average spectral distortion (SD) less than 1 dB. Accordingly, we proposed to utilize a variable-stage vector quantizer (VSVQ) to design a variable-rate MELP speech coder. Specifically, we insert an experimentally determined threshold after each stage of the VSVQ to determine whether the SD requirement is satisfied. When the answer is yes, the procedure of the VSVQ is stop to save the bits for the following VQ stages.
Our experimental results showed that the speech quality of the proposed variable-rate MELP coder is very close to that of FS-1017 standard 2.4 kbps MELP coder. When the average bit rate is around 2.1 kbps, there is no audible difference between the FS-1017 and the proposed variable-rate MELP coders. The experimental results also showed that the Signal-to-Difference Ratio (SDR) between the synthetic speech of the FS-1017 and that of the proposed variable-rate MELP coder is as high as 85 dB. The structure of the variable-stage vector quantizer we proposed in this research has been proved to be a success in MELP coder. We believe that it also has high potential to be used in other types of speech coders to extend their usage.
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LIU, ZHENG-YI, and 劉正義. "Vector sum CELP for variable bit rate coding." Thesis, 1991. http://ndltd.ncl.edu.tw/handle/25309396114575261258.
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Chen, Chin Ping, and 陳錦屏. "A Variable Bit Rate LD-CELP Speech Coder." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/96728154266187513605.
Full textAbstract:
碩士<br>大同大學<br>通訊工程研究所<br>89<br>Since speech signals have time-varying characteristics consisting of different types of segments, and furthermore since silence may occupy a large portion in conversation, the design of a variable bit rate (VBR) speech coder is certainly an attractive approach to retain overall high voice quality at a low average bit rate. The possible forms for VBR speech coders include using different degrees of coding resolutions for voiced/unvoiced/silence segments or using a closed-loop voice quality evaluation to select the bit-rate. The aim of this thesis is thus to develop a VBR speech coder based on the Low-Delay Code Excited Linear Prediction (LD-CELP) at 16 kb/s, which was standardized by the International Telecommunication Union (ITU) as recommendation G.728 in 1992. The proposed system is designed to retain the main structure and various analysis frames of the LD-CELP, and thus keeps the feature of low coding delay unchanged.
Through investigating the codevector use frequency (CUF) of the LD-CELP excitation codebook, we found that the CUF’s of both the shape codebook and the gain codebook spread across a very large dynamic range. Our experimental results further showed that the CUF of the shape codebook is dependent on individual users and converges in 3 seconds. Based on this observation, the proposed system is designed, for a specific a user, by first calculating the CUF of the initial 3 seconds and then constructing reduced-sized excitation codebooks by gathering codevectors with high CUF’s. By this means, excitation codebook of required sizes can be constructed to ensure lower down the average bit rate at price of little degradation of overall voice quality. In addition to the 10-bit excitation codebook (corresponding to bit rate of 16 kb/s) of the original LD-CELP, the proposed VBR speech coder contains 5 other reduced-sized excitation codebooks of 9 to 5 bits, which corresponding to transmission bit rates from 14.4 to 8 kb/s. Our simulation results showed that, for a similar case of 14.4 kb/s, our new system outperforms the dual rate (16/14.4 kb/s) LD-CELP speech coder proposed by the previous researcher, Zhang.
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謝明昇. "A Smoothing Algorithm in Variable Bit Rate Streaming." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/ftw5k7.
Full textAbstract:
碩士<br>國立交通大學<br>電信工程系所<br>93<br>With the increase of bandwidth and the progress in image compression, it has become acceptable to watch real-time high quality video through the network.. VBR compression can gain better quality but it introduces burstier traffic. This will cause a burden to the network service, making the bandwidth usage inefficient. Smoothing algorithm can deal with this problem; it can make the traffic less burstier. The optimal smoothing algorithm can optimizes the traffic in terms of the variance. In this thesis, first we introduce and analyze this algorithm; then we simulate it on real video traces, discussing the relation between buffer size and statistical data, such as peak frame size and variance. Second, we introduce the minimum unit(byte) in transmission rate to this algorithm and solve the problems it meets with. Then we simulate this algorithm on the same video trace and discuss its characteristics. Finally, we compare the original algorithm and the modified one.
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彭英欽. "A perceptually tuned subband video codec for lowconstant bit-rate (CBR) and variable bit-rate (VBR)transmission." Thesis, 1994. http://ndltd.ncl.edu.tw/handle/89394244636790023876.
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Yung-Hen, Lee. "Energy-Efficient Considerations on a Variable Bit-rate Device." 2006. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-0702200613350100.
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Huang, Zhi Wei, and 黃智瑋. "Improvement of the Variable Bit-Rate Residual-Transformed Vocoder." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/55067139591931198312.
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Hu, Chung-Rong, and 胡展榮. "Design and TMS320C30 Implementation of Variable Bit-Rate Vocoders." Thesis, 1997. http://ndltd.ncl.edu.tw/handle/93882534452822632089.
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Lee, Yung-Hen, and 李永恆. "Energy-Efficient Considerations on a Variable Bit-rate Device." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/04040098361426485741.
Full textAbstract:
碩士<br>國立臺灣大學<br>資訊工程學研究所<br>94<br>Dynamic power management has been adopted in many systems to reduce the power consumption by changing the system state dynamically. This thesis explores energy efficiency for systems equipped with PCI Express devices, which are designed for low power consumption and high performance, compared to corresponding PCI devices. We propose a dynamic power management mechanism and a management policy for energy-efficiency considerations. A case study is exploited for a variable-bit-rate (VBD) LAN device under the PCI Express specification to provide dynamic packet transmission support. Simulation results show that the proposed mechanism and policy would reduce the system energy consumption substantially.
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Lee, Wei-Jiang, and 李維疆. "Design and evaluation of variable bit-rate residual-transformed vocoders." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/66865703670908488848.
Full textAbstract:
碩士<br>國立成功大學<br>電機工程研究所<br>83<br>In this thesis, a new vocoder scheme for coding speech at low
bit rate(4.8 Kb/s) with good quality is proposed. The key idea
here is to perform "Hadamard Transform" or "Discrete Cosine
Transform" (DCT) upon residual sequence of frame-length speech,
then coding it by vector quantization(VQ) procedure. For voiced
sound, the residual signal means the residue after subtracting
the period component from original speech, the period is
decided from the previously coded speech frames. For unvoiced
sound, the residual signal denotes the inverse linear
predictive (LP) filtering of the speech. Furthermore, we
evaluate our algorithm by using the develop- ment tools for
TMS320C30 floating point digital signal process- or. The real-
time evaluation system includes the C30 software assembly codes
and hardware interfacing circuits design. The advantage of
proposed vocoder reduce the computations which makes the
algorithm more suitable to be realized a real-time duplex(
encoder and decoder) system implemented in single digi- tal
signal processor. Another merit of this vocoder is that, it can
offer concurrently 9.6 Kb/s or other bit rates to achieve
higher quality speech without cost of any computational burden.
In other words, the proposed vocoder can offer variable bit-
rated speech in order to provide different speech qualities
whenever the bandwidth of channel is adequate.
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Lee, Sho-chi, and 李修齊. "Strategic of Flow Control for Variable-Bit-Rate Video Streams." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/27831596173454546011.
Full textAbstract:
碩士<br>國立屏東科技大學<br>資訊管理研究所<br>86<br>Stream-like delivery of videos, compressed into a sequence of variable size frames, presents a grand challenge on the design of video-on-demand systems, in particular,in the aspect of flow control. An efficient video flow control,which intends to take the maximum advantage of system resources (such as
buffer space and network bandwidth) to satisfy the real-time constraint of the data delivery imposed on the end applications, has to be implemented on the level of the end-to-end data path. The end-to-end data path basically involves three processes: server producing data, network transmitting data and client consuming data. Their individual data processing characteristics,determining mutual interaction, need to be carefully studied in order to achieve the efficient video flow control.
In this thesis, we investigate a simple buffer-driven flow control model,where the precedence of processes is simply controlled by the status of server or client buffers and the channel reserved is assumed to operate at a constant-bit-rate to minimize the network management overheads. We present the model in a geometric model, which offers us a convenient visual way to systemically analyze the problem. We propose an optimal algorithm of finding the minimum bandwidth with the provision of server/client data processing model and their buffer sizes. In particular, we study the way of server producing data from a more realistic perspective by assuming a circular lagged resumption property, where a stopped stream can only be resumed its data retrieval when the server runs into to the next time slot reserved for it. This property, quite common to the video servers employing data striping technique, presents a unique problem in performing the buffer-driven flow control.
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"Adaptation of variable-bit-rate compressed video for transport over a constant-bit-rate communication channel in broadband networks." Chinese University of Hong Kong, 1995. http://library.cuhk.edu.hk/record=b5888495.
Full textAbstract:
by Chi-yin Tse.<br>Thesis (M.Phil.)--Chinese University of Hong Kong, 1995.<br>Includes bibliographical references (leaves 118-[121]).<br>Chapter 1 --- Introduction --- p.1<br>Chapter 1.1 --- Video Compression and Transport --- p.2<br>Chapter 1.2 --- VBR-CBR Adaptation of Video Traffic --- p.5<br>Chapter 1.3 --- Research Contributions --- p.7<br>Chapter 1.3.1 --- Spatial Smoothing: Video Aggregation --- p.8<br>Chapter 1.3.2 --- Temporal Smoothing: A Control-Theoretic Study。 --- p.8<br>Chapter 1.4 --- Organization of Thesis --- p.9<br>Chapter 2 --- Preliminaries --- p.13<br>Chapter 2.1 --- MPEG Compression Scheme --- p.13<br>Chapter 2.2 --- Problems of Transmitting MPEG Video --- p.17<br>Chapter 2.3 --- Two-layer Coding and Transport Strategy --- p.19<br>Chapter 2.3.1 --- Framework of MPEG-based Layering --- p.19<br>Chapter 2.3.2 --- Transmission of GS and ES --- p.20<br>Chapter 2.3.3 --- Problems of Two-layer Video Transmission --- p.20<br>Chapter 3 --- Video Aggregation --- p.24<br>Chapter 3.1 --- Motivation and Basic Concept of Video Aggregation --- p.25<br>Chapter 3.1.1 --- Description of Video Aggregation --- p.28<br>Chapter 3.2 --- MPEG Video Aggregation System --- p.29<br>Chapter 3.2.1 --- Shortcomings of the MPEG Video Bundle Scenario with Two-Layer Coding and Cell-Level Multiplexing --- p.29<br>Chapter 3.2.2 --- MPEG Video Aggregation --- p.31<br>Chapter 3.2.3 --- MPEG Video Aggregation System Architecture --- p.33<br>Chapter 3.3 --- Variations of MPEG Video Aggregation System --- p.35<br>Chapter 3.4 --- Experimental Results --- p.38<br>Chapter 3.4.1 --- Comparison of Video Aggregation and Cell-level Multi- plexing --- p.40<br>Chapter 3.4.2 --- Varying Amount of the Allocated Bandwidth --- p.48<br>Chapter 3.4.3 --- Varying Number of Sequences --- p.50<br>Chapter 3.5 --- Conclusion --- p.53<br>Chapter 3.6 --- Appendix: Alternative Implementation of MPEG Video Aggre- gation --- p.53<br>Chapter 3.6.1 --- Profile Approach --- p.54<br>Chapter 3.6.2 --- Bit-Plane Approach --- p.54<br>Chapter 4 --- A Control-Theoretic Study of Video Traffic Adaptation --- p.58<br>Chapter 4.1 --- Review of Previous Adaptation Schemes --- p.60<br>Chapter 4.1.1 --- A Generic Model for Adaptation Scheme --- p.60<br>Chapter 4.1.2 --- Objectives of Adaptation Controller --- p.61<br>Chapter 4.2 --- Motivation for Control-Theoretic Study --- p.64<br>Chapter 4.3 --- Linear Feedback Controller Model --- p.64<br>Chapter 4.3.1 --- Encoder Model --- p.65<br>Chapter 4.3.2 --- Adaptation Controller Model --- p.69<br>Chapter 4.4 --- Analysis --- p.72<br>Chapter 4.4.1 --- Stability --- p.73<br>Chapter 4.4.2 --- Robustness against Coding-mode Switching --- p.83<br>Chapter 4.4.3 --- Unit-Step Responses and Unit-Sample Responses --- p.84<br>Chapter 4.5 --- Implementation --- p.91<br>Chapter 4.6 --- Experimental Results --- p.95<br>Chapter 4.6.1 --- Overall Performance of the Adaptation Scheme --- p.97<br>Chapter 4.6.2 --- Weak-Control verus Strong-Control --- p.99<br>Chapter 4.6.3 --- Varying Amount of Reserved Bandwidth --- p.101<br>Chapter 4.7 --- Conclusion --- p.103<br>Chapter 4.8 --- Appendix I: Further Research --- p.103<br>Chapter 4.9 --- Appendix II: Review of Previous Adaptation Schemes --- p.106<br>Chapter 4.9.1 --- Watanabe. et. al.'s Scheme --- p.106<br>Chapter 4.9.2 --- MPEG's Scheme --- p.107<br>Chapter 4.9.3 --- Lee et.al.'s Modification --- p.109<br>Chapter 4.9.4 --- Chen's Adaptation Scheme --- p.110<br>Chapter 5 --- Conclusion --- p.116<br>Bibliography --- p.118
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Kim, Hyun Soo. "Speech analysis techniques useful for low or variable bit rate coding /." 2005. http://www.library.unsw.edu.au/~thesis/adt-NUN/public/adt-NUN20050803.233219/index.html.
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Chang, Tien Yu, and 張殿宇. "On Variable Bit Rate Video Sources for Asynchronous Transfer Mode Transmissions." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/51149626811408231553.
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Yen, Jui-Yu, and 嚴瑞友. "Effective Moving Object Detection Over Variable Bit-Rate Wireless Video Streaming." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/28nn8v.
Full textAbstract:
碩士<br>國立臺北科技大學<br>電腦與通訊研究所<br>100<br>Motion detection plays an important role in video surveillance system. Video communications over wireless networks can easily suffer from network congestion or unstable bandwidth, especially for embedded application. A rate control scheme produces variable bit-rate video streams to match the available network bandwidth. However, effective detection of moving objects in variable bit-rate video streams is a very difficult problem. This paper proposes an advanced approach based on the counter-propagation network through artificial neural networks to achieve effective moving object detection in variable bit-rate video streams. The proposed method is composed of two important modules: a various background generation module and a moving object extraction module. The proposed various background generation module is employed in order to generate the adaptive background model which can express properties of variable bit-rate video streams. After an adaptive background model is generated by using the various background generation module, the proposed moving object extraction module is employed to detect moving objects effectively from both low-quality and high-quality video streams. Lastly, the binary motion detection mask can be generated as the detection result by the output value of the counter-propagation network. In this paper, we compare our method with other state-of-the-art methods. To demonstrate the performance of our proposed method in regard to object extraction, we analyze qualitative and quantitative comparisons in real-world limited bandwidth networks over a wide range of natural video sequences. The overall results show that our proposed method substantially outperforms other state-of-the-art methods by Similarity and F1 accuracy rates of 83.34% and 89.71%, respectively.
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CHANG, CHANG HUNG, and 張宏昌. "Evaluating Performance of Hierarchal Link-Sharing Mechanisms with Real-Time Variable Bit-Rates Traffics under Linux." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/57147666495273475689.
Full textAbstract:
碩士<br>大葉大學<br>資訊工程學系碩士在職專班<br>95<br>Because of the popularization of the internet network and development of the multimedia technology, there are a lot of the pronunciation and communication of media and control the protocol like RTP through the internet network, Session Initiation Protocol (SIP) [1] ,etc. are regarded as the protocol of linking up of the
materials by development, these protocols are all for set up , revise , finish the control of conversation line , make internet network exchange by simple materials information, rise to real time video to transmit. Use internet network is substitute a
traditional real time video to transmit , can save effectively not merely to come, and can use a lot of extra functions , such as interactive video and internet phone ,etc. But the real time video package is not like other use serving the package produced , can pass last time because network droped , wide getting enough package caused of the factor lose frequently again. Because the pronunciation package is instant materials, the package does not have any help to it to spread again, so it is the best
method not to allow it and to lose as much as possible.Because the network application service appears like the mushrooms after rain, how be can frequently among the wide resource all kinds of package and frequently wide to it assigns to be a basic solution properly in a limited one.
With the wide-band popularization that surfed the Net, the application of the network moves towards the diversification and develops too. Various kinds of network application, to serving the demand for quality differently, for instance: On-line to is it need instant interdynamic of to talk, multimedia bunch flow audio-visual to is it need a large amount of frequently wide to broadcast. If network user can distribute the frequently wide amount and frequently wide priority of use according to different characteristics that use , can make rational and effective
application to resources of the network. Link-Sharing structure based on classification has been attracted attention over these several years under Linux, with Class Based Queuing in Link-Sharing structure (CBQ) [4] [5] and Hierarchical Token Bucket (HTB ) [6], in order to make comparisons in fact commonly. In Link-Sharing structure is done in fact , generally think that HTB deals with the
package and wide managerial ability is all superior to CBQ frequently . But few documents serve the getting instant variable speed under the circumstances that HTB and CBQ are done in fact there can be intact test and analysis, so this text flow
media and frequently wide to is it make systemic test and discussion to share stratum type to instant bunch, content visit HTB and CBQ flow test efficiency of media to instant bunch under testing environment to demand to include, HTB and CBQ happen frequently widely while sharing , can reach media Delay and request
for Jitter that bunch real time video .
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Ding, Jen-Wen, and 丁建文. "An efficient block allocation strategy for variable bit rate continuous media data." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/97132284137730468098.
Full textAbstract:
碩士<br>國立成功大學<br>工程科學系<br>86<br>The advances in storage, computing, and communication
technologiesin the past decade have made it feasible to provide
on-line accessto a variety of information resources (such as
newspaper, image,video clips, audio, etc.) over high speed
networks. One of thefundamental technologies to construct such
an information retrievalsystem is continuous media (CM) servers,
which provide continuousmedia data for clients. The main design
goal of a CM server is toimprove the number of supported users
of a CM server. In the pastfew years, many techniques have been
developed to achieve thisdesign goal. One of the most important
developed techniques is datastriping, which increase the
availability of the CM data of a CMserver. According to previous
studies, the most widely appliedstriping strategy is round-robin
permutation (RRP), which allocatesconstant-sized data blocks to
a number of hard disks in a round-robinfashion. However,
applying RRP on a variable-bit-rate (VBR) CM serverwill result
in performance decay due to the unbalanced loads on diskscaused
by the VBR nature of CM data. Unfortunately, owing to thenature
of compression algorithms, compressed media data are usuallyVBR.
Thus, it is inevitable for RRP to encounter this problem.In this
thesis, we propose a novel block allocation strategy,
calledconstant time permutation (CTP), which takes the VBR
nature of CMfiles into consideration. CTP has more balanced
loads on disks andcan achieve a higher efficiency than RRP. Both
analytic models andtrace-based simulations are constructed for
CTP and RRP to evaluatetheir performance. Both analysis and
simulation results show thatCTP can obtain more balanced loads
on disks and can support at least20% more users than RRP. In
addition, the performance of CTP isindependent of the scale of
the system, while that of RRP declinesas the system scale
enlarges.
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Chang, Chin-Yane, and 張俊彥. "Design and Implementation of a Channel Manager for Variable Bit Rate Based Data Transmission." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/16998789219006567588.
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碩士<br>國立交通大學<br>電子研究所<br>83<br>In this thesis, we present a channel manager(CM) for variable
bit rate (VBR) based data transmission structure, so that
channel utilization ratio can be enhanced. That is, more
sources use the same network channel to transmit their data
under this CM structure. Also, we design two VLSI modules of
CM: one is a high speed sorter for the priority selection in
this CM structure, and the other is a data encryption chip for
communication security. We also present automation CAD
viewpoints for the design flow under development in our
research Lab, where we briefly discuss the design methodology
and verification approach based on the available CAD tools.
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XU, MING-XIAN, and 徐明憲. "Modeling and analysis of variable bit-rate video sources over asynchronous transfer mode networks." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/26603370660726359064.
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Chen, Zhi Wei, and 陳志維. "A perceptually tuned three-dimensional subband video codec for variable bit-rate (VBR) transmission." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/54530276977737901885.
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