Delivery of low rate isochronous streams over the digital video broadcasting/Digital Audio-Visual Council cable television protocol

This paper examines the capacity and performance characteristics of the digital video broadcasting (DVB)/Digital Audio-Visual Council (DAVIC) cable television protocol for the delivery of low rate isochronous streams for a cable population of up to 700 nodes. Streams (ranging from 8 to 128 kbps) suitable for timing critical services such as compressed/uncompressed voice (e.g., VoIP: G711 and G.7231), audio and low quality video, were considered in order to study the effects on channel capacity when using reservation and fixed access for the delivery of timing critical services. The analysis focuses on the,performance of the upstream channel, which is the limiting factor of community antenna television (CATV) networks and is critical in the delivery of services to individual subscribers on demand. Simulation results indicated that such streams, within the given protocol limitations, can be supported for a particular system population with trade-offs in terms of system throughput and channel utilization. Network capacity, in terms of the number of simultaneous streams supported and link utilization, is significantly affected by packet size. Analysis of the results indicated that for different streams, packet sizes and combined with header suppression, the benefits from the use of fixed access is essential for the support of timing critical services.     

Variable bit rate transmission of television signals

An estimate of the advantage of variable bit rate over fixed bit rate transmission of television signals has been obtained by recording and processing 735 48-frame sequences during 72 hours of broadcasting. This advantage is calculated to be between 3.4 and 3.8     

Optimal bit rate conversion of MPEG-2 video bit streams

An optimal bit rate conversion scheme for MPEG-2 video bit streams, based on a drift free video transcoder, is proposed. It is shown that optimally transcoded bit streams produce better picture quality than both a cascade of decoder-encoder and than a standard encoder using the original pictures, at the same reduced bit rates     

Online traffic smoothing for delivery of variable bit rate media streams

Traffic smoothing for delivery of online variable bit rate (VBR) media streams is one of the most important problems in designingmultimedia systems. Given an available client buffer, a window size, and a window-sliding size, previous window-based online smoothing methods have tried to reduce the peak bandwidth allocated in each window. However, as bandwidths allocated in different windows are minimized independently, these methods require a large peak bandwidth for transmitting an entire stream. In this paper, a new window-based method is proposed. It introduces two new ideas, the dynamic windowsliding size and the aggressive workahead, for delivery of online VBR media streams. Our aggressive and dynamic window sliding (ADWS) method can automatically decide on the suitable window-sliding sizes for different windows. Thus, the allocated peak bandwidth can be further reduced. By examining various media streams, ADWS is shown to be effective and efficient. Considering the online transmission of the movieStar Wars with a 90-kB client buffer, ADWS yields 13% less in peak bandwidth, compared with the bestknown window-based online smoothing algorthm SLWIN(1). Its computation cost (the window-sliding number) is 75% of that required by SLWIN(1).This work was partially supported by NSC, Taiwan, under grants. NSC88-2213-E-001-011, NSC88-2213-E-001-012, and NSC88-2213-E-001-025. A primitive version of this work appeared in IEEE INFOCOM'99 [7].     

Analysis of low bit rate image transform coding

Calculations based on high-resolution quantizations prove that the distortion rate D(R¯) of an image transform coding is proportional to 2^-2R when R¯ is large enough. In wavelet and block cosine bases, we show that if R¯<1 bit/pixel, then D(R¯) varies like R¯^1-2γ, where γ remains of the order of 1 for most natural images. The improved performance of embedded codings in wavelet bases is analyzed. At low bit rates, we show that the compression performance of an orthonormal basis depends mostly on its ability to approximate images with a few nonzero vectors     

极低误码下的误码率估计问题

在极低误码率的情况下由于测得的误码点很少，由实测的点来直接估计误码率这一方法变得不可靠。本文提出一种方法，通过牺牲测量值的精度来换取结果的可靠性，即用误码率的上界来描述误码特性。本文还给出了误码率的上界、实际测量得到了误码点数和表示误码率上界可靠性的门限值之间的定量关系。     

Recovery analysis of the DOCSIS protocol after service disruption

Wide scale power outages can cause serious disruptions to digital services over community antenna television (CATV) networks. This can result in long service recovery times for some or all of the cable modems (CMs). In this paper, the recovery mechanisms of the Data Over Cable Service Interface Specification (DOCSIS) protocol, due to service disruption, are studied. We propose a new scheme for the ranging of modems, namely persistent ranging algorithm (PRA) and two different simulation models have been developed for this purpose. The simulation results show that the static nature of the registration and initialization parameters can introduce an excessive delay in the recovery time of the system. Minimum recovery times of 820 s and 975 s (two ranging opportunities/second) for network architectures with 400 CMs and 500 CMs, respectively, on a single upstream channel were obtained. In order to validate the accuracy of the simulation models a set of experiments was performed on a CATV network testbed. The testbed results revealed that the simulation models produced are realistic and offer reliable estimates of large-scale network topologies. Extensions to the DOCSIS MAC protocol in which the status of the system prior to disruption is used have reported a decrease of up to 42% in the recovery time of the system     

H.263甚低码率视频编码标准的码率控制方法

主要提出ＩＴＵ组织的Ｈ．２６３甚低码率视频压缩编码标准输出码率的控制方案。着重分析了一种通过调整宏观量化器的量化参数来控制Ｈ．２６３编码器输出码率的方法,它能满足在ＰＳＴＮ网上低时延中的码率输出要求。     

混合激励线性预测低速率语音编码研究

为了满足数字通信及其他商业应用的需求,语音压缩编码技术得到迅速发展。近年来主流的低速率语音编码方案主要基于LPC-10,混合激励线性预测（MELP）,多带激励编码（MBE）,正弦变换编码（SCI）,波形内插编码（WI）。大多都工作在2．4kb／s速率下。作为一种重要的低速率语音编码算法。MELP算法对LPC-10编码方案进行大量改进,引入混合激励,非周期脉冲,残差付氏幅度谱,脉冲散布和自适应谱滤波5个特征。实验结果表明,该混合激励线性预测编码在2．4kb／s上得到了更好的合成语音,并使得合成语音能更好地拟合自然语音。     

混合激励线性预测低速率语音编码研究

为了满足数字通信及其他商业应用的需求,语音压缩编码技术得到迅速发展.近年来主流的低速率语音编码方案主要基于LPC-10,混合激励线性预测(MELP),多带激励编码(MBE),正弦变换编码(SCI),波形内插编码(WI),大多都工作在2.4 kb/s速率下.作为一种重要的低速率语音编码算法,MELP算法对LPC-10编码方案进行大量改进,引入混合激励,非周期脉冲,残差付氏幅度谱,脉冲散布和自适应谱滤波5个特征.实验结果表明,该混合激励线性预测编码在2.4 kb/s上得到了更好的合成语音,并使得合成语音能更好地拟合自然语音.     

A financial model of cable television with a simulation analysis of the 1993 rate regulations

As an illustration of how a widely accepted financial analysis format can be developed into a simulation tool to investigate economic issues for telecommunications systems, a pro forma cash flow model of cable television is developed using regression analysis of market demand and operating costs, and engineering analysis of capital costs. The model can be used to simulate financial performance for a cable television system to investigate the impact of different market conditions, system or service configurations, and pricing structures. Results of some model simulations indicate that cable systems significantly benefit from higher consumer density, benefit somewhat from system size, and may have some monopoly power with regard to price. Large urban systems seem to be favored by the FCC's 1993 benchmark prices.     

Trellis decoding of Reed-Solomon codes in low bit rate speechcommunications

The residual redundancy that persists in the transmitted parameters of a low bit rate speech coder are exploited to reduce the computational complexity of a Reed-Solomon (RS) trellis decoder. The use of RS codes for mobile telephony provides the opportunity for avoiding channel interleaving and so reduces one-way delay     

Efficient transmission of MPEG‐2 video bit streams over HIPERLAN/2

This paper studies the transmission of MPEG‐2 video bit streams over HIPERLAN/2 based wireless local area networks (WLANs). HIPERLAN/2, a WLAN standard introduced by ETSI (the European Telecommunications Standards Institute), is well known for its ability to operate under different physical (PHY) modes (modulation schemes) and adaptively switch its modes within a single transmission session. Each physical mode has certain throughput and error resilience associated with it. The presence of multiple physical modes can be exploited to achieve Quality of Service (QoS) guarantees for application parameters such as throughput, bit error rate (BER), jitter etc. In this paper, we first study the relationship of different PHY modes with BER and throughput. Based on this study, different content based marking algorithms are presented and analyzed that partitions the input video bitstream into multiple segments, where each segment is deemed suitable for transmission over HIPERLAN/2 operating under a specific PHY mode. We show that for a given video bit stream, the use of multiple PHY modes chosen based on the characteristics of underlying video data yields superior throughput and better picture quality in terms of pixel signal to noise ratio (PSNR). From the power consumption point of view, the marking algorithms can achieve desired throughput or PSNR with lower signal to interference ratio (SIR) levels thus saving battery power and extending the overall battery life. Copyright © 2004 John Wiley & Sons, Ltd.     

Performance characterisation of the MCNS DOCSIS 1.0 CATV protocolwith prioritised first come first served scheduling

The Multimedia Cable Networks Systems (MCNS) Data Over Cable Service Interface Specifications (DOCSIS) is established as the primary cable network data communications standard. The head-end scheduling algorithm is not defined within the standard but it is the key function for providing the required performance capability. Computer simulation, using the Common Simulation Framework (CSF) 12 version of the CableLabs DOCSIS 1.0 computer model, has been used to predict the upstream system throughput and mean access delay. A prioritised first come first served scheduling algorithm has been considered to provide a baseline reference set of performance statistics against which other algorithms can be compared. The simulations, with theoretical confirmation, have shown that the maximum sustainable system throughput is 1965 kbps for a channel capacity of 2560 kbps and a packet size of 1500 octets, whereas for packet size of 100 octets the maximum sustainable throughput is only 1550 kbps. The mean access delay is found to vary between 10-900 ms depending on the offered load and assuming that the offered load does not exceed the capacity of the channel. Excess offered load causes service starvation according to the assigned priority     

Delay analysis of continuous bit rate traffic over an ATM network

A packet multiplexer is modeled for continuous bit rate (CBR) traffic in an asynchronous transfer mode (ATM) network as an nD/D/1 queue. The efficiencies of various algorithms for finding the delay distribution are compared. In particular, a new algorithm is proposed whose time complexity is O(n²), where n is the number of voice sources being multiplexed. The use of the central limit theorem can reduce the time complexity to O(n) for large n . An asymptotic formula is found whose time complexity is independent of n and it works well (for practical purposes) over a wide range of parameter values. The authors examine and comment on the use of the M/D/1 results as an approximation. In addition to comparing the performances of these algorithms, they show that the buffer requirements for such a queue are significantly less than the theoretical maximum (even when the requirement on the call disruption probability is very low). This result has important implications in the design of buffer size. The buffer requirement is relatively insensitive to the design criterion (call disruption probability)     

低码率活动图像帧间预测编码的研究

主要研究适于窄带低码率活动图像的帧间预测编码算法。依据Ｈ．２６３建议中的算法和编码方案，着重论述了三种先进的帧间预测模式：无约束运动矢量模式、高级预测模式和ＰＢ帧模式，并分别对其编、解码的效果作出计算机仿真及分析。     

Lossy multiplexing of low bit rate speech in thin route telephony

To alleviate congestion at thin route multiplexers such as are used in VSAT systems, some transmitters are forced to drop speech packets. Receiver reconstruction of the lost speech invariably produces some degradation of quality. The authors exploit the knowledge of dropped packets at the transmitter to mitigate this degradation     

Trial results of novel bit error rate monitor scheme over skywave HF link

Details are given of an error monitoring technique applicable to any digital communication system. An example application of the technique to a live skywave HF link using parallel-tone modems is described. The performance of the scheme is presented and discussed for this channel.<>     

Policy allocation for transmission of embedded bit streams over noisy channels with feedback - [Transactions letters]

An efficient policy allocation algorithm for the transmission of embedded bit streams over noisy channels with feedback is proposed. The transmission is based on the type- II hybrid ARQ/FEC protocol and uses a nested sequence C of channel codes to protect the packets. There are also constraints on the total bit budget and on the allowed number of retransmissions per packet. The allocation algorithm assigns different protection policies, each policy being a subset of C, to different packets to maximize the average number of correctly received source bits. We study the performance and the complexity of the proposed scheme through the transmission of images encoded by JPEG2000 over mobile channels with correlated Rayleigh fading. We demonstrate by simulations that the proposed multiplepolicy scheme provides significant improvements over a purely FEC scheme with no feedback and also the existing fixed-policy schemes. Our results show that feedback is particularly helpful for poor channel conditions and that the proposed scheme is very robust against changes in the channel signal-to-noise ratio (SNR) and the mobile speed.     

Digital television adoption: Comparing the adoption of digital terrestrial television with the adoption of digital cable in Taiwan

This study employed Rogers’ model to compare the diffusion of digital terrestrial television with the diffusion of digital cable in Taiwan. A telephone survey, which yielded 753 valid questionnaires, was conducted to collect data. The results of this study show that the factors affecting the intention to adopt digital terrestrial television differed widely from those affecting the intention to adopt digital cable. The diffusion of digital cable was discovered to be generally congruent with the prediction of Rogers’ model, while the diffusion of digital terrestrial television was not. This second finding reveals a limitation in Rogers’ model, namely, its alleged pro-innovation bias. It may be that the respondents in this study looked favorably on digital cable because they considered it, and not digital terrestrial television, to be an inevitable technology. Furthermore, this study found that awareness played an important role in respondents’ adoption of digital television. This study therefore suggests that, to accelerate digital conversion, Taiwan’s government should establish policies to educate people about digital television.