PERFORMANCE EVALUATION OF NON-BLOCKING ATM SWITCHES UNDER VARIOUS TRAFFIC AND BUFFERING SCHEMES

Broadband integrated services digital networks (BISDN) are designed to offer a variety of services with bit rates ranging from several kb/s (e.g. teleactions) to hundreds of Mb/s (e.g. HDTV), and in some cases approaching Gb/s (e.g. in interconnections of high-speed LANs). A multiplicity of rates and the burstiness of traffic sources lead naturally to systems based on the fast packet switching (or asynchronous transfer mode) concept. The requirements of data buffering and high-speed processing of packet headers have resulted in a plethora of proposals for ATM switching nodes and an equal number of ways for evaluating them. In particular, the class of non-blocking architectures of ATM switches has received the most attention from the research community. This paper reviews this class of architecture with emphasis on contention resolution methods and communication traffic performance. The contention resolution methods are divided into four categories: input buffering, output buffering, shared buffering, and hybrid buffering. The communication traffic characteristics are divided into two categories: uniform traffic and bursty traffic.     

Performance Model of a WIMAX 2.0 All-IP 4G System

4G is promising a wireless broadband with data rates up to 1Gbps. The two candidate technologies for 4G are the Advanced Long Term Evolution (Advanced LTE) which is based on the 3GPP standards and the WiMAX 2.0 based on the IEEE 802.16 family of standards. The common feature of both technologies is that they will provide All-IP connectivity with flexible bit rates and quality of service guarantees for multiple classes of services including voice, mainly using voice over IP, data and video services. Most of the performance studies of 4G technologies use highly complex and sophisticated simulations due to the multiple complexity factors in investigating 4G technologies such as All-IP flexible bit rates, adaptive coding and modulation as well as the multi-services provided. These factors usually make any modelling attempt very difficult. This paper presents a numerical/analytical model for a 4G WiMAX cell based on a multi-dimensional Continuous-Time Markov Chain (CTMC) model. Performance measures were derived for the key performance indicators such as throughput and average bit rate per cell and per service class. By assuming minimum acceptable bit rates for certain quality of service guarantees, we derived measures for blocking probabilities. The model has been formulated and solved using MOSEL-2 (Modelling Specification and Evaluation Language) which captures the key features of a 4G system that affect services at session/call level. The resuls obtained from the model using sample parameters show that, the model can provide very useful insight to system behavior and can give good first indication to the performance of such a complex system.     

Fixed versus adaptive admission control in direct broadcast Satellite networks with return channel systems

In this paper, as part of the adaptive resource allocation and management (ARAM) system (Alagoz, 2001), we propose an adaptive admission control strategy, which is aimed at combating link congestion and compromised channel conditions inherent in multimedia satellite networks. We present the performance comparisons of a traditional (fixed) admission control strategy versus the new adaptive admission control strategy for a direct broadcast satellite (DBS) network with return channel system (DBS-RCS). Performance comparisons are done using the ARAM simulator. The traffic mix in the simulator includes both available bit rate (ABR) traffic and variable bit rate (VBR) traffic. The dynamic channel conditions in the simulator reflect time variant error rates due to external effects such as rain. In order to maximize the resource utilization, both for fixed and adaptive approaches, assignment of the VBR services are determined based on the estimated statistical multiplexing and other system attributes, namely, video source, data transmission, and channel coding rates. In this paper, we focus on the admission control algorithms and assess their impact on quality-of-service (QoS) and forward link utilization of DBS-RCS. We show that the proposed adaptive admission control strategy is profoundly superior to the traditional admission control strategy with only a marginal decrease in QoS. Since the ARAM system has several parameters and strategies that play key roles in terms of the performance measures, their sensitivity analysis are also studied to verify the above foundations.     

A Performance Study of Delta Modulation Systems for Voiceband Data Signals

In this paper the performance of delta modulation (DM) systems have been studied by computer simulation when the input signal to the DM coder is a voiceband data signal. First, the parameter values of three DM systems, linear DM (LDM), constant factor DM (CFDM), and continuously variable slope DM (CVSD) has been optimized for 4800 bits/s differential phase shift keying (DPSK) signal. Then, the performance of the three DM systems have been studied for ideal and noisy channels. It has been found that the peak signal-to-quantization-noise ratio (SQNR) is nearly the same regardless of coding scheme used, but CFDM yields the widest dynamic range. In a noisy channel, however, CFDM is very sensitive to channel bit errors. Considering the overall performance, CVSD appears to be the best among the three DM systems studied. Also, the performances of DM's have been compared with those of PCM and DPCM systems. In addition, we have studied the effect of DM quantization noise on modem bit error rate by the Monte Carlo simulation method. It is possible to transmit a 4800 bits/s DPSK signal at a bit error rate below 10^-5by CVSD with the rate of 32 kbits/s.     

Variable Frame Rate Transmission: A Review of Methodology and Application to Narrow-Band LPC Speech Coding

We review the variable frame rate (VFR) transmission methodology that we developed, implemented, and tested during the period 1973-1978 for efficiently transmitting LPC vocoder parameters extracted from the input speech at a fixed frame rate. In the VFR method, parameters are transmitted only when their values have changed sufficiently over the interval since their preceding transmission. We explored two distinct approaches to automatic implementation of the VFR method. The first approach bases the transmission decisions on comparisons of the parameter values of the present frame and the last transmitted frame. The second approach, which is based on a functional perceptual model of speech, compares the parameter values of all the frames that lie in the interval between the present frame and the last transmitted frame against a linear model of parameter variation over that interval. The application of VFR transmission to the design of narrow-band LPC speech coders with average bit rates of 2000-2400 bits/s is also considered. The transmission decisions are made separately for the three sets of LPC parameters, pitch, gain, and spectral parameters, using separate VFR schemes. A formal subjective spccch quality test of six selected LPC coders is described, and the results are presented and analyzed in detail. It is shown that a 2075 bit/s VFR coder produces speech quality equal to or better than that of a 5700 bit/s fixed frame rate coder.     

基于二维非负矩阵分解的1kb/s WI语音编码算法

 本文针对波形内插（WI）语音编码模型和参数量化等技术进行了研究,并最终提出了一种基于二维非负矩阵分解的1kb/s波形内插（2DNMF-WI）语音编码算法. 文中采用二维非负矩阵分解（2D-NMF）方法来分解语音特征波形（CW）,该分解方法在行和列两个方向上同时压缩CW幅度谱矩阵的维数,使得CW幅度谱矩阵降维后得到的编码矩阵维数较小,易于量化. 此外,在甚低速率语音编码中,由于没有足够的比特数来描述编码参数,往往很难得到高质量的合成语音. 本算法采用两帧联合编码、帧间后向预测三级矢量量化、离散余弦变换（DCT）和分裂式矩阵量化等技术来降低编码速率和改善音质. 非正式主观听觉测试显示,1kb/s 2DNMF-WI编码器合成语音的质量稍差于2kb/s的NMF-WI语音编码算法.     

Introduction of very high bit rates for NFC and RFID

This article gives a detailed insight into the very high bit rates (VHBR) technology for the data transfer with NFC and RFID. In the first part enhancements and changes to the ISO/IEC 14443 and related standards are discussed. In the second part an analysis of different bit rates with focus on communication parameters and bit error rates are provided.     

IEEE 802.11 goodput analysis for mixed real-time and data traffic for home networks

Increasingly, wireless networks are being used to provide connection services for devices running applications with very different quality of service requirements. Although this issue has been addressed by the IEEE 802.11e standard, the fact is that most networks deployed in home/office environments nowadays use IEEE 802.11a/b/g standard devices. Unfortunately, administrators often do not set configuration parameters of network devices to maximize resources performance, thus providing poor quality of service. In this paper, two IEEE 802.11a/b/g analytical performance evaluation models for mixed traffic Ad Hoc and infrastructure WLANs are presented, assuming that some network devices are executing single applications, like VoIP, videoIP or network browsing. In our analysis, network devices are grouped according to the expected traffic pattern of the applications they are running. Then, global and individual group goodput are calculated assuming a congested network. Based upon the outcome of this analysis for different settings of the device configuration parameters, it is shown that the performance of a standard home/office IEEE 802.11 wireless network can be significantly improved by choosing appropriate values of these parameters.     

Diversity Performance of a Wavelet-Packet-Based Multicarrier Multicode CDMA Communication System

In this paper, we propose and analyze a novel multicarrier (MC) multicode (MCD) code-division multiple-access (CDMA) system employing wavelet packets (WPs) for modulation. This system can achieve robust performance against multipath fading due to the localization of WPs in the time and frequency domains. The analytical framework is presented, and the system performance with diversity is evaluated by means of bit error rates and the outage probability . Since WPs have lower sidelobes compared to sinusoidal carriers, our system is very effective in reducing the problem of intercarrier interference. The effects of system parameters (e.g., order of diversity, fading parameters, and WP type) were investigated. The major contribution is to compare the performance of the system to that of the MC/MCD-CDMA system that is based on sinusoidal carriers. The results reveal a considerable performance improvement of our proposed system over the MC/MCD-CDMA system.     

Recent Advances in Ultrahigh Bit Rate ETDM Transmission Systems

Increase of Internet traffic and introduction of triple-play services force operators to increase network capacity at moderate costs. Introduction of higher electronic time-division multiplexing (ETDM) channel bit rate targets reduce the cost per bit for the transmission due to lower power consumption, smaller footprint, less management effort, and complexity of the systems. Improved performance of electronic and optoelectronic components allows for research on ETDM bit rates beyond 40 Gb/s, which is currently the highest standardized channel bit rate for optical telecommunication networks. In this paper, an overview of recent progress in high-speed ETDM technology for 80 Gb/s and beyond and results of high-speed ETDM transmission experiments are given. Currently, the speed of electronics enables ETDM systems with line rates of 80/85 Gb/s and even 100 Gb/s, which is expected to be the next generation of Ethernet in data communication     

Very low-bit-rate segmentation-based video coding using contour andtexture prediction

The most efficient video coding standard for low bit rates (around 64 kb/s) is the H.261 algorithm recommended by ITU-TS. However, in certain applications such as mobile audiovisual communications and videophone through PSTN, the available transmission bandwidth is very limited. Therefore codecs working at very low bit rates are required. The paper presents a segmentation-based video coding algorithm that can work at rates as low as 10 kb/s. A novel representation of the contour information using a number of control points is proposed to estimate the contour shapes and locations from the previous frame by using the motion information. The texture parameters are also predicted and only the residual values are entropy coded. In addition two novel postprocessing techniques for edge-profile smoothing and jagged-edge rectification are described     

A nonblocking architecture for broadband multichannel switching

The paper investigates multichannel switching as a promising alternative to traditional single-channel switching where virtual paths established in a switch are between a single input channel and a single output channel. A particular non-blocking condition is derived for flip networks, which is exploited to realize a multichannel switching architecture that supports an arbitrary number of channel groups. The architecture is internally nonblocking and bufferless. Using one flip network recursively a number of times based on the number of channel groups, the resulting architecture becomes efficient in the sense that the cross point complexity is O(N log₂ N) for N inputs. Other distinguishing features are the abilities to provide multicasting, superrate switching (i.e., rates that exceed the capacity of a single channel are accommodated), multirate switching (i.e., bit pipes of different rates are supported simultaneously), multiple performance requirements (i.e., services with different performance requirements are treated accordingly), and fair access to all inputs (i.e., no input is systematically discriminated against). In multichannel switching, cells belonging to a single session can traverse multiple channels. Providing the cell sequencing integrity becomes a challenging issue. The architecture proposed in the paper accomplishes the task without employing any cell resequencing mechanism     

基于8位微控制器的参数自调整模糊控制DC/DC变换器

提出了一种基于8位微控制器实现的用于控制DC／DC变换器输出电压的参数自调整模糊控制方法。用参数自调整的策略来优化常规模糊控制器，提高了DC／DC变换器的控制性能。实验结果表明，所提出的新控制方法不仅具有良好的控制性能，而且容易用成本较低的8位微处理器来实现。     

Strategies to improve the performance of very low bit rate speech coders and application to a variable rate 1.2 kb/s codec

This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.     

Sequentially Adaptive Backward Prediction in ADPCM Speech Coders

Several questions concerning the performance in ADPCM systems of sequentially adaptive backward predictors based on the adaptive gradient and Kalman-type algorithms are addressed. Using a Jayant-type adaptive quantizer, it is shown that for bit rates less than 16 kbits/s with second order predictors and for bit rates less 18.4 kbits/s with fourth order predictors, backward-adaptive predictors have a definite performance advantage over fixed-tap predictors, since the latter may cause system divergence. For higher bit rates, the adaptive gradient predictor offers no advantage over a second order fixed-tap predictor; however, the Kalman predictor produces a substantial performance increment over the fixed-tap predictor. It is also shown that the Kalman predictor maintains a significant advantage over the adaptive gradient predictor for all bit rates from 12.8 to 32 kbits/s. Finally, it is noted that the ADPCM system divergence that occurs for fixed, multiple-tap predictors and a Jayant quantizer is caused by predictor mismatch with the input signal coupled with the infinite quantizer memory. This problem can be corrected by a modification to the quantizer adaptation logic.     

Optical network design with mixed line rates

Future telecommunication networks employing optical wavelength-division multiplexing (WDM) are expected to be increasingly heterogeneous and support a wide variety of traffic demands. Based on the nature of the demands, it may be convenient to set up lightpaths on these networks with different bit rates. Then, the network design cost could be reduced because low-bit-rate services will need less grooming (i.e., less multiplexing with other low-bit-rate services onto high-capacity wavelengths) while high-bit-rate services can be accommodated on a wavelength itself. Future optical networks may support mixed line rates (say over 10/40/100 Gbps). Since a lightpath may travel a long distance, for high bit rates, the effect of the physical impairments along a lightpath may become very significant (leading to high bit-error rate (BER)); and the signal’s maximum transmission range, which depends on the bit rate, will become limited.In this study, we propose a novel, cost-effective approach to design a mixed-line-rate (MLR) network with transmission-range (TR) constraint. By intelligent assignment of channel rates to lightpaths, based on their TR constraint, the need for signal regeneration can be minimized, and a “transparent” optical network can be designed to support all-optical end-to-end lightpaths. The design problem is formulated as an integer linear program (ILP). A heuristic algorithm is also proposed. Our results show that, with mixed line rates and maximum transmission range constraints, one can design a cost-effective network.     

Video Compression Using Conditional Replenishment and Motion Prediction

A study of a low-rate monochrome video compression system is presented in this paper. This system is a conditional-replenishment coder that uses two-dimensional Walsh-transform coding within each video frame. The conditional-replenishment algorithm works by transmitting only the portions of an image that are changing in time. This system is augmented with a motion-prediction algorithm that measures spatial displacement parameters from frame to frame, and codes the data using these parameters. A comparison is made between the conditional-replen-ishment system with, and without, the motion-prediction algorithm. Subsampling in time is used to maintain the data rate at a fixed value. Average bit rates of 1 bit/picture element (pel) to 1/16 bit/pel are considered. The resultant performance of the compression simulations is presented in terms of the average frame rates produced.     

Design and simulation of an efficient adaptive delta modulationembedded coder

An embedded coding version of hybrid companding delta modulation (HCDM) is described that operates from 16 to 48 kb/s in 8 kb/s steps. The embedded HCDM coder employs the explicit noise coding technique to transmit an adaptive PCM (APCM) coded version of the HCDM reconstruction error signal as a supplementary bit stream that may be partly or wholly deleted in transmission. SNR performance with speech input depends critically on the design of the supplemental APCM code and two new coding algorithms are investigated. In algorithm 1, the basic cue for step size adaptation is obtained from the RMS slope energy of the HCDM output whereas in algorithm 2, the HCDM reconstruction error is logarithmically compressed before quantisation and the basic step size is derived from peak input magnitudes. Instantaneous adaptation for both algorithms is achieved by using step size multipliers which are optimised for operation at single fixed bit rates and also for decoding with an unknown number of input bit deletions. Simulation results show that SNR performance is significantly enhanced using either algorithm and a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 kb/s to 16 kb/s. On the whole, the SNR performance of the embedded HCDM system is superior in comparison with conventional HCDM     

Respiratory sounds compression

Recently, with the advances in digital signal processing, compression of biomedical signals has received great attention for telemedicine applications. In this paper, an adaptive transform coding-based method for compression of respiratory and swallowing sounds is proposed. Using special characteristics of respiratory sounds, the recorded signals are divided into stationary and nonstationary portions, and two different bit allocation methods (BAMs) are designed for each portion. The method was applied to the data of 12 subjects and its performance in terms of overall signal-to-noise ratio (SNR) values was calculated at different bit rates. The performance of different quantizers was also considered and the sensitivity of the quantizers to initial conditions has been alleviated. In addition, the fuzzy clustering method was examined for classifying the signal into different numbers of clusters and investigating the performance of the adaptive BAM with increasing the number of classes. Furthermore, the effects of assigning different numbers of bits for encoding stationary and nonstationary portions of the signal were studied. The adaptive BAM with variable number of bits was found to improve the SNR values of the fixed BAM by 5 dB. Last, the possibility of removing the training part for finding the parameters of adaptive BAMs for each individual was investigated. The results indicate that it is possible to use a predefined set of BAMs for all subjects and remove the training part completely. Moreover, the method is fast enough to be implemented for real-time application.