Broadband ISDN, using asynchronous transfer mode, are expected to carry traffic of different classes, each with its own set of traffic characteristics and performance requirements. To achieve the quality of service in ATM networks, a suitable buffer management scheme is needed. In this paper, we propose a buffer management scheme using a priority service discipline to improve the delay time of delay-sensitive class and the packet loss ratio of loss-sensitive class. The proposed priority scheme requires simple buffer management logic and minor processing overhead. We also analyze the delay time and the packet loss ratio for each class of service. The results indicate that the required buffer size of the proposed priority scheme is reduced and the delay time of each class of service is controlled by a parameter. If the control parameter is appropriately chosen, the quality of service of each class is improved.

Several research institutions in Europe have developed set-ups for wide-band mobile radio communication measurements. The performance and evaluation has been coordinated in the framework of the cooperation in the field of scientific and technical research within the committees COST 207 COST 231. New parameters have been defined to improve the insight into performance limits of digital radio communication systems which are caused by propagation phenomena. The definitions of these new parameters are presented in the paper. Channel sounders developed in Norway, Denmark, the United Kingdom, Switzerland, France and Germany are described. They are based on considerably different technical principles for evaluation and recording of the measured results. A few results gained in European measurement campaigns are also presented.

A new type of analog-to-digital (A/D) converter is introduced. The structure is based on a magnitude-preserving quadrature mirror filter (QMF) bank where the analysis bank is composed on IIR switched-capacitor (SC) filters. The analog output samples of the analysis filters are converted into digital form using individual A/D converters and combined by an IIR digital filter synthesis filter bank. This A/D converter is useful in applications where only the magnitude of the spectrum of the analog signal needs to be preserved. The structure incorporates the advantages of sub-band coding and reduces considerably the effect of mismatches among the sub-band A/D converters. In addition, the proposed scheme leads to an increase in the conversion speed by a factor of M when an M-channel QMF bank is used. An illustrative example verifying the good performance of the proposed approach is included.

A multicarrier modulation is considered as an effective technique in high speed digital transmission under the multipath fading. In this paper, we theoretically analyze the bit error rate (BER) performance of the multicarrier modulation/differential detection scheme, and show the trade-offs between the BERs and the number of carriers or the guard period to clarify the optimum values to minimize the BER in the number of carriers and the guard period.

A field trial was conducted to evaluate the technical performance of land mobile message communication in different environments. The OmniTRACS system and the Ku-band JCSAT satellite were utilized as the mobile communications system and the satellite, respectively. The trial took place in September 1990 at different areas in Japan. Data collected correspond to about 65 hours of operation, during which a large number of messages were sent via the satellite. Two land mobile terminals operated simultaneously, each terminal having a function of generating messages automatically which simulates a large volume of traffic corresponding to about 50 terminals. Thus, the system was evaluated under the condition that 100 mobile terminals were in operation. Obtained data have been analyzed with a particular focus on the message transmission correlating with actual environments. The analysis was done by classifying environments into five categories: overall condition, type of roads, terrain, areas and weather conditions. The average transmission count per message experienced under all conditions is equal to 1.432 for forward messages transmitted from the hub station to mobiles, and 1.157 for return messages transmitted from mobiles to the hub station. With respect to the classification by the type of roads, for enample it becomes obvious that the performance is generally good except along roads of North-South orientation through dense urban areas. It is concluded that the message communications from/to mobiles are feasible in a wide range of environments, with the performance of success essentially depending on the visibility of satellite.

This paper proposes a new fault tolerant intercommunication scheme for real-time operations and three new interconnection networks to construct a fault tolerant multi-processor system for pipeline processings. The proposed intercommunication scheme using bank memory switching technique has an advantage to make a fault tolerant pipeline system so that it can detect any failure caused in a processing element of the system. In addition, it can overcome conventional problems caused in interconnection circuits to flow data with one way direction such as a pipeline processing.

An acoustic echo canceller that also cancels room noise is proposed. This system has an additive (noise reference) input port, and a noise canceller (NC) precedes the echo canceller (EC) in a cascade configuration. The adaptation control problem for the cascaded echo and noise canceller is solved by controlling the adaptation process to match the occurrence of intermittent speech/echo; the room noise is a stationary signal. A simulation shows that adaptation using the NLMS algorithm is very effective for the echo and noise cancellation. Sub-band cancelling techniques are utilized. Noise cancellation is realized with a lower band EC. Hardware is implemented and its performance evaluated through experiments under a real acoustic field. The combination of the EC with NC maintains excellent performance at all echo to room noise power ratios. It is shown that the proposed canceller overcomes the disadvantages traditionally associated with ECs and NSc.

The realization of acoustic inverse filter is often difficult because of the non-minimum phase property and the long time duration of the impulse response of the acoustic enclosure. However, if the signals are divided into a large number of sub-bands, many of the sub-bands are found to be invertible. The invertibility of a sub-band signal depends on the zero distribution of the transfer function in the z-plane. In a multi-microphone system, the transfer functions between the sound source and the mirophones have different zero distributions. The method proposed here, taking advantage of the differences of zero distributions, selects the best invertible microphone in each sub-band, and reconstructs the full band signal by summing up the inverse filtered sub-band signals of the best microphones. The quality of the dereverberated signal using the proposed inverse filtering approach is improved with increasing number of microphones and sub-bands. When seven microphones are used and the number of sub-bands is 513, the quality of the dereverberated speech signals are almost the same with the original ones even when the revergeration time is about one second. The introduction of multi-microphones in addition to sub-band processing provides a new way of dealing with the non-minimum phase problem in deconvolution.

The effectiveness of traffic shaping for VBR traffic is analyzed. Evaluation results prove that traffic shaping can improve link efficiency for most forms of bursty VBR traffic and that link efficiency gains of more than 250% can be expected without the shaping delay imposing any significant QOS deterioration. Traffic shaping increases the link efficiency to about 80% for traffic with short burst repetition periods. The traffic shaping techniques and analytical results described herein can be employed in the traffic management of future B-ISDN/ATM networks.

This paper deals with a network architecture based on a backbone network, using ATM switches (ATM-SW) and ATM Cross-Connect Systems (ATM-XC). The backbone network is efficiently utilized by multiple-routing scheme. The performance of the network is controlled, exploiting the concept of Virtual Paths (VP) in ATM technology. The network is controlled by allocating the bandwidth of VPs so as to minimize the worst call blocking probability of all ATM-SW pairs, under the constraints of the ATM-SW capacities and the bandwidths of transmission paths in the backbone network. To improve network performance, we use a trunk reservation scheme among service classes. We propose a heuristic approach to solve the problem of non-linear integer programming. Evaluation of the proposed optimization scheme, in comparison to other optimal methods, shows the efficiency of the present scheme.

A Chebyshev type bandpass filter using four TM01δ-mode dielectric rod resonators oriented axially in a high-Tc superconductor cylinder is designed with 3 dB bandwidth 36 MHz at 11.958 GHz. The single resonator which contains a Ba (MgTa) O3 ceramic rod of εγ=24 and a YBa2Cu3Oy bulk cylinder is designed to realize temperature coefficient of f0, τf=0 ppm/K at 20 K. The unloaded Q, Qu measured at 20 K is 150,000 which is higher than Qu=100,000 for a TM01δ-mode resonator with a copper cylinder. When the constructed filter is cooled from room temperature to below 50 K, the center frequency shifted only 5 MHz which corresponds to τf=1.5 ppm/K and the insertion loss IL0 at the center freqency reduced from 3.0 dB to about 0 dB, the designed value of which is 0.04 dB, which is too small to be measured accurately.

This paper analyzes the performance of an ATM cell multiplexer with a two level MMPP input on a discrete-time basis. We approximated the input process as a simple MMPP model. We developed an MMPP/D/1/K queueing model for the ATM cell multiplexer, and employed an analytic approach for the evaluation of cell loss probability. We verified the accuracy of the results using computer simulation. We applied the above analytic method to connection admission control (CAC) of the ATM network. The resulting connection admission control scheme employs the concept of the "effective bandwidth" and table-look-up procedure. We confirmed through a computer simulation that the proposed connection admission control scheme outperforms the peak bandwidth allocation scheme with respect to link utilization.

This paper discusses the bit-rate compression of super high definition still images with subband coding. Super high definition (SHD) images with more than 20482048 pixels or resolution are introduced as the next generation imaging system beyond HDTV. In order to develop bit-rate reduction algorithms, an image evaluation system for super high definition images is assembled. Signal characteristics are evaluated and the optimum subband analysis/synthesis system for the SHD images is clarified. A scalar quantization combined with run-length and Huffman coding is introduced as a conventional subband coding algorithm, and its coding performance is evaluated for SHD images. Finally, new coding algorithms based on block Huffman coding and entropy coded vector quantization are proposed. SNR improvement of 0.5 dB and 1.0 dB can be achieved with the proposed block Huffman coding and the vector quantization algorithm, respectively.

An integrated multiplexer in intermediate node is analyzed. The multiplexer is modeled as a system with multiple synchronous servers (channels) and having two kinds of customers. Between the two, one is wideband (WB) and the other is narrowband (NB); they are queueable with the same deterministic service time. The WB customer is given higher priority of channel access than the NB. To incorporate the delay constraint of WB, we use a simple instant discarding scheme for WB. As a result, the system states defined just after the beginning of a slot form an one-dimensional embedded Markov chain. This makes the analysis computationally tractable. The performance measures such as queue length distribution, average blocking probability, and average waiting time are obtained, particularly, the waiting time distribution. Some interesting numerical examples are discussed. Simulation results are also provided to help verify the validity of analysis.

A novel coding scheme using orthonormal wavelet transform is proposed. Various forms of transform coding and subband coding are first reviewed. Then a wavelet coding method is proposed adopting a new approach similar to the one used for transform coding. The approach differs to conventional ones which considers wavelet coding as a class of subband coding. Simulation work is carried out to evaluate the proposed coding method. Significant improvement is obtained in subjective quality, and some improvement is also obtained in signal to noise ratio. Wavelet coding is still in its early stage of development, but can be considered to be a promising technique for image coding.

One of the problems with subband image coding is the increase in image sizes caused by filtering. To solve this, it has been proposed to process the filtering by transforming input sequence into a periodic one. Then filtering is implemented by circular convolution. Although this technique solves the problem, there are very strong restrictions, i.e., limitation on the filter type and on the filter bank structure. In this paper, development of this technique is presented. Consequently, any type of linear phase FIR filter and any structure of filter bank can be used.

This paper discusses and evaluates, from the viewpoints of definition, analysis, and performance, frequency assignment schemes that enable the efficient assignment of multiple-bandwidth carriers on the transponder in SCPC/FDMA systems with demand assignment operation. The system considered handles carriers of two different bandwidths, and assigns only consecutive slots on the transponder band to broadband carriers. Three types of frequency assignment schemes are proposed, each of which incorporates one or both of two assignment concepts: (1) pre-establishment of assignment priorities on the transponder band, and (2) establishment of broadband slots to guide broadband carrier assignment. Following a definition of the schemes, equations are derived to theoretically analyze performance factors such as call loss for the narrowband and broadband carriers, and system utilization efficiency. Finally, theoretical performance calculated for various traffic and system conditions are presented and evaluated, for the purpose of comparison between the three schemes. Computer simulation results are also presented, to demonstrate the accuracy of the derived equations and to supply data for models too large for theoretical computation. Main results obtained are as follows. (1) Regardless of traffic or system conditions, the assignment scheme incorporating both assignment priorities and broadband slots shows the best performance in terms of broadband call loss and system utilization efficiency. (2) The establishment of broadband slots improves performance when the ratio of broadband traffic to the total traffic volume is high, but worsens performance when the narrowband traffic ratio is higher. (3) All aspects of performance improve with the increase of the total number of assignable slots on the transponder band.

A new error rate formula for narrowband Differential Quaternary Phase Shift Keyed system in a Rayleigh fading channel is obtained in closed-form. The formula predicts a non-zero error probability for noiseless reception. As predicted, the computed error rates approach some constant or floor values as the signal-to-noise ratio is increased beyond a certain limit. In the presence of various Doppler frequency shifts, an IF filter bandwidth of about one times the symbol rate is found to lead to a minimum error probability prior to the appearence of the error rate floor.

Very high resolution images with more than 2,000*2.000 pels will play a very important role in a wide variety of applications of future multimedia communications ranging from electronic publishing to broadcasting. To make communication of very high resolution images practicable, we need to develop image coding techniques that can compress very high resolution images efficiently. Taking the channel capacity limitation of the future communication into consideration, the requisite compression ratio will be estimated to be at least 1/10 to 1/20 for color signals. Among existing image coding techniques, the sub-band coding technique is one of the most suitable techniques. With its applications to high-fidelity compression of very high resolution images, one of the major problem is how to encode high frequency sub-band signals. High frequency sub-band signals are well modeled as having approximately memoryless probability distribution, and hence the best way to solve this problem is to improve the quantization of high frequency sub-band signals. From the standpoint stated above, the work herein first compares three different scalor quantization schemes and improved permutation codes, which the authors have previously developed extending the concept of permutation codes, from the aspect of quantization performance for a memoryless probability distribution that well approximates the real statistical properties of high frequency sub-band signals, and thus demonstrates that at low coding rates improved permutation codes outperform the other scalor quatization schemes and that its superiority decreases as its coding rate increases. Moreover, from the results stated above, the work herein, develops a rate-adaptive quantization technique where the number of bits assigned to each subblock is determined according to the signal variance within the subblock and the proper quantization scheme is chosen from among different types of quantization schemes according to the allocated number of bits, and applies it to the high-fidelity encoding of sub-band signals of very high resolution images to demonstrate its usefulness.

This paper describes the configuration and performance of a stable, high compression video coding scheme suitable for broadcast quality. This scheme was developed for application to high quality image packet transmission in Asynchronous Transfer Mode (ATM) networks. There are two problems in implementing image packet transmission in ATM networks, namely the achievement of a compression scheme with high coding efficiency, and the achievement of an effective compensation method for cell loss. We describe a scheme which resolves both these problems. It comprises the division of a two-dimensional spectral image signal into several sub-bands. In the case of the high frequency band, block-matching interframe prediction and Discrete Cosine Transform (DCT) are applied to achieve high compression ratio, while intraframe DCT coding is applied to the baseband. This scheme, moreover, provides a stable compensation for cell loss. It is shown that, based on this system, an original image signal of 216Mbit/s is compressed to about 1/10, and a high quality reconstructed image stable to cell loss is obtained.