This paper proposes an FFT (Fast Fourier Transform) interference detection for interference suppression which combines notch filtering and FEC (forward error correction) to improve the Pe (probability of error) performance degradation due to co-channel interference in digital satellite communication systems. The proposed FFT interference detection scheme can determine the co-channel interference carrier frequency, power, and bandwidth precisely by using the power detection threshold suitable for the desired signal power spectrum, and the notch filter characteristic can be set according to the results. The interference suppression with the proposed scheme achieves the degradation in required Eb/No to only 1.0 dB at a Pe of 10-4 compared to that with the optimum notch filter (ideal detection) in unknown CW (continuous wave) and FM (frequency modulation) co-channel interference environments. Moreover, the proposed scheme improves the required Eb/No by 6.5 dB compared to that without a notch filter in an FM interference environment with interference carrier frequency offset normalized by the desired signal clock rate of 0.52, desired to undesired (interference) signal power ratio of 3 dB and interference bandwidth at 10 dB down power point from the peak normalized by the desired signal clock rate of 0.25.

This paper describes the performance of a video communication system over mobile radio channels. Mobile channel quality changes rapidly due to various factors. When compressed video data is transmitted through these channels, it is indispensable to employ an error control scheme because reconstructed video quality is seriously degraded by channel error. To control this error, an automatic repeat request (ARQ) scheme is often employed, however, this incurs a cost. The benefit of a non-degraded reconstructed video sequence is offset by the transmission delay due to ARQ retransmission. We apply to a video communication system a selective-repeat ARQ which is combined with the coding control scheme to reduce the transmission delay. We evaluate the quality of the reconstructed video sequence and transmission delay using computer simulations and make clear its applicability over Rayleigh and Nakagami-Rican fading channels and intersymbol interference.

In this paper, the evaluation of a hybrid acquisition performance has been considered for the pilot signal in direct sequence code division multiple access (DS/CDMA) forward link. The hybrid acquisition is introduced by the combination of two schemes, parallel and serial acquisitions. The mean acquisition time of the hybrid acquisition scheme is derived to consider both case 1 (the correct code-phase offsets ae included in one subset) and case 2 (the correct code-phase offsets exist at the boundary of two subsets), which are caused by the distribution of the correct code-phase offsets between two subsets. Detection, false alarm, and miss probabilities are derived for the cases of multiple correct code-phase offsets and multipath Rayleigh fading channel. Results are provided for the acquisition performance with respect to system design parameters such as postdetection integration length in the search and verification modes, subset size, and number of I/Q noncoherent correlators. Also, comparision between hybrid acquisition and parallel acquisition under the same hardware complexity is provided in terms of the minimum mean acquisition time.

The performances of an SET required for integration are discussed. Conventional SETs had several problems such as large leakage current, insufficient voltage gain and so on. To overcome these problems, a new SET utilizing Schottky barriers as tunnel junctions is proposed. Its current characteristics and Coulomb-blockade conditions are calculated and the effectiveness for an integrated device is discussed.

In this paper, we propose a quasi-optimum multiuser detector using co-channel interference cancellation technique in an asynchronous code-division multiple-access communication system, and evaluate its performance by computer simulations. In the proposed detector, maximum likelihood sequence estimation is performed to compare the original received signal with replicas of the signal which are produced from the demodulation data bit sequence of a co-channel interference canceller. In several conditions, the proposed detector is compared with the co-channel interference canceller, and it is shown that the average bit error rate characteristics of the propose detector are improved considerably.

Applying ARQ to real time video communication can significantly increase transmission delay due its retransmission operations. We analyze this delay and propose an adaptive error control scheme that uses acknowledgment from the receiver to reduce the delay. We evaluate this scheme using a computer simulation and show that the proposed scheme can reduce the delay by controlling the amount of video data by changing the quantization step size and video frame skipping. It also offers acceptable video quality as confirmed by a subjective evaluation test.

It is highly desirable to develop an efficient and flexible dynamic channel assignment algorithm in order to realize an integrated traffic TDMA mobile radio communication network. In this paper, an integrated traffic TDMA system is studied in which transmission of voice and data are assumed to occupy one and n time slots in each TDMA frame, respectively. In general, there are two types of channel (time slot) assignment algorithms: the partitioning algorithm and the sharing algorithm. However, they are not well-suited to the multimedia traffic consisting of various information sources that occupy different number of slots per frame. In this paper, assuming that voice is much more sensitive to transmission delay than data, an algorithm based on the sharing algorithm with flexible tima slot management scheme is proposed. Our method tries to vary the number of data slots adaptively so as to improve the quality of servive of voice calls and the system capacity. Computer simulations show the good performance of the proposed algorithm when compared to conventional channel assignment algorithms.

Adaptive maximum likelihood (ML) detection implemented by the Viterbi algorithm (VA) is proposed for the reception of MPSK signals in frequency nonselective fast Rayleigh fading. M-state VA, each state in the VA trellis represents a signal constellation point, is used. Diversity reception is incorporated into the structure of Viterbi decoding. The pilot symbol (unmodulated carrier) is periodically inserted to terminate the trellis so that the phase ambiguity of the detected data sequence is avoided. Applying the per-survivor processing principle (PSPP), adaptive ML detection performs adaptive channel estimation using a simple linear predictor at all trellis states in parallel. The predictor coefficient is stochastically adapted without requiring a priori knowledge of fading channel statistics, based on a recursive least-squares (RLS) adaptation algorithm, to match changes in the statistical properties of the channel (i.e., AWGN of fast/slow fading) using both data and pilot symbols. Simulations of 4PSK signal transmission demonstrate that the proposed adaptive ML detection scheme can track fast fading, thus significantly reducing the irreducible bit error rate (BER) due to Doppler spread in the fading channel. It is also shown that adaptive ML detection provides BER performance close to ideal coherent detection (CD) in AWGN channels.

A steady-state current-voltage characteristics of fully-depleted surrounding gate transistor (FD-SGT) with short channel effects, such as threshold voltage lowering and channel length modulation, is analyzed. First, new threshold voltage model of FD-SGT, which takes threshold voltage lowering caused by decreasing channel length into consideration, are proposed. We express surface potential as capacitance couple between channel and other electrodes such as gate, source and drain. And we analyze how surface potential distribution deviates from long channel surface potential distribution with source and drain effects when channel length becomes short. Next, by using newly proposed model, current-voltage characteristics equation with short channel effects is analytically formulated for the first time. In comparison with a three-dimensional (3D) device simulator, the results of newly proposed threshold voltage model show good agreement within 0.011 V average error. And newly formulated current-voltage characteristics equation also shows good agreement within 0.95% average error. The results of this work make it possible to clear the device designs of FD-SGT theoretically and show the new viewpoints for future ULSI's with SGT.

Dynamic Channel Assignment (DCA), which improves the efficiency of channel use in cellular mobile communication systems, requires finding an available channel for a new call after the call origination. This causes the delay which is defined as the time elapsing between call origination and completion of the channel search. For system planning, it is important to evaluate the delay characteristic of DCA because the delay corresponds to the waiting time of a call and influences service quality. It is, however, difficult to theoretically analyze the delay characteristics except its worst case behavior. The time delay of DCA has not been theoretically analyzed. The objective of this paper is analyzing the distribution and the mean value of the delay theoretically. The theoretical techniques in this paper are based on the techniques for analyzing the blocking rate performance of DCA.

The demand for data communication over Personal Handy-phone System (PHS) is expected to rapidly increase in the near future. Some applications based on the circuit-switched services have been recently developed. However, the packet-switched service is better than the circuit-switched service for personal data communications in terms of the flexible utilization of radio resources. In this paper, we propose PHS with packet data communications system (PHS-PD), which has four system concepts; (i) to supprot the Internet access, (ii) to realize compatibility with circuit-switching services, (iii) to share the common radio channels with circuit-switched calls, and (iv) to utilize idle time slots for packet data. Moreover, a novel packet channel structure for sharing radio resources with circuit-switched calls is introduced. Although packet data are transferred using common radio resources, the proposed channel structure prevents any degradation in call loss performance of the circuit-switching service. An evaluation of the maximum packet transmission rate shows that PHS-PD can offer a data communication rate of 20.1 kbps even if circuit-switched calls are in progress. Furthermore, up to 83.6 kbps is possible if circuit-switched calls are quiescent. It is also shown that enough capacity for a practical e-mail service can be ensured by PHS-PD even if the degradation of throughput performance due to packet collisions on random access channels is considered.

This paper proposes an adaptive coding rate and process gain control technique with channel activation function to realize a CDMA based radio subsystem for multi-media communication services that include two types of media, i.e., fixed size data such as the computer data and still image, and constant bit rate data such as voice and video. The proposed system achieves high throughput data transmission for the fixed size data by controlling the process gain and coding rate according to the variation of the channel. Moreover, to adopt the constant bit rate data, the proposed system also employs a channel activation technique. Computer simulation confirms that the proposed system is very effective for multi-media communication services.

We propose two multipriority reservation protocols for wavelength division multiplexing (WDM) networks. The network architecture is a single-hop with control channel-based passive star topology. Each station is equipped with two pairs of laser and filter. One pair of laser and filter is always tuned to wavelength λ0 for control and the other pair of laser and filter can be tuned to any of data wavelengths, λ1, λ2, ..., λN. According to the access methods of the control channel, one protocol is called slotted ALOHA-based protocol and the other protocol is called TDM-based protocol. The two protocols have the following properties. First, each of them has its own priority control scheme which easily accommodates multipriority traffics. Second, they can be employed in the network with limited channels, i.e. the number of stations in the system is not restricted by the number of data channels. Third, they are conflict-free protocols. By using a reservation scheme and a distributed arbitration algorithm, channel collision and destination conflict can be avoided. For the performance point of view, the TDM-based protocol gives an optimal solution for the priority control. However it is less scalable than the slotted ALOHA-based protocol. The slotted ALOHA-based protocol also performs good priority control even though it is not an optimal solution. We analyze their performances using a discrete time Markov model and verify the results by simulation.

Many efforts are currently underway to design wideband mobile communication systems. In this letter, we clarify the received signal level characteristics for wideband mobile radio channels in line-of-sight (LOS) microcells. We conduct several urban-area field experiments to measure the received signal levels for various receiver bandwidths from 300 kHz to 30 MHz and the power delay profile. The experimental results show that the fading depth of the received signal decreases as the normalized rms delay spread, defined as the product of receiver bandwidth and rms delay spread, increases. These results are useful in designing wideband microcell systems for urban areas.

We have investigated the BER performance of TC 8PSK with 2 branch SC and MRC diversities on spatially correlated Rayleigh fading channel. The upper bounds using the transfer function bounding technique are derived several numerical results are shown. Although the correlation between branches causes signal-to-noise (SNR) loss (relative to uncorrelated fading case) for SC and MRC diversities, the diversity can lead to achieve the diversity gain compared to the system without diversity. It is found that the diversity gain of 4-state TC 8PSK is larger than 8-state TC 8PSK. It is also shown that the BER performance of TC 8PSK is decreased as the antenna separation is decreased.

A Reed-Solomon coded Type-I Hybrid ARQ scheme based on a Selective-Repeat (SR) ARQ with multicopy retransmission is proposed for mobile/personal satellite communication systems of a transmitter and a receiver both with the finite buffer. The performance of the proposed scheme on fading channels is analyzed. The basic idea of the strategy is the use of two modes; the SR mode and the multicopy mode. In the latter mode, erroneous blocks stored in the transmitter buffer are alternatively retransmitted multiple times when ν consecutive retransmissions in the SR mode are received in error. Numerical and simulation results for ν1 show that the proposed scheme presents better performance than the conventional SR+ST scheme 2 of the 2N block buffer by Miller and Lin.

In this paper, we evaluate the throughput performance of CDMA Slotted ALOHA systems. To improve the throughput performance, we employ the Quasi-synchronous sequences and the Modified Channel Load Sensing Protocol as an access control procedure. As a result, we found a good throughput by the QS-sequences. By employing MCLSP, we can keep the maximum throughput even in high offered load and in the presence of a long access timing delay, which is one of the issue in satellite packet communication systems.

New interference cancellation technique using time division reference signal is proposed for optical synchronous code-division multiple-access (CDMA) systems with modified prime sequence codes. In the proposed system one user in each group is not allowed to access the network at each time, and this unallowable user's channel is used as a reference signal for other users in the same group at the time. The performance of the proposed system using an avalanche photodiode (APD) is analyzed where the Gaussian approximation of the APD output is employed and the effects of APD noise, thermal noise, and interference for the receiver are included. The proposed cancellation techniqus is shown to be effective to improve the bit error probability performance and to alleviate the error floor when the number of users and the received optical power are not appreciably small.

In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in the Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. By subtracting the estimated short-time noise spectrum from the input spectrum, not only the mean value of the noise spectrum but also the component varying around the mean value can be reduced. This method can be interpreted as a partial construction of the adaptive beamformer where only the amplitude of the short-time noise spectrum is estimated, while the adaptive beamformer is equivalent to the estimator of the complex short-time noise spectrum. By limiting the estimation to the amplitude spectrum, the proposed system achieves better performance than the adaptive beamformer in the case when the number of sound sources exceeds the number of microphones.

Two simple and high performance multichannel distributed queue dual bus (MDQDB) protocols based on the wavelength division multiplexing (WDM) network technology are proposed, and the network architecture and operations are presented. To be suited for a high speed network, they inherit the advantages of the DQDB protocol such as node simplicity, network flexibility and distributed operations of individual nodes. The network capacity can also be greatly increased with marginal increase of node complexity. Simulation has been done to estimate the performances such as throughput and average access delays for individual nodes. The influence of the bandwidth balancing mechanism on the two protocols is considered at medium, high, and overload conditions. We also investigate the fairness characteristics in asymptotic conditions for various initial states.