Unlike most of the previous work for smart antennas that covered each area individually (antenna-array design, signal processing and communications algorithms and network throughput), this paper may be considered as a review of comprehensive effort on smart antennas that examines and integrates antenna array design, the development of signal processing algorithms (for angle of arrival estimation and adaptive beamforming), strategies for combating fading, and the impact on the network throughput. In particular, this study considers problems dealing with the impact of the antenna design on the network throughput. In addition, fading channels and tradeoffs between diversity combining and adaptive beamforming are examined as well as channel coding to improve the system performance.

In December 2001, the authors started two kinds of experiments on the meteor burst communication (MBC) in Antarctica to study the ability of MBC as a communication medium for data collection systems in that region. In the first experiment, a continuous tone signal is transmitted from Zhongshan Station. The signal received at Syowa Station (about 1,400 km apart) is recorded and analyzed. This experiment is aimed to study basic properties of the meteor burst channel in that high latitude region. On the other hand, the second experiment is designed to estimate data throughput of a commercial MBC system in that region. A remote station at Zhongshan Station tries to transfer data packets each consisting of 10 data words to the master station at Syowa Station. Data packets are generated with five minutes interval. In this paper, we explain the experiments, briefly examine the results of the first year (from April 2002 to March 2003), and put forward the plan for the experiments in the second and third year. From the data available thus far, we can see that 1) the sinusoidal daily variation in the meteor activity typical in middle and low latitude regions can not be clearly seen, 2) non-meteoric propagations frequently dominate the channel especially during night hours, 3) about 60% of the generated data packets are successfully transferred to the master station within two hours delay even though we are now operating the data transfer system only for five minutes in each ten minutes interval, etc.

Diagonal algebraic space time (DAST) block codes was proved to achieve the full transmit diversity over a quasi-static fading channel and to maintain 1 symbol/s/Hz. When the number of transmit antennas employed is larger than 2, DAST codes outperform the codes from orthogonal design with the equivalent spectral efficiency. However, due to the limitation on the signal constellation with complex integer points, no good 3bits/symbol DAST block code was given previously. In this paper, we propose a general form of 8-star-PSK constellations with integer points and present some theoretical results on the performance of the equivalent 8-star-PSK modulations. By using our proposed 8-star-PSKs, we present a searching algorithm to construct DAST codes with 3 bits per symbol under some criteria and investigate their performances over flat Rayleigh fading channels. It is shown that (5,2) 8-star-PSK scheme has a comparable performance to conventional 8PSK over additive white Gaussian noise (AWGN) channel and the corresponding DSAT codes constructed can achieve significant performance gain over flat Rayleigh fading channel.

In this letter, DAB system employing multiple antenna scheme is considered for high capacity transmission. In the MIMO-DAB system, the transmission rate is augmented efficiently by increasing the number of antennas, and thus the multi-channel estimation process is required. So, this letter proposes a new phase reference symbol structure which can be adapted to the multi-channel estimation, and investigates the channel estimation performance based on the semi-blind processing using the time-domain windowing. Simulation results have been shown that the MIMO-DAB system with multi-antennas can achieve the high-rate transmission for multimedia broadcasting.

In this paper, we propose a new parallel algorithm for cellular radio channel assignment problem that can help the expanded maximum neural network escape from local minima by introducing a transient chaotic neurodynamics. The goal of the channel assignment problem, which is an NP-complete problem, is to minimize the total interference between the assigned channels needed to satisfy all of the communication needs. The expanded maximum neural model always guarantees a valid solution and greatly reduces search space without a burden on the parameter-tuning. However, the model has a tendency to converge to local minima easily because it is based on the steepest descent method. By adding a negative self-feedback to expanded maximum neural network, we proposed a new parallel algorithm that introduces richer and more flexible chaotic dynamics and can prevent the network from getting stuck at local minima. After the chaotic dynamics vanishes, the proposed algorithm then is fundamentally reined by the gradient descent dynamics and usually converges to a stable equilibrium point. The proposed algorithm has the advantages of both the expanded maximum neural network and the chaotic neurodynamics. Simulations on benchmark problems demonstrate the superior performance of the proposed algorithm over other heuristics and neural network methods.

Decision-directed, pilot-symbol-aided channel estimation (PSACE) for coded orthogonal frequency division multiplexing (COFDM) systems has structurally unavoidable processing delay owing to the generation of new reference data. In a fast fading environment, the channel condition which varies during the delay induces channel estimation error. This paper proposes a method of reducing this estimation error. In this method, channel equalization is performed for the received signal twice. One is done as pre-equalization with the delayed estimates of channel frequency response in order to update them periodically. At the same moment, the other is done as post-equalization for the received signal that is delayed by the processing delay time, with the same estimates as the pre-equalization. By the proposed method, more accurate channel estimation can be realized without significant output delay. Computer simulations are performed by utilizing the IEEE 802.11a packet structure of 24 Mbit/s. The result shows that the proposed OFDM transmission scheme having the delay time of 20 µs offers 2.5 dB improvement in the required Eb/N0 at PER = 10-2 in the ESTI-BRAN model C Rayleigh fading channel with fd = 500 Hz.

Multi-Input Multi-Output (MIMO) systems, which utilize multiple antennas at both the receiver and transmitter, promise very high data rates in a rich scattering environment. It was proven in literature that with optimal power allocation, MIMO eigenmode transmission system (EMTS) is optimal because MIMO capacity is maximized. However, the performance of MIMO EMTS is very sensitive to the accuracy of channel state information and thus it is of practical importance to analyze its performance when channel state information is corrupted under realistic system and propagation conditions. In this paper, we lower bound the mutual information of MIMO EMTS with imperfect channel estimation and delayed quantized feedback in a spatially correlated continuous fading channel. Our results showed that this lower bound is tight and can serve as a comprehensive guide to the actual performance of MIMO EMTS under practical operating conditions.

A Generalized Symbol-rate-increased (GSRI) Pragmatic Trellis coded Type-I Hybrid ARQ based on a Selective-Repeat (SR) ARQ with multicopy (MC) retransmission (SR+MC scheme) for high speed mobile satellite communication system is analyzed. The SR+MC ARQ is a suitable scheme for mobile satellite systems and further improvement of the throughput performance can be expected by an additional combination of an error control coding. In this paper, we investigate the performance of the SR+MC scheme employing GSRI Pragmatic TCM. GSRI TC-MPSK can arbitrarily set the bandwidth expansion ratio keeping higher coding gain than conventional TCM scheme. Also Pragmatic TCM has an advantage in that the modulation level can be easily changeable. By changing the modulation level and the bandwidth expansion ratio, this scheme can optimize the performance according to the channel conditions. Numerical and simulation results show that the GSRI Trellis Coded Type-I Hybrid ARQ presents better performance than conventional Pragmatic Trellis Coded Type-I Hybrid ARQ.

A new scheme for watermarking based on vector quantization (VQ) over a binary symmetric channel is proposed. By optimizing VQ indices with genetic algorithm, simulation results not only demonstrate effective transmission of watermarked image, but also reveal the robustness of the extracted watermark.

An adaptive filtering algorithm based on the sliding window criterion is known to be very attractive for violent changing environments. In this paper, a new sliding window linearly constrained recursive least squares (SW-LC-RLS) algorithm based on the modified minimum mean squared error (MMSE) structure is devised for the RAKE receiver in direct sequence spread spectrum code-division multiple access (DS-SS CDMA) system over multipath fading channels, where the channel estimation scheme is accomplished at the output of adaptive filter. The proposed SW-LC-RLS algorithm has the advantage of having faster convergence property and tracking ability, and can be applied to the environments, where the narrowband interference is joined suddenly to the system, to achieve desired performance. Via computer simulation, we show that the performance, in terms of mean square errors (MSE), signal to interference plus noise ratio (SINR) and bit error rate (BER), is superior to the conventional LC-RLS and orthogonal decomposition-based LMS algorithms based on the MMSE structure.

This paper proposes an analytical model to demonstrate the benefit of data service in wireless networks using dynamic multi-channel scheme with channel de-allocation. The performance of a system providing buffers to voice calls to reduce the raised voice blocking probability caused by data contention is investigated. The effect of the cell dwell time and overlap area with adjacent cells on system performance are studied. All free channels are allocated to data users dynamically. For those data users using more than one channel, channels would be de-allocated for new requests, voice or data. Buffers are provided for voice calls to reduce the voice blocking probability caused by data packets contention. Handoff calls are given priority to be queued in the front of the buffer instead of providing guard channels to reduce their dropping probability. Meanwhile, the reneging time for new calls and the handoff dwell time for handoff calls are considered in our analysis to obtain an appropriate amount of buffer to voice. To compensate the blocking probability in data, guard channels are provided for data traffic. Numerical results show that the dynamic multi-channel scheme with possible de-allocation, compared with the single channel scheme, can enhance data traffic performance significantly in terms of the mean transmission time and blocking probability. A system providing an appropriate amount of buffer to voice traffic and giving priority to queued handoff calls can indeed reduce new call blocking probability and handoff call dropping probability. In addition, the proposed scheme can reduce the incomplete transmission probability of data packets.

In this paper, we analyzed the performance of dynamic resource allocation with channel de-allocation and buffering in cellular networks. Buffers are applied for data traffic to reduce the packet loss probability while channel de-allocation is exploited to reduce the voice blocking probability. The results show that while buffering data traffic can reduce the packet loss probability, it has negative impact on the voice performance even if channel de-allocation is exploited. Although the voice blocking probability can be reduced with large slot capacity, the improvement decreases as the slot capacity increases. On the contrary, the packet loss probability increases as the slot capacity increases. In addition to the mean value analysis, the delay distribution and the 95% delay of data packets are provided.

We consider vector channel estimation for differentially modulated systems with transmit diversity. We propose a blind estimation scheme to yield superior performance to that of the pilot-aided estimation. The proposed scheme utilizes the repeated transmission of space-time coded data symbols. We compare the performance of the proposed blind scheme with the pilot-based estimation scheme via computer simulations.

The relative window method provides quantitative crosstalk delay degradation for the post-layout timing analysis in deep sub-micron VLSI design. However, to the best of our knowledge, the relative window method has not been applied to the crosstalk minimization in gridded channel routing problem. Most conventional crosstalk optimizers only use the coupling length to estimate the crosstalk. In this paper, we present a post-layout timing driven crosstalk optimizer based on the relative window method. According to the relative signal arrival time and the coupling length, we define a delay degradation graph to describe the crosstalks between nets in a routing solution. Our optimization goal is to maximize the time slack by iteratively improving the delay degradation graph without increasing the channel height. Benchmark data consistently show that our post-layout timing driven crosstalk optimizer can further improve the routing solution obtained by a conventional crosstalk optimizer.

Recently, Muramatsu proposed source coding algorithms that use the randomness of a past sequence. The technique of his source coding algorithms is one method of constructing codes from the technique of random coding. By using his technique, we propose a channel coding algorithm with random numbers which can be observed by both the encoder and the decoder where the random numbers are independent of the messages to be transmitted. Then the proposed coding algorithm can transmit messages over a discrete memoryless channel up to the channel capacity with an arbitrarily small decoding error rate and arbitrarily small bits of random numbers per message transmission asymptotically.

We consider a blind estimation of the vector channel for systems with receive diversity. The objective of this paper is to reduce the complexity of the conventional subspace-based method in vector channel estimation. A reduced-complexity estimation scheme is proposed, which is based on selecting a column of the covariance matrix of the received signal vectors. The complexity and performance of the proposed scheme is investigated via computer simulations.

The accuracy of channel estimation significantly affects the performance of coherent rake receiver in DS-CDMA systems. It is desirable for improved channel estimation to employ a channel estimation filter (CEF) whose bandwidth is adjustable to the channel condition. In this paper, we consider the use of moving average (MA) FIR filters as the CEF since it is simple to implement and can provide relatively good receiver performance. First, we optimize the tap size of the MA FIR CEF so as to minimize the mean squared error of the estimated channel impulse response. For practical applications, we propose a low-complexity adaptive channel estimator (ACE), where the tap size of the MA FIR CEF is adjusted based on the estimated channel condition by exploiting the correlation characteristics of the received pilot signal. Numerical results show that the use of the proposed ACE can provide the receiver performance comparable to that of Wiener CEF without exact a priori information on the operating condition.

We have proposed a new low-IF transceiver architecture to simultaneously achieve both a small chip area and good minimum input sensitivity. The distinctive point of the receiver architecture is that we replace the complicated high-order analog filter for channel selection with the combination of a simple low-order analog filter and a sharp digital band-pass filter. We also proposed a high-speed convergence AGC (automatic gain controller) and a demodulation block to realize the proposed digital architecture. For the transceiver, we further reduce the chip area by applying a new form of direct modulation for the VCO. Since conventional VCO direct modulation tends to suffer from variation of the modulation index with frequency, we have developed a new compensation technique that minimizes this variation, and designed the low-phase noise VCO with a new biasing method to achieve large PSRR (power-supply rejection ratio) for oscillation frequency. The test chip was fabricated in 0.35-µm BiCMOS. The chip size was 3 3 mm2; this very small area was realized by the advantages of the proposed transceiver architecture. The transceiver also achieved good minimum input sensitivity of -85 dBm and showed interference performance that satisfied the requirements of the Bluetooth standard.

This paper proposes a new method, EC-MDC that can detect and correct bit errors in the bitstream generated by multiple description coding. The proposed method generates two sub-bitstreams having a few redundancies as much as conventional multiple description coding. If a sub-bitstream at one side has bit error, the bit error can be corrected by using sub-bitstream of the other side. In BER-SNR experiments, reconstruction quality of the proposed method shows about 11dB higher than that of the conventional MDC at BER < 10-3 when a sub-bitstream is corrupted.

In this paper, we proposed a novel two-stage channel estimation (2S-CE) method. In contrast to conventional channel estimation methods, this method makes the maximum use of the information contributed by the known data in every transmission burst. In the first stage, the least-squares (LS) algorithm was used to estimate the channel impulse response (CIR) based on the normal training sequence. Then the maximum channel memory was estimated and used to locate the uncorrupted data in the guard interval. In the second stage, the uncorrupted data together with the normal training sequence were sent to the LS algorithm again to obtain the fine-tuned CIR. To verify the efficiency of the proposed 2S-CE method, both a theoretical analysis and computer simulations have been done. Computer simulation results confirm the analysis results and demonstrate that the proposed 2S-CE method outperforms a conventional single-stage channel estimation method.