Wireless network systems introducing both of the cellular concept and the ad-hoc concept have been proposed. Communication between two nodes in a cell is guaranteed by relaying capability of the base station in these systems. Additionally, two nodes can directly communicate with each other while they are close to each other. We call this type of network a two-hop wireless network. The teletraffic performance of this network depends on various parameters such as the size of a cell, location of nodes, the communication range of nodes, channel assignment schemes, teletraffic behavior and so on. The purpose of this paper is to theoretically analyze the teletraffic performance of the network, which has been evaluated by computer simulation, by introducing a simple model. This paper shows a technique to analyze the performance in this model. Also, this paper considers the range in which the two-hop wireless network works well for the efficient use of channels.

This paper discusses self-similarity in cell dwell time of a mobile terminal, the discovery of which was described in our previous paper, and its effects on teletraffic of mobile communication networks. We have evaluated various teletraffic statistics, such as cell dwell time and channel occupancy time, of a mobile terminal based on measurements of motion for various types of vehicles. Those results show that cell dwell time follows a long-tailed log-normal distribution rather than the exponential distribution that has been used for modeling. Here, we first elaborate on self-similarity in cell dwell time of various vehicles. We then evaluate self-similarity in channel occupancy time. For future mobile multimedia communication systems employing a micro-cell configuration, it is anticipated that data communication will be the main form of communication and that call holding time will be long. For such cases, we have shown that channel occupancy time will be greatly affected by the cell dwell time of the mobile terminal, and that self-similarity, a characteristic that is not seen in conventional systems, will consequently appear. We have also found that hand-off frequently fails as self-similarity in cell dwell time of a mobile terminal becomes stronger.

Call demand suddenly and greatly increases in the aftermath of a major disaster, because people want to check on their families and friends in the stricken area. Many call attempts in mobile cellular systems are blocked due to the limited radio frequency resources. In this paper, as a solution to this problem, limiting the holding time of calls is investigated and a dynamic holding time limit (DHTL) method, which varies the holding time limit dynamically based on the number of call attempts, is proposed. The effect of limiting the holding time is investigated first using a computer simulation with a constant and heavy traffic load model. This simulation shows that the average holding time of calls is decreased as the holding time limit is reduced. But it also shows limiting the holding time decreases the number of calls blocked and forced call terminations at handover considerably. Next, a simple estimation method for the holding time limit, which reduces the blocking rate to the normal rate for increasing call demand, is described. Finally, results are given of a simulation, which show that the DHTL method keeps good performance for a sudden and great traffic load fluctuation condition.

In this paper, we introduce a concept of minimum bandwidth guaranteed service model for mobile multimedia. In this service model, service is defined in the context of the guaranteed minimum bandwidth and the residual service share. Each flow under this service model is guaranteed with its minimum bandwidth and provided with more in proportion to the residual service share if there is leftover bandwidth. The guaranteed minimum bandwidth assures a flow to keep minimum tolerable quality regardless of the network load, while the leftover bandwidth enhances the quality of service according to the application's adaptivity and the user's interest. We show that the minimum bandwidth guaranteed service model could be implemented by a two-folded wireless packet scheduler consisting of a guaranteed scheduler and a sharing scheduler. Wireless channel condition of each flow is considered in scheduling so that wireless resource can be distributed only to the flows of good channel state, improving total wireless link utilization. We evaluate the service model and the scheduling method by simulation and implementation.

We developed a distributed control algorithm to solve the problem of a trade-off between transient response and stability. We applied it to a congestion control algorithm for transmitting best-effort packets such as transmission control protocol (TCP) packets over the Internet. A new transmission power control algorithm suitable for transmitting best-effort packets over the wireless Internet was also developed using the distributed control algorithm. We showed that in a steady state, TCP connections can use the bandwidth efficiently over both wired and wireless Internet when the proposed control algorithms are used. The transient response was also evaluated and it was found that the packet transmission rate and the transmission power adjusted by the proposed control algorithms converge to a steady state faster than when adjusted by conventional control algorithms while maintaining the stability of network systems.

We derived the design requirements that wireless systems and congestion control algorithms must satisfy to transmit best-effort Internet protocol (IP) packets over wireless systems. We proved that, if these requirements are satisfied, congestion control algorithms are robust against unfairness in the systems and can provide near-maximum throughputs in various environments. From the viewpoint of the design requirements, we investigated the effect of automatic repeat request (ARQ) on the throughputs of best-effort IP connections, and showed why ARQ can improve the throughputs while too large a number of retransmissions degrade them. We also investigated the effect of variance in packet transmission rates and clarified what kind of congestion control algorithm degrades the throughputs.

In this paper, we propose an RTP/UDP/IP header compression method, Multiple-Reference Compression (MRC), which is designed for mobile multimedia communications. MRC is a compression method that calculates differences from the multiple reference headers that have already been sent and inserts them into a compressed header. The receiver can decompress the compressed header as long as at least one of the reference headers is correctly received and decompressed. MRC improves robustness against packet losses compared with CRTP defined in IETF RFC2508, and imposes less overheads and computational burden than robust header compression (ROHC) defined in RFC3095. We also implemented MRC and other header compression algorithms into our mobile testbed, and conducted multimedia streaming experiments over the testbed. The results of the experiments show that MRC offers the same level of packet loss rate as Legacy RTP for both audio and video streams, and provides better media quality than Legacy RTP and CRTP on error-prone radio links. Header compression robust against packet losses is expected as a key technology for VoIP and multimedia streaming services over 3G and future mobile networks.

We propose a new analytical model of Transmission Control Protocol (TCP) in wireless environments where transmission errors occur frequently. In our proposed model, we consider the exponential increase of a congestion window and the exponential increase of a timeout back-off. Finally, we have clarified the behavior of TCP mechanisms of different versions and TCP throughput characteristics analytically. From our result, the behavior of TCP mechanisms is different in each implementation version. These differences mean that the required characteristics of wireless links are different in each implementation version. Therefore, our proposed model is a base analysis of designing wireless link mechanisms.

This paper investigates the cell search time performance of our previously proposed three-step cell search method in a two-cell site environment by laboratory and field experiments supporting asynchronous cell site operation, which is one of the most notable features of wideband direct sequence code division multiple access (W-CDMA) mobile communications. The cell search methods used in the paper are based on the ongoing third generation partnership project (3GPP), in which our original scheme was refined with respect to several points in order to reduce the complexity of the receiver. The experimental results demonstrate that the method achieves the fast cell search time of less than one second in real multipath-fading channels. The cell search is accomplished in less than approximately 700 msec at 90% of the detection probability when 4.7% and 0.5% of the total transmit power of a cell site is assigned to the common pilot channel (CPICH) and synchronization channels (SCHs), respectively, in a two-cell site environment. We also elucidate that the cell search time at the detection probability of 90% using time switched transmit diversity (TSTD) is decreased by approximately 100 msec compared to that without TSTD in low-mobility environments such as the average vehicular speed of 5 km/h with a transmit power assignment of the CPICH of 4.7%.

This paper proposes a multipath interference canceller (MPIC) for orthogonal code multiplexed channels in the W-CDMA forward link and evaluates by computer simulation the improvement in BER performance owing to the multipath interference (MPI) suppression effect obtained by the MPIC. The simulation results show that a one-stage MPIC, which removes the MPI from the common pilot channel (PICH), common control channel (CCH), and synchronization channel (SCH), achieves a sufficient MPI suppression effect, and that the required received E_b/N₀ of the traffic channel (TCH) at the average BER of 10^-3 using the MPIC for the common channels is decreased by approximately 6.5 dB compared to that with a matched filter (MF)-based Rake receiver (the transmit power ratio of each common channel to TCH: Δ_PICH/TCH=0 dB, Δ_CCH/TCH=5 dB, Δ_SCH/TCH=3 dB, without fast transmit power control (TPC) and antenna diversity reception). We also show that by using MPIC, the required transmit E_b/N₀ at the average BER of 10^-3, when the ratio of the target E_b/I₀ of the 9-interfering users to desired user is Δ_Int/Des=6 dB with fast TPC, is increased by only approximately 0.6 dB compared to that when Δ_Int/Des=0 dB. This implies that the preferential MPI suppression from high-rate TCHs that abates the increase in complexity in a mobile terminal is effective in increasing the link capacity in the forward link.

In this paper, the co-existence of TDMA and W-CDMA spectrum sharing system (TDMA/W-CDMA overlaid system) with cellular architecture is discussed. In this system, both systems share the same frequency band to improve the spectrum efficiency. Overall rate, bit error ratio (BER) and spectrum efficiency of the system are calculated for the forward link (down-link) in the presence of AWGN channel. Taking into account the path loss and shadow fading loss in this system with cellular architecture, W-CDMA applying interference cancellation (IC) shows a substantial difference in spectrum efficiency, the overlaid system can provide a greater overall rate and higher spectrum efficiency than a single multiple access-based system such as TDMA system or W-CDMA system. The interference cancellation can significantly improve BER of the spectrum overlaid system.

This paper proposes a signal-to-interference power ratio (SIR) measurement method that employs the transmission gap of a data channel (TGDC) interval for precise link adaptation, in order to eliminate the influence of severe multipath interference (MPI) from a shared packet channel and to decrease further the instantaneous variations in interference components for high-speed packet transmission in the forward link using adaptive data modulation associated with the multipath interference canceller (MPIC). Computer simulation results elucidate that the required received signal energy per chip-to-background noise power spectrum density ratio (E_c/N₀) based on the SIR measurement employing TGDC at the throughput of 4.2 Mbps is decreased by approximately 2.0 dB compared to the conventional method without TGDC using chip-based interference power measurement for the number of paths L = 1, and by approximately 1.5 dB compared to the conventional method using symbol-based interference power measurement for L = 2, respectively. Therefore, we show that the adaptive data modulation with the SIR measurement exploiting the TGDC interval achieves almost the maximum (i.e., almost ideal selection) throughput, without changing the SIR measurement method according to the propagation conditions such as the number of multipaths.

This paper presents a comparison of the throughput performance employing hybrid automatic repeat request (HARQ) with packet combining, such as Type-I with packet combining (simply Chase combining hereafter) and Type-II (Incremental redundancy hereafter), using turbo coding in a multipath fading channel in high speed downlink packet access (HSDPA). We apply a multipath interference canceller (MPIC) to remove the influence of severe multipath interference. Link level simulation results show that the maximum throughput using Incremental redundancy with 64QAM is improved by approximately 5-8% compared to that using Chase combining, and that the required average received signal energy of 12 code channels per chip-to-background noise spectrum density (E_c/N₀) at the throughput of 4 Mbps with Incremental redundancy is decreased by approximately 1.0 dB rather than that with Chase combining when the vehicular speed is higher than approximately 30 km/h. Furthermore, we elucidate based on the system level simulation that although no improvement is obtained in a slow mobility environment such as the average vehicular speed of 3 km/h, the achieved throughput of Incremental redundancy is increased by approximately 5-6% and 13% for the average vehicular speed of 30 km/h and 120 km/h, respectively, compared to that with Chase combining.

We have measured the non line-of-sight (NLOS) propagation characteristics of microwave frequencies in an urban environment with a base station antenna situated well above the surrounding buildings. When these characteristics are compared with the results of measurements made in the same environment with a low base station antenna height, it can be seen that with a low base station antenna height the attenuation coefficient varies greatly between line-of-sight (LOS) and NLOS environments, whereas with a high base station antenna height there is no variation of this sort. This is because the waves arriving NLOS environments from a high base station antenna do so primarily as a result of rooftop diffraction, and the path loss does not vary much over regions of equal distance between the base station and mobile station. We have confirmed that the frequency characteristics of relative loss in NLOS environments with a high antenna height follow a relationship of 22.8 log f, which is more or less the same as the characteristic for the UHF band. By modifying the frequency terms of the Sakagami model (used for UHF band) based on this trend to allow it to handle microwave frequencies, a close correspondence is seen between the results of actual measurements and the values predicted by the extended model.

We investigate the relationship between microwave path-loss characteristics and line-of-sight (LOS) blocking in an urban environment with a low base-station antenna using LOS-blocking measurement equipment that we have developed. Changes in path loss, traffic conditions, and LOS-blocking caused by vehicles were measured simultaneously. It was found that path loss exhibits a Rayleigh distribution even in a LOS environment if the amount of traffic is such that LOS- blocking occurs for 80% of the time or more, but the other case path loss exhibits a Nakagami-Rice distribution. It was also found that ratio of coherent wave level to envelope level (c/r) depends heavily on rate of road traffic flow.

In order to clarify the receiver sensitivity of remote keyless entry systems operating at 314 MHz, the radiation characteristics of a cabin antennas must be studied. For this purpose, electromagnetic simulations are very useful but the multiple reflected waves present in the cabin must be accurately estimated. No prior study examined the radiation characteristics of cabin antennas. This paper introduces an accurate simulation method. A MM (Method of Moments) simulator is selected because it can simulate multiple reflected waves. Initial studies are being performed to find the calculation requirements and ensuring convergence of multiple reflected wave structures. Next, calculation requirements for car analyses such as convergence parameters and computer abilities are studied. The calculation time is 11 hours by a personal computer. 1.1 GB of RAM is also needed. Many calculated results such as three-dimensional radiation patterns and induced currents on the car's body are obtained. The radiation pattern varies with antenna position, and polarization changes are shown. In order to ensure that the results are accurate, measurements are made on a one seventh scale model of a car. The good agreement of measured and calculated results verifies the validity of the simulation method. The influence of multiple reflected waves on the radiation patterns is also experimentally studied. The conclusion is that the MM method is shown to be useful in estimating the radiation pattern of cabin antennas. Interesting calculation results are also shown.

This paper investigates the optimum adaptive antenna array beam forming (AAA-BF) configuration considering the diversity effect provided by transmit diversity (TD) in a multipath fading channel in the W-CDMA forward link. Computer simulation results show that the required transmit signal energy per bit-to-background noise power spectrum density ratio (E_b/N₀) at the average block error rate (BLER) of 10^-2 using AAA-BF coupled with TD is decreased by approximately 1.0 dB compared to that of AAA-BF assuming the identical number of total antennas when the capacity, i.e., the number of simultaneously accommodated users with the transmission power proportional to the symbol rate, normalized by processing gain, Pg, is below approximately 20%. However, we find that in an interference-limited channel, when the capacity normalized by Pg is above approximately 30%, AAA-BF employing all antennas accommodates a larger capacity compared to AAA-BF coupled with TD because of a sufficient interference suppression effect due to a much narrower beam width despite the absence of the antenna diversity effect. This paper also elucidates in a multi-cell model that AAA-BF employing all antennas can accommodate approximately 1.5 times more users with the average BLER of 10^-2 and with the outage probability of 5%, compared to the case with AAA-BF coupled with TD, when the total number of antennas is 8.

In this paper, we propose the code orthogonalizing filter (COF) based adaptive array antenna using sample matrix inversion with common correlation matrix (CCM-SMI) of time domain signals for multicarrier DS/CDMA systems. The conventional array antenna system calculates the weight using the correlation matrix of individual subcarrier's signals. On the other hand, our proposed system calculates the weight using the correlation matrix of time domain signals before FFT operation, so it can reduce the calculation time and the complexity of weight calculation than the conventional scheme, to maintain the system performance. Moreover, we consider the code orthogonalizing filter to reduce the demerit of adaptive array antenna system using sample matrix inversion algorithm with common correlation matrix that requires heavy computational complexity while the signal environment frequently changes. Our proposed system obtains more accurate channel response vector using COF than that of the conventional CCM-SMI based on the matched filter, without increasing the matrix size. The performance is evaluated in term of bit error probability. From the analysis and simulation results, it is shown that our proposed scheme achieves better BER performance than that of the conventional system.

This paper proposes adaptive antenna array transmit diversity (AAA-TD) in the W-CDMA forward link with frequency division duplexing (FDD), based on adaptively-generated receiver antenna weights in the reverse link, which only track the changes in the average signal-to-interference power ratio (SIR) and direction of arrival (DOA) but with the calibration of the phase/amplitude variations of the parallel RF receiver/transmitter circuits corresponding to the number of array antennas. The laboratory and field experimental results exploiting AAA-TD are presented to show the strong multipath interference (MPI) suppression effect especially from high-rate users with large transmission power. Laboratory experiments elucidate that by using AAA-TD with four antennas, the required transmitted SIR before multiplying the transmitter antenna weights at the average BER of 10^-3 is decreased by approximately 13 dB compared to that with one omni-directional antenna transmitter. Field experiments also show that although an error floor above 10^-2 is observed with one omni-directional antenna transmitter when the transmitted SIR is -12 dB due to severe MPI, no error floor is observed when employing 4-antenna AAA-TD and the loss of the required received signal power at the average BER of 10^-3 from the single-user case is suppressed to below approximately 5 dB. Therefore, we show that AAA-TD is very effective in suppressing severe MPI especially from high rate users with large transmission power due to its adaptive main lobe and null steering.

This paper elucidates the optimum bandwidth per sub-carrier in the reverse link for multicarrier (MC)/DS-CDMA using a 10 to 80-MHz bandwidth in a multipath fading channel with numerous resolved multipaths, taking into account all major effects, i.e., the improvement in the Rake time diversity effect and the degradation in the path search and the channel estimation due to multipath interference (MPI). In the paper, we assume a broadband channel model with the maximum delay time of up to approximately 1 µsec simulating a microcell with the radius of less than 1 km in an urban area. The simulation results clarify that the improvement in the radio link performance is almost saturated at a bandwidth greater than approximately 40 MHz when the spreading factor of the channel is SF=32, and the best performance is achieved at the bandwidth of approximately 20-40 MHz when SF=4, employing two-branch antenna diversity reception (an average equal power delay profile and an exponential decay power delay profile are assumed, where the number of multipaths is changed from 12 to 48 for both profiles). This is generated by the tradeoff between the improvement in the Rake time diversity effect and the increased MPI in addition to the degradation in accuracy of the path search and channel estimation associated with a lower average received signal-to-interference plus background noise power ratio. Therefore, we conclude that MC/DS-CDMA, where each sub-carrier has the bandwidth of approximately 20-40 MHz, is one of the most promising candidates for broadband packet wireless access in the reverse link.

This paper proposes a pilot channel assisted minimum mean square error (MMSE) combining scheme in orthogonal frequency and code division multiplexing (OFCDM) based on actual signal-to-interference power ratio (SIR) estimation, and investigates the throughput performance in a broadband channel with a near 100-MHz bandwidth. In the proposed MMSE combining scheme, the combining weight of each sub-carrier component is accurately estimated from the channel gain, noise power, and transmission power ratio of all the code-multiplexed channels to the desired one, by exploiting the time-multiplexed common pilot channel in addition to the coded data channel. Simulation results elucidate that the required average received signal energy per bit-to-noise spectrum density ratio (E_b/N₀) for the average packet error rate (PER) = 10^-2 is improved by 0.6 and 1.2 dB by using the proposed MMSE combining instead of the conventional equal gain combining (EGC) in a 24-path Rayleigh fading channel (exponential decay path model, maximum delay time is approximately 1 µsec) in an isolated cell environment, when the number of multiplexed codes = 8 and 32, respectively, with the spreading factor of 32. Furthermore, when the average received E_b/N₀ = 10 dB, the achievable throughput, i.e., the number of simultaneously multiplexed codes for the average PER = 10^-2 in the proposed MMSE combining, is increased by approximately 1.3 fold that of the conventional EGC.

In this paper, we focus on an OFDM peak power reduction method that uses clipping and filtering. This method can reduce the peak power of OFDM via clipping, and can also reduce the out-band emission via filtering, even if a nonlinear amplifier is used. However, the filtering causes peak power regeneration. For purposes of reducing the effect of peak power regeneration, we propose an adaptive clipping level control method for OFDM peak power reduction, as part of a technique using clipping and filtering. In this method, the clipping level is optimized by checking the peak power regrowth which is caused by inserting a filter, by using a multi-stage filtering simulator. Thus the peak power is adjusted to the target PAPR. If the target PAPR is decided to be the saturation power of an amplifier, the bit error rate performance is improved without increasing the out-band emission. In order to confirm the effectiveness of the proposed system, we evaluate its performance by using computer simulation.

Methods that precisely compensate for propagation distortion using pilot symbols are widely used in mobile communications. We describe such a pilot-symbol-assisted technique for precise compensation of flat fading and frequency offset. This technique provides a wide range of offset compensation. Conventional methods using fast Fourier transform (FFT) compensate for both slow and fast fading, but their tolerable range of frequency offset is very limited. By composing a system with an approximate frequency estimator, we can estimate and compensate for fading and a large frequency offset at the same time. The estimation and compensation are carried out by periodic pilot symbols and no other index sequences are needed. This method enables high-data-rate transmission. We describe the method and provide a theoretical analysis for the compensable range of fading and frequency offset for a transmission frame structure with pilot symbols. Then, we evaluate the method by computer simulation.

In an orthogonal frequency division multiplexing (OFDM) system, the bit error performance is degraded in the presence of multiple propagation paths whose excess delays are longer than the Guard Interval (GI), because the orthogonality between subcarriers cannot be maintained. Therefore, the GI has to be long enough for an expected delay spread of the channel. On the other hand, a long GI causes a decrease in transmission efficiency. In this paper, we propose a new OFDM demodulation method with a variable effective symbol duration, in order to improve the bit error performance in the presence of multipaths whose excess delays are longer than the GI. The proposed method can realize more stable radio communication systems under a multipath propagation environment even if a propagation path whose excess delay is longer than the GI exists. In other words, the proposed method can improve transmission efficiency without performance degradation by a shortened GI under the same environment. The principle of the proposed method is explained, and the bit error probability of the proposed method is analyzed theoretically in an AWGN channel and a multipath fading channel. The performance of the proposed method is then evaluated by computer simulation. The results show that the proposed method improves the system availability under more various multipath fading environments without changing the system parameters.

MIMO communication systems have been studied as a way to increase system capacity and to improve spectrum efficiency. This paper proposes a novel branch metric generation method for soft-decision Viterbi decoding in coded OFDM systems employing SDM (OFDM-SDM) systems, one form of MIMO communication, that employ the maximum likelihood decision (MLD) algorithm. In coded OFDM systems, the branch metric for forward error correction (FEC) is normally calculated based on received sub-carrier power level and the constellation point likelihood. This conventional approach fails in OFDM-SDM systems. In the proposed method, branch metric is derived based on the likelihood calculated by MLD algorithm, which improves FEC performance effectively. Then, availability of coded OFDM-SDM to realize wireless LANs with data rate above 100 Mbit/s in a bandwidth of 20 MHz is evaluated in office environments using required CNR calculated by computer simulation. The evaluation indicates that OFDM-SDM has the possibility of realizing 100 Mbit/s in office environments.

In this paper, we propose a new coded orthogonal frequency division multiplex (COFDM)-based ubiquitous antenna system, which is composed of multiple radio base stations (RBSs) deployed over the service area and Radio-on-Fiber (RoF) link that connects RBSs to the central control station (CCS). The proposed system is capable of receiving multiple mobile terminals simultaneously operating at the same frequency by making effective use of joint detection. However, the propagation delay due to the RoF link could be a major problem for realizing the ubiquitous antenna system. In order to overcome this delay problem, we assume that the guard interval of COFDM is longer than the delay difference. Furthermore, in order to improve BER performance in a multipath Rayleigh fading channel, we also propose the MSE normalization scheme followed by the use of an MMSE-based joint detector. Computer simulation results show that the proposed system can improve the frequency utilization efficiency of the broadband wireless access system.

In evaluating the teletraffic of mobile communication networks, it is important to model the motion of terminals. In the previous migration model, mobility characteristics of terminals, such as cell dwell time, have been expressed by a single probability distribution. In this paper, we discuss the modeling of the cell dwell time of terminals in each cell. Using measured data we show that cell dwell time differs from cell to cell and follows log-normal distributions rather than conventional exponential distributions.

Upper bounds on the bit error rate (BER) for maximum likelihood (ML) decoding are derived in convolutional coded parallel combinatorial spread spectrum (PC/SS) systems over additive white Gaussian noise (AWGN) channels. PC/SS systems can achieve higher data transmission than conventional multicode SS systems. To make the derivation tractable, we put a uniform interleaver between a convolutional encoder and a PC/SS transmitter. Since the PC/SS transmitter is employed as the "inner encoder," the bounds are obtained in a similar manner of the derivation in serially concatenated codes through a uniform interleaver. Two different error patterns in the PC/SS system are considered in the performance analysis. Numerical results show that the derived BER bounds are sufficiently accurate. It is found that the coded PC/SS systems outperform coded all-code-parallel DS/SS systems under the same data rate conditions if the number of pre-assigned PN codes increases.

This paper presents a new method of designing a dither matrix based on simulated annealing. An obtained dither matrix (halftone screen/mask) is appropriate for press printing. Because of several physical reasons, halftoning for press printing is more difficult than halftoning for electronic displays, or ink-jet printers. Even if stochastic dispersed-dot screening (so-called FM-screening) is one of the best solutions for halftoning, that is not appropriate for press printing. On the other hand, classical periodic clustered-dot screening (so-called AM-screening) is more important and is widely used even now. We recognize unfavorable quality of AM-screening, but we can not ignore its productive stability in printing section. The proposed halftone dither matrix has aperiodic clustered-dot pattern, and size of cluster can be controlled by a weighting parameter of a cost function. We will obtain a dither matrix which consists of clustered-dots. Some characteristics of the design algorithm and halftoned images are investigated in detail. As a result, the fact that an obtained dither matrix is superior to AM-screen and comparable to FM-screen in visual quality, and the matrix is comparable to AM-screen and superior to FM-screen in press printability is confirmed.

A noise suppression algorithm with high speech quality based on weighted noise estimation and MMSE STSA is proposed. The proposed algorithm continuously updates the estimated noise by weighted noisy speech in accordance with an estimated SNR. The spectral gain is modified with the estimated SNR so that it can better utilize the improvement in noise estimation. With a better noise estimate, a more correct SNR is obtained resulting in the enhanced speech with low distortion. Subjective evaluation results show that five-grade mean opinion scores of the new algorithm with and without a speech codec are improved by as much as 0.35 and 0.40 respectively, compared with either the original MMSE STSA or the EVRC noise suppression algorithm.

New methods for digital image watermarking based on the characteristics of vector quantization (VQ) are proposed. In contrast with conventional watermark embedding algorithms to embed only one watermark at a time into the original source, we present one algorithm to embed multiple watermarks for copyright protection. The embedding and extraction processes are efficient for implementing with conventional VQ techniques, and they can be accomplished in parallel to shorten the processing time. After embedding, embedder would output one watermarked reconstruction image and several secret keys associated with the embedded watermarks. These secret keys are then registered to the third party to preserve the ownership of the original source in order to prevent the attackers from inserting counterfeit watermarks. Simulation results show that under no attacks, the embedded watermarks could be perfectly extracted. If there are some intentional attacks in our simulation, all the watermarks could survive to protect the copyrights. Therefore, we are able to claim the robustness, usefulness, and ease of implementation of our algorithm.

In this semi-tutorial paper, the reliability-based decoding approaches using the reprocessing of the most reliable information set are investigated. This paper somehow homogenizes and compares former different studies, hopefully improving the overall transparency, and completing each one with tricks provided by the others. A couple of sensible improvements are also suggested. However, the main goal remains to integrate and compare recent works based on a similar general approach, which have unfortunately been performed in parallel without much efforts of comparison up to now. Their respective (dis)advantages, especially in terms of average or maximum complexity are elaborated. We focus on suboptimum decoding while some works to which we refer were developed for maximum likelihood decoding (MLD). No quantitative error performance analysis is provided, although we are in a position to benefit from some qualitative considerations, and to compare different strategies in terms of higher or lower expected error performances for a same complexity. With simulations, however, it turns out that all considered approaches perform very closely to each other, which was not especially obvious at first sight. The simplest strategy proves also the fastest in terms of CPU-time, but we indicate ways to implement the other ones so that they get very close to each other from this point of view also. On top of relying on the same intuitive principle, the studied algorithms are thus also unified from the point of view of their error performances and computational cost.

A new blocking artifact reduction algorithm is proposed that uses block classification and feedforward neural network filters in the spatial domain. At first, the existence of blocking artifact is determined using statistical characteristics of neighborhood block, which is then used to classify the block boundaries into one of four classes. Thereafter, adaptive inter-block filtering is only performed in two classes of block boundaries that include blocking artifact. That is, in smooth regions with blocking artifact, a two-layer feedforward neural network filters trained by an error back-propagation algorithm is used, while in complex regions with blocking artifact, a linear interpolation method is used to preserve the image details. Experimental results show that the proposed algorithm produces better results than the conventional algorithms.

A plane drawing of a graph is called a floorplan if every face (including the outer face) is a rectangle. A based floorplan is a floorplan with a designated base line segment on the outer face. In this paper we give a simple algorithm to generate all based floorplans with at most n faces. The algorithm uses O(n) space and generates such floorplans in O(1) time per floorplan without duplications. The algorithm does not output entire floorplans but the difference from the previous floorplan. By modifying the algorithm we can generate without duplications all based floorplans having exactly n faces in O(1) time per floorplan. Also we can generate without duplications all (non-based) floorplans having exactly n faces in O(n) time per floorplan.

On the basis of generalized theory of system design the behavior of the different design trajectories in the design phase space was analyzed. An additional acceleration effect of the design process has been discovered by the analysis of various design strategies with different initial points. This effect can be understood well on the basis of the elaborated design methodology by means of the different design trajectory analysis. This effect is displayed for all analyzed circuits and it reduces additionally the total computer time for the system design.

In this letter a low-complexity and low-power realization of the 2D discrete-cosine-transform and its inverse (DCT/IDCT) is presented. A VLSI circuit based on the Chen algorithm with the distributed arithmetic approach is described. Furthermore low-power design techniques, based on clock gating and data driven switching activity reduction, are used to decrease the circuit power consumption. To this aim, input signal statistics have been extracted from H.263/MPEG verification models. Finally, circuit performance is compared to known software solutions and dedicated full-custom ones.

In this paper a low-complexity and high-resolution algorithm to estimate the magnitude of complex numbers is presented. Starting from a review of previous art, the new algorithm has been derived to improve precision performance without any penalty in hardware complexity. As a case example, a semi-custom VLSI implementation for 10 bit 2's complement input data has been performed. A mean square error and mean error performance improvement of nearly one order of magnitude has been demonstrated for an hardware complexity increase of roughly 34% with respect to previously presented solutions.