In mobile broadcasting environments, an information server periodically broadcasts a set of data items to a large mobile client population at every broadcast cycle and mobile clients retrieve the data items they need upon arrival at the broadcast channel. In such environments, the cost of data delivery is independent of the number of clients. Many applications such as auctions and stock quotes perform read-only transactions that require the clients to read consistent and current data for accurate result. Previous concurrency control mechanisms designed for such environments ensure that the clients read consistent data, but they cannot ensure latest-bound currency which allows the clients to read the latest or most up-to-date data. In this paper, we propose an efficient concurrency control mechanism that ensures latest-bound currency as well as update consistency, which is appropriate for the mobile broadcasting environments. To ensure latest-bound currency, the server computes control information using "virtual" broadcast cycles. This control information is also used for checking update consistency. Thus, the proposed mechanism allows all data committed in current broadcast cycle to be broadcast. We have performed simulation experiments to measure transaction aborts in order to evaluate the performance of the proposed mechanism. The result confirms that the proposed mechanism produces no stale reads and also shows that the proposed mechanism generates less transaction aborts than previous mechanisms, which implies that we can get higher data currency without increasing data inconsistency.

In this paper, we propose a Call Admission Control (CAC) scheme for the Direct Sequence-Code Division Multiple Access (DS-CDMA) systems supporting voice and data services and analyze the Erlang capacity under the proposed CAC scheme. Service groups are classified by Quality of Service (QoS) requirements such as the required Bit Error Rate (BER) and information bit rate, and Grade of Service (GoS) requirement such as required call blocking probability. Different traffics require different system resources based on their QoS requirements. In the proposed CAC scheme, some system resources are reserved exclusively for handoff calls to have high priority over new calls. Additionally, the queueing is allowed for both new and handoff data traffics that are not sensitive to delay. As a performance measure of the suggested CAC scheme, Erlang capacity is introduced. For the performance analysis, a four-dimensional Markov chain model is developed. As a numerical example, Erlang capacity of an IS-95B type system is depicted, and optimum values of system parameters such as the number of reservation channels and queue lengths are found. Finally, it is observed that Erlang capacity is improved more than 2 times by properly selecting the system parameters with the proposed CAC scheme. Also, the effect of handoff parameters on the Erlang capacity is observed.

This paper presents the design, implementation and test results of a complete wideband CDMA modem with a modulator and a demodulator for use in next-generation cell sites. The modem is based on Japanese proposal for the next generation W-CDMA air interface. The modulator has a flexible architecture, which can transmit the data of different physical channels such as common control physical channels, and dedicated physical channels by setting the channel selection registers of the modulator. The modulator performs the transmission power control digitally according to the physical channel setting. The receiver consists of a searcher, four fingers and a combiner. The searcher supports path selected search operation considering antenna diversity which is employed to reduce the effect of excessively deep fades, and early dump operation to speed up the searching process. Since the simulation results to determine the number of finger showed that the system with more than four fingers had no essential performance improvement, demodulation performed with four-finger rake receiver. The proposed cell site modem was implemented with FPGAs and verified by the board-level experiment. Experiments have verified that the modem fully complied with the specifications.

To improve the reliability and efficiency of multicast transmissions in wireless systems, a novel retransmission procedure is desired. In this paper, the representative acknowledgment scheme for reliable wireless multicast communications is proposed that offers quite a low packet loss rate. The proposed protocol carries out retransmissions in the datalink layer within the wireless region, and retransmissions do not affect the traffic in the wired region. The representative acknowledgment scheme employs both positive acknowledgment (ACK) and negative acknowledgment (NACK) to achieve reliable multicast transmissions and reduces the number of responses to be returned by forming groups of stations in the cell. One of the members in a group, called a representative station, returns a response for a received data frame while the others return a NACK if necessary. With this scheme, reliable multicast transmissions are enabled in wireless communications without spending much time as in conventional reliable multicast protocols. The performance of the proposed protocol is evaluated by numerical analyses and by computer simulation. The results show that 30% or more decrease in transmission time is achieved in a typical wireless environment.

We propose a p-persistent protocol for slow-frequency-hopped slotted random access networks, assuming that all the users know the number of backlog users in a slot. The proposed protocol delays new packet transmission until the number of users with a packet to be retransmitted decreases to the threshold or less. Performance of the proposed protocol is evaluated with a two-dimensional Markov chain for systems with finite population in terms of throughput efficiency and the average transmission delay. Numerical results show that the proposed protocol can achieve better performance by suppressing new packet transmission, compared to the conventional frequency-hopped slotted ALOHA. The optimum value of the threshold is also numerically derived.

To satisfy huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes two different procedures of multi-channel multiple access schemes with the slotted ALOHA operation for both data and voice traffic and presents an exact analysis to numerically evaluate the performance of the systems. In scheme I, there is no limitation on access between data transmissions and voice transmissions, i.e., all channels can be accessed by all transmissions. In scheme II, a channel reservation policy is applied, where a number of channels are used exclusively for voice packets while the remaining channels are used for both data packets and voice packets. We call the system using scheme I "Non-reservation system" and call the system using scheme II "Reservation system. " Performance characteristics we obtained include loss probability for voice traffic, average packet delay for data traffic and channel utilization for both traffic. The performance of the schemes and the effects of the design parameters are numerically evaluated and compared to a wide-bandwidth conventional single-channel slotted ALOHA system with single data traffic. The analysis presented in this paper will be not only useful for the performance evaluation and the optimum design of multi-channel multi-traffic systems in wireless environments, but also applicable to evaluate other performance measures in priority networks, cellular mobile radio networks or multi-hop wireless networks.

In any ill-conditioned information-transfer system, as in long-distance communication, we often must construct feedback confirmation channels, in order to confirm that informations received at destinations are correct. Unfortunately, for such systems, undesirable uncertain fluctuations may be induced not only into forward communication channels but also into feedback confirmation channels, and it is such difficult that transmitters always confirm correct communications. In this paper, two fuzzy-set-valued mappings are introduced into both the forward communication channel and the feedback confirmation channel, separately, and overall system-behaviors are discussed from the standpoint of functional analysis, by means of fixed point theorem for a system of generalized equations on fuzzy-set-valued mappings. As a result, a good mathematical condition is successfully obtained, for such information-transfer systems, and fine-textured estimations of solutions are obtained, at arbitrary levels of values of membership functions.

This paper describes a new method for estimating the direction of arrival (DOA) using a nonlinear microphone array system based on complementary beamforming. Complementary beamforming is based on two types of beamformers designed to obtain complementary directivity patterns with respect to each other. In this system, since the resultant directivity pattern is proportional to the product of these directivity patterns, the proposed method can be used to estimate DOAs of 2(K-1) sound sources with K-element microphone array. First, DOA-estimation experiments are performed using both computer simulation and actual devices in real acoustic environments. The results clarify that DOA estimation for two sound sources can be accomplished by the proposed method with two microphones. Also, by comparing the resolutions of DOA estimation by the proposed method and by the conventional minimum variance method, we can show that the performance of the proposed method is superior to that of the minimum variance method under all reverberant conditions.

A combination of a software and a systolic hardware implementation for the Quasi Arithmetic compression algorithm is presented. The hardware is implemented as a pipeline hardware implementation. The implementation doesn't change the the algorithm. It just split it into two parts. The combination of parallel software and pipeline hardware can give very fast compression without decline of the compression efficiency.

Orthogonal Frequency Division Multiplexing-Direct Sequence/Code Division Multiple Access (OFDM-DS/CDMA) systems provide frequency diversity gain avoiding inter symbol interference (ISI) in a frequency selective fading channel. However, path diversity gain can not be obtained by using conventional OFDM-DS/CDMA schemes. This paper proposes a new multiple antenna transmission system with combined path diversity and frequency diversity. Signal of each antenna is delayed by several chips to create artificial path diversity as well as frequency diversity of multi-carrier transmission in which can then be combined by using a RAKE receiver. Therefore multiple antenna transmission scheme creates a path diversity effect on uncorrelated signals in multi-carriers from each antenna. The received uncorrelated signals can be processed by Maximum Ratio Combining (MRC) diversity without ISI at a RAKE receiver even when we use FFT modulation. As a result, we can obtain combined path diversity and frequency diversity gain effectively by the RAKE system with the combination of multiple antennas.

In this paper, we propose an efficient channel estimation technique for orthogonal frequency-division multiplexing (OFDM) systems with transmitter diversity. The proposed technique estimates uniquely all channel frequency responses needed in space-time coded OFDM systems using "comb-type" training symbols. The computational complexity of the proposed technique is reduced dramatically, compared with the previous minimum mean-squared error (MMSE) technique, due to the processing made all in the frequency-domain. Also, several other techniques for mitigating random noise effect and tracking channel variation are discussed to further improve the performance of the proposed approach. The performances of the proposed approach are demonstrated by computer simulation for mobile wireless channels.

We examine delay performance of packets from constant bit rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the Σ Di/G/1 queue with vacations. Our analysis allows heterogeneous service time and heterogeneous interarrival time. Thus, we can get the impact of packet length of a stream on the delay time of other streams. We then give various numerical results for enterprise multimedia networks, which include voice, video and data communication services. From our quantitative evaluation, we conclude that packet length of video traffic has large influence on the delay time of voice traffic while voice traffic gives a little impact on the delay time of video traffic.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

A low sidelobe two-beam waveguide slot array is designed and measured. The antenna structure should be symmetrical for realizing two symmetrical beams which imposes restriction in slot design for the sidelobe and the gain. The slot coupling distribution is optimized numerically for side-lobe suppression under the condition of the structural symmetry. The first side-lobe level is minimized for the specific antenna efficiency in the continuous source model. This synthesis is reinforced by the full wave slot analysis using the method of moments. The design is confirmed by experiments using a one-dimensional array at 12 GHz and the good agreements between the predictions and the measurements are observed.

High-speed digital data transmission services with mobile equipment are becoming available. Though the visual signal is one of the expected media to be used with such transmission capabilities, the bandwidth of visual signal is much broader than the provided transmission bandwidth in general. Therefore efficient video encoding algorithms have to be introduced. The ITU-T Recommendation H.263 and ISO/IEC MPEG-4 are very powerful encoding algorithms for a wide range of video sequences. But a large amount of bits are generated in highly active scenes to encode them using such conventional methods. This results in frame skipping and degradation of decoded picture quality. In order to keep these degradations as low as possible, we proposed a Dynamic Resolution Conversion (DRC) method of the prediction error. In the method, a reduced resolution encoding is carried out when the input scene is highly active. Simulation results show that the proposed scheme can improve both coding frame rate and picture quality in a highly active scene. We also present in this paper that some analysis for the performance of the DRC method under the error prone environment that is inevitable with mobile communications.

One of the disadvantages of compressed data is their vulnerability, that is, even a single corrupted bit in compressed data may destroy the decompressed data completely. Therefore, Variable-to-Fixed length Arithmetic Coding, or VFAC, with error detecting capability is discussed. However, implementable error recovery method for compressed data has never been proposed. This paper proposes Burst Error Recovery Variable-to-Fixed length Arithmetic Coding, or BERVFAC, as well as Error Detecting Variable-to-Fixed length Arithmetic Coding, or EDVFAC. Both VFAC schemes achieve VF coding by inserting the internal states of the decompressor into compressed data. The internal states consist of width and offset of the sub-interval corresponding to the decompressed symbol and are also used for error detection. Convolutional operations are applied to encoding and decoding in order to propagate errors and improve error control capability. The proposed EDVFAC and BERVFAC are evaluated by theoretical analysis and computer simulations. The simulation results show that more than 99.99% of errors can be detected by EDVFAC. For BERVFAC, over 99.95% of l-burst errors can be corrected for l 32 and greater than 99.99% of other errors can be detected. The simulation results also show that the time-overhead necessary to decode the BERVFAC is about 12% when 10% of the received words are erroneous.

The normalized least mean square (NLMS) algorithm has the drawback that the convergence speed of adaptive filter coefficients decreases when the reference signal has high auto-correlation. A technique to improve the convergence speed is to apply the decorrelated reference signal to the calculation of the gradient defined in the NLMS algorithm. So far, only the effect of the improvement is experimentally examined. The convergence property of the adaptive algorithm to which the technique is applied is not analized yet enough. This paper first defines a cost function properly representing the criterion to estimate the coefficients of adaptive filter. The name given in this paper to the adaptive algorithm exploiting the decorrelated reference signal, 'normalized least mean EE' algorithm, exactly expresses the criterion. This adaptive algorithm estimates the coefficients so as to minimize the product of E and E' that are the differences between the responses of the unknown system and the adaptive filter to the original and the decorrelated reference signals, respectively. By using the cost function, this paper second specifies the convergence condition of the normalized least mean EE' algorithm and finally presents computer simulations, which are calculated using real speech signal, to demonstrate the validity of the convergence condition.

This paper proposes a unified coding algorithm for the lossless and near-lossless compression of still color images. The proposed unified color image coding scheme can control the Peak Signal-to-Noise Ratio (PSNR) of the reconstructed image while the level of distortion on the RGB plane is suppressed to within a preset magnitude. In order to control the PSNR, the distortion level is adaptively changed at each pixel. An adaptive quantizer to control the distortion is designed on the basis of psychovisual criteria. Finally, experiments on Super High Definition (SHD) images show the effectiveness of the proposed algorithm.

In this letter, the bandwidth adaptation of an adaptive IIR multiline enhancer is explored. In addition to the problem of bias cancellation of the main structure, different aspects of the proposed filter such as noise equivalent bandwidth and optimal bandwidth are considered and compared with the half power bandwidth of the adaptive multiline enhancer. Since the center frequency of the multiple sinusoids of the input signal is estimated with no bias, with the assumption that the center frequency of the incoming signal is accurately adapted, the error surface of the algorithm is calculated analytically as a function of the filter bandwidth. Computer simulations are used to compare optimum and adapted bandwidths.

The hardware implementation of a proposed high dimensional discrete torus knot code was successfully realized on an ASIC chip. The code has been worked on for more than a decade since then at Aichi Prefectural University and Nagoya Institutes of Technology, both in Nagoya, Japan. The hardware operation showed the ability to correct the errors about five to ten times the burst length, compared to the conventional codes, as expected from the code configuration and theory. The result in random error correction was also excellent, especially at a severely degraded error rate range of one hundredth to one tenth, and also for high grade characteristic exceeding 10-6. The operation was quite stable at the worst bit error rate and realized a high speed up to 50 Mbps, since the coder-decoder configuration consisted merely of an assemblage of parity check code and hardware circuitry with no critical loop path. The hardware architecture has a unique configuration and is suitable for large scale ASIC design. The developed code can be utilized for wider applications such as mobile computing and qualified digital communications, since the code will be expected to work well in both degraded and high grade channel situations.