A channel equalization technique on a time division duplex CDMA/TDMA system for wireless multimedia networks is investigated, and the bit error rate performance of the system is theoretically analyzed. The assumed network connects mobile terminals to a node of ATM based high speed LAN through a radio central unit. Only human interface facilities are implemented into the terminal so that users access integrated services through the node of the network. The uplink (from a mobile terminal to a radio central unit) employs a CDMA scheme to transmit human interface signals and the downlink employs a TDMA scheme to transmit display interface signals. Both the CDMA and the TDMA signals occupy the same frequency band. To mitigate bit error rate degradation due to fading, the radio central unit estimates the impulse response of the channel from the received CDMA signals and subtracts the replica signal to cancel the major intersymbol interference (ISI) component. Numerical results using the Nakagami-m fading model and recent propagation measurements show that the proposed TPC technique compensates the fading attenuation and the proposed CEQ cancels the major ISI component. The bit error rate performance of the downlink with the proposed CEQ is superior to that with the DFE by 12dB of the symbol SNR at the BER=10-6 over a specular channel, and the system with the proposed CEQ achieves a BER=10-6 at the symbol SNR=12dB. Furthermore, the channel equalizer is implemented without increases in complexity of the terminal because all the processing on the equalization is carried out only in the radio central unit.

To support high bit rate and high quality indoor radio communication systems, we have to solve intersymbol interference (ISI) problem caused by frequency-selective fading. Recently multi-carrier modulation technique is considered to be one of the effective methods for this problem. In this paper we propose Partial Frequency ARQ (Automatic Repeat reQuest) system which can achieve effective ARQ scheme for multi-carrier packet communication. This system operates partial retransmission of erroneous power faded packets, and it is superior to the traditional ARQ systems. Furthermore two different protocols are examined for this system: Static Carrier Assignment (SCA) and Dynamic Carrier Assignment (DCA). By computer simulation we found that DCA method can achieve better performance than SCA in terms of both throughput and packet transmission delay.

Standard Site Method (SSM) is theoretically analyzed using matrix representations to examine its validity and develop an improved method. The analysis reveals that the SSM yields an antenna factor specifically related to the effective load impedance presented by the cable and associated devices which are disconnected from the antenna during the SSM site attenuation measurements. Therefore, an additional conversion is required to determine the desired antenna factor under actual load conditions. It is also concluded that the SSM is not applicable to antennas having height-dependent antenna factors. In addition, the SSM correction factors are found to be theoretically inappropriate. Uncertainty of the antenna factor obtained using the SSM is discussed and the required antenna separation distance is investigated. To improve the existing SSM, it is proposed that both transmitting and receiving antennas are placed at the same height during the site attenuation measurements. Experiments exhibit the superiority of the improved method.

The back-propagation algorithm has been applied to many fields, and has shown large capability of neural networks. Many people use the back-propagation algorithm together with a momentum term to accelerate its convergence. However, in spite of the importance for theoretical studies, theoretical background of a momentum term has been unknown so far. First, this paper explains clearly the theoretical origin of a momentum term in the back-propagation algorithm for both a batch mode learning and a pattern-by-pattern learning. We will prove that the back-propagation algorithm having a momentum term can be derived through the following two assumptions: 1) The cost function is Enαn-µEµ, where Eµ is the summation of squared error at the output layer at the µth learning time and a is the momentum coefficient. 2) The latest weights are assumed in calculating the cost function En. Next, we derive a simple relationship between momentum, learning rate, and learning speed and then further discussion is made with computer simulation.

This paper presents a newly developed architecture for a highly parallel DSP suited for realtime image reaognition. The programmable DSP was designed for a variety of image recognition systems, such as computer vision systems, character recognition systems and others. The DSP consists of functional units suited for image recognition: a SIMD processing core, a hierarchical bus, an Address Generation Unit, Data Memories, a DMA controller, a Link Unit, and a Control Unit. The high performance of 3.2GOPS is realized by the eight-parallel SIMD core with a optimized pipeline structure for image recognition algorithms. The DSP supports flexible data transfers including an extraction of lacal images from raster scanned image data, a table-loop-up, a data-broadcasting, and a data-shifting among processing units in the SIMD core, for effective execution of various image processing algorithms. Hence, the DSP can process a 55 spatial filtering for 512512 images within 13.1 msec. Adopting the DSP to a Japanese character recognition system, the speed of 924 characters/sec can be achieved for feature extractions and feature vectors matchings. The DSP can be integrated in a 14.514.5 mm2 single-chip, using 0.5 µm CMOS technology. In this paper, the key features of the architecture and the new techniques enabling efficient operation of the eight parallel processing units are described. Estimation of the performance of the DSP is also presented.

This paper describes modifications to a previously proposed 8-kb/s 4-ms-delay CELP speech coding algorithm with a view to improving the speech quality while maintaining low delay and only moderately increasing complexity. The modifications are intended to improve the effectiveness of interframe pitch lag prediction and the sub-optimality level of the excitation coding to the backward adapted synthesis filter by using delayed decision and joint optimization techniques. Results of subjective listening tests using Japanese speech indicate that the coded speech quality is significantly superior to that of the 8-kb/s VSELP coder which has a 20-ms delay. A method that reduces the computational complexity of closed-loop 3-tap pitch prediction with no perceptible degradation in speech quality is proposed, based on representing the pitch-tap vector as the product of a scalar pitch gain and a normalized shape codevector.

A down sampling technique was applied to signal processing of fiber optic gyroscopes with optical phase modulation. The technique shifts the frequency spectrum of the gyroscopic signal down to low frequencies, and lowers the speed requirements for analog-to-digital (A/D) conversion and numerical operations. A single-chip digital signal processor (DSP) with a built-in A/D converter and timers was used to demonstrate the proposed technique. The DSP internally generated a phase modulation signal and sampling trigger timing. The reference signals for digital lock-in discrimination of gyroscopic spectrum are generated by using an external binary counter, and their phases were adjusted optimally by DSP software. The DSP compensated for fluctuations in laser source intensity and phase modulation index, using the signal spectrum extracted, and linearized the gyroscopic response. The measured resolution of rotation detection was 0.9 deg/s (with a full scale of 100 deg/s) and it agreed with the resolution in A/D conversion.

This paper describes a system that can enchance the speech quality degradation due to severe band limitation during speech transmission. We have already proposed a spectrum widening method that utilizes aliasing in sampling rate conversion and digital filtering for spectrum shaping. This paper proposes a new method that offers improved performance in terms of the spectrum distortion characteristics. Implementation procedures are clarified, and its performance is discussed. The proposed method can effectively enhance speech quality.

A multi-dimensional shaping scheme based on multi-level Maximum Average Weight (MAW)-codes is presented. One can reduce the average energy of transmitted signal, by using low energy signal points more frequently than high energy ones. The proposed scheme employs a multi-dimensional region of 2,4,6 and 8 dimensions; these regions are selected using a multi-level MAW-code. A multi-level MAW-code is a q-ary code and has unequal probability of the occurrence of a symbol. The scheme can achieve a shaping gain of 0.6-1.0 dB with small constellation expansion ratio and peak to average energy ratio. This scheme is based on a two-level table look up algorithm. Therefore, the less complexity of encoding/decoding can be realized.

This paper proposes a modified normalized LMS algorithm based on a long-term average of the reference input signal power. The reference input signal power for normalization is estimated by using two leaky integrators with a short and a long time constants. Computer simulation results compare the performance of the proposed algorithm with some previosuly proposed adaptive-step algorithms. The proposed algorithm converges faster than the conventional adaptive-step algorithms. Almost 30dB of the ERLE (Echo Return Loss Enhancement), which is comparable to the conventional algorithms, is achieved in noisy environments.

This paper condiders a problem of logecal configuration in reconfigurable VCDN (Virtual Circuit Data Networks) which is analyzed through a mimimax approach, and its objective is to minimize the largest delay on any logical link, measured in both queueing delay and propagation delay. The problem is formulated as a 0/1 mixed integer programming and analyzed by decomposing it into two subproblems, called routing and dimensioning problems, for which an efficient hauristic algorithm is proposed in an iterating process made beween the two subproblems for solution improvement. The algorithm is tested for its performance eveluation.

A new variable step size Least Mean Square (LMS) FIR adaptive filter algorithm (VSS-CC) is proposed. In the VSS-CC algorithm the step size adjustment (α) is controlled by using the correlation between the output error (e(n)) and the adaptive filter output ((n)). At small times, e(n) and (n) are correlated which will cause a large α providing faster tracking. When the algorithm converges, the correlation will result in a small size α to yield smaller misadjustments. Computer simulations show that the proposed VSS-CC algori thm achieves a better Echo Return Loss Enhancemen (ERLE) than a conventional NLMS Algorithm. The VSS-CC algorithm was also compared with another variable step algorithm, achieving the VSS-CC a better ERLE when the additive noise is incremented.

Keyword spotting is a fundamental approach to recognizing/understanding naturally and spontaneously spoken language. To spot acoustic events such as keywords, an overall spotting system, comprising acoustic models and decision thresholds, primarily needs to be optimized to minimize all spoting errors. However, in most conventional spotting systems, the acoustic models and the thresholds are separately and heuristically designed: There has not necessarily been a theoretical basis that has allowed one to design an overall system consistently. This paper introduces a novel approach to spotting, by proposing a new design method called Minimum SPotting Error learning (MSPE). MSPE is conceptually based on a recent discriminative learning theory, i.e., the Minimum Classification Error learning/Generalized Probabilistic Descent method (MCE/GPD); it features a rigorous framework for minimizing spotting error objectives. MSPE can be used in a wide range of pattern spotting applications, such as spoken phonemes, written characters as well as spoken words. Experimental results for a Japanese consonant spotting task clearly demonstrate the promising future of the proposed approach.

This paper presents a theoretically best algorithm within the framework of our image noise model for reconstructing 3-D from two views when all the feature points are on a planar surface. Pointing out that statistical bias is introduced if the least-squares scheme is used in the presence of image noise, we propose a scheme called renormalization, which automatically removes statistical bias. We also present an optimal correction scheme for canceling the effect of image noise in individual feature points. Finally, we show numerical simulation and confirm the effectiveness of our method.

An adaptive digital filter with adaptive sampling phase is proposed. The structure of the filter makes use of an adaptive delay device at the input of the filter. The algorithm is derived to determine the value of the delay and the filter coefficients by minimizing MSE (mean square error) between the desired signal and the filter output. The computer simulation of the convergence of the proposed adaptive filter with the input of sinusoidal wave and BPSK modulated wave are shown. According to the simulation, the MSE of the proposed adaptive delay algorithm is lower than that of the conventional LMS algorithm.

Data flow is a paradigm for concurrent computations in which a collection of parallel processes communicate asynchronously. For nondeterministic data flow networks many semantic models have been defined, however, it is complex to reason about the semantics of a network. In this paper, we introduce a transformation between data flow networks and the LOTOS specification language to make available theories and tools developed for process algebras for the semantic analysis based on traces of the networks. The transformation does not establish a one-to-one mapping between the traces of a data flow network and the LOTOS specification, but maps each network in a specification which usually contains more traces. The obtained system specification has the same set of traces as the corresponding network if they are finite, otherwise also non fair traces are included. Formal analysis and verification methods can still be applied to prove properties of the original data flow network, allowing in case of networks with finite traces to prove also network equivalence.

An optoelectronic technique to control both the amplitude and phase of a radio frequency (RF) signal is presented that uses two electrically controllable birefringence mode nematic liquid-crystal spatial light modulators (ECB mode nematic LC-SLMs). An experimental circuit was built and its performance was examined. The intensity could be changed down to -25 dB, and a phase shift of up to 240 degrees was achieved, by changing LC-SLM supplied voltages. Carrier-to-noise ratio (CNR) and intermodulation characteristics of an RF signal were measured. It was, for the first time, found that CNR was not degraded by the amplitude control and phase shift performed by the LC-SLMs.

The simulation of a specific absorption rate (SAR) with a temperature distribution becomes more important in the treatment planning for microwave hyperthermia. The simulation technique can also be used to estimate SAR distribution inside human body under hazardous electromagnetic (EM) field circumstances. In the simulation, to use exact permittivity of biological tissues becomes very important to obtain accurate SAR distribution. The permittivity of the medium is very sensitive to the temperature. Therefore, it is considered that the SAR distribution is also very sensitive to the tissue temperature. In this paper, SAR distribution is calculated using FDTD method considering tissue temperature under the electromagnetic (EM) field irradiation. Simulations of temperature distribution are also performed using heat transfer equation. In addition, temperature depending blood flow is taking into account to obtain temperature depending SAR distribution. The results can be used to estimate temperature depending heat generation which can be applied such as microwave hyperthermia treatment.

This paper describes the multiple access performance of parallel combinatory spread spectrum (PC/SS) communication systems in nonfading and Rayleigh fading multipath channels. The PC/SS systems can provide the high-speed data transmission capability by transmitting multiple pseudo-noise sequences out of a pre-assigned sequence set. The performance is evaluated in terms of average bit error rate (BER) by numerical computation. In nonfading white gaussian channel, the PC/SS systems are superior to conventional direct sequence spread spectrum (DS/SS) systems under the identical spreading factor condition. In Rayleigh fading channel, the performance of the PC/SS system without diversity is poorer than that of the DS/SS system. By including the explicit and implicit diversity, the performance of the PC/SS system becomes better than that of conventional DS/SS systems. A longer spreading sequence is assignable to a PC/SS system having the spreading factor equal to that in the conventional DS/SS system. Hence, the error control coding is easily. It is found that the PC/SS systems including diversity and Reed-Solomon coding improves the multiple access performance.

A transmission power control technique on a TDD-CDMA/TDMA system for wireless multimedia networks is proposed. The assumed network connects mobile terminals to a node of an ATM based high speed LAN through a radio central unit. Only human interface facilities are implemented into the terminal so that users access integrated services through the node of the network. The uplink (from a mobile terminal to a radio central unit) employs a CDMA scheme to transmit human interface signals (2.4kbit/s) and the downlink employs a TDMA scheme to transmit display interface signals (24 Mbit/s). Both the CDMA and the TDMA signals occupy the same frequency band. To mitigate bit error rate degradation due to the fading, the radio central unit estimates the impulse response of the channel from the received CDMA signals and controls the transmission power of the TDMA signals to compensate the fading attenuation. The bit error rate performance of the downlink with the proposed transmission power control is theoretically analyzed under several fading conditions. Numerical results using the Nakagami-m fading model and recent propagation measurements show that the proposed power control technique compensates the fading attenuation and improves the bit error rate performances. The bit error rate of the downlink is reduced from 10-2 to 10-5 at the symbol SNR of 20dB by employing the proposed transmission power control, which is less sensitive to the severity of the fading. Furthermore, the proposed transmission power control is implemented without increasing the terminal complexity because all the processing on the power control of the downlink is carried out only in the radio central unit.