In this paper, we propose an adaptive algorithm for an IIR adaptive line enhancer (ALE) based on constrained FIR ADF. The two constraints that are considered are the unit norm constraint and zeros constraint. By incorporating these two constraints, we show that the effect of a white disturbance signal on the convergence of the ALE is eliminated in the mean sense. The mathematical analysis for exact modeling also indicates that there exists a solution, which gives perfect estimation of the frequencies of the sinusoidal signals. Furthermore, by using a sufficiently small step size and introducing a control mechanism, the convergence of a second order ALE is guaranteed even if there is a sudden change in the frequency of the input sinusoidal signal. Simulation results that are presented, verify these properties.

To support video streaming over the current Internet, a media server usually needs to perform adaptive streaming in combination with congestion control. While existing streaming schemes are mainly designed for particular congestion algorithms and rate shaping techniques, this paper proposes a general buffer-driven adaptive streaming scheme based on control theory. Our scheme can be applied in combination with different congestion control algorithms and different rate shaping (and source coding) techniques. It is also amenable to analysis and easy to implement. Analytical and simulation results show that the proposed scheme gives satisfactory video quality under various Internet conditions while fully utilizing the available network bandwidth.

In this paper, a novel chrominance up-sampling algorithm for color post-processing is described. This scheme exploits so-called inter-chrominance coherence, i.e., luminance and chrominance signals share the same structural information. Usually luminance has higher spatial resolution than chrominance in compression coding standards. So the idea is to up-sample chrominance signals using the structural information extracted from the luminance.

This paper proposes a new blind algorithm effective for multiuser detection with an adaptive array antenna. The conventional blind algorithm, known as the Constant Modulus Algorithm (CMA), has two major drawbacks: (i) the convergence speed is not sufficiently fast for usual applications in mobile communications, and (ii) the algorithm is very likely to lock on the path with the largest received power, which means the signal with the second largest power can hardly be extracted. This paper introduces the Recursive Least Squares algorithm for CMA (RLS-CMA) in order to speed the convergence up, and additionally introduces the concept of the orthogonal projection into CMA so as to extract signals with weak power. The proposed CMA with Orthogonal Projection (CMA-OP) sequentially calculates the weight vector of each user under a constraint that the weight vector should be orthogonal to the estimated array response vectors of previously extracted users. Computer simulations demonstrate that the proposed scheme can operate properly in the Rayleigh fading channels under the two-user condition.

This paper proposes an adaptive radio parameter control scheme that utilizes an optimum radio parameter set comprising the maximum number of retransmissions in hybrid automatic repeat request (HARQ) in addition to the data modulation and channel coding scheme (MCS) according to the Quality of Service (QoS) requirements (i.e., the required packet error rate and delay) and propagation conditions such as the delay spread in the forward link of Orthogonal Frequency and Code Division Multiplexing (OFCDM) broadband wireless access. We elucidate by simulation evaluation that most of the optimum MCSs are common regardless of the delay requirement of traffic data, i.e., common between non-real time (NRT) and real-time (RT) class data. Concretely, the three MCSs of QPSK with the coding rate of R=1/2, 16QAM with R=1/2 and 3/4 are optimum ones, although the additional MCS of QPSK with R=1/3 is effective only for the RT class data in the lower received average received signal energy per symbol-to-background noise power density ratio (Es/N0) region. Furthermore, application of a much higher MCS set, 16QAM with R=5/6 and 64QAM with R=3/4, in addition to the three common MCSs improves the throughput under much higher Es/N0 conditions in a small delay spread environment. The simulation results show that the delay requirement, i.e., the maximum number of retransmissions, in HARQ does not affect the key radio parameter such as MCS, because of informative results such as a smaller number of retransmissions associated with a less-efficient MCS achieves a higher throughput than does using a more highly-efficient MCS allowing a larger number of retransmissions. Consequently, it is concluded that the proposed adaptive radio parameter control according to the QoS requirements substantially results in the selection of the optimum MCS irrespective of the delay requirement except for the extreme case where no retransmissions are allowed and for special propagation channel conditions.

A 3D display using super high-density multi-view images should enable reproduction of natural stereoscopic views. In the super multi-view display system, viewpoints are sampled at an interval narrower than the diameter of the pupil of a person's eye. With the parallax produced by a single eye, this system can pull out the accommodation of an eye to an object image. We are now working on a real-time view-interpolation system for the super multi-view 3D display. A multi-view camera using convergence capturing to prevent resolution degradation captures multi-view images of an object. Most of the data processing is used for view interpolation and rectification. View interpolation is done using a high-speed image-processing board with digital-signal-processor (DSP) chips or single instruction stream and multiple data streams (SIMD) parallel processor chips. Adaptive filtering of the epipolar plane images (EPIs) is used for the view-interpolation algorithm. The multi-view images are adaptively interpolated using the most suitable filters for the EPIs. Rectification, a preprocess, converts the multi-view images in convergence capturing into the ones in parallel capturing. The use of rectified multi-view images improves the processing speed by limiting the interpolation processing in EPI.

In this paper, we evaluate the average bit error rate (BER) and bits per symbol (BPS) performances of CDMA system employing the adaptive modulation according to the position of mobile station since the performance of CDMA system is highly dependent on user location especially in the downlink, and further we evaluate the overall performance by considering the user distribution at the level of cell. To evaluate the performance of the considered system in the forward link, we derive the distribution of the received signal to noise ratio (SNR) by Mobile Station located at arbitrary cell position with which we investigate the average BER and BPS performances of the common shared channel, one of the downlink channels of beyond IMT-2000 systems such as 1xEV-DO, 1xTEME and HSPDA. As a result, we observe that the CDMA system adopting the adaptive modulation can get about 2.47 bits per symbol at the cell level, while satisfying the required BER performance over all cell area.

This paper proposes a symbol-by-symbol-based multilevel transmit power control (MTPC) scheme for orthogonal frequency division multiplexing (OFDM) based adaptive modulation system (AMS) to achieve high quality broadband wireless transmission for high mobility terminals. In the proposed system, delay profile for each OFDM symbol is estimated by linearly extrapolating previously received delay profile information (DPI) sequence to improve tracking ability of OFDM based AMS with MTPC to the fast fading variation. Moreover, 2-branch reception diversity is applied to reduce dynamic range and variation speed of the multipath fading. Computer simulation confirms that the proposed system is effective in supporting higher mobility terminals with keeping high transmission quality.

The adaptive base station antenna is an attractive candidate for establishing high-speed and highly-reliable wireless communication systems. From a commercial viewpoint, since the cost and complexity of adaptive antennas depend on the number of elements, optimizing the antenna configuration while considering the propagation environment is necessary to reduce the number of elements. This paper first presents the Angle of Arrival (AOA) characteristics of delayed waves in a street microcell environment, typically used in urban microcell systems. Then the antenna configuration and antenna spacing suitable for the street microcell are investigated utilizing bit error rate (BER) performance simulations using practical delay profiles and AOAs. The effectiveness of bidirectional elements with respect to the BER performance is also investigated. As the results, we found that broadside array with the spacing of 2.5 wavelengths is suitable for adaptive base station antennas for high data-rate wireless systems placed in a street microcell environment. We also found that bidirectional elements alleviate the BER degradation due to the grating lobe of the antenna with wide element spacing without increasing the antenna size.

This paper presents laboratory and field experimental results of the coherent adaptive antenna array diversity (CAAAD) receiver employing receiver antenna-weight generation common to all Rake-combined paths (hereafter path-common weight generation method) in the W-CDMA reverse link, in order to elucidate the suitability of the path-common weight generation method in high-elevation antenna environments such as cellular systems with a macrocell configuration. Laboratory experiments using multipath fading simulators and RF phase shifters elucidate that even when the ratio of the target Eb/I0 of the desired to interfering users is Δ Eb/I0=-12 dB, the increase in the average transmit Eb/N0 employing the CAAAD receiver coupled with fast transmission power control (TPC) using outer-loop control from that for Δ Eb/I0=0 dB is within only 1.0 dB owing to the accurate beam and null steering associated with fast TPC. Furthermore, field experiments demonstrate that the required transmission power at the average block error rate (BLER) of 10-2 employing the CAAAD receiver with four antennas is reduced by more than 2 dB compared to that using a four-branch space diversity receiver using maximum ratio combining (MRC) with the fading correlation between antennas of 0 when Δ Eb/I0=-15 dB and that the loss in the required transmission power of the CAAAD receiver in the same situation as that in a single-user environment is approximately 1 dB. The field experimental results in an actual propagation environment suggest that the CAAAD receiver is effective in suppressing multiple access interference, thus decreasing the required transmission power when the gap in the direction of arrival between the desired user and interfering users is greater than approximately 20 degrees.

This paper proposes an Orthogonal Frequency Division Multiplexing system with Grouping Adaptive Modulation method (GAM-OFDM). The salient feature of the proposed system is to enable the reduction of required transmission bits for adaptive modulation information (AMI) that is required in the demodulation process at the receiver. This paper also proposes an efficient AMI transmission method for the GAM-OFDM system to enable the efficient transmission of AMI bits by using only two preamble symbols, and the Multi-Carrier Spectrum Spreading (MC-SS) technique to achieve the excellent performance of AMI transmission even under severe multi-path fading environments. This paper presents the various computer simulation results to verify the performance of proposed GAM-OFDM system.

Many types of adaptive algorithms based on the MMSE criterion for co-channel interference suppression in DS/CDMA systems have been studied in great detail. However, these algorithms have such a problem that the training speed is greatly dropped under the strong near-far problem. In this paper, we propose and analyze an adaptive filter based on the Maximum Signal to Interference and Noise Ratio (MSINR) criterion, called adaptive MSINR filter. This filter is basically equivalent to the adaptive filter based on the MMSE criterion. However, due to the structual difference, the convergence speed is greatly improved. Specifically, the de-spreading vector in this filter is so renewed as to maximize the Signal to Interference and Noise Ratio (SINR) by minimizing the de-spread interference and noise power under the condition that the de-spread desired signal power keeps constant. So the proposed filter uses the estimated interference and noise signal calculated by subtracting the estimated desired signal from the received signal. It is just the reason why the adaptive MSINR filter shows remarkable convergence speed. And to satisfy the constant signal power condition, the projection matrix onto the orthogonal complement of the desired signal space is used for the de-spreading vector. For the proposed filter, we analyze the convergence modes and also investigate the de-spread interfernce and noise power for calculating the theoretical SINR curve. Then, we conduct some computer simulations in order to show the difference between this filter and the conventional one in terms of the SINR convergence speed. As the result, we confirm that the adaptive filter based on the MSINR criterion achieves significant progress in terms of the SINR convergence speed.

This paper describes a rhythm pattern accuracy diagnosis system based on the rhythm pattern matching algorithm and a diagnosis feedback method by employing the SVM technique. A beat rhythm pattern is recorded by a PC and analyzed with an algorithm including cluster-analysis-based pattern matching. Rhythm performance is represented by a performance feature vector, which features note length deviation, note length instability, and tempo instability. The performance feature vector is effective for objectively evaluating the accuracy of rhythm patterns objectively. In addition, this system has the music experts' knowledge base, which is calculated from the performance feature vectors associated with the experts' subjective evaluation by listening to the performance. The system generates both an objective measuring report, and experts' comments for learners. Reproductivity of experts' comments is statistically indicated to be excellent for eight rhythm patterns, two tempo levels, and eight users. Reliability of experts' comments are also described considering the threshold of the decision function of SVM. Subjective evaluation of the system is carried out by fifteen users by a questionnaire using the SD method. As a result of factor analysis for the sixteen questions, four factors named "Audio-visual representation," "User-friendliness," "Reliability," and "Window representation," are extracted. Users' four factor scores indicate that the system is reliable and easy to use.

An adaptive modulated orthogonal frequency division multiplexing (OFDM) radio transmission scheme that enables efficient data transmission in multipath fading environments is newly proposed. This scheme can be used in standardized multimedia mobile access systems such as ETSI-BRAN, and ARIB-MMAC. It is based on estimating the delay spread and the carrier-to-noise power density ratio (C/N0). The estimation is done using channel estimation symbols that are inserted into the frames of the standard OFDM radio transmission scheme. Computer simulations show that the estimation method results in an average BER close to those when propagation characteristics are perfectly estimated. Furthermore, when the adaptive OFDM transmission scheme is based on BPSK, QPSK or 16 QAM, the average BER is almost close to that when BPSK-OFDM is only used, and the average transmission rate is 1.8 times as high. Using an error-correction code based on convolutional code results in an average BER lower than that with the BPSK- and QPSK-OFDM schemes.

A postprocessing algorithm is presented for blocking artifact reduction in block-coded images using the adaptive filters along the pattern of neighborhood blocks. Blocking artifacts appear as irregular high-frequency components at block boundaries, thereby reducing the noncorrelation between blocks due to the independent quantization process of each block. Accordingly, block-adaptive filtering is proposed to remove such components and enable similar frequency distributions within two neighborhood blocks and a high correlation between blocks. This type of filtering consists of inter-block filtering to remove blocking artifacts at the block boundaries and intra-block filtering to remove ringing noises within a block. First, each block is classified into one of seven classes based on the characteristics of the DCT coefficient and MV (motion vector) received in the decoder. Thereafter, adaptive intra-block filters, approximated to the normalized frequency distributions of each class, are applied adaptively according to the various patterns and frequency distributions of each block as well as the filtering directions in order to reduce the blocking artifacts. Finally, intra-block filtering is performed on those blocks classified as complex to reduce any ringing noise without blurring the edges. Experimental tests confirmed the effectiveness of the proposed algorithm.

A blind equalizer which uses the differential constant modulus algorithm (DCMA) is introduced. An anchored FIR equalizer applied to a first-order autoregressive channel and updated according to the DCMA is shown to converge to the inverse of that channel regardless of the initial tap-weights and the gain along the direct path.

This paper presents the theoretical analysis of subband adaptive array combining cyclic prefix transmission scheme (SBAA-CP) in multipath fading environment. The exact expressions for optimal weights, array outputs and the output signal to interference plus noise ratio (SINR) are derived. The analysis shows that use of the cyclic prefix data transmission scheme can significantly improve the performance of subband adaptive array (SBAA). An example of implementing SBAA-CP as a software antenna is also presented.

This paper presents an adaptive burst-mode M-QAM modem architecture suitable for variable rate broadband wireless packet data networks. The core signal processing functions for the modem are common to all constellations resulting in an efficient hardware architecture for field programmable gate array (FPGA) implementation.

Computing the Eigen Value Decomposition (EVD) of a symmetric matrix is a frequently encountered problem in adaptive (or smart or software) antenna signal processing, for example, super resolution DOA (Direction Of Arrival) estimation algorithms such as MUSIC (MUltiple SIgnal Classification) and ESPRIT (Estimation of Signal Parameters via Rotational Invariance Technique). In this paper the hardware architecture of the fast EVD processor of symmetric correlation matrices for the application of an adaptive antenna technology such as DOA estimation is proposed and the basic idea is also presented. Cyclic Jacobi method is well known for the simplest algorithm and easily implemented but its convergence time is slower than other factorization algorithm like QR-method. But if considering the fast parallel computation of the EVD with a hardware architecture like ASIC (Application Specific Integrated Circuit) or FPGA (Field Programmable Gate Array), the Jacobi method can be a appropriate solution, since it offers a quite higher degree of parallelism and easier implementation than other factorization algorithms. This paper computes the EVD using a Jacobi-type method, where the vector rotations and the angles of the rotations are obtained by CORDIC (COordinate Rotation DIgital Computer). The hardware architecture suitable for ASIC or FPGA with fixed-point arithmetic is presented. Because it consists of only shift and add operations, this hardware friendly feature provides easy and efficient implementation. In this paper, the computational load, the estimate of circuit scale and expected performance are discussed and the validation of fixed-point arithmetic for the practical application to MUSIC DOA estimation is examined.

This paper investigates an accurate channel estimation method using the common pilot channel (CPICH) in addition to a dedicated pilot channel (PICH) when the fading correlation between the dedicated PICH and CPICH is high, and clarifies the area in which the proposed channel estimation method is effective for adaptive antenna array transmit diversity (AAA-TD) in the forward link. Computer simulation results elucidate that although a more precise channel estimation is possible by using the primary-CPICH (P-CPICH) transmitted from an omni-directional antenna in addition to the dedicated PICH for the area where the distance, d, between a base station and a mobile terminal is longer than approximately 200 m, no improvement is obtained for the area where the value of d is shorter than approximately 200 m. Meanwhile, by employing the secondary-CPICH (S-CPICH) transmitted with several directional beams in addition to the dedicated PICH, the required average received Eb/N0 at the average BER of 10-3 is decreased by approximately 0.4 (0.2-0.4) dB compared to the channel estimation method using only the dedicated PICH regardless of the value of d when the number of antennas is 4 (8).