This paper presents a simple but high resolution DOA estimation method using second-order differential of MUSIC spectrum. MUSIC method is paid attention as one of "Superresolution" DOA estimation methods because of their brilliant characteristics, however MUSIC also has the problem of estimation accuracy in severe environments like low SNR, small number of snapshots, or incident waves from closely-spaced angles. Especially the case of two or more incident waves from closely-spaced angles, MUSIC often fails in making spectrum peaks that leads inaccurate DOA estimation. We pay attention to the fact that the second-order differential of MUSIC spectrum makes negative peaks around the original DOAs even when MUSIC spectrum does not make peaks there. We try to estimate DOAs not by MUSIC spectrum but by the second-order differential of the MUSIC spectrum, and to find its peaks for being estimated DOAs. The performance of the present method is evaluated in compared with MUSIC and Root-MUSIC methods through computer simulations and experiments.

This paper proposes a new blind adaptive MLSE equalizer for frequency selective mobile radio channels. The proposed equalizer performs channel estimation for each survivor path of the Viterbi algorithm (VA), and restricts the number of symbol candidates for the channel estimation in order to reduce prohibitive complexity. In such channel estimation, autocorrelation matrices of the symbol candidates are likely to become singular, which increases the estimation error. To cope with the singularity, the proposed equalizer employs a recursive channel estimation algorithm using the Moore-Penrose generalized inverse of the autocorrelation matrix. As another problem, the blind channel estimation can yield plural optimal estimates of a channel impulse response, and the ambiguity of the estimates degrades the BER performance. To avoid this ambiguity, the proposed equalizer is enhanced so that it can take advantage of the fractional sampling. The enhanced equalizer performs symbol-spaced channel estimation for each fractional sampling phase. This equalizer combines separate channel estimation errors, and provides the sum to the VA processor as the branch metric, which tremendously reduces the probability that a correct estimate turns into a false one. Computer simulation demonstrates the effectiveness of the proposed equalizers in the frequency selective fading channels.

This paper proposes low-complexity blind detection for orthogonal frequency division multiplexing (OFDM) systems with the differential space-time block code (DSTBC) under time-varying frequency-selective Rayleigh fading. The detector employs the maximum likelihood sequence estimation (MLSE) in cooperation with the blind linear prediction (BLP), of which prediction coefficients are determined by the method of Lagrange multipliers. Interpolation of channel frequency responses is also applied to the detector in order to reduce the complexity. A complexity analysis and computer simulations demonstrate that the proposed detector can reduce the complexity to about a half, and that the complexity reduction causes only a loss of 1 dB in average E_b/N₀ at BER of 10^-3 when the prediction order and the degree of polynomial approximation are 2 and 1, respectively.

During these years, we have been focusing on developing ultra high-data-rate wireless access systems for future wireless multimedia communications. One of such kind of systems is called DPC-OF/TDMA (dynamic parameter controlled orthogonal frequency and time division multiple access) which targets at beyond 100 Mbps data rate. In order to support higher data rates, e.g., several hundreds of Mbps or even Gbps for future wireless multimedia applications (e.g., streaming video and file transfer), it is necessary to enhance DPC-OF/TDMA system based on MIMO-OFDM (multiple-input multiple-output orthogonal frequency division multiplexing) platform. In this paper, we propose an enhanced DPC-OF/TDMA system based on Multi-Layer MIMO-OFDM scheme which combines both diversity and multiplexing in order to exploit potentials of both techniques. The performance investigation shows the proposed scheme has better performance than its counterpart based on full-multiplexing MIMO-OFDM scheme. In addition to the Exhaustive Detection (EXD) scheme which applies the same detection algorithm on each subcarrier independently, we propose the Reduced-Complexity Detection (RCD) scheme. The complexity reduction is achieved by exploiting the suboptimal Layer Detection Order and subcarrier correlation. The simulation results show that huge complexity can be reduced with very small performance loss, by using the proposed detection scheme. For example, 60.7% complexity can be cut off with only 1.1 dB performance loss for the 88 enhanced DPC-OF/TDMA system.

Distributed Video Coding (DVC), based on the theorems proposed by Slepian-Wolf and Wyner-Ziv, is attracting attention as a new paradigm for video compression. Some of the DVC systems use intra-frame compression based on discrete cosine transform (DCT). Unfortunately, conventional DVC systems have low affinity with DCT. In this paper, we propose a wavelet-based DVC scheme that utilizs current JPEG 2000 standard. Accordingly, the scheme has scalability with regard to resolution and quality. In addition, we propose two methods to increase the coding gain of the new DVC scheme. One is the introduction of a Gray code, and the other method involves optimum quantization. An interesting point is that though our proposed method uses Gray code, it still achieves quality scalability. Tests confirmed that the PSNR is increased about 5 [dB] by the two methods, and the PSNR of the new scheme (with methods) is about 1.5-3 [dB] higher than that of conventional JPEG 2000.

An encryption scheme is proposed that considers hierarchies in media, such as text, images, sound, and so on, in a composite multimedia content to enable versatile access control. In the proposed scheme, a content provider has only one managed key (the master key) for a particular composite multimedia content, and an user who is permitted to access a reserved content entities in the composite content receives only one key that is subordinately generated from the master key. Another key generated from the identical master key is delivered to another user, and this permits the user to access different entities. This scheme introduces a new key concept, namely "unusable key," to keep all entities encrypted in a particular medium and to simultaneously decrypt several entities in other media. The other new key, "numbering key," is also used in this scheme to support simultaneous partial decryption of multiple images that are coded with a scalable coding technology. Simulation results show the effectiveness of the proposed scheme; in particular, the length of the managed master key and that of keys to be delivered to users are small.

In recent years, digital watermarking has become a popular technique for hiding information in digital images to help protect against copyright infringement. In this paper we develop a high quality and robust watermarking algorithm that combines the advantages of block-based permutation with that of neighboring coefficient embedding. The proposed approach uses the relationship between the coefficients of neighboring blocks to hide more information into high frequency blocks without causing serious distortion to the watermarked image. In addition, an extraction method for improving robustness to mid-frequency filter attacks is proposed. Our experimental results show that the proposed approach is very effective in achieving perceptual imperceptibility. Moreover, the proposed approach is robust to a variety of signal processing operations, such as compression (JPEG), image cropping, sharpening, blurring, and brightness adjustments. The robustness is especially evident under blurring attack.

A novel pedestrian tracking scheme based on a particle filter is proposed, which adopts a skeleton model of a pedestrian for a state space model and distance transformed images for likelihood computation. The 6-stick skeleton model used in the proposed approach is very distinctive in representing a pedestrian simply but effectively. By the experiment using the real sequences provided by PETS, it is shown that the target pedestrian is tracked adequately by the proposed approach with a simple silhouette extraction method which consists of only background subtraction, even if the tracking target moves so complicatedly and is often so cluttered by other obstacles that the pedestrian can not be tracked by the conventional methods. Moreover, it is demonstrated that the proposed scheme can track the multiple targets in the complex case that their trajectories intersect.

3-D sound using head-related transfer functions (HRTFs) is applicable to embedded systems such as portable devices, since it can create spatial sound effect without multichannel transducers. Low-order modeling of HRTF with an IIR filter is effective for the reduction of the computational load required in embedded applications. Although modeling of HRTFs with IIR filters has been studied earnestly, little attention has been paid to sound movement with IIR filters, which is important for practical applications of 3-D sound. In this paper, a practical method for sound movement is proposed, which utilizes time-varying IIR filters and variable delay filters. The computational cost for sound movement is reduced by about 50% with the proposed method, compared to conventional low-order FIR implementation. In order to facilitate efficient implementation of 3-D sound movement, tradeoffs between the subjective quality of the output sound and implementation parameters such as the size of filter coefficient database and the update period of filter coefficients are also discussed.

Existing standard speech coders can provide high quality speech communication. However, they tend to degrade the performance of automatic speech recognition (ASR) systems that use the reconstructed speech. The main cause of the degradation is in that the linear predictive coefficients (LPCs), which are typical spectral envelope parameters in speech coding, are optimized to speech quality rather than to the performance of speech recognition. In this paper, we propose a speech coder using mel-frequency cepstral coefficients (MFCCs) instead of LPCs to improve the performance of a server-based speech recognition system in network environments. To develop the proposed speech coder with a low-bit rate, we first explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel errors. As a result, we propose an 8.7 kbps MFCC-based CELP coder. It is shown that the proposed speech coder has a comparable speech quality to 8 kbps G.729 and the ASR system using the proposed speech coder gives the relative word error rate reduction by 6.8% as compared to the ASR system using G.729 on a large vocabulary task (AURORA4).

We propose a gradient-limited affine projection algorithm (GL-APA), which can achieve fast and double-talk-robust convergence in acoustic echo cancellation. GL-APA is derived from the M-estimation-based nonlinear cost function extended for evaluating multiple error signals dealt with in the affine projection algorithm (APA). By considering the nonlinearity of the gradient, we carefully formulate an update equation consistent with multiple input-output relationships, which the conventional APA inherently satisfies to achieve fast convergence. We also newly introduce a scaling rule for the nonlinearity, so we can easily implement GL-APA by using a predetermined primary function as a basis of scaling with any projection order. This guarantees a linkage between GL-APA and the gradient-limited normalized least-mean-squares algorithm (GL-NLMS), which is a conventional algorithm that corresponds to the GL-APA of the first order. The performance of GL-APA is demonstrated with simulation results.

In this paper, theoretical properties of deinterlacer banks are analyzed. Deinterlacer banks are novel filter banks in the sense that a progressive video sequence is separated into two progressive video sequences of a half frame rate and, furthermore, interlaced sequences are produced as intermediate data. Unlike the conventional filter banks, our deinterlacer banks are constructed in a way unique to multidimensional systems by using invertible deinterlacers, which the authors have proposed before. The system is a kind of shift-varying filter banks and it was impossible to derive the optimal bit-allocation control without any equivalent parallel filter banks. This paper derives an equivalent polyphase matrix representation of the whole system and its equivalent parallel structure, and then shows the optimal rate allocation for the deinterlacer banks. Some experimental results justify the effectiveness of the optimal rate allocation through our theoretical analysis.

In nature an unbalanced clock tree exists in a SoC because the clock sinks of IPs have distinct input capacitive loads and internal delays. The construction of a bottom-up RLC clock tree with minimal clock delay and zero skew is crucial to ensure good SoC performance. This study proves that an RLC clock tree construction always has no zero skew owing to skew upward propagation. Specifically, this study proposes the insertion of two unit-size buffers associated with the binary search for a tapping point into each pair of subtrees to interrupt the non-zero skew upward propagation. This technique enables reliable construction of a buffered RLC clock tree with zero skew. The effectiveness of the proposed approach is demonstrated by assessing benchmarks.

In this paper, we propose an effective asynchronous datapath synthesis system to optimize statistical performance of asynchronous systems. The proposed algorithm is a heuristic method which simultaneously performs scheduling and resource binding. During the design process, decisions will be made based on the statistical schedule length analysis. It is demonstrated that asynchronous datapaths with the reduced mean total computation time are successfully synthesized for some datapath synthesis benchmarks.

Digital watermarking is a technique that aims at hiding a message signal in a multimedia signal for copyright claim, authentication, device control, or broadcast monitoring, etc. In this paper, we focus on embedding watermarks into still images, where the watermarks themselves can be binary sequences or grayscale images. We propose to scramble the watermark bits with pseudo-noise (PN) or orthogonal codes before they are embedded into an image. We also try to incorporate error correction coding (ECC) into the watermarking scheme, anticipating reduction of the watermark bit error rate (WBER). Due to the similarity between the PN/orthogonal-coded watermarking and the spread spectrum communication, it is natural that, following similar derivations regarding data BER in digital communications, we derive certain explicit quantitative relationships regarding the tradeoff between the WBER, the watermark capacity (i.e. the number of watermark bits) and the distortion suffered by the original image, which is measured in terms of the embedded image's signal-to-noise ratio (abbreviated as ISNR). These quantitative relationships are compactly summarized into a so-called tradeoff triangle, which constitutes the major contribution of this paper. For the embedding of grayscale watermarks, an unequal error protection (UEP) scheme is proposed to provide different degrees of robustness for watermark bits of different degrees of significance. In this UEP scheme, optimal strength factors for embedding different watermark bits are sought so that the mean squared error suffered by the extracted watermark, which is by itself a grayscale image, is minimized while a specified ISNR is maintained.

The 2-dimensional (2-D) Hilbert curve is a one-to-one mapping between 2-D space and one-dimensional (1-D) space. It is studied actively in the area of digital image processing as a scan technique (Hilbert scan) because of its property of preserving the spacial relationship of the 2-D patterns. Currently there exist several Hilbert scan algorithms. However, these algorithms have two strict restrictions in implementation. First, recursive functions are used to generate a Hilbert curve, which makes the algorithms complex and computationally expensive. Second, both sides of the scanned rectangle must have same size and each size must be a power of two, which limits the application of the Hilbert scan greatly. In this paper, a Pseudo-Hilbert scan algorithm based on two look-up tables is proposed. The proposed method improves the Hilbert scan to be suitable for real-time processing and general application. The simulation indicates that the Pseudo-Hilbert scan can preserve point neighborhoods as much as possible and take advantage of the high correlation between neighboring lattice points. It also shows competitive performance of the Pseudo-Hilbert scan in comparison with other scan techniques.

Localization of mobile terminals has received considerable attention in wireless communications. In this letter, we present a covariance shaping least squares (CSLS) estimator using time-of-arrival measurements of the signal from the mobile station received at three or more base stations. It is shown that the CSLS estimator yields better performance than the other LS estimators at low signal-to-noise ratio conditions.

In this paper, based on the variation of the modified Booth encoding method, an efficient modified Booth multiplier design method for predetermined coefficient groups is proposed. In the case of pulse-shaping filter design used in CDMA, it is shown that by the proposed method, area and power consumption can be reduced up to 44% and 48%, respectively, compared with the conventional designs. Also, it is shown that in the case of 128-point radix-2⁴ FFT, the area and power consumption can be reduced by 18% and 36%, respectively.

Local weight distribution is the weight distribution of minimal codewords in a linear code. We give the local weight distribution of the (256, 93) third-order binary Reed-Muller code. For the computation, a coset partitioning algorithm is modified by using a binary shift invariance property. This reduces the time complexity by about 1/256 for the code. A necessary and sufficient condition for minimality in Reed-Muller codes is also presented.