IEICE Transactions on Fundamentals of Electronics, Communications and Computer Sciences
Online ISSN : 1745-1337
Print ISSN : 0916-8508
Volume E96.A, Issue 4
Displaying 1-15 of 15 articles from this issue
Regular Section
  • Masahito TOGAMI, Yohei KAWAGUCHI, Yasunari OBUCHI
    Article type: PAPER
    Subject area: Engineering Acoustics
    2013 Volume E96.A Issue 4 Pages 749-761
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    This paper proposes a novel multichannel speech enhancement technique for reverberant rooms that is effective when noise sources are spatially stationary, such as a projector fan noise, an air-conditioner noise, and unwanted speech sources at the back of microphones. Speech enhancement performance of the conventional multichannel Wiener filter (MWF) degrades when the Signal-to-Noise Ratio (SNR) of the current microphone input signal changes from the noise-only period. Furthermore, the MWF structure is computationally inefficient, because the MWF updates the whole spatial beamformer periodically to track switching of the speakers (e.g. turn-taking). In contrast to the MWF, the proposed method reduces noise independently of the SNR. The proposed method has a novel two-stage structure, which reduces noise and distortion of the desired source signal in a cascade manner by using two different beamformers. The first beamformer focuses on noise reduction without any constraint on the desired source, which is insensitive to SNR variation. However, the output signal after the first beamformer is distorted. The second beamformer focuses on distortion reduction of the desired source signal. Theoretically, complete elimination of distortion is assured. Additionally, the proposed method has a computationally efficient structure optimized for spatially stationary noise reduction problems. The first beamformer is updated only when the speech enhancement system is initialized. Only the second beamformer is updated periodically to track switching of the active speaker. The experimental results indicate that the proposed method can reduce spatially stationary noise source signals effectively with less distortion of the desired source signal even in a reverberant conference room.
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  • Taizo SUZUKI, Hirotomo ASO
    Article type: PAPER
    Subject area: Digital Signal Processing
    2013 Volume E96.A Issue 4 Pages 762-768
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    This paper presents an M-channel (M=2n (n ∈ N)) integer discrete cosine transforms (IntDCTs) based on fast Hartley transform (FHT) for lossy-to-lossless image coding which has image quality scalability from lossy data to lossless data. Many IntDCTs with lifting structures have already been presented to achieve lossy-to-lossless image coding. Recently, an IntDCT based on direct-lifting of DCT/IDCT, which means direct use of DCT and inverse DCT (IDCT) to lifting blocks, has been proposed. Although the IntDCT shows more efficient coding performance than any conventional IntDCT, it entails many computational costs due to an extra information that is a key point to realize its direct-lifting structure. On the other hand, the almost conventional IntDCTs without an extra information cannot be easily expanded to a larger size than the standard size M=8, or the conventional IntDCT should be improved for efficient coding performance even if it realizes an arbitrary size. The proposed IntDCT does not need any extra information, can be applied to size M=2n for arbitrary n, and shows better coding performance than the conventional IntDCTs without any extra information by applying the direct-lifting to the pre- and post-processing block of DCT. Moreover, the proposed IntDCT is implemented with a half of the computational cost of the IntDCT based on direct-lifting of DCT/IDCT even though it shows the best coding performance.
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  • Guifen TIAN, Xin JIN, Satoshi GOTO
    Article type: PAPER
    Subject area: Digital Signal Processing
    2013 Volume E96.A Issue 4 Pages 769-779
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    High Efficiency Video Coding (HEVC) outperforms H.264 High Profile with bitrate saving of about 43%, mostly because block sizes for hybrid prediction and residual encoding are recursively chosen using a quadtree structure. Nevertheless, the exhaustive quadtree-based partition is not always necessary. This paper takes advantage of all-zero residual blocks at every quadtree depth to accelerate the prediction and residual encoding processes. First, we derive a near-sufficient condition to detect variable-sized all-zero blocks (AZBs). For these blocks, discrete cosine transform (DCT) and quantization can be skipped. Next, using the derived condition, we propose an early termination technique to reduce the complexity for motion estimation (ME). More significantly, we present a two-dimensional pruning technique based on AZBs to constrain prediction units (PU) that contribute negligibly to rate-distortion (RD) performance. Experiments on a wide range of videos with resolution ranging from 416×240 to 4k×2k, show that the proposed scheme can reduce computational complexity for the HEVC encoder by up to 70.46% (50.34% on average), with slight loss in terms of the peak signal-to-noise ratio (PSNR) and bitrate. The proposal also outperforms other state-of-the-art methods by achieving greater complexity reduction and improved bitrate performance.
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  • Guifen TIAN, Xin JIN, Satoshi GOTO
    Article type: PAPER
    Subject area: Digital Signal Processing
    2013 Volume E96.A Issue 4 Pages 780-789
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    The quadtree-based variable block sized prediction makes the biggest contribution for dramatically improved coding efficiency in the new video coding standard named HEVC. However, this technique takes about 75-80% computational complexity of an HEVC encoder. This paper brings forward an adaptive scheme that exploits temporal, spatial and transform-domain features to speed up the original quadtree-based prediction, targeting at high resolution videos. Before encoding starts, analysis on utilization ratio of each coding depth is performed to skip rarely adopted coding depths at frame level. Then, texture complexity (TC) measurement is applied to filter out none-contributable coding blocks for each largest coding unit (LCU). In this step, a dynamic threshold setting approach is proposed to make filtering adaptable to videos and coding parameters. Thirdly, during encoding process, sum of absolute quantized residual coefficient (SAQC) is used as criterion to prune useless coding blocks for both LCUs and 32×32 blocks. By using proposed scheme, motion estimation is performed for prediction blocks within a narrowed range. Experiments show that proposed scheme outperforms existing works and speeds up original HEVC by a factor of up to 61.89% and by an average of 33.65% for 4kx2k video sequences. Meanwhile, the peak signal-to-noise ratio (PSNR) degradation and bit increment are trivial.
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  • Yosuke SUGIURA, Arata KAWAMURA, Youji IIGUNI
    Article type: PAPER
    Subject area: Digital Signal Processing
    2013 Volume E96.A Issue 4 Pages 790-795
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    This paper proposes a new adaptive comb filter which automatically designs its characteristics. The comb filter is used to eliminate a periodic noise from an observed signal. To design the comb filter, there exists three important factors which are so-called notch frequency, notch gain, and notch bandwidth. The notch frequency is the null frequency which is aligned at equally spaced frequencies. The notch gain controls an elimination quantity of the observed signal at notch frequencies. The notch bandwidth controls an elimination bandwidth of the observed signal at notch frequencies. We have previously proposed a comb filter which can adjust the notch gain adaptively to eliminate the periodic noise. In this paper, to eliminate the periodic noise when its frequencies fluctuate, we propose the comb filter which achieves the adaptive notch gain and the adaptive notch bandwidth, simultaneously. Simulation results show the effectiveness of the proposed adaptive comb filter.
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  • Jin Ho HAN, Jong Hwan PARK, Dong Hoon LEE
    Article type: PAPER
    Subject area: Cryptography and Information Security
    2013 Volume E96.A Issue 4 Pages 796-806
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    Broadcast encryption scheme with personalized messages (BEPM) is a new primitive that allows a broadcaster to encrypt both a common message and individual messages. BEPM is necessary in applications where individual messages include information related to user's privacy. Recently, Fujii et al. suggested a BEPM that is extended from a public key broadcast encryption (PKBE) scheme by Boneh, Gentry, and Waters. In this paper, we point out that 1) Conditional Access System using Fujii et al.'s BEPM should be revised in a way that decryption algorithm takes as input public key as well, and 2) performance analysis of Fujii et al.'s BEPM should be done depending on whether the public key is transmitted along with ciphertext or stored into user's device. Finally, we propose a new BEPM that is transmission-efficient, while preserving O(1) user storage cost. Our construction is based on a PKBE scheme suggested by Park, Kim, Sung, and Lee, which is also considered as being one of the best PKBE schemes.
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  • Changyong PAN, Linglong DAI, Zhixing YANG
    Article type: PAPER
    Subject area: Communication Theory and Signals
    2013 Volume E96.A Issue 4 Pages 807-813
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    Time domain synchronous orthogonal frequency division multiplexing (TDS-OFDM) has higher spectral efficiency than the standard cyclic prefix OFDM (CP-OFDM) OFDM by replacing the random CP with the known training sequence (TS), which could be also used for synchronization and channel estimation. However, TDS-OFDM requires suffers from performance loss over fading channels due to the iterative interference cancellation has to be used to remove the mutual interferences between the TS and the useful data. To solve this problem, the novel TS based OFDM transmission scheme, referred to as the unified time-frequency OFDM (UTF-OFDM), is proposed in which the time-domain TS and the frequency-domain pilots are carefully designed to naturally avoid the interference from the TS to the data without any reconstruction. The proposed UTF-OFDM based flexible frame structure supports effective channel estimation and reliable channel equalization, while imposing a significantly lower complexity than the TDS-OFDM system at the cost of a slightly reduced spectral efficiency. Simulation results demonstrate that the proposed UTF-OFDM substantially outperforms the existing TDS-OFDM, in terms of the system's achievable bit error rate.
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  • Tomohiro KAWAMURA, Takafumi KANAZAWA, Toshimitsu USHIO
    Article type: PAPER
    Subject area: Concurrent Systems
    2013 Volume E96.A Issue 4 Pages 814-820
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    Evolutionary stability has been discussed as a fundamental issue in single-criterion games. We extend evolutionarily and neutrally stable strategies to multicriteria games. Keeping in mind the fact that a payoff is given by a vector in multicriteria games, we provide several concepts which are coincident in single-criterion games based on partial vector orders of payoff vectors. We also investigate the hierarchical structure of our proposed evolutionarily and neutrally stable strategies. Shapley had introduced concepts such as strong and weak equilibria. We discuss the relationship between these equilibria and our proposed evolutionary stability.
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  • Lingjuan WU, Ryan KASTNER, Bo GU, Dunshan YU
    Article type: LETTER
    Subject area: Engineering Acoustics
    2013 Volume E96.A Issue 4 Pages 821-823
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    Design of acoustic modem becomes increasingly important in underwater sensor networks' development. This paper presents the design of a reconfigurable acoustic modem, by defining modulation and demodulation as reconfigurable modules, the proposed modem changes its modulation scheme and data rate to provide reliable and energy efficient communication. The digital system, responsible for signal processing and control, is implemented on Xilinx Virtex5 FPGA. Hardware and software co-verification shows that the modem works correctly and can self-configure to BFSK and BPSK mode. Partial reconfiguration design method improves flexibility of algorithm design, and slice, LUT, register, DSP, RAMB are saved by 17%, 25%, 22%, 25%, 25% respectively.
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  • Tatsuki HYODO, Gaku ASAKURA, Kiwamu TSUKADA, Masashi KATO
    Article type: LETTER
    Subject area: Noise and Vibration
    2013 Volume E96.A Issue 4 Pages 824-825
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    This letter proposes an analog active noise control (ANC) circuit with an all-pass filter (APF). To improve performance of the previously reported analog ANC circuit, we inserted an APF to the circuit in order to fit phases of a noise and an electrical signal in the circuit. As a result, we confirmed improvement of the noise canceling effect of the analog ANC circuit.
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  • Kyung-In KANG, Kyun-Sang PARK, Jong-Tae LIM
    Article type: LETTER
    Subject area: Systems and Control
    2013 Volume E96.A Issue 4 Pages 826-829
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    In this letter, we consider the ultimate boundedness of the singularly perturbed system with measurement noise. The composite controller is commonly used to regulate the singularly perturbed system. However, in the presence of measurement noise, the composite controller does not guarantee the ultimate boundedness of the singularly perturbed system. Thus, we propose the modified composite controller to show the ultimate boundedness of the singularly perturbed system with measurement noise.
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  • Hua ZHAO, Songfeng LU, Yan LIU
    Article type: LETTER
    Subject area: Algorithms and Data Structures
    2013 Volume E96.A Issue 4 Pages 830-832
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    Fused Blocked Pattern Matching is a kind of approximate matching based on Blocked Pattern Matching, and can be used in identification of fused peptides in tumor genomes. In this paper, we propose a new algorithm for fused blocked pattern matching. We give a comparison between Julio's solution and ours, which shows our algorithm is more efficient.
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  • Dan ZOU, Yong DOU, Rongchun LI
    Article type: LETTER
    Subject area: Algorithms and Data Structures
    2013 Volume E96.A Issue 4 Pages 833-834
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    We present a new approach for sparse Cholesky factorization on a heterogeneous platform with a graphics processing unit (GPU). The sparse Cholesky factorization is one of the core algorithms of numerous computing applications. We tuned the supernode data structure and used a parallelization method for GPU tasks to increase GPU utilization. Results show that our approach substantially reduces computational time.
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  • Masashi NOMURA, Shigemasa TAKAI
    Article type: LETTER
    Subject area: Concurrent Systems
    2013 Volume E96.A Issue 4 Pages 835-839
    Published: April 01, 2013
    Released on J-STAGE: April 01, 2013
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    In this paper, we study decentralized supervisory control of timed discrete event systems, where we adopt the OR rule for fusing local enablement decisions and the AND rule for fusing local enforcement decisions. For any specification language satisfying a certain assumption, we propose a method for constructing a decentralized supervisor that achieves its sublanguage. The proposed method does not require computing the achieved sublanguage.
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