• Title/Summary/Keyword: SNR(Signal to noise ratio)

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A time delay estimation method using canonical correlation analysis and log-sum regularization (로그-합 규준화와 정준형 상관 분석을 이용한 시간 지연 추정에 관한 연구)

  • Lim, Jun-Seok;Pyeon, Yong-Gook;Lee, Seokjin;Cheong, MyoungJun
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.4
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    • pp.279-284
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    • 2017
  • The localization of sources has a numerous number of applications. To estimate the position of sources, the relative time delay between two or more received signals for the direct signal must be determined. Although the GCC (Generalized Cross-Correlation) method is the most popular technique, an approach based on CCA (Canonical Correlation Analysis) was also proposed for the TDE (Time Delay Estimation). In this paper, we propose a new adaptive algorithm based on CCA in order to utilized the sparsity in the eigenvector of CCA based time delay estimator. The proposed algorithm uses the eigenvector corresponding to the maximum eigenvalue with log-sum regularization in order to utilize the sparsity in the eigenvector. We have performed simulations for several SNR(signal to noise ratio)s, showing that the new CCA based algorithm can estimate the time delays more accurately than the conventional CCA and GCC based TDE algorithms.

Robust Audio Watermarking Algorithm with Less Deteriorated Sound (음질 열화를 줄이고 공격에 강인한 오디오 워터마킹 알고리듬)

  • Kang, Myeong-Su;Cho, Sang-Jin;Chong, Ui-Pil
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.7
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    • pp.653-660
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    • 2009
  • This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.

Singular Value Decomposition based Noise Reduction Technique for Dynamic PET I mage : Preliminary study (특이값 분해 기반 Dynamic PET 영상의 노이즈 제거 기법 : 예비 연구)

  • Pyeon, Do-Yeong;Kim, Jung-Su;Baek, Cheol-Ha;Jung, Young-Jin
    • Journal of radiological science and technology
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    • v.39 no.2
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    • pp.227-236
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    • 2016
  • Dynamic positron emission tomography(dPET) is widely used medical imaging modality that can provide both physiological and functional neuro-image for diagnosing various brain disease. However, dPET images have low spatial-resolution and high noise level during spatio-temporal analysis (three-dimensional spatial information + one-dimensional time information), there by limiting clinical utilization. In order to overcome these issues for the spatio-temporal analysis, a novel computational technique was introduced in this paper. The computational technique based on singular value decomposition classifies multiple independent components. Signal components can be distinguished from the classified independent components. The results show that signal to noise ratio was improved up to 30% compared with the original images. We believe that the proposed computational technique in dPET can be useful tool for various clinical / research applications.

A Study on Optimal Bit Loading Algorithms for Discrete MultiTone ADSL (DMT 변조방식을 사용하는 ADSL에서의 최적 비트 할당 방식 연구)

  • 이철우;박광철;윤기방;장수영;김기두
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.39 no.4
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    • pp.395-402
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    • 2002
  • In the conventional public switched telephone network(PSTN), there are various types of modulation that can be used in ADSL to offer fast data communication, two of which are CAP(Carrierless Amplitude Phase) and DMT(Discrete MultiTone). As we consider the current situation, DMT is getting more predominant in the market than CAP. One of the reasons is that it gives high performance in spite of its high complexity Since DMT divides the full range of bandwidth into 256 sub-channels, it can be highly adaptive in the circumstances, where the problems of attenuation and noise caused by the propagation distance are very crucial. In this paper, a new bit loading algorithm for DMT modulation is proposed. The proposed algorithm can be efficiently implemented in a way that it requires less computation than the conventional modulation techniques. In contrast to the conventional algorithms which perform sorting processing, the proposed algorithm uses look-up tables to reduce the repetition of calculation. Consequently, it is shown that less processing time and lower complexity can be achieved.

Multi-Cell Transmit Diversity Scheme for OFDMA Systems (OFDMA 시스템을 위한 다중 셀 전송 다양성 기법)

  • Seo, Bangwon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37A no.9
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    • pp.721-727
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    • 2012
  • Since a conventional multi-cell transmit diversity scheme depends on the feedback from the user for the channel gain information, its performance gets to severely degrade when the channel varies fast due to the high mobility of the user. Also, transmit power of the base station cannot be fully used in the conventional scheme because only one transmit antenna is used for data transmission. In this paper, we propose a multi-cell transmit diversity scheme appropriate for fast fading channel. In the proposed scheme, channel-independent precoding vector is applied over all transmit antennas and different precoding vectors are applied for neighboring subcarriers so that the received signal is avoided to experience deep fading over multiple neighboring subcarriers. Simulation results show that the proposed scheme has better detector output signal-to-noise ratio (SNR) and bit error rate (BER) performances than the conventional scheme.

BER Performance Analysis for Adaptive Cooperation Scheme with Decode-and-Forward Relay-Selection (복호 후 전달 릴레이 선택을 이용한 적응형 협력 기법의 BER 성능분석)

  • Vu, Ha Nguyen;Kong, Hyung-Yun
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.11A
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    • pp.831-843
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    • 2009
  • In this paper, we propose a new adaptive cooperation scheme with multi-relay nodes which achieves higher performance and spectral efficiency than that of some conventional cooperative schemes. The relay-selection is applied to choose the most potential relay among K ones. Afterward, the instantaneous signal-to-noise ratio (SNR) differences between S-D, S-R and R-D channels are considered for adaptive selection between the direct and the cooperation transmission strategy. In the proposed adaptive protocol, if the direct link is of high quality, the source will transmit to destination directly with all power consumption. Otherwise, the source broadcasts the signal with a lower power and requires the help of the chosen relay if it decodes correctly, else the source will transmit again with remaining power. Firstly, the spectral efficiency is derived by calculating the probability of each mode. Subsequently, the BER performance for the adaptive cooperation scheme is analyzed by considering each event that one of K relays is selected and then making the summation of all. Finally, the numerical results are presented to confirm the performance enhancement offered by the proposed schemes.

Performance Analysis of Frame Synchronization and Structure Detection Utilizing Multiple Frames of the DVB-S2 Satellite Broadcasting System (다수개 프레임을 활용한 DVB-S2 위성방송 시스템의 프레임 동기 및 구조 검출 성능 분석)

  • Kim, Sang-Tae;Kang, Seok-Heon;Sung, Won-Jin
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.2A
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    • pp.139-147
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    • 2008
  • DVB-S2 (Digital Video Broadcasting-Satellite, Version 2) system transmits frames which adapt their structures based on the channel conditions, thus requiring simultaneous detection of the start of the Same (SoF) and the frame structure at the initial acquisition stage of the receiver. Also, a very low value of the minimum operating signal-to-noise ratio (SNR) for the acquisition necessitates a method utilizing multiple received frames to meet the required performance. In this paper, performance of joint time synchronization and frame structure detection methods using multiple DVB-S2 frames is evaluated by deriving the detection error probability. In particular, we evaluate the performance and complexity variations when the soft- and hard-decision values of the signal correlation output are used, present the synchronization parameters to optimize the performance, and verify the analysis results via computer simulations.

AT-DMB Reception Method with Eigen-space Beamforming Algorithm (고유 공간 빔형성 알고리즘을 이용한 AT-DMB 수신 방법)

  • Lee, Jae-Hong;Choi, Seung-Won
    • Journal of Broadcast Engineering
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    • v.15 no.1
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    • pp.122-132
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    • 2010
  • AT-DMB system has been developed to increase data rate up to double of conventional T-DMB in the same bandwidth while maintaining backward compatibility. The AT-DMB system adopted hierarchical modulation which adds BPSK or QPSK signal as enhancement layer to existing DQPSK signal. The enhancement layer signal should be small enough to maintain backward compatibility and to minimize the coverage loss of conventional T-DMB service coverage. But this causes the enhancement layer signal of AT-DMB susceptible to fading effect in transmission channel. A turbo code which has improved error correction capability than convolutional code, is applied to the enhancement layer signal of the AT-DMB system for compensating channel distortion. However there is a need for other solutions for better reception of AT-DMB signal in receiver side without increasing transmitting power. In this paper, we propose adaptive array antenna system with Eigen-space beamforming algorithm which benefits beamforming gain along with diversity gain. We analyzed the reception performances of AT-DMB system in indoor and mobile environments when this new smart antenna system and algorithm is introduced. The computer simulation results are presented along with analysis comments.

Comparison of ICA Methods for the Recognition of Corrupted Korean Speech (잡음 섞인 한국어 인식을 위한 ICA 비교 연구)

  • Kim, Seon-Il
    • 전자공학회논문지 IE
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    • v.45 no.3
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    • pp.20-26
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    • 2008
  • Two independent component analysis(ICA) algorithms were applied for the recognition of speech signals corrupted by a car engine noise. Speech recognition was performed by hidden markov model(HMM) for the estimated signals and recognition rates were compared with those of orginal speech signals which are not corrupted. Two different ICA methods were applied for the estimation of speech signals, one of which is FastICA algorithm that maximizes negentropy, the other is information-maximization approach that maximizes the mutual information between inputs and outputs to give maximum independence among outputs. Word recognition rate for the Korean news sentences spoken by a male anchor is 87.85%, while there is 1.65% drop of performance on the average for the estimated speech signals by FastICA and 2.02% by information-maximization for the various signal to noise ratio(SNR). There is little difference between the methods.

A Beamforming Method for a Perturbed Linear Towed Array (비선형 형상 견인 어레이를 위한 빔형성 기법)

  • 김승일;도경철;오원천;윤대희;이충용
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.5
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    • pp.478-484
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    • 2002
  • Linear towed arrays (LTA) have a nonlinear shape due to tow vessel motion, ocean swells and currents. By reasons of nominally linear shape, various towed array shape estimation techniques have been developed since the perturbed shape cause the error in target detection. In this paper,, we propose the beamforming method for the perturbed LTA with simple structure. The proposed method linearizes a nonlinear phase of steering vector with position information measured by two reference sensors. It can be proved using some properties of Markov transition matrix, and iteration number of linearization process is decided by variance of cross phase difference. As a result of computer simulation in the ocean environment, beampattern of the proposed method is almost same with the ideal case in my type of array shape. In the signal-to-noise ratio (SNR) performance simlation, the DOA estimation performance of the proposed beamforming method is evaluated, and the comparison with Bartlett beamformer of the LTA shows that the proposed method can estimate. the spatial characteristic of sources more accuracy.