• Title/Summary/Keyword: frequency bin

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Frequency Bin Alignment Using Covariance of Power Ratio of Separated Signals in Multi-channel FD-ICA (다채널 주파수영역 독립성분분석에서 분리된 신호 전력비의 공분산을 이용한 주파수 빈 정렬)

  • Quan, Xingri;Bae, Keunsung
    • Phonetics and Speech Sciences
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    • v.6 no.3
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    • pp.149-153
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    • 2014
  • In frequency domain ICA, the frequency bin permutation problem falls off the quality of separated signals. In this paper, we propose a new algorithm to solve the frequency bin permutation problem using the covariance of power ratio of separated signals in multi-channel FD-ICA. It makes use of the continuity of the spectrum of speech signals to check if frequency bin permutation occurs in the separated signal using the power ratio of adjacent frequency bins. Experimental results have shown that the proposed method could fix the frequency bin permutation problem in the multi-channel FD-ICA.

Blind Source Separation U sing Variable Step-Size Adaptive Algorithm in Frequency Domain

  • Park Keun-Soo;Lee Kwang-Jae;Park Jang-Sik;Son Kyung Sik
    • Journal of Korea Multimedia Society
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    • v.8 no.6
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    • pp.753-760
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    • 2005
  • This paper introduces a variable step-size adaptive algorithm for blind source separation. From the frequency characteristics of mixed input signals, we need to adjust the convergence speed regularly in each frequency bin. This algorithm varies a step-size according to the magnitude of input at each frequency bin. This guarantee of the regular convergence in each frequency bin would become more efficient in separation performances than conventional fixed step-size FDICA. Computer simulation results show the improvement of about 5 dB in signal to interference ratio (SIR) and the better separation quality.

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Post-Processing of IVA-Based 2-Channel Blind Source Separation for Solving the Frequency Bin Permutation Problem (IVA 기반의 2채널 암묵적신호분리에서 주파수빈 뒤섞임 문제 해결을 위한 후처리 과정)

  • Chu, Zhihao;Bae, Keunsung
    • Phonetics and Speech Sciences
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    • v.5 no.4
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    • pp.211-216
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    • 2013
  • The IVA(Independent Vector Analysis) is a well-known FD-ICA method used to solve the frequency permutation problem. It generally works quite well for blind source separation problems, but still needs some improvements in the frequency bin permutation problem. This paper proposes a post-processing method which can improve the source separation performance with the IVA by fixing the remaining frequency permutation problem. The proposed method makes use of the correlation coefficient of power ratio between frequency bins for separated signals with the IVA-based 2-channel source separation. Experimental results verified that the proposed method could fix the remaining frequency permutation problem in the IVA and improve the speech quality of the separated signals.

New Frequency-domain GSC using the Modified-CFAR Algorithm (변형된 CFAR 알고리즘을 이용한 새로운 주파수영역 GSC)

  • Cho, Myeong-Je;Moon, Sung-Hoon;Han, Dong-Seog;Jung, Jin-Won;Kim, Soo-Joong
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.36S no.2
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    • pp.96-107
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    • 1999
  • The generalized sidelobe cancellers(GSC's) ar used for suppressing an interference in array radar. The frequency-domain GSC's have a faster convergence rate than the time-domain GSC's because they remove the correlation between the interferences using a frequency-domain least mean square(LMS) algorithm. However, we have not fully used the advantage of the frequency-domain GSC's since we have always updated the weights of all frequency bins, even the interferer free frequency bin. In this paper, we propose a new frequency-domain GSC based on constant false-alarm rate(CFAR) detector, of which GSC adaptively determine the bin whose weight is updated according to the power of each frequency bin. This canceller updates the weight of only updated according to the power of each frequency bin. This canceller updates the weight of only the bin of which the power is high because of the interference signal. The computer simulation shows that the new GSC reduces the iteration number for convergence over the conventional GSC's by more than 100 iterations. The signal-to-noise ration(SNR) improvement is more than 5 dB. Moreover, the number of renewal weights required for the adaptation is much fewer than that of the conventional one.

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수직방향 집중하중 상태의 외팔보 거동에 대한 선형 및 비선형적 해석 비교

  • Go, Jeong-U;Bin, Yeong-Bin
    • Proceeding of EDISON Challenge
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    • 2015.03a
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    • pp.239-245
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    • 2015
  • In this paper, to examine the difference between the linear and non-linear static, dynamic analysis for a structure, a cantilevered beam was used. Then, an external transverse static and dynamic loads were applied at the free end of the beam. Classical theories were used for the linear analysis and the EDISON CSD solver, co-rotational dynamic FEM program, was used for nonlinear analysis. In the static analysis, effects of the load for the beam deflection were observed in the linear and nonlinear analysis. Then, normalized displacement of tip of the beam was predicted for different frequency ration and a significant difference was obtained in the vicinity of the resonant frequency. In addition, effects of frequency and time for the beam deflection were investigated to find the frequency delay.

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Noise Suppression Using Normalized Time-Frequency Bin Average and Modified Gain Function for Speech Enhancement in Nonstationary Noisy Environments

  • Lee, Soo-Jeong;Kim, Soon-Hyob
    • The Journal of the Acoustical Society of Korea
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    • v.27 no.1E
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    • pp.1-10
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    • 2008
  • A noise suppression algorithm is proposed for nonstationary noisy environments. The proposed algorithm is different from the conventional approaches such as the spectral subtraction algorithm and the minimum statistics noise estimation algorithm in that it classifies speech and noise signals in time-frequency bins. It calculates the ratio of the variance of the noisy power spectrum in time-frequency bins to its normalized time-frequency average. If the ratio is greater than an adaptive threshold, speech is considered to be present. Our adaptive algorithm tracks the threshold and controls the trade-off between residual noise and distortion. The estimated clean speech power spectrum is obtained by a modified gain function and the updated noisy power spectrum of the time-frequency bin. This new algorithm has the advantages of simplicity and light computational load for estimating the noise. This algorithm reduces the residual noise significantly, and is superior to the conventional methods.

Recommendation for the modelling of Donnell shell: The relationship between non-local parameter and frequency

  • Mohamed A. Khadimallah;Muzamal Hussain;Elimam Ali;Sehar Asghar;Abdelouhed Tounsi
    • Computers and Concrete
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    • v.32 no.2
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    • pp.165-172
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    • 2023
  • The vibration analysis of armchair, zigzag and chiral double-walled carbon nanotubes has been developed by inserting the nonlocal theory of elasticity into thin shell theory. First Donnell shell theory is employed while exercising wave propagation approach. Scale effects are realized by using different values of nonlocal parameters under certain boundary conditions. The natural frequencies have been investigated and displayed for various non-local parameters. It is noticed that on increasing nonlocal parameter, the frequency curve tends to decrease. The frequency estimates of clamped-free boundary condition are less than those of clamped-clamped and simply supported computations. The frequency comparisons are presented for armchair, zigzag and chiral nanotubes. The software MATLAB is used to extract the frequencies of double walled carbon nanotubes.

Vocal Separation Using Selective Frequency Subtraction Considering with Energies and Phases (에너지와 위상을 고려한 선택적 주파수 차감법을 이용한 보컬 분리)

  • Kim, Hyuntae;Park, Jangsik
    • Journal of Broadcast Engineering
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    • v.20 no.3
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    • pp.408-413
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    • 2015
  • Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. The specific method is to make the original sound accompaniment to remove only the voice of the singer in the singer music album. In this paper, a system to separate vocal components from music accompaniment for stereo recordings were proposed. Proposed system consists of two stages. The first stage is a vocal detection. This stage classifies an input into vocal and non vocal portions by using SVM with MFCC. In the second stage, selective frequency subtractions were performed at each frequency bin in vocal portions. In this case, it is determined in consideration not only the energies for each frequency bin but also the phase of the each frequency bin at each channel signal. Listening test with removed vocal music from proposed system show relatively high satisfactory level.

A Blind Watermarking for 3-D Mesh Sequence Using Temporal Wavelet Transform of Vertex Norms (꼭지점 좌표 벡터 크기값의 시간축 웨이블릿 변환을 이용한 3차원 메쉬 시퀀스의 블라인드 워터마킹)

  • Kim, Min-Su;Cho, Jae-Won;Prost, Remy;Jung, Ho-Youl
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.3C
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    • pp.256-268
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    • 2007
  • In this paper, we present a watermarking method for 3-D mesh sequences. The main idea is to transform vertex norm with the identical connectivity index along temporal axis using wavelet transform and modify the distribution of wavelet coefficients in temporally high (or middle) frequency frames according to watermark bit to be embedded. All vertices are divided into groups, namely bin, using the distribution of coefficients in low frequency frames. As the vertices with the identical connectivity index over whole frames belong to one bin, their wavelet coefficients are also assigned into the same bin. Then, the watermark is embedded into the wavelet coefficients of vertex norm. Due to the use of the distribution, our method can retrieve the hidden watermark without any information about original mesh sequences in the process of watermark detection. Through simulations, we show that the proposed is flirty robust against various attacks that are probably concerned in copyright protection of 3-D mesh sequences.

Modification-robust contents based motion picture searching method (변형에 강인한 내용기반 동영상 검색방법)

  • Choi, Gab-Keun;Kim, Soon-Hyob
    • 한국HCI학회:학술대회논문집
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    • 2008.02a
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    • pp.215-217
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    • 2008
  • The most widely used method for searching contents of mot ion picture compares contents by extracted cuts. The cut extract ion methods, such as CHD(Color Histogram Difference) or ECR(Edge Change Ratio), are very weak at modifications such as cropping, resizing and low bit rate. The suggested method uses audio contents for indexing and searching to make search be robust against these modification. Scenes of audio contents are extracted for modification-robust search. And based on these scenes, make spectral powers binary on each frequency bin. in the time-frequency domain. The suggested method shows failure rate less than 1% on the false positive error and the true negative error to the modified(using cropping, clipping, row bit rate, addtive frame) contents.

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