D.H. Youn

Yonsei University, Seoul, Seoul, South Korea

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Publications (25)36.51 Total impact

  • J. Lee, Y.-C. Park, D.-H. Youn
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    ABSTRACT: Recently, a variable step-size affine projection (VSS-AP) algorithm has been introduced. The algorithm provides faster convergence rate and lower misadjustment but has heavy computational complexity. Proposed is a new variable step-size pseudo affine projection (VSS-PAP) algorithm, which not only has a much lower complexity than the VSS-AP algorithm but also provides performance comparable with the conventional algorithm. Simulation results confirm fast convergence rate and small misalignment of the proposed algorithm with less computational complexity.
    Electronics Letters 02/2008; · 1.04 Impact Factor
  • J.H. Lee, Y.C. Park, D.H. Youn
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    ABSTRACT: An efficient method for controlling the echo in a continuously variable slope delta modulation (CVSD) mobile communication channel is proposed. By utilising the symmetry of the coder pairs, the proposed method shortens the echo path length and reduces the nonlinear distortion inherent in the coders. An additional advantage for the VLSI implementation is obtained using the combined bit-stream. The proposed method shows an improved echo return loss enhancement (ERLE) performance and reduced complexity compared to the conventional method.
    Electronics Letters 06/2000; · 1.04 Impact Factor
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    ABSTRACT: The filtered-x LMS algorithm and its modified versions have been successfully applied in suppressing acoustic noise such as single and multiple tones and broadband random noise. This paper presents an adaptive algorithm based on the filtered-x LMS algorithm which may be applied in attenuating tonal acoustic noise. In the proposed method, the weights of the adaptive filter and estimation of the phase shift due to the acoustic path from a loudspeaker to a microphone are computed simultaneously for optimal control. The algorithm possesses advantages over other filtered-x LMS approaches in three aspects: (1) each frequency component is processed separately using an adaptive filter with two coefficients, (2) the convergence parameter for each sinusoid can be selected independently, and (3) the computational load can be reduced by eliminating the convolution process required to obtain the filtered reference signal. Simulation results for a single-input/single-output (SISO) environment demonstrate that the proposed method is robust to the changes of the acoustic path between the actuator and the microphone and outperforms the filtered-x LMS algorithm in simplicity and convergence speed.
    The Journal of the Acoustical Society of America 08/1998; 104(1):248-54. · 1.65 Impact Factor
  • Y.C. Park, H.T. Sung, D.H. Youn
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    ABSTRACT: The authors present an adaptive eigenfilter implementing the optimum moving target indication (MTI) processor. The method presented shows a fast convergence capability as well as better steady-state performance than the prediction error filter (PEF). Moreover, the arithmetic efficiency of the method results from using modular processing architecture which takes advantage of the Gram-Schmidt (GS) processor
    Electronics Letters 09/1996; · 1.04 Impact Factor
  • B.J. Kwak, K.M. Kim, I.W. Cha, D.H. Youn
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    ABSTRACT: The authors derive expressions of the output signal-to-interference plus noise ratio (SINR) of the linearly constrained narrowband beamformer in noncoherent situations using a vector approach. Numerical results are included
    IEEE Transactions on Antennas and Propagation 11/1993; · 2.33 Impact Factor
  • Y.C. Park, W.K. Kim, D H Youn
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    ABSTRACT: The authors propose an adaptive processor implementing an optimum moving target indicator (MTI) in the sense of maximizing the improvement factor (IF). A multichannel scheme for preventing coefficient fluctuation due to the presence of target echoes or impulse interferences is also discussed. The proposed eigenfilter recursively estimates the minimum eigenvector of the input clutter covariance matrix based on the inverse power method (IPM). In the proposed scheme, the IPM iteration is performed in a numerically efficient way using systolic processing architectures. Simulation results demonstrate that the convergence speed of the eigenfilter is faster than that of the prediction error filters and IF values of the eigenfilter always approach the optimum. The multichannel adaptive eigenfilter is shown to be a robust scheme preventing coefficient fluctuation in the presence of target echoes
    Acoustics, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on; 05/1993
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    ABSTRACT: An efficient algorithm detecting the presence of a fetal QRS complex is presented. The proposed fetal QRS detection method computes the averaged magnitude of the difference between the fetal ECG signal and the reference signal to detect the fetal QRS event. The detected fetal QRS complexes are exponentially averaged to generate the template signal which can track the slowly varying shape of the fetal ECG signal. As an effort to obtain improved detection performances, two approaches of normalizing the fetal ECG signal and the template are considered.
    IEEE Transactions on Biomedical Engineering 09/1992; 39(8):868-71. · 2.35 Impact Factor
  • K.M. Kim, Y.C. Park, I.W. Cha, D.H. Youn
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    ABSTRACT: An adaptive digital filter structure which can be used in multichannel noise-canceling applications is described. The proposed structure is obtained by creating a hybrid from the lattice and the escalator (Gram-Schmidt) structures, with the coefficients being updated using the least mean square (LMS) algorithm. As an application, the proposed adaptive multichannel digital filter is applied to implement an adaptive generalized sidelobe canceler
    IEEE Transactions on Signal Processing 08/1992; · 2.81 Impact Factor
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    ABSTRACT: Not Available
    Consumer Electronics, 1992. Digest of Technical Papers. ICCE., IEEE 1992 International Conference on; 07/1992
  • K.M. Kim, I.W. Cha, D.H. Youn
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    ABSTRACT: The convergence rate for the adaptive weights to reach the optimum value in an adaptive array system depends on the eigenvalue spread ratio of the autocovariance matrix. How the eigenvalue spread ratio in a generalized sidelobe canceller (GSC) is affected by the various parameters is studied. Expressions for the output power of the GSC in coherent situations are derived, and numerical results are included
    IEEE Transactions on Antennas and Propagation 05/1992; · 2.33 Impact Factor
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    ABSTRACT: A discrete cosine transform (DCT)-based block adaptive image data compression scheme is proposed for a home-use digital video cassette recorder (VCR). The technique maintains a high picture quality at a moderate bit rate. To attain a reasonable picture quality in high-speed playback, a segment forming method and a recording sequence of the encoded data were developed. The coding constraints for high-speed playback are reviewed. The coding scheme proposed for bit rate reduction in a digital VCR is described. Computer simulation results are presented. The simulation results show that the bit rate reduction scheme could attain not only subjectively good picture quality in normal-speed playback, but also an acceptable picture quality in high-speed playback
    IEEE Transactions on Consumer Electronics 01/1992; 38(3):236-242. · 1.09 Impact Factor
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    ABSTRACT: Not Available
    Consumer Electronics, 1991 IEEE International Conference on; 07/1991
  • K.M. Kim, I.W. Cha, D.H. Youn
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    ABSTRACT: An adaptive multichannel digital filter structure that can be used in multichannel noise-cancelling applications is described. The proposed structure is a hybrid of the lattice and the escalator (Gram-Schmidt) structures, and the coefficients are updated using the least-mean-square algorithm. In the lattice-escalator hybrid structure, complete decorrelation of multiple reference signals is achieved by successively generating forward prediction error vectors via the lattice structure and decorrelating the elements of the error vectors using escalator linear combiners at each lattice stage. Simulation results indicate that the convergence speed of the hybrid structure is as fast as the escalator realization
    Acoustics, Speech, and Signal Processing, 1990. ICASSP-90., 1990 International Conference on; 05/1990
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    ABSTRACT: In this paper, Korean language recognition using the phoneme is studied. The experiment is carried out by dividing 545 isolated words into phonemes. Using linear prediction coefficients the recognition rate of consonants, vowels, and end-consonants are , respectively. Recognition rate of isolated words combined with the phonemes is . Itakura-saito distortion measure is used to phoneme segmentation and phoneme recognition.
    The Journal of the Acoustical Society of Korea. 01/1989; 8(5).
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    ABSTRACT: The authors describe the implementation of a real-time fetal ECG (electrocardiogram) monitoring system in which an adaptive multichannel noise canceller is realized using the TMS32020 programmable digital signal processor. An ECG signal from the electrode placed on the mother's abdomen and three ECGs from those on the chest are applied as the desired signal and the reference inputs, respectively, of the multichannel filter. The coefficients of the filter are updated using the LMS (least-mean-square) algorithm, such that the output of the multichannel filter copies the maternal ECG embedded in the abdominal ECG. The enhanced fetal ECG is obtained by subtracting the filter output from the abdominal ECG, and the difference signal is recorded
    Circuits and Systems, 1988., IEEE International Symposium on; 07/1988
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    T.H. Park, D.H. Youn, I.W. Cha
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    ABSTRACT: The authors update delta modulation digital filter weights using the LMS (least-mean-squares) and the SIGN algorithms to realize an adaptive digital filter without multiplication operations. It is shown that using the SIGN algorithm results in an adaptive filter that can be implemented using the simple up/down counting operations. Learning curves demonstrating convergence properties of the algorithms for a system identification problem are presented
    Proceedings of the IEEE 06/1988; · 6.91 Impact Factor
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    ABSTRACT: This paper introduces an adaptive nonlinear filtering algorithm that uses the sequential regression(SER) method to update the second order Volterra filter coefficients in a recursive way. Conventionally, the SER method has been used to invert large matrices which result from direct application of Wiener filter theory to the Volterra filter. However, the algorithm proposed in this paper uses the SER approach to update the least squares solution which is derived for Gaussian input signals. In such an algorithm, the size of the matrix to be inverted is smaller than that of conventional approaches, and hence the proposed method is computationally simpler than conventional nonlinear system identification techniques. Simulation results are presented to demonstrate the performance of the proposed algorithm.
    The Journal of the Acoustical Society of Korea. 01/1988; 7(4).
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    D.N. Kim, I.W. Cha, D.H. Youn
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    ABSTRACT: It is proposed to use a pair of frequency-domain adaptive digital filters to estimate the magnitude squared coherence (MSC) functions of two signals. Such a method requires less computations than the LMS-MSC algorithm in which the least mean square (LMS) algorithm is applied in the time domain to compute the coefficients of a pair of adaptive digital filters. The frequency-domain adaptive digital filtering algorithms considered in this paper include the constrained frequency domain LMS (CFLMS) and the unconstrained frequency domain LMS (UFLMS) algorithms. The performance of the proposed methods are compared with those of the LMS-MSC algorithm.
    The Journal of the Acoustical Society of Korea. 01/1988; 7(2).
  • D.H. Youn, V.J. Mathews, S.H. Cho
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    ABSTRACT: An efficient method for spectral analysis of non-stationary signals is proposed. The signals are modelled as time-varying autoregressive (AR) processes and the relevant AR parameters are obtained from the adaptive lattice predictor coefficients minimizing the forward/backward prediction errors. In this paper it is proposed to use the sign algorithm, which computes the pertinent coefficients using only the signs of the forward/backward prediction errors at each stage. This method requires less number of multiplications than other adaptive lattice predictor algorithms. The performance of this method is compared with that of the adaptive lattice predictor using the least mean square algorithm for the problems of estimating instantaneous frequencies of FM signals via computer simulations.ZusammenfassungVorgeschlagen wird eine leistungsfa¨hige Methode zur Spektral-Analyse nicht-stationa¨rer Signale. Die Signale werden als zeit-variable autoregressive (AR) Prozesse betrachtet, wobei die beno¨tigten AR Parameter aus denjenigen Koeffizienten eines adaptiven gitter-Pra¨diktors gewonnen werden, welche die Vorwa¨rts/Ru¨ckwart-Fehler minimalisieren. In disem Beitrag wird vorgeschlagen, den Vorzeichen-Algorithmus zu benutzen, welcher die Koeffizienten einzig aus den Vorzeichen der aus jeder Stufe resultierenden Vorwa¨rts/Ru¨ckwa¨rts-Fehler berechnet. Diese Methode beno¨tigt weniger Multiplikationen als andere adaptive gitter-Pra¨diktions-Algorithmen. Die Leistungsfa¨higkeit dieser Methode wird mit derjenigen des auf dem kleinsten Fehlerquadrat-Algorithmus basierenden adaptiven gitter-Pra¨diktor verglichen, und zwar in der Anwendung auf Probleme, welche sich in der Scha¨tzung der momentanen Frequenzen eines FM-signals via Computer-Simulation ergeben.RésuméUne me´thode efficace pour l'analyse spectrale des signaux non stationnaires est propose´e. Les signaux sont mode´lise´s comme des processus autoregressifs (AR) variant dans le temps et les parame`tres AR relevants sont obtenusa`partir des coefficients du pre´dicteura`treilli adaptif, minimisant les erreurs de pre´diction en avant et en arrie`re. Dans cet article, il est propose´d'utiliser l'agorithme de signe qui calcul les coefficients pertinants en utilisant seulement les signes des erreurs de pre´diction en avant et an arrie`rea`chaquee´tape. Cette me´thode ne´cessite un nombre de multiplication infe´rieura`celui des algorithmes de pre´dictiona`treilli adaptatif. La performance de cette me´thode est compare´e avec celle des pre´dicteursa`treilli adaptatif en utilisant l'algorithme quadratique moyen minimum pour des proble`mes d'estimation de la fre´quence instantane´e des signaux module´s en fre´quence, par simulation sur ordinateur.
    Signal Processing. 01/1986;
  • Dae Youn, Jin-Geol Kim
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    ABSTRACT: The purpose of this paper is to present some experimental results estimating instantaneous frequencies of FM (frequency modulation) signals using the short-time Fourier transform (STFT) method. The relevant STFT coefficients were computed using a bank of simple one-pole low-pass filters which estimates the cross correlation of a given signal and complex sinusoids. Also, it is shown that computing the STFT coefficients is equivalent to finding the least square solution which best approximates the given signal as a weighted sum of complex sinusoids.
    IEEE Transactions on Acoustics Speech and Signal Processing 03/1985;

Publication Stats

93 Citations
36.51 Total Impact Points

Institutions

  • 1990–2008
    • Yonsei University
      • Department of Electrical and Electronic Engineering
      Seoul, Seoul, South Korea
  • 1993
    • Concordia University–Ann Arbor
      Ann Arbor, Michigan, United States
  • 1983–1986
    • University of Iowa
      • Department of Electrical and Computer Engineering
      Iowa City, IA, United States
  • 1980–1984
    • Kansas State University
      Kansas, United States