• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.3
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• pp.119-128
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• 2017

In this paper, the wideband direction finding algorithm and system design method for short duration signal such as frequency hopping or burst signal is presented. The polyphase filterbank that it is possible for the near perfect reconstruction was used as a pre-processing and in each subband power measurement was performed to determine whether the presence of a signal and finally general direction finding algorithm was performed. In addition, various experiments was performed using Matlab Simulink and collected data from wideband receiver to verification of the proposed algorithm.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.3
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• pp.120-126
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• 2005

This paper presents the 3D wavelet based video compression system using quad-tree algorithm. The 3D wavelet based video compression system removes the temporal correlation of the input sequences using the motion compensation filter and decomposes the spatio-temporal subband using the spatial wavelet transform. The proposed system allocates the higher bit rate to the low frequency image of the 3D wavelet sequences and improves the 0.64dB PSNR performance of the reconstructed image in comparison with that of H.263. In addition to the limitation on the propagation of the motion compensation error by the 3D wavelet transform, the proposed system progressively transmits the input sequence according to the resolution and rate scalability.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.1
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• pp.26-31
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• 2012

The highest frequency band of wavelet decomposition band is divided into 4 equal subbands and by the minimum power of the subbands and by the monotonic transform, the level adapted threshold is obtained. The adapted threshold is applied to the soft threshold technique to denoise high and middle frequency band noise of image signals. And the results of PSNRs are compared with the results obtained by the VisuShrink technique and by the technique using the monotonic transform and the weight value. The results showed the validity of this technique.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.7 no.5
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• pp.418-429
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• 1996

The number of bits per message and the number of tones in the frequency-hopping sequence are determined by the available bandwidth and the data rate of each user. These parameters in turn determine the tone duration which strongly influences the vulnerability of the system to transmission distortions. In this paper, an address code which is assigned to each individual user was employed in order to reduce the collisions or hit. Also the frequency band is divided into several subbands and each user transmits multitone frequency per subband per chip. And the new system which is to increase the duration of each tone by increasing the total number of system frequencies that has been proposed. It is found that an optimum value bit, tone, number of frequencies per chirp can improve the err performance. This flexibility slightly increases maximum efficiecy and makes the the system less vulnerable to multipath delay. So, It is found that as the nuber of user increased 50%, the efficiency as a tuncion of the bandwidth to user'rate ratio improve 20%.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.150-155
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• 2014

본 논문에서는, 원거리 음성인식을 위한 서브밴드 필터링 기반의 빠르고 효율적인 에코제거 시스템을 제안한다. 제안하는 에코제거 시스템은 우선 채널간 유사도 (correlation) 가 높을 경우 적응필터가 오작동하는 것을 방지하기 위해 spatial decorrelation 을 적용하게 된다. 그리고 tree 형태를 가지는 IIR filterbank 기반의 subband 구조를 채택함으로써, 적은 차수로도 효과적인 analysis, synthesis 필터링을 수행할 수 있도록 한다. 이 과정에서 불가피하게 발생하는 서브 밴드간 spectral aliasing은 notch filter를 적용해 해결할 수 있다. 또한 적응 필터로는 improved proportionate normalized least-mean-square (IP-NLMS) 알고리즘을 사용해 수렴속도 및 에코제거 성능에서 우수함을 확인하였다. 마지막으로 decision-directed estimation 기반의 residual echo suppressor를 적용해 잔여 에코를 제거하게 된다. 본 논문에서는 각 단계를 구성하게 된 이론적인 배경을 소개하고, 실제 에코가 존재하는 환경에서 ERLE, 원거리 음성 인식률, computational complexity를 통해 제안하는 에코제거 시스템의 효과를 입증하도록 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.817-820
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• 2002

인간과 기계와의 인터페이스에 있어서 궁극적 목표는, 인간과 기계가 마치 사람과 사람이 대화하듯 자연스런 인터페이스가 이루어지도록 하는데 있다. 이에 본 논문에서는 사람의 음성속에 깃든 6개의 기본 감정을 인식하는 알고리듬을 제안하고자 한다. 이를 위하여 뛰어난 주파수 분해능력을 갖고 있는 웨이블렛 필터뱅크를 이용하여 음성을 여러 개의 서브밴드로 나누고 각 밴드에서 특징점을 추출하여 감정을 이식하고 이를 최종적으로 융합, 단일의 인식값을 내는 다중의사 결정 구조를 갖는 알고리듬을 제안하였다. 이를 적용하여 실제 음성 데이타에 적용한 결과 기존의 방법보다 높은 90%이상의 인식률을 얻을 수 있었다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.305-308
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• 2000

암의 진행 정도를 판단하기 위한 암세포 조직영상의 분석은 그 대상이 되는 영상의 다양성과 잡음으로 인해 정확한 분석이 어렵다. 특히, 암의 진행 정도를 판단하는데 있어서 중요한 요인인 세포핵의 variation에 따른 order/disorder 정도를 객관적 수치로 정량화하기 위해서는, 각 기(stage)에 따른 암의 진행정도를 가장 잘 나타낼 수 있는 특징값 추출이 필수적이다. 본 논문에서는 가장 유효한 특징값을 추출하기 위하여, 공간 영역과 주파수 영역에서 그 지역적 특징을 잘 나타내는 wavelet 변환을 적용한 후, 분할 된 서브 밴드 중 고대역 서브 밴드에서 질감 특징을 추출하고, 추출 된 질감 특징값들이 암의 진행 정도에 따른 각 집단간에 유의한 차이를 나타내는지에 대한 유의성을 검증하기 위하여, 다변량 통계학적 분석 방법을 사용하여 비교분석 하였다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.58-68
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• 2000

This paper proposes a sinusoidal modeling of polyphonic audio signals. Sinusoidal modeling which has been applied well to speech and monophonic signals cannot be applied directly to polyphonic signals because a window size for sinusoidal analysis cannot be determined over the entire signal. In addition, for high quality synthesized signal transient parts like attacks should be preserved which determines timbre of musical instrument. In this paper, a multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in sinusoidal modeling a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. The improved dynamic segmentation is proposed which shows better performance about transients and reduced computation. For various polyphonic audio signals the result of simulation shows the suggested sinusoidal modeling can model polyphonic audio signals without loss of perceptual quality.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.37-41
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• 2002

In this paper, we made shorter EDT(early decay time) of room reverberation curve using multiple-channel. The speech signal was processed inverse filtering with full-band and sub-band in the basis MINT, and then the multiple-channel adaptive filters were used LMS (Least Mean Square) and NLMS (Normalized Least Mean Square) algorithm. Experimental results, we could get 1/3 of time reduction at 20dB level in the reverberation curve using full-band NLMS when two microphones were used. Also, it is shown that the speech articulation was improved 80% from the test listeners with the speech, which was to shorten EDT by MINT in the subjective assessments using real room impulse response.