• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.513-516
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• 2001

마이크로폰을 배열을 이용한 음성개선 기법 중에서 저주파 영역에서의 높은 상관성과 고주파 영역에서의 spatial aliasing을 동시에 고려하기 위하여 대수적인 선형 마이크로폰 배열을 사용하고 웨이브렛 도메인에서의 Wiener 필터에 기반한 postfiltering을 수행하는 방법이 제안된 바 있는데[l], 본 논문에서는 이 방법의 문제점을 분석하고 해결방안을 제시하였다. 제안한 알고리즘을 사용하여 시뮬레이션한 결과, 마이크에 입력되는 음성신호의 SNR이 0dB일 때와 l0dB일 때, 기존의 알고리즘에 비해 약 1.7dB와 2.5dB의 성능개선이 있었으며, 청취실험을 통해서도 음질의 향상을 확인할 수 있었다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.425-428
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• 2002

본 논문에서는 해양에서 다중경로를 통하여 수신되는 음파의 도달시간을 정확히 파악하기 위한 방법을 제시하였다. 음파 도달 시간을 파악하기 위한 방법으로는 정합 필터 방법 및 웨이브렛 방법을 도입하였으며 각각 모의 수신신호 및 실관측 수신신호에 대해 적용하여 식별 성능을 분석하였다.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.12
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• pp.67-75
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• 1999

In this paper, we present an ASIC(Application Specific Integrated Circuit) design of wavelet transform filter Wavelet transform is used in lots of application fields which include image compression, because it has an excellent energy compaction. The operation characteristic and performance of wavelet transform filter are analyzed by using verilog-HDL(Hardware Description Language). In this paper, the designed wavelet transform filter uses line memory to improve data processing rate. Generally, when it reads and writes data of DRAM by using Fast Page Mode, input and output processing is very fast in horizontal direction but substantially slow in vertical direction. The use of line memory solves this low speed processing problem. As a result, though the size of the chip is getting larger, processing time for an image frame becomes 4.66ms. Generally, since the limit of 1 frame processing time on the data of TV video is 33ms, so it is appropriate for TV video.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.30-37
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• 1996

In the communication system, adaptive equalizer using LMS algorithm has slow convergence rate in spite of effectiveness and simplicity. In this paper, we designed the wavelet transform based adaptive equalizer to overcome this problem. The performance of this new approach is compared with that of the time domain LMS algorithm by convergence rate with respect to change of channel distortion and filter order. As a result, the wavelet transform based adaptive equalizer shows the improvements in the speed of convergence compared with LMS algorithm based adaptive equalizer.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Journal of Broadcast Engineering
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• v.1 no.2
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• pp.108-117
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• 1996

In this paper we propose a Wavelet- based compression algorithm of HDTV video signals for the purpose of recording HDTV signals in the digital VTR. Comparing to the DCT- based compression method, which only yields unrecognizable DCT coefficients, the low frequency components of Wavelet coefficients maintain recognizable spatial domain information. So, it is more suitable for various VTR operations such as editing and multi-speed mode operations. Also, the adopted Wavelet filter can be Implemented with simple shift operations, which can reduce the computational complexities substantially. The quality of reconstructed HDTV signals with a 4:1 compression ratio turns out to be good enough for the studio use.

• Journal of the Korean earth science society
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• v.32 no.6
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• pp.575-585
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• 2011

The effect of wavelet pair choice in the compression of the satellite images is studied. There is a trade-off between compression rate and perception quality. The encoding ratio is used to express the compression rate, and Peak Signal-to-Noise Ratio (PSNR) is also used for the perceptional performance. The PSNR and the encoding ratio are not matched well for the images with various wavelet pairs, but the tendency is remarkable. It is hard to find the pattern of PSNR for sampled images. On the other hand, there is a pattern of the variation range of the encoding ratio for each image. The satellite images have larger values of the encoding ratio than those of nature images (close range images). Depending on the wavelet pairs, the PSNR and the encoding ratio vary as much as 13.2 to 21.6% and 16.8 to 45.5%, respectively for each image. For Synthetic Aperture Radar (SAR) images the encoding ratio varies from 16 to 20% while for the nature images it varies more than 40% depending on the choice of wavelet pairs. The choice of wavelet for the compression affects the nature images more than the satellite images. With the indices such as the PSNR and the encoding ratio, the satellite images are less sensitive to the choice of wavelet pairs. A new index, energy concentration ratio (ECR) is proposed to investigate the effect of wavelet choice on the satellite image compression. It also shows that the satellite images are less sensitive than the nature images. Nevertheless, the effect of wavelet choice on the satellite image compression varies at least 10% for all three kinds of indices. However, the important of choice of wavelet pairs cannot be ignored.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2074-2084
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• 2001

la this paper, a new digital watermarking algorithm using wavelet transform in frequency domain is suggested. The wavelet coefficients of low frequency subband are utilized to embed the watermark, After the original image is transformed using discrete wavelet transform, their coefficients are transformed into efficient1y in Sequency domain. DCT and FFT transforms are utilized in this processing. Watermark image of general image format is transformed using DCT and the hiding watermark into wavelet coefficients is equally distributed in frequency domain. Next, these wavelet coefficients are performed with inverse transform. The detection process of watermark is performed with reverse direction to insertion process. In this paper, we developed core watermark technologies which are a data hiding technology to hide unique logo mark which symbolizes the copyright and a robust protection technology to protect logo data from external attack like as compression, filtering, resampling, cropping. The experimental results show that two suggested watermarking technologies are invisible and robust.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.89-92
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• 2000

필터뱅크를 기반으로 하는 이산 웨이브렛 멀티톤(DWMT) 데이터 전송 시스템은 전송 채널의 영향으로 발생하는 심벌간 및 부채널간의 간섭 잡음에 민감하다. 본 논문에서 제안한 시간 영역 등화기(TEQ)는 전송 채널 응답의 길이를 단축시키는 TEQ-S와 전송 채널의 주파수 특성을 향상시키는 TEQ-C등 2단계로 이루어져 있다. 제안한 시간영역 등화기(TEQ)를 DWMT 데이터 전송 시스템에 적용하면 많은 간섭 성분을 감소시킬 수 있어 하드웨어적으로 보다 간단한 FEQ로도 잔여 간섭 잡음을 효과적으로 제거할 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2369-2381
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• 1999

In this paper, we propose an iterative algorithm for reducing the blocking artifact in block transform-coded images by using a wavelet transform. In the proposed method, an image is considered as a set of one-dimensional horizontal and vertical signals and one-dimensional wavelet transform is utilized in which the mother wavelet is the first order derivative of a Gaussian like function. The blocking artifact is reduced by removing the blocking component, that causes the variance at the block boundary position in the first scale wavelet domain to be abnormally higher than those at the other positions, using a minimum mean square error (MMSE) filter in the wavelet domain. This filter minimizes the MSE between the ideal blocking component-free signal and the restored signal in the neighborhood of block boundaries in the wavelet domain. It also uses local variance in the wavelet domain for pixel adaptive processing. The filtering and the projection onto a convex set of quantization constraint are iteratively performed in alternating fashion. Experimental results show that the proposed method yields not only a PSNR improvement of about 0.56-1.07 dB, but also subjective quality nearly free of the blocking artifact and edge blur.