• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.47 no.6
• /
• pp.28-38
• /
• 2010

In order to remove the noise contained in recorded analog video, it is important to recognize the real characteristics and strength of the noise. This paper presents an efficient training-based noise reduction method for recorded analog video after analyzing the noise characteristics of analog video captured in a real broadcasting system. First we show that there is non-negligible noise correlation in recorded analog video and describe the limitations of the traditional noise estimation and reduction methods based on additive white Gaussian noise (AWGN) model. In addition, we show that auto-regressive (AR) model considering noise correlation can be successfully utilized to estimate and synthesize the noise contained in the recorded analog video, and the estimated AR parameters are utilized in the training-based noise reduction scheme to reduce the video noise. Experiment results show that the proposed method can be efficiently applied for noise reduction of recorded analog video with non-negligible noise correlation.

• 한국정보통신설비학회:학술대회논문집
• /
• 2005.08a
• /
• pp.115-119
• /
• 2005

In this paper, we propose a method that search the sweet spot of array antenna beam, and keep it for fast speed transmission in millimeter wave on single array antenna link. We use TDD(Time Division Duplex) as transfer method, and it transfers the control data of antenna. The proposed method is the modified genetic algorithm which selects a superior initial group through slave-processing in order to resolve the local solution of genetic algorithm. The efficiency of the proposed method is verified by means of simulations with white Gaussian noise and not on single array antenna link.

• The Journal of the Acoustical Society of Korea
• /
• v.37 no.5
• /
• pp.271-275
• /
• 2018

There are several methods for estimating the time delay between incoming signals to two sensors. Among them, the GCC-PHAT (Generalized Cross Correlation-Phase Transform) method, which estimates the relative delay from the signal whitening and the cross-correlation between the different signal inputs to the two sensors, is a traditionally well known method for achieving stable performance. In this paper, we have identified a part of GCC-PHAT that can improve the periodicity. Also, we apply the auto-correlation method that is widely used as a method to improve the periodicity. Comparing the proposed method with the GCC-PHAT method, we show that the proposed method improves the mean square error performance by 5 dB ~ 15 dB at the SNR above 0 dB for white Gaussian signal source and also show that the method improves the mean square error performance up to 15 dB at the SNR above 2 dB for the color signal source.

• The Journal of the Acoustical Society of Korea
• /
• v.16 no.5
• /
• pp.112-121
• /
• 1997

In this paper, we study a model parameter compensation method for noise-robust speech recognition. We study model parameter compensation on a sentence by sentence and no other informations are used. Parallel model combination(PMC), well known as a model parameter compensation algorithm, is implemented and used for a reference of performance comparision. We also propose a modified PMC method which tunes model parameter with an association factor that controls average variability of gaussian mixtures and variability of single gaussian mixture per state for more robust modeling. We obtain a re-estimation solution of environmental variables based on the expectation-maximization(EM) algorithm in the cepstral domain. To evaluate the performance of the model compensation methods, we perform experiments on speaker-independent isolated word recognition. Noise sources used are white gaussian and driving car noise. To get corrupted speech we added noise to clean speech at various signal-to-noise ratio(SNR). We use noise mean and variance modeled by 3 frame noise data. Experimental result of the VTS approach is superior to other methods. The scheme of the zero order VTS approach is similar to the modified PMC method in adapting mean vector only. But, the recognition rate of the Zero order VTS approach is higher than PMC and modified PMC method based on log-normal approximation.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.22 no.1
• /
• pp.33-39
• /
• 2018

An additive noise reduction algorithm for a mass spectrum analyzer is proposed. From the measured ion signal, we first used an estimated threshold from the mode of the measured signal to eliminate background noises with the white Gaussian characteristics. Also, a signal block corresponding to each mass index is constructed to perform a second order curve fitting and a linear approximation to signal block. In this process, the effective signal block composed of only the ion signal can be reconstructed by removing the impulsive noises and the sample signals which are insufficient to be viewed as normal ion signals. By performing curve fitting on the effective signal block, the noise-free mass spectrum can be obtained. To evaluate the performance of the proposed method, a simulation was performed using the signals acquired from the development equipment. Simulation results show the validity of the threshold setting from the mode and the superiority of the proposed curve fitting and linear approximation based noise canceling algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.10C
• /
• pp.1433-1443
• /
• 2004

The power line modem proposed in this paper transmits the remote control signal using CSK(Code Shift Keying) and DS/SS method. The CSK technique provides the increased capacity of transmission and robustness towards noise. Besides, the DS/SS technique provides protection against narrow-band Gaussian interference and multi-path interference. The modem supports full-duplex communication using FDD(Frequency Division Duplex) and the modem structure for forward link is same with that for reverse link. To switch each sub-controlled unit smoothly, 4/$\pi$-DQPSK is adopted for noncoherent demodulation. The PN code for spreading spectrum seues to divide each group which consists of sub-controlled units and Walsh code is used for the M-ary CSK technique. Each block is designed and verified with TMS320C5402 DSP. We show the superiority of the proposed method by analyzing numerically the system performance for the factors of the DS/SS and CSK method ullder additive white Gaussian noise and PBI.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.11B
• /
• pp.1872-1878
• /
• 2000

Turbo Code는 반복 부호 알고리듬을 사용함으로써 백색 가우시안 잡음(AWGN)채널 환경하에서 Shannon의 한계에 가까운 성능을 보이는 오류정정 방식으로 제안되었다. 그러나 Turbo code는 반복복호로 인해 매 복호시마다 큰 인터리버와 복호기를 거쳐야 하기 때문에 수신과정에서 커다란 지연을 요구하게 된다. 따라서 차세대 무선 멀티미디어 통신에서 실시간 음성서비스나 화상서비스를 제공하는데 어려움이 많다. 본 논문에서는 기존의 터보 복호기를 변형하여 매 복호시 각각의 복호기에서 LLR 출력시퀀스를 발생시킴으로써 반복 복호 횟수를 줄이는 방법을 제안하였다. 이렇게함으로서 기존의 Toubo code가 갖는 성능은 크게 변화시키지 않으면서 각각의 정보프레임을 가변적으로 복호함으로서 반복 복호로 인한 시간 지연을 줄였다.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.597-600
• /
• 2000

Turbo Code는 반복 복호알고리듬을 사용함으로써 백색 가우시안 잡음(AWGN)채널 환경에서 Shannon의 한계에 가까운 성능을 보이는 오류정정 방식으로 제안되었다. 그러나 Turbo code는 반복복호로 인해 매복호시마다 큰 인터리버와 복호기를 거쳐야 하기 때문에 수신과정에서 커다란 지연을 요구하게 된다. 따라서 차세대 무선 멀티미디어 통신에서 실시간으로 음성서비스나 화상서비스를 제공하는데 어려움이 많다. 본 논문에서는 기존의 터보 복호기를 변형하여 매 복호시 각각의 복호기에서 출력시퀀스를 발생시킴으로서 반복 복호 횟수를 줄이는 방법을 제안하였다. 이렇게 함으로서 기존의 Turbo code가 갖는 성능은 크게 변화시키지 않으면서 각각의 정보프레임을 가변적으로 복호함으로서 반복복호로 인한 시간 지연을 줄일수 있었다.

• Proceedings of the IEEK Conference
• /
• 1998.06a
• /
• pp.102-105
• /
• 1998

차세대 이동통신 시스템을 위해 활발히 연구되고 있는 MC-CDMA 방식은 OFDM과 DS-CDMA의 장점을 결합하여 심벌간 간섭과 페이딩에 강하고 IFFT/FFT를 이용하여 효과적으로 변조/복조부를 구현할 수 있다는 장점을 갖는다. 하지만, 송신 신호가 다중 레벨 진폭 특성을 갖기 때문에 손신기에서 사용되는 고출력 증폭기의 포화 특성으로 인해 심각한 비선형 왜곡이 발생한다. 본 논문에서는 OFDM에서 이러한 비선형 왜곡의 보상을 위해 우리가 제안한 고정점 반복 기반의 사전왜곡기를 MC-CDMA 시스템에 적용하고, 이의 우수한 성능을 부가성 백색 가우시안 잡음 채널에서 64-FFT/IFFT을 이용하여 변조 및 복조를 수행하는 MC-CDMA 시스템에 대한 모의실험을 통해 확인하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2011.05a
• /
• pp.559-562
• /
• 2011

In this research, We derived Recursive Least Squares(RLS) algorithm with adaptive maximum-likelihood channel estimate for digital pulse amplitude modulated sequence in the presence of intersymbol interference and additive white Gaussian noise. RLS algorithms have better convergence characteristics than conventional algorithms, LMS (Least Mean Squares) algorithms.