• Journal of the Institute of Convergence Signal Processing
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• v.1 no.1
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• pp.7-14
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• 2000

Recently, the technologies of adaptive noise cancellation(ANC) are developed fast and widely due to the highly sophisticated digital signal processing algorithms and the high-speed communication networks and devices. But, thousand numbers of the adaptive filter taps are required to obtain the satisfying results in the fields of the adaptive noise cancellation and echo cancellation. In the paper, performance comparisons based on the real-time processing between frequency domain adaptive filter(FDAF) and multi-delay frequency domain adaptive filter(MDF) are carried. Those algorithms provide us with the reductions of the computational burdens and the increase of the convergence rate for the lengthy Fill adaptive filters. The time delay due to the long taps of FDAF can be reduced by adopting the MDF algorithms. The conventional ANC and cross talks ANC using FDAF are implemented on the dSP ACE 1103 real-time signal processing board.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.10-15
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• 1998

Application of the Filtered-X LMS adaptive filter to active noise cancellation requires to estimate the transfer characteristics between the output and the error signal of the adaptive canceler. In this paper, we derive an adaptive cancellation algorithm and analyze is convergence behavior when the acoustic noise is assumed to consist of multiple sinusoids. The results of the convergence analysis of the Filtered-X LMS algorithm indicate that the effects of parameter estimation inaccuracy on the convergence behavior of the algorithm are characterize by two distinct components : Phase estimation error and estimated magnitude. In particular, the convergence of the Filtered-X LMS algorithm is show to be strongly affected by the accuracy of the phase response estimate. Simulation results of the algorithm are presented which support the theoretical convergence analysis.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.225-228
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• 2000

It is well known that a Frequency Domain Adaptive Filter (FDAF) converges faster than a Time Domain Adaptive Filter (TDAF) even when the input signal is colored such as a speech signal. We have proposed the FDAF using the Modified Discrete Fourier Transform Pair (MDFTP) and its realization and effectiveness has been confirmed through the computer simulations. In this paper, we apply the FDAF using the MDFTP to the noise cancellation system. The proposed system is based on the Adaptive Line Enhancer (ALE) and utilizes single microphone; therefore it is suitable for the portable electronic equipment. Moreover, we propose to utilize the MDFT for detecting of the pitch in the speech because the number of data points in the MDFT must be equal to the pitch to confirmed that the noise can be removed to near the level of SNR.

• Proceedings of the KIEE Conference
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• 1995.07b
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• pp.904-906
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• 1995

Based on the detection of the size of datas of multipath channel characterization, we propose a new algorithm. called the impulse size based adaptive median filter(ISMF), for ghost cancellation system. The ISMF consists of two levels. The first one is the impulse noise size detection level and the second one is the adaptive median filtering level to remove the impulse noise detected. Test results confirm that the proposed ISMF removes impulse noise due to multipath channel characterization while preserving signal as well as ghosts so that the LMS algorithm performs effectively.

• Proceedings of the Korea Information Processing Society Conference
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• 2021.11a
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• pp.1203-1206
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• 2021

소음은 난청, 스트레스 등의 원인이 된다. 본 연구에서는 ANC(Active Noise Cancellation)을 바탕으로, 기술적인 방법을 통해 소음을 저감 시키는 스피커를 구현하였다. ANC 란 소음 주파수의 위상을 180° 변환하여 주파수와 레벨이 동일한 역 소음을 발생시켜 주변 소음을 저감, 차단하는 기술이다. 현재 시중 제품들에 적용되는 일반적인 ANC 의 경우, 피드백(Feedback) 방식이라는 점과 시간 지연(Time gap)이 발생한다는 한계가 있다. 이를 보완하기 위해 AI 학습으로 소음을 미리 예측하여 시간 지연을 줄이는 방법을 고안했다. 순환 신경망(RNN)의 장기의존성 문제를 해결하는 시계열 예측 딥러닝 알고리즘인 LSTM(Long Short-Term Memory Network) 모델을 사용하였다. 또한, AI 학습 효율을 향상시킬 수 있는 하드웨어 장비들을 활용하였다.

• International Journal of Highway Engineering
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• v.17 no.6
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• pp.11-18
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• 2015

PURPOSES : The purpose of this thesis is to evaluate the effectiveness of an active noise cancellation (ANC) system in reducing the traffic noise level against frequencies from the predictive model developed by previous research. The predictive model is based on ISO 9613-2 standards using the Noble close proximity (NCPX) method and the pass-by method. This means that the use of these standards is a powerful tool for analyzing the traffic noise level because of the strengths of these methods. Traffic noise analysis was performed based on digital signal processing (DSP) for detecting traffic noise with the pass-by method at the test site. METHODS : There are several analysis methods, which are generally divided into three different types, available to evaluate traffic noise predictive models. The first method uses the classification standard of 12 vehicle types. The second method is based on a standard of four vehicle types. The third method is founded on 5 types of vehicles, which are different from the types used by the second method. This means that the second method not only consolidates 12 vehicle types into only four types, but also that the results of the noise analysis of the total traffic volume are reflected in a comparison analysis of the three types of methods. The constant percent bandwidth (CPB) analysis was used to identify the properties of different frequencies in the frequency analysis. A-weighting was applied to the DSP and to the transformation process from analog to digital signal. The root mean squared error (RMSE) was applied to compare and evaluate the predictive model results of the three analysis methods. RESULTS : The result derived from the third method, based on the classification standard of 5 vehicle types, shows the smallest values of RMSE and max and min error. However, it does not have the reduction properties of a predictive model. To evaluate the predictive model of an ANC system, a reduction analysis of the total sound pressure level (TSPL), dB(A), was conducted. As a result, the analysis based on the third method has the smallest value of RMSE and max error. The effect of traffic noise reduction was the greatest value of the types of analysis in this research. CONCLUSIONS : From the results of the error analysis, the application method for categorizing vehicle types related to the 12-vehicle classification based on previous research is appropriate to the ANC system. However, the performance of a predictive model on an ANC system is up to a value of traffic noise reduction. By the same token, the most appropriate method that influences the maximum reduction effect is found in the third method of traffic analysis. This method has a value of traffic noise reduction of 31.28 dB(A). In conclusion, research for detecting the friction noise between a tire and the road surface for the 12 vehicle types needs to be conducted to authentically demonstrate an ANC system in the Republic of Korea.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.2
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• pp.308-315
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• 2001

In this paper, a new power amplifier is proposed for reduction of amplified RX band noise signals as well as intermodulation distortion signals using feedforward technique. This power amplifier is implemented for IMT-2000 basestation TX frequency band. Both TX band intermodulation distortion signals and RX band noise signals are reduced by controlling variable attenuator, phase shifter and error amplifier. The proposed power amplifier, which contains two loops-intermodulation distortion signals cancellation loop and RX band noise signals cancellation loop, can provide duplexer with low TX path insertion loss for various wireless communication systems due to choice of loose RX attenuation characteristic. The principle of the proposed amplifier is described graphically based on the conceptual schematic diagram. A two-tone test for power amplifier is done at 2.14GHz with frequency spacing of 5MHz, and RX band rejection test is done over RX full band of 60MHz with 1.95GHz center frequency. Experimental results represent that the cancellation performance of intermodulation distortion signals and RX band noise signals are more than 3 1dB and 21dB, respectively.

• Journal of KSNVE
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• v.9 no.3
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• pp.600-606
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• 1999

A reactive type acoustic filter is developed based on the phase cancellation effect which is occurring in the plane wave propagation through the two paths where the cross sectional areas are reversely changing. The theory is experimentally validated by the use of a cylindrical duct and an inserted hollowed cone of which vertex part is eliminated. Noise attenuation and the filtered frequency are dependent on the area variation and the effective length of the filter. Experimental comparison shows that the filtered frequencies of 1st and 2nd mode are lower than the analytical prediction due to the mass loading effects, and the 3rd mode is in good agreement. The proposed filter can be applied as an in-duct noise filter for improving the sound quality in a narrow space for various industrial applications.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.727-730
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• 2002

This paper proposes the interference cancellation improvement in smart antenna system by using Applebaum array covariance matrix adjustment. This technique adds the specific adjustable multipliers with both desired signal covariance matrix and interference signal covariance matrices in order to overcome some disadvantages and improve the interference cancellation efficiency of Applebaum array. It is based on the desired and undesired signal power or desired signal-to-interference-plus-thermal noise ratio (SINR). As the result from demonstration, the proposed technique can improve and increase the interference cancellation efficiency in smart antenna better than the conventional technique.

• Journal of electromagnetic engineering and science
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• v.9 no.2
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• pp.61-65
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• 2009

Successive interference cancellation(SIC) for the uplink of MC-CDMA mobile communications systems is an effective method to improve performance. We propose a successive interference cancellation(SIC) technique for the uplink of MC-CDMA mobile systems with multiple receive antennas. The destination uses optimum combining(OC) to combine the signals from an OFDM Demodulator with multiple receiving antennas, and applies SIC processing to the combined signals. Achieved interference cancellation order is depends on the signal to interference-plus-noise ratio (SINR) at the output of the optimum combiner. Monte-Carlo simulations are employed to verify the proposed technique.