• Proceedings of the KIEE Conference
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• 2005.07a
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• pp.75-77
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• 2005

전력선통신 시스템은 일반적인 통신시스템과는 달리 전력선에 연계된 다양한 부하기기들의 동작 및 변동에 따른 잡음이 발생한다. 특히 부하의 기동, 정지 시 발생하는 임펄스 잡음은 통신성능 저하의 주된 요인임으로 통신성능 향상과 신뢰성 높은 데이터 전송을 위하여 임펄스 잡음 제거와 임펄스 잡음 영향을 감쇠하기 위한 방안이 절실히 요구되고 있다. 따라서, 본 논문에서는 전력선 통신시스템에서 전송된 데이터를 오류 없이 복원해 내기 위하여 적응형 정합필터링 기법을 제안하고, 수신 측에서 임펄스잡음에 따른 최적의 탭 계수를 선택해 효과적으로 임펄스잡음을 제거하는 기법을 확립하였다. 또한, 제시한 알고리즘에 다양한 임펄스 잡음의 모델을 반영하여 비트오율에 대한 모의실험을 행하고 결과를 도출함으로써 제안 방식의 유용성과 신뢰성을 확인하였다.

• Journal of the Korean Society for Precision Engineering
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• v.26 no.12
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• pp.146-153
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• 2009

One of the most important issues in the wearable healthcare sensors is to minimize the motion artifacts in the vital signals for continuous monitoring. This paper presents a reflected type photoplethysmograph (PPG) sensor for monitoring heart rates at the artery of the wrist. Active noise cancellation algorithm was applied to compensate the distorted signals by motions with Least Mean Square (LMS) adaptive filter algorithms, using acceleration signals from a MEMS accelerometer. Experiments with a watch type PPG sensor were performed to validate the proposed algorithm during typical daily motions such as walking and running. The developed sensor is suitable for ubiquitous healthcare system and monitoring vital arterial signals during surgery.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.185-188
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• 2000

현대의 잡음이 많은 환경에서 적응 음향반향제거기는 배경잡음의 영향으로 원활한 통화환경을 제공할 수 없다. 이러한 문제점을 해결하기 위하여 음향반향 제거와 더불어 배경잡음을 제거하는 결합구조의 적응 음향반향제거기가 제안되었다. 본 논문에서는 기존의 결합구조가 가지는 단점을 보완하여 적응 음향반향제거기의 성능을 향상시켰다. 제안한 결합구조는 적응 음향잡음제거기의 기준입력 신호를 적응 음향잡음제거기의 오차신호와 같게 구성함으로서 배경잡음 신호뿐만 아니라 잔여반향 신호도 효율적으로 제거할 수 있다. 성능 평가를 위한 실험결과, 제안한 방법이 기존의 방법에 비하여 ERLE 성능이 수렴 구간에서 3㏈ 이상 향상되었음을 확인하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2020.07a
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• pp.346-347
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• 2020

본 논문에서는 ANC 시스템의 빠른 수렴을 위한 적응 필터의 초기화 방법을 제안한다. 기존 ANC 시스템은 적응 필터의 계수를 0으로 초기화한다. 이러한 초기화 방법은 일반적으로 발생하는 외부 소음의 특성을 고려하지 않은 방법으로 ANC의 수렴 소요시간이 길다. 이와 같은 문제를 해결하고자 본 논문에서는 핑크 노이즈를 입력으로 ANC를 수행하여 얻은 적응 필터의 계수 값을 초기값으로 사용하는 새로운 초기화 방법을 제안한다. 제안한 방법으로 여러 잡음에 대해 실험한 결과, 낮은 초기 에러를 갖고 기존 방법보다 빠르게 수렴하는 것을 확인하였다. 또한, 기존 방법에서 수렴하지 못한 일부 소음에 대해서도 수렴하는 것을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.73-84
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• 1992

This paper studies noise subtraction methods and distance measures for speech recognition in a noisy environment, and investigates noise robustness of the distance measures applied to the problem of isolated word recognition in white Gaussian and colored noise (vehicle noise) environments. Noise subtraction methods which can be used as a pre-processor for the speech recognition system, such as the spectral subtraction method, autocorrelation subtraction method, adaptive noise cancellation and acoustic beamforming are studied, and distance measures such and Log Likelihood Ratio ($d_{LLR}$), cepstral distance measure ($d_{CEP}$), weighted cepstral distance measure ($d_{WCEP}$), spectral slope distance measure ($d_{RPS}$) and cepstral projection distance measure ($d_{CP},\;d_{BCP},\;d_{WCP},\;d_{BWCP}$) are also investigated. Testing of the distance measures for speaker-dependent isolated word recognition in a noisy environment indicate that $d_{RPS}\;and\;d_{WCEP}$ which weigh higher order cepstral coefficients more heavily give considerable performance improvement over $d_{CEP}and\;d_{LLR}$. In addition, when no pre-emphasis is performed, the recognizer can maintain higher performance under high noise conditions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• 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.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.1
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• pp.18-25
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• 2018

In this paper, we present a low-power implementation of the multi-channel hearing aid system using a general-purpose DSP chip. The system includes an acoustic amplification algorithm based on Wide Dynamic Range Compression (WDRC), an adaptive howling canceller, and a single-channel noise reduction algorithm. To achieve a low-power implementation, each algorithm is re-constructed in forms of integer program, and the integer program is converted to the assembly program using BelaSigna(R) 250 instructions. Through experiments using the implementation system, the performance of each processing algorithm was confirmed in real-time. Also, the clock of the implementation system was measured, and it was confirmed that the entire signal processing blocks can be performed in real time at about 7.02MHz system clock.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• Journal of Sensor Science and Technology
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• v.27 no.3
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• pp.192-198
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• 2018

Pulse oximetry is a non-invasive method for monitoring how much oxygenated hemoglobin is present in the blood. The principle of pulse oximetry is based on the red infrared light adsorption characteristics of oxygenated and deoxygenated hemoglobin. Even through the convenience of a pulse oximeter, its weak signal-to-noise ratio against motion artifacts and low perfusion makes it difficult to be accepted by execs devices. Several researchers have suggested the use of an adaptive noise cancellation (ANC) algorithm. They have demonstrated that ANC is feasible for reducing the effects of motion artifacts. Masimo Corporation developed a discrete saturation transformation (DST) algorithm that uses a reference signal and ANC. In commercial devices, it is very hard to escape it because Masimo's patents are very powerful and a better method is yet to be developed. This study proposes a new method that can measure noise saturation as well as accurate oxygen saturation from signals with high motion artifacts without using ANC and DST. The proposed algorithm can extract a normal signal without noise from a signal with motion artifacts. The reference signal from a pulse oximeter simulator was used for the evaluation of our proposed algorithm and achieved good results.