• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.680-685
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• 2009

In this paper, we propose a efficient noise power estimation technique for an integrated acoustic echo and noise suppression system in a frequency domain. The proposed method uses speech absence probability (SAP) derived from the microphone input signal as the smoothing parameter updating noise power to reduce the noise power estimation error resulted from the distortions in the unified structure where the noise suppression (NS) operation is placed after the acoustic echo suppression (AES) algorithm. Therefore, in the proposed approach, the smoothing parameter based on SAP derived from the input signal instead of echo-suppressed signal should stop updating noise power estimates during the distorted noise spectrum periods. The performance of the proposed algorithm is evaluated by the objective test under various environments and yields better results compared with the conventional scheme.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.161-164
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• 2000

본 논문에서 제안한 동시통화 검출기는 기존의 전대역에서 이루어지던 상호상관계수를 이용한 동시통화 검출기의 검출성능을 향상시키기 위하여 웨이브릿변환된 각각의 서브밴드 내에서 동시통화 및 반향경로를 구별하여 효율적으로 검출할 수 있도록 구성하였다. 서브밴드 동시통화 검출기 사용으로 동시통화 시에 발생하는 적응필터의 계수 발산을 막음으로써 시스템의 안정성을 높이고, 근단화자 신호가 원단화자에게 더 유쾌하게 들릴 수 있게 함으로써 원활한 통화환경을 제공할 수 있도록 구현하였다.

• The Journal of the Acoustical Society of Korea
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• v.17 no.7
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• pp.74-78
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• 1998

본 논문에서는 고속의 변환속도를 갖는 파이프라인드 방식과 저전력 특성을 갖는 축차 비교 방식 구조를 혼용하여 고속, 저전력 아날로그-디지털 변환기를 설계하였다. 제안 된 구조는 축차 비교 방식의 변환에서 비교기를 파이프라인드 구조로 연결하여 홀드된 주기 에 비교기의 기준 전위를 전 비교기의 출력값에 의해 변환하도록 하여 고속 동작이 가능하 도록 하였다. 제안된 구조에 의해 8비트 아날로그 디지털 변환기를 0.8㎛ CMOS공정으로 HSPICE를 이용하여 시뮬레이션한 결과, INL/DNL은 각각 ±0.5/±1이었으며, 100kHz 사인 입력 신호를 10MS/s로 샘플링 하여 DFT측정 결과 SNR은 41dB를 얻을 수 있었다. 10MS/s의 변환 속도에서 전력 소모는 4.14mW로 측정되었다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.121-128
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• 2003

Acoustic Echo Cancellation(m) is very active research topic having many applications like teleconference and hands-free communication and it employs Double Talk Detector(DTD) to indicate whether the near-end speaker is active or not. However. the DTD is very sensitive to the variation of acoustical environment and it sometimes provides wrong information about the near-end speaker. In this paper, we are focusing on the development of robust DTD algorithm which is a basic building block for reliable AEC system. The proposed AEC system consists of delayless subband AEC and narrow-band DTD. Delayless subband AEC has proven to have excellent performance of echo cancellation with a low complexity and high convergence speed. In addition, it solves the signal delay problem in the existing subband AEC. On the other hand, the proposed narrowband DTD is operating on low frequency subband. It can take most advantages from the narrow subband such as a low computational complexity due to the down-sampling and the reliable DTD decision making procedure because of the low-frequency nature of the subband signal. From the simulation results of the proposed narrowband DTD and wideband DTD, we confirm that the proposed DTD outperforms the wideband DTD in a sense of removing possible false decision making about the near-end speaker activity.

• Journal of the Korean Society for Nondestructive Testing
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• v.16 no.1
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• pp.34-45
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• 1996

The signal pattern recognition program which can perform signal acquisition and processing, the extraction and selection of features, the classifier design and the evaluation, is developed and applied to the classification of artificial defects in the weld zone of Austenitic STS304. The neural network classifier is compared with the linear discriminant function classifier and the empirical Bayesian classifier. The signal through a broadband sensor is compared with that through a resonance type sensor. In recognition rate, the neural network classifier is best, and the signal through a broadband sensor is better.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.9 no.5
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• pp.25-37
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• 2010

This study is aimed at enhancing mobility rights for the transportation underprivileged that has been made light of relatively compared to normal people. In order to do this, we've suggested having ITS (Intelligent Traffic System) built and improving satisfaction through the test operation of its main system. The existing sound signal device for the visually handicapped has one problem with managing it. Because, the people in charge of it had to visit each problematic site directly to maintain and fix some problems every time it was out of order. Moreover, it couldn't provide sustainable services about voice guidance and the visually handicapped had to control it by either confirming the location of buttons that were installed on the pillar of traffic light and then pressing one of them or using a remote controller on their own. In order to improve such inconveniences, we have created a new typed sound signal device for the visually handicapped by applying the cutting-edge wireless technology based on ergonomics considering actual road situations. Such technology enables it report the status of signal device and light to them by using its voice guidance system automatically every time they have access to it. Additionally, we've already introduced it to a couple of test areas and then known the fact that they recognized traffic situation more conveniently and safely compared to the existing sound signal device. That is above average in terms of satisfaction. In addition to that, we've provided LTS (Location Tracking System - Location-based service intended for elementary students) by utilizing the existing wireless infrastructure and founded the fact that about 87% of their parents were satisfied with the service based on LTS.

• The Journal of the Acoustical Society of Korea
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• v.30 no.8
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• pp.461-467
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• 2011

Accurate classification of input signals is the key prerequisite for variable bit-rate coding, which has been introduced in order to effectively utilize limited communication bandwidth. Especially, recent surge of multimedia services elevate the importance of speech/music classification. Among many speech/music classifier, the ones based on support vector machine (SVM) have a strong selling point, high classification accuracy, but their computational complexity and memory requirement hinder their way into actual implementations. Therefore, techniques that reduce the computational complexity and the memory requirement is inevitable, particularly for embedded systems. We first analyze implementation of an SVM-based classifier on embedded systems in terms of execution time and energy consumption, and then propose two techniques that alleviate the implementation requirements: One is a technique that removes support vectors that have insignificant contribution to the final classification, and the other is to skip processing some of input signals by virtue of strong correlations in speech/music frames. These are post-processing techniques that can work with any other optimization techniques applied during the training phase of SVM. With experiments, we validate the proposed algorithms from the perspectives of classification accuracy, execution time, and energy consumption.

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

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.361-364
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• 1998

주파수 합성기는 주로 PLL을 이용하여 설계하는데, PLL(Phase-lock loop)이란 출력신호 주파수를 항상 일정하게 유지하도록 구성된 주파수 부귀환 회로로써 기본적인 구성은 위상출력기, 저역통과필터, 전압 제어 발진기로 이루어진다. 이런 PLL의 기본적인 구성에 프로그래머블카운터를 VCO의 출력단에 부가하여 구성한 형태가 주파수합성기이다. 이 주파수합성기의 출력을 프로그래머블 디바이더에 입력하기 전에 주파수를 낮출 필요가 있는데, 현재 슈퍼헤테로다인 다운 컨버터방식과 프리스케일러방식과 펄스 스웰로 카운터를 사용하는 방식 등의 3가지 방법이 있다. 본 논문에서는 펄스 스웰로 카운터 방식의 주파수 합성기를 MATLAB의 GUI환경과 병행하여 시뮬레이션 과정을 통한 동작특성을 이해하고, 한 화면에서 이루어지는 조작에 의해 모든 주파수 합성기의 요소를 관찰할 수 있도록 모델링하였다. 그리고, 모델링한 주파수합성기와 실제 주파수합성기에서 예상되는 출력과 비교하여 그 결과에 있어서 얼마나 유사한지 살펴보았다.

• Journal of the Korean Society for Nondestructive Testing
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• v.21 no.1
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• pp.46-53
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• 2001

The purpose of this paper is to develop a new feature selection method for AE signal classification. The neural network of back propagation algorithm is used. The proposed feature selection method uses the difference between feature coordinates in feature space. This method is compared with the existing methods such as Fisher's criterion, class mean scatter criterion and eigenvector analysis in terms of the recognition rate and the convergence speed, using the signals from the defects in welding zone of austenitic stainless steel and in the metal contact of the rotary compressor. The proposed feature selection methods such as 2-D and 3-D criteria showed better results in the recognition rate than the existing ones.