• 한국통신학회논문지
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• 제30권1C호
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• pp.54-60
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• 2005

본 논문은 지연 추정 기능을 갖는 적응 마이크로폰 어레이 알고리즘을 제안한다. Generalized sidelobe canceller (GSC)의 적응 차단 행렬이 각 센서간의 상호 시간 지연을 추정할 수 있다는 것을 보임으로써, 제안한 시스템은 적응 차단 행렬을 목적 신호의 차단 뿐 아니라 각 센서의 시간 지연 추정을 위해 사용한다. 이로 인해, 제안한 시스템은 GSC 구조만을 사용하면서. 시간 지연 추정기를 외부의 전처리기로 사용하는 기존의 시스템과 같은 성능을 얻을 수 있다. 실제 환경에서의 실험 결과를 통해 제안한 시스템의 성능이 기존의 시스템과 유사함을 확인하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(2)
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• pp.293-296
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• 2000

In this paper, we designed a 192-tap echo canceller for a modem using VHDL. To evaluate errors, we used adaptive algorithm. We adopted pipeline technique and realized delay-taps with RAMs. We simulated this design using Altera MAX+PLUS II.

• 한국통신학회논문지
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• 제21권8호
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• pp.2033-2042
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• 1996

In this papre, we describe a recently implemented echo canceller for digital cellular communication of Code Division Multiple Access(CDMA) that features time sharing of digital signal processor(DSP) over four channels in one DSP to reduce per channel costs. In the Public Land Mobile Network(PLMN), it is important to cancel the echo reflected from the Public Switched Telephone Network(PSTN) side. In case of digital mobile system, the round-trip delay of the echo is in excess of about 180 milliseconds due to frame-by-frame voice coding. It is necessary to cancel the echo in PLMN. We have developed a multi-channel echo canceller tht operates with Time Switch Module in a Mobile Switching Center(MSC). The general echo canceller needs PCM trunk interface circuits and the tone detection and disabling circuits, but the multi-channel echo canceller linked with Time Switch Module does not need them. Therefore we could develop the effective and economical echo canceller.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• 제어로봇시스템학회논문지
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• 제5권7호
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• pp.800-807
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• 1999

A Periodic disturbance canceller is proposed to compensate for the periodic disturbance due to cutting process in a CNC machining center. The periodic disturbance canceller estimates the Periodic disturbance without phase delay. This is achieved by using linear phase low-pass filter and frequency response reciprocal filter of plant at the frequency of the periodic disturbance. This method is implemented in the position control system of the CNC machining center with general disturbance compensators in order to compensate for both the frictional force and the periodic disturbance. The experimental results are described to show its effectiveness.

• 음성과학
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• 제8권2호
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• pp.73-81
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• 2001

In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

• 한국전자파학회논문지
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• 제16권8호
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• pp.825-834
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• 2005

본 논문에서는 인가되는 두 입력 단자의 신호들을 동일 군속도를 가지게 하면서 신호 상쇄를 가능하게 하는 새로운 구조의 신호 상쇄기를 제안하였고, 이를 적용한 feedforward 전력 증폭기를 제작하였다. 종전의 신호 상쇄기는 동작 주파수 전 대역에서 위상 정합과 군속도 지연 정합을 동시에 만족시킬 수 없으나 제안된 동일 군속도 지연 신호 상쇄기는 위상 정합과 군속도 지연 정합을 동시에 만족시킬 수 있다. 위상과 군속도 지연 정합은 광대역 신호 상쇄를 가능하게 한다. 제안한 동일 군속도 지연 신호 상쇄기를 적용한 feedforward 전력 증폭기의 주 신호 제거 루프에서는 $2,140{\pm}100$ MHz 대역에서 26.3 dB 이상의 신호 상쇄를 얻었고, 혼변조 왜곡 신호 제거 루프에서는 15.2 dB 이상의 신호 상쇄를 얻었다. 또한 2,115 MHz와 2,165 MHz인 2-tone 신호를 인가하여 (C/I)를 20.8 dB 개선시켰다.

• 대한전자공학회논문지SP
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• 제39권2호
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• pp.128-136
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• 2002

본 논문에서는 이동통신 단말기의 음성부호화기에 탑재할 수 있도록 TeakLite DSP Core를 이용한 음향반향제거기(Acoustic Echo Canceller)를 실시간으로 구현하였다. 음성부호화기에서 음향반향제거기가 사용할 수 있는 연산량의 제한때문에 적응필터는 NLMS(Normalized Least Mean Square) 알고리즘을 이용한 FIR 필터를 사용하였다. 먼저 음향반향제거기를 부동소수점 C-언어로 구현한 다음 고정소수점 시뮬레이션을 통하여 고정소수점 연산으로 바꾸었다. 그리고 고정소수점 연산 결과를 기반으로 어셈블리 언어로 프로그램을 작성하고 최적화 과정을 거쳐 실시간으로 동작하도록 하였다. 최종적으로 구현된 반향제거기는 프로그램 메모리가 624 words이고 데이터 메모리는 811 words이었다. 샘플링 주파수를 8 ㎑로 하였을 때, 32 msec의 반향경로 지연시간에 해당되는 256 차수의 필터를 이용한 경우에는 14.12 MIPS의 연산량을, 16 msec의 반향경로 지연시간에 해당되는 128 차수의 필터를 이용한 경우에는 9.00 MIPS의 연산량을 필요로 하였다.

• 한국멀티미디어학회논문지
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• 제19권2호
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• pp.189-199
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• 2016

Many implantable hearing aids are being developed as alternatives to conventional hearing aids which has inconveniences for use and social stigma that make hearing-impaired people avoid to wear it. Particularly, the fully-implantable middle ear hearing devices (F-IMEHD) are being actively studied for mixed or sensorineural hearing impaired people. In development of F-IMEHD, the most difficult problem is improving the performance of implantable microphone. Recently, Cho et al. have studied the tympanic membrane installed microphone which has better sensitivity and is easier to operate on patient than the microphone implanted under the skin. But, it may cause howling problem due to the feedback signal via oval window and ossicle chain from the transducer on round window in the middle ear cavity, therefore, a feedback canceller is necessary. In this paper, we designed NLMS (normalized least mean square) adaptive feedback canceller for F-IMEHD with tympanic membrane installed microphone and a transducer implemented at round window, and computer simulation was performed to verify its operation. The designed adaptive feedback canceller has a delay filter, a 64 point FIR fixed filter and a 8-tap adaptive FIR filter. Computer simulation of the feedback path is modeled by using the data obtained through human cadaver experiment.

• 전자공학회논문지B
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• 제30B권12호
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.