• Journal of Korea Multimedia Society
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• v.11 no.3
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• pp.281-291
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• 2008

In this paper, we propose a new double talk detection algorithm which detects near end signals with less degradation, tracking echo path variation of echo canceler simultaneously. Our method makes use of a cross-correlation between channel input signals and estimated error signals and a normalized cross-correlation between microphone input signals and estimated error signals. By combing thresholds for these cross-correlations pertinently, this algorithm discriminates between variation of echo path and occurrence of double talk. These two cross-correlation are used to detect double talk periods, tracking echo path variation. During the detection period, adjustive adaptive filter is ceased to prevent the echo canceler from being disturbed by near end signals. Also, the echo canceler will still be kept on for tracking any variation in echo path. Through computer simulation results, it was confirmed that the proposed algorithm shows better performance, tracking echo path variation and detecting the double talk periods, than the Ye et. al's and the NLMS algorithms from ERLE viewpoint.

• 전자공학회논문지 IE
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• v.44 no.4
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• pp.1-10
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• 2007

In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.291-297
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• 2004

In this paper, a new structure for the AEC(Acoustic Echo Canceller) is proposed in which echo signal components that can be created in mobile communications is effectively eliminated. Block Data Flow Architecture is a parallel architecture that achieves high performance, high efficiency, high throughput, and almost linear speed up. The proposed architecture employs AEC and is implemented using the TMS320C6711 for real-time applications. The proposed AEC shows improved performance by eliminating echoes at 55ms delay path. Since the proposed AEC can also be implemented in Firmware, it is believed to effectively work on various types of echoes if it is applied on CDMA mobile devices. The TMS320C6711 shows much better performance comparing to previous DSPs. For experimental verifications, filtering operation using adaptive algorithm is performed on TMS320C6711 board and error signals resulted from computations are monitored on PC, and then the performance of the implemented AEC is verified through ERLE computation. According the results of simulation, good characteristic of 100dB are shown after 500 sampling data.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.520-528
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• 2003

This paper proposes a method of implementing synthetic stereo and an acoustic echo cancellation algorithm for multiple participant conference system. Synthetic stereo is generated by HRTF and two loudspeakers. A robust adaptive algorithm for synthetic stereo echo cancellation is proposed to reduce the weight misalignment due to near-end speech signals and ambient noises. The proposed adaptive algorithm is modified version of SMAP algorithm and the coefficients of adaptive filter is updated with cross correlation of input and estimation error signal normalized with sum of the autocorrelation of input signal and the power of the estimation error signal multiplied with projection order. This is more robust to projection order and ambient noise than conventional SMAP. Computer simulation show that the proposed algorithm effectively attenuates synthetic stereo acoustic echo.

• Journal of Korea Multimedia Society
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• v.3 no.6
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• pp.604-610
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• 2000

A stereo acoustic echo canceller has significant cross-correlation between each channel signal. As the result, the adaptive filter coefficient that estimates the acoustic echo path of a receiving room can misconverge to the path or converge slowly. In this paper, a new preprocessor using absolute difference in stereo signals is proposed to reduce cross-correlations and to improve the removal efficiency of the stereo acoustic echo. Compared to the previous preprocessor using a half wave rectifier, the newly proposed preprocessor showed better performance according to computer simulation. In addition, when the paths of transmitting room were changed the performance of the proposed preprocessor was not degraded.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2388-2396
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• 2012

This paper is on an simplified orthogonal projection method which cancel the acoustic echo signals in the stereo acoustic echo canceller. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances for signals with the high auto-correlation, and has small computational quantities. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it was proved that the stereo acoustic echo canceller adopting the proposed algorithm shows about 3dB more high ERLE than the NLMS algorithm for the white noise signals, and 5dB for the colored voice signals.

• Proceedings of the Korea Multimedia Society Conference
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• 2001.06a
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• pp.124-127
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• 2001

핸즈프리 음성 통신에 있어 음향 반향 제거 시스템은 필수적인 요소이다. 본 논문에서는 음향 반향을 효과적으로 제거하기 위한 새로운 시변 적응 상수를 갖는 적응 알고리즘을 제안한다. 제안하는 알고리즘은 최적 필터의 직교 원리를 이용하여 적응 알고리즘의 적응 상수를 시변한다. 직교 원리에 따라 적응 상수를 시변하면 적응 초기에는 적응 필터가 반향 경로를 추정하지 못해서 큰 값을 가지지만 수렴 후에는 적응 필터의 입력 신호와 추정 오차 신호간의 상호 상관도가 작아져서 정상 상태 오차가 작으며 근단화자 등 의 주변 잡음 신호에 적응필터 계수가 오조정되는 것을 줄일 수 있다. 본 논문에서 음향반향 시뮬레이션을 통해서 제안하는 적응 알고리즘이 기존의 방법에 비해서 우수함을 보인다.

• Electronics and Telecommunications Trends
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• v.10 no.2 s.36
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• pp.101-109
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• 1995

음향반향제거기술은 현재 널리 사용되고 있는 핸드프리 전화기, 영상회의 시스템 등 앞으로의 멀티미디어 통신단말에서 그 실용성을 높이고 부가가치를 높이는데 공통적으로 요구되는 필수적인 기반기술이다. 따라서 본 고에서는 적응 신호 처리를 이용한 음향반향제거기술의 기본원리, 적응알고리듬 그리고 국제 표준화 활동 및 실제 구현에 대한 동향을 기술하여, 이 분야에 있어서 우리의 자립기술 개발을 유도할 수 있는 바탕이 될 수 있도록 분석 자료로서 활용코자 한다.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.927-930
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• 2001

본 논문에서는 웨이블렛 변환 알고리듬을 이용 한 음향반향 제거 기법을 제안하였다. 일반적으로 LMS 알고리듬은 간단하지만 그 수렴특성이 고유치 분포율과 탭 수에 의해 결정되는 단점이 있고 MDF (multidelay filter)를 이용한 적응 알고리듬은 블록단위로 수행되는 LMS를 수렴속도 측면에서 보완해주지만 긴 차수일때는 FFT에 비례하여 연산량이 늘어나는 단점이 있다. 본 논문은 웨이블렛의 직교성을 이용한 변환 적응 알고리듬을 음향반향 제거기에 도입하여 그 성능을 비교·평가하였다. 모의실험을 통해 시간영역 LMS, MDF 알고리듬과 비교하여 제안한 알고리듬의 수렴성능이 우수함을 보였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.04a
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• pp.129-132
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• 2000

본 논문에서는 음향 반향 제거기의 새로운 동시 통화 검출기를 제안하여 근단화자신호에 의한 적응 필터의 계수 오조정을 줄이고자 하였다. 제안한 동시 통화 검출기는 마이크 입력 신호의전력과 추정 오차 신호와 마이크 입력 신호간의 상호 상관의 비를 이용하여 동시 통화를 검출하는 방법이다. 제안한 동시 통화 검출기는 주변 잡음이 없는 상황에서는 다른 동시 통화 검출기와 비교하여 우수한 성능을 보였다. 그리고 전체적인 음향 반향 제거기의 향상을 가져왔다.