• 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.

• 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.

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.189-192
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• 1998

전화망을 이용한 음성인식 시스팀에서 출력신호의 반향에 의한 입력신호의 음질 열화현상을 막기위해서 적응디지털 필터를 이용한 반향제거기가 필수적이다. 대표적인 적응 필터 알고리즘인 LMS와 NLMS 들을 각각 이용한 적응 반향제거기들과 입/출력신호의 상관계수를 이용한 개선된 적응 필터 알고리즘의 성능을 비교하였다. 개선된 알고리즘의 경우 NLMS 알고리즘의 빠른 수렴특성을 가지면서도 더블톡(double talk)구간에서의 음질왜곡 현상을 LMS보다 개선시켰다.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.44-50
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• 2002

Conventional acoustic echo cancelers using ES(Exponentially weighted Stepsize) algorithm have simple operational configuration and fast convergence speed batter then NLMS algorithm, but they are very weak in external noise because ES algorithm updates filter taps using an average energy reduction rate of room impulse response in specific acoustical condition. So, a new configuration of acoustic echo canceler with stepsize generator and selector is proposed in this thesis. The proposed stepsize generator and selector improve conventional acoustic echo canceler's weakness in external noise and improve the system robustness. The stepsize generator generates additional stepsize value using moving averager, which is the residual noise energy of error signal multiplied by constant ${\gamma}$. The stepsize selector selects the stepsize value that has better performance in an acoustic echo canceler using a coefficient decision factor ${\Delta}_{differ}$ The simulation results show that the proposed algorithm reduces residual error by 5[dB] to 10[dB], improves misadjustment regardless of external noise's SNR. 

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.4C
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• pp.343-349
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• 2010

In this paper, we study real time AEC (acoustic echo canceller) based on NLMS adaptive filter. Proposed method improves conversation quality by enhancing the performance of AEC during double talk section and reduces the power consumption by controling the adaption operation of NLMS adaptive filter. Proposed method examines the convergence of the NLMS adaptive filter, stores the estimated echo path and chooses operation of NLMS adaptive filter. Furthermore if double talk is detected, the proposed AEC utilizes the stored echo path optionally considering missed double talk time. When the proposed AEC is used, the performance of the AEC is enhanced although the simple double talk detector is used and the power consumption of the AEC is reduced.

• 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 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.