• Journal of Broadcast Engineering
• /
• v.5 no.1
• /
• pp.123-129
• /
• 2000

This paper addresses a new double detection algorithm which is based on the fuzzy control in the adaptive echo canceller of communication system. In this method, the two input of the fuzzy inference for detecting double talk condition are used. The one is the cross-correlation coefficient between the error signal and the primary signal which is the summed signal of the real echo signal and the near-end signal. The other is the cross-correlation coefficient between the estimation error signal and the primary signal. The fuzzy controller made a fuzzification for two inputs by the membership functions of trapezoid and them became the composition using inference rules. The composed result is defuzzificated by the center gravity method. The output is compared with two threshold values to detect double talk and echo path variation effectively. It is confirmed by computer simulation that this fuzzy double talk detector is able to track echo path variation accurately.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.13 no.5
• /
• pp.203-208
• /
• 2013

In this paper, we proposed a performance improvement method of the double talk detector which can operate before the echo canceller converges. Microphone input signal is filtered by the linear prediction filter and this filtered signal is used for detection. The coefficients of the linear prediction filter are given by the far-end talker signal. During single talk, filtered signal has low power since the characteristics of the echo signal is similar with those of the far-end talker signal. But, during double talk, the filtered signal does not have low power because the signal of different characteristics is included in the microphone signal. Double talk is detected by this difference. Simulations using real speech signals verified that the proposed method outperformed the conventional methods.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.2
• /
• pp.121-128
• /
• 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 information and communication convergence engineering
• /
• v.8 no.6
• /
• pp.640-644
• /
• 2010

When we talk with hands-free in a car or noisy lobby, the performance of the echo canceller degrade because background noise added to echo caused by the distance from mouth to microphone is relatively long. It gives a reason for necessity of noise-robust and high convergence speed adaptive algorithm. And if acoustic echo canceller operated not perfectly, residual signal going through the echo canceller to far-end speaker remains residual echo, which degrade quality of talk. To solve this problem, post-processing needed to remove residual echo ones more. In this paper, we propose a new acoustic echo canceller, which has noise robust and high convergence speed, linked with linear predictor as a post-processor. By computer simulation, it is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.541-544
• /
• 2000

본 논문은 부밴드별 반향제거 필터 뱅크를 구현하기 위한 가중 Overlap-add 적응필터를 이용한 단일 채널 음향 반향 제거기를 제안한다. 기준 입력 신호의 고유치 분포율에 의존하여 수렴 특성이 결정되는 NLSM알고리즘을 사용하여 전대역 처리 과정에서 발생하는 수렴성능의 저하를 방지하고, 효율적인 블록별 병렬 처리가 가능한 부밴드 처리기법인 가중 Overlap-add 방식을 적용한 적응 반향제거기의 성능을 고찰한다 또한 본 논문에서는 동시 통화 검출을 위한 전용 필터와 에너지 비교 방법을 동시에 사용하는 새로운 형태의 동시통화 검출 기법을 제안한다.

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

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.213-216
• /
• 2002

본 논문은 원거리 회의 시스템이나 차량 내 핸즈프리 통화 시 필연적으로 발생하는 음향 반향을 제거하기 위해 새로운 알고리즘을 제안한다. 본 논문에서 제안된 음향반향 제거 시스템은 Delayless 서브밴드 음향반향 제거기와 협대역 동시통화 검출기로 구성된다. Delayless Subband 적응 음향 반향 제거기는 적은 계산 량과 높은 수렴속도로 음향 반향 제거 성능이 뛰어난 것으로 알려져 있으며 본 논문은 이를 이용해 안정적인 음향 반향 제거를 위해서 협대역 Subband 내에서 동시통화 검출기를 구현한다. 기존의 광대역 동시 통화 검출기에 비해 본 논문에서 제안된 협대역 동시통화 검출기는 저주파 Subband 대역에서만 동시통화 검출을 수행하여 down-sampling으로 인한 계산 량 감소와 저주파 특성을 가지는 Subband 대역의 신호 특성으로 인한 신뢰성 있는 통화 상태 정보를 제공함으로서 전체적인 음향반향제어 시스템의 성능을 향상시킬 수 있도록 하였다. 본 연구에서 제안된 음향반향 제어 시스템의 성능은 다양한 컴퓨터 시뮬레이션을 통하여 입증하도록 한다.

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

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.35S no.11
• /
• pp.184-193
• /
• 1998

In this paper, a real-time acoustic echo canceller is implemented using only PC's CPU without extra help from a DSP chip. The adaptive digital filter is designed efficiently so that it can be implemented in real-time and has a proper cancellation performance. It is proposed that a new double talk detector consumes a small computational complexity and guarantees the fast detection and robust operation. The real-time acoustic echo canceller consists of the full-duplex sound card and 166 MHz Pentium PC, and requires less than 10% CPU time.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.39C no.12
• /
• pp.1346-1351
• /
• 2014

We propose a novel approach for stereophonic acoustic echo suppression using spectro-temporal filtering based on soft decision. Unlike the conventional approaches estimating the echo pathes directly, the proposed technique can estimate stereo echo spectra without any double-talk detector. In order to improve the estimation of echo spectra, the extended power spectrum density matrix and echo overestimation control matrix are applied on this method. In addition, this echo suppression technique is based on soft decision technique using speech absence probability in STFT domain. Experimental results show that the proposed method improves compared with the conventional approaches.