• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1106-1113
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• 2000

In this, paper, a DFT based polyphase filter bank for the fractional ratio oversampling is proposed. Proper fractional oversampling ratio gives lower aliasing than the critical sampling and, at the same time, lower computational load than the integer ratio oversampling. In addition, filter bank design becomes easier by the reduced aliasing effect of fractional ratio oversampling. Proposed fractional ratio oaversampling polyphase structure is applied to a subband adaptive filter for acoustic echo cancellation where long adaptive filter are ofter required. Echo cancellation results show that fractional ratio oversampling gives comparable performance to the integer ratio oversampling with less computational load.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.93-100
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• 1999

확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.5
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• pp.369-378
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• 1988

In this paper, the performance of the timing jitter which has great effects on the echo canceller that can be used for full-duplex digital transmission on two-wire subscriber loops is analyzed. The power spectrum of timing jitter is about 8.9dB lower in the AMI input format than in the Polar-NRZ-L input format. The performance of the echo canceller also has been shown improved by 4dB when the input signal is in the AMI format.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.135-138
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• 1999

본 연구는 2개의 마이크로폰을 이용한 지연 시간(TOOA) 추정에 대한 연구로서 기존의 Cross-Correlation 알고리듬과 Roth Processor 알고리듬에 Onset 알고리듬을 적용하영 기존 알고리듬의 성능 개선에 대한 연구를 수행하였다. 작은 사무실OR나 회의실내 주변 잡음과 음향반향이 동시에 존재한다는 가정 하에 Onset 알고리듬은 마이크로폰의 수신신호에 포함되어 있는 반향 신호를 효율적으로 마스킹하여 반향 현상을 제거함으로서 정확한 시간 지연 추정을 가능하게 한다. Onset을 적용한 Cross-Correlation 과 Roth Processor 알고리듬의 우수성을 입증하기 위하여 7개의 SNR 환경에서 총 420번의 컴퓨터 모의 실험을 수행하였으며 실험 결과 SNR이 508 이상에서는 Onset 알고리듬을 적용한 알고리듬이 Onset을 적용하지 않은 알고리듬에 비해 우수함을 입증할 수 있었다.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.631-634
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• 1998

본 논문은 음향반향제거기에서 동시통화구간을 검출하고 처리하는 알고리듬에 대하여 논하고 있다. 새로운 동시통화처리 알고리듬은 송신신호와 잔류신호간의 상호 상관계수를 이용하여 동시통화 여부 및 구간을 검출하고 이 구간에서는 반향제거기의 계수적응을 중지하도록 하는 것이다. 본 알고리듬을 사용함으로서 동시통화구간의 검출에 소요되는 계산량이 대폭 감소할 뿐만 아니라 비정상상태의 원인이 동시통화에 의한 것인지 아니면 반항경로응답의 변화에 의한 것인지를 구별할 수가 있다. 모의실험을 수행하여 본 알고리듬과 기존의 것과의 성능을 비교, 검토하였다. 실험 결과 몬 알고리듬을 적용하므로서 동시통화 후에도 계수가 발산하지한ㅎ고 안정적으로 정상상태를 유지함을 확인하였다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1996.10a
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• pp.736-739
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• 1996

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

• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.6
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• pp.628-633
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• 2014

This paper proposes a performance improvement method of the conventional double talk detection method which can operate before convergence of the echo canceller. The proposed method estimates the coefficients of the linear predictive coding(LPC) filter by using the primary input signal. The time-varying threshold for double talk detection is determined based on the LPC filter gain of the primary input signal level. The proposed method can reduce not only false detection rate which means wrong detection of single talk as double talk but also double talk detection delay. Computer simulation was performed using a long-term real speech signals. It is shown that the proposed method improves the conventional method in terms of lowering the false detection rate and shortening the detection delay.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

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