• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06c
• /
• pp.179-182
• /
• 1998

본 논문에서는 보청기의 다채널 구조를 이용하여 음향 궤환 제거의 대역폭과 대역의 위치를 조절할 수 있는 새로운 알고리듬을 제안하였다. 제안한 알고리듬에서는 음향 궤환 제거기가 보청 알고리듬의 각 주사수 대역별로 따로 연결되어 있기 때문에 주파수 대역과의 연결에 따라 특정 주파수 대역에서만 음향 궤환이 이루어지게 할 수도 있고, 기존의 음향 궤환 제거 방식과 같이 전 주파수 대역에서도 음향 궤환 제거를 할 수 있으므로 보다 효과적이고 유연한 알고리듬이라 할 수 있다. 따라서 성능면에서도 기존의 알고리듬과 같거나 특별한 조건하에서는 더 나은 성능을 보인다. 제안한 알고리듬에 대해 3개의 채널을 가지는 보청기 구조와 8개의 채널을 가지는 보청기 구조에서 실험을 행하였다. 음향 궤환 경로는 문헌의 자료를 참고하여 2가지를 만들어 이들 음향 궤환 경로의 특성이 집중되어 있는 대역으로 제한하여 음향궤환 제거를 한 결과 전주파수 대역에서 음향 궤환 제거를 한 경우보다 이들 제한된 대역내에서는 더 음향 궤환이 잘 이루어졌다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.1C
• /
• pp.96-102
• /
• 2010

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.7C
• /
• pp.413-420
• /
• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.4
• /
• pp.38-46
• /
• 1999

In this paper, two methods were proposed to improve the performance of adaptive feedback cancellation in hearing aids. One is “Orthogonality principle acoustic feedback cancellation method(Orthogonality principle method)” to track optimal solution with monitoring the instantaneous gradient, the other is a method using the CLMS algorithm(CLMS method). In many simulation conditions, adaptive feedback cancellation method proposed in this paper was much better than Greenberg method by Sum-method LMS algorithm which is known the most excellent method by now in case of system mismatch, SNR and segmental SMR. Also. Orthogonality principle method is as good as CLMS method in terms of adaptive feedback cancellation in many simulation conditions.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.699-702
• /
• 2009

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows the improved SNR of about more than 20 dB.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.11C
• /
• pp.911-916
• /
• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.38A no.11
• /
• pp.955-966
• /
• 2013

We propose two feedback interference cancellers(FICs) to improve the performance of the equalization digital on-channel repeater(EDOCR) with the FIC for the ATSC DTV broadcasting system. The FIC estimates the feedback channel between Tx. and Rx. antennas of the repeater by cross-correlating the reference signal and the feedback signal. Since there is a DC pilot which ruins the white property of the ATSC DTV signals, the FIC cannot estimate the feedback channel accurately. To solve the problem, the structural method which uses an additional DC pilot free reference for feedback channel estimation and the algorithmic method based on the digital signal processing which whitens the ATSC DTV signals and performs the feedback cancellation in the whitened signal domain. Simulation results show that the proposed two FICs show better feedback cancellation performance than the conventional FIC.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2008.11a
• /
• pp.193-196
• /
• 2008

지상파 DMB 동일 채널 중계기에서는 송신 안테나로부터 수신 안테나로 재입력되는 궤환 신호가 존재한다. 이 궤환 신호는 적절히 제거되지 않을 경우 수신 신호에 간섭으로 작용하게 되어 중계기 시스템의 안정성이 크게 떨어지게 된다. 과거에 진행된 연구에서는 궤환 채널을 추정하여 간섭을 제거시키는 궤환 신호 제거기(feedback canceller) 를 사용하여 이와 같은 문제를 해결하였다. 하지만 기존의 추정 알고리즘은 간단한 구현에 비해 수렴 성능과 속도 면에서 만족할만한 성능을 얻을 수 없었으며, 신호 간 상관도 제거를 위해 충분한 시간 지연을 해야 하는 단점이 있다. 본 논문은 궤환 채널 추정에 적합한 PNLMS (Proportionate Normalized Least Mean Squares) 적응형 알고리즘을 적용해 새로운 궤환 신호 제거 기법을 제안하였으며, 전산 모의 실험을 통해 이와 같은 문제를 효과적으로 해결할 수 있음을 보여준다.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.14 no.6
• /
• pp.1249-1256
• /
• 2019

In this paper, we propose a new algorithm to remove acoustic feedback and noise in hearing aids. Instead of using the conventional FIR structure, this algorithm is a deep learning algorithm using neural network adaptive prediction filter to improve the feedback and noise reduction performance. The feedback canceller first removes the feedback signal from the microphone signal and then removes the noise using the Wiener filter technique. Noise elimination is to estimate the speech from the speech signal containing noise using the linear prediction model according to the periodicity of the speech signal. In order to ensure stable convergence of two adaptive systems in a loop, coefficient updates of the feedback canceller and noise canceller are separated and converged using the residual error signal generated after the cancellation. In order to verify the performance of the feedback and noise canceller proposed in this study, a simulation program was written and simulated. Experimental results show that the proposed deep learning algorithm improves the signal to feedback ratio(: SFR) of about 10 dB in the feedback canceller and the signal to noise ratio enhancement(: SNRE) of about 3 dB in the noise canceller than the conventional FIR structure.

• Journal of Biomedical Engineering Research
• /
• v.25 no.4
• /
• pp.315-321
• /
• 2004

In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.