• The Transactions of The Korean Institute of Electrical Engineers
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• v.64 no.12
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• pp.1756-1760
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• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.1
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• pp.129-139
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• 1989

During a signal is being transmitted, an interference signal can be introduced through an unknown channel. In these cases, an adaptive system, so called adaptive noise canceller, can restore the original signal from the corrupted signal by first identifying the unknown interference channel on the minimum mean square error criteron, and then by cancelling the interference signal using the identified interference channel. Whereas this method is quite effective when the a priori knowledges about the characteristics of the interference signal and of the intrference channel are unknown or time-varyng, but has a drawback that the presence of the original signal has a severe effect on the optimum value of the interference channel to be identified on the miniumum mean square eror criterion In this paper an adaptive noise canceller and its algorithms are introduced that can restore the original signal more accurately especially when the correlatedness between the original signal and the interference signal is small.

• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.163-170
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• 2015

In this paper, it is shown that the performance of the frequency-domain multi-reference narrowband noise canceller is determined by the narrowband component to the broadband disturbance power ratio in the reference signals. To overcome this problem, a new narrowband ANC is proposed, where the update of the adaptive filter is determined based on SNR of the reference inputs being measured using the magnitude squared coherence (MSC) between the primary and the reference signals. Simulation results show that the proposed ANC has superior performance over the conventional one.

• Journal of Digital Convergence
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• v.10 no.10
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• pp.277-282
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• 2012

The speech recognition system can not quickly adapt to varied environmental noise factors that degrade the performance of recognition. In this paper, the echo noise robust HMM learning model using average estimator LMS algorithm is proposed. To be able to adapt to the changing echo noise HMM learning model consists of the recognition performance is evaluated. As a results, SNR of speech obtained by removing Changing environment noise is improved as average 3.1dB, recognition rate improved as 3.9%.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1998.04a
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• pp.678-683
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• 1998

Combustion instability by thermoacoustic feedback incite strong low frequency noise and vibration which damage the system and provoke the environmental problems. Therefore, it is necessary to control the thermoacoustic oscillation. In the way of controlling the instability, active control method using adaptive algorithm is applied. In this study, active noise control method using anti-sound technique is selected, whose principle is cancelling the noise with the addition of opposite phase sound. At first, simulation is performed to confirm the stability of controller, and after that control of combustion instability is carried out to get cancellation of 20-30dB SPL.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• Journal of KSNVE
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• v.8 no.6
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• pp.1037-1042
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• 1998

This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. We also Propose a novel control algorithm (AFC/CAFC) for tonal noise cancellation. The proposed algorithm estimates the magnitude and phase of the tonal noise. The algorithm uses the steepest descent method for the phase/magnitude estimation. Performances of tile CAFC algorithm are presented in comparison with those by the AFC algorithm based on computer simulations and experiments.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.4
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• pp.491-498
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• 2012

Recently, many papers have been proposed in order to improve the OFDM system performance in T-DMB DOCR (Digital On Channel Repeater), by using removing the feedback signal so that the transmitter power can be increased or by using the equalizer to remove ICI. Despite these efforts, however, signal quality at the receiving terminal has not been improved because of constellation smearing in T-DMB DOCR. In this paper, in order to suppress constellation smearing, we propose an effective equalizer algorithm that can improve system performance. We perform adaptive channel estimation and non-coherent decision directed noise cancellation method that can estimate the channel subsequently during data symbols period in the frequency domain. So we can obtain better quality of the signal at the receiving terminal. In order to secure QoS(Quality of Service) required in T-DMB handsets, we evaluate SNR and BER in T-DMB DOCR(Digital On Channel Repeater) and verified by simulation. In this simulation results, this system is satisfied the performance of BER=$10^{-5}$ at less than SNR=14 dB at the receiver after compensation of phase noise -18 dBc.

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

• Journal of KSNVE
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• v.11 no.1
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• pp.68-75
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• 2001

A method for actively controlling dynamic reaction forces in flexible structures subject to persistent excitations is presented. Since reaction forces are not directly measured in flexible structures, reaction forces are estimated by using the Kalman filter. The estimated reaction force is used as an error signal in the adaptive feedforward disturbance cancellation controller. In order to compensate the static effect of the truncated modes in the reaction forces, the residual flexibility matrix is used with the Kalman filter. The paper presents the formulation of the reaction forces in conjunction with the Kalman filter estimator and the adaptive feedforward controller. The results show that the dynamic reaction forces at its supports in a flexible beam test rir are well suppressed.