• Speech Sciences
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• v.12 no.3
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• pp.79-89
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• 2005

Acoustic feedback between microphone and receiver can be effectively cancelled adaptive feedback cancellation algorithm. Although many speech sounds have non-Gaussian distribution, most algorithms were tested with speech like sounds whose distribution were Guassian type. In this paper, we proposed an adaptive feedback cancellation algorithm based on independent component analysis (ICA) for digital hearing aid. The algorithm was tested with not only Gaussian distribution but also Laplacian distribution. We verified that the proposed algorithm has better acoustic feedback cancelling performance than conventional normalized root mean square (NLMS) algorithm, especially speech like sounds with Laplacian distribution.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.

• Speech Sciences
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• v.13 no.3
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• pp.133-149
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• 2006

This study attempted to compare acoustic parameters, physiological observation and perceptual evaluation values obtained from the treatment and control groups in order to find out which of the self voice feedback therapies was better and which methods to train them were more effective. The experimental group carried out various self voice feedback therapies while the control group did only vocal hygiene. The acoustic measurement and voice manipulation for providing the patients visual, auditory feedback were done by a speech analysis software, Praat. The authors designed vocal hygiene, abdominal respiration and Praat self voice feedback therapies and applied them to 15 patients while applying only one vocal hygiene to 15 of the control group. For the purpose of examining the degree of their voice improvement after the treatment, pre- mid- and final evaluations were made for the two groups at the beginning, the 6th week and immediately after the 8th treatment session. Results of this study were as follows: The treatment group showed much improvement after receiving the voice treatment. In particular, acoustical and physiological indices from the optical endoscopy, pitch variation(Jitter), amplitude variation (Shimmer), maximum phonation time(MPT), and psychoacoustic evaluation showed statistically significant improvements over the control groups.

• Journal of KSNVE
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• v.4 no.4
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• pp.497-505
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• 1994

This paper presents a design scheme of the active noise absorber that consists of the feedforward and feedback controller. The feedback controller aims to increase damping for the specific acoustic mode. The feedforward controller synthesizes the input signal coherent with the primary noise source in order to attenuate the noise field in the broad frequency range. The feedforward controller is adapted to the variation of acoustic plants using the proposed algorithm which compensates the effect of feedback link. Experimental results demonstrate that the proposed method is effective for the active control of band-limited noise fields in the enclosure.

• The Journal of the Acoustical Society of Korea
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• v.4 no.1
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• pp.16-21
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• 1985

Both shifiting the input signal's frequencies by a fixed frequency and compressing the input signal's bandwidth have been known to be effective in improving the stability margin of public adress systems operating in reverberant spaces. This paper describes the effect of an alternative approach of improving the acoustic-feedback stability and yet maintaining speech inteligibility by bandwidth compression and expansion. Conditions are derived for this technizue to be realized and an experimental system has been made - up. A series of experiments has been performed in small spaces and the results have shown that more than 5dB improvement can be obtained in the stability margin.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.6
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• pp.1249-1256
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• 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 KSNVE
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• v.2 no.2
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• pp.107-116
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• 1992

An adaptive signal processing technique is implemented for the active noise cancellation of the plane acoustic wave propagating in a duct. To avoid the instability caused by the acoustic feedback from the control speaker to the detect microphone, an off-line modeling of the acoustic feedback plant is done using the FIR filter. Auxiliary path required for the filtered-x LMS algorithm is modeled as well. Before going into the experiments, a simulation is carried out under the same conditions with experiments. The simulation shows that the longer the length of the adaptive filter is, the better the results are achieved. Experiments have been carried out at lower audio frequency range (50 - 400Hz), and the results are in good agreements with those of simulation study. As a results of this adaptive noise control, around 50dB is reduced for a pure tone noise, and for a bandlimited noise with the bandwidth of 316Hz, a maximum of 30dB noise reduction is attained.

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

• Speech Sciences
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• v.14 no.2
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• pp.65-71
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• 2007

The severe-profound hearing impaired children have various disorders in everday communication due to the lack of hearing feedback. Especially, their speech produced unstable voice, omission and distortion of articulation, pitch break, cul-de-sac voice, and so on so that they were difficult to accurately deliver an intended message. This study attempts to analyze the acoustic characteristics of 4 vowel sounds produced by 35 severe-profound hearing impaired children using CSL(Computerized Speech Lab, Model 4300b). The formant data were obtained from the spectrogram and analyzed data by 12 formant filter and auto-correlation among the formants. Results showed that the hearing impaired children's formant values came out very high. They produced the vowels at the mode of hypertension with unstable voice. In order to improve their speech, they would need some adequate auditory feedback.

• Journal of Communications and Networks
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• v.18 no.3
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• pp.387-396
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• 2016

Multiuser communication has been an important research area of underwater acoustic communications and networking. This paper studies the use of adaptive orthogonal frequency-division multiple access (OFDMA) in a downlink scenario, where a central node sends data to multiple distributed nodes simultaneously. In practical implementations, the instantaneous channel state information (CSI) cannot be perfectly known by the central node in time-varying underwater acoustic (UWA) channels, due to the long propagation delays resulting from the low sound speed. In this paper, we explore the CSI feedback for resource allocation. An adaptive power-bit loading algorithm is presented, which assigns subcarriers to different users and allocates power and bits to each subcarrier, aiming to minimize the bit error rate (BER) under power and throughput constraints. Simulation results show considerable performance gains due to adaptive subcarrier allocation and further improvement through power and bit loading, as compared to the non-adaptive interleave subcarrier allocation scheme. In a lake experiment, channel feedback reduction is implemented through subcarrier clustering and uniform quantization. Although the performance gains are not as large as expected, experiment results confirm that adaptive subcarrier allocation schemes based on delayed channel feedback or long term statistics outperform the interleave subcarrier allocation scheme.