• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.469-474
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• 2018

As urban population increases, research on urban environmental noise is getting more attention. In this study, we classify the abnormal noise occurring in traffic situation by using a deep learning algorithm which shows high performance in recent environmental noise classification studies. Specifically, we classify the four classes of tire skidding sounds, car crash sounds, car horn sounds, and normal sounds using convolutional neural networks. In addition, we add three environmental noises, including rain, wind and crowd noises, to our training data so that the classification model is more robust in real traffic situation with environmental noises. Experimental results show that the proposed traffic sound classification model achieves better performance than the existing algorithms, particularly under harsh conditions with environmental noises.

• Journal of Internet Computing and Services
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• v.11 no.5
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• pp.1-8
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• 2010

This paper presentsthe performance study of blind equalizer algorithms for impulsive-noise environments based on Gaussian kernel and constant modulus error(CME). Constant modulus algorithm(CMA) based on CME and mean squared error(MSE) criterion fails in impulsive noise environment. Correntropy blind method recently introduced for impulsive-noise resistance has shown in PAM system not very satisfying results. It is revealed in theoretical and simulation analysis that the maximization of zero-error probability based on CME(MZEP-CME) originally proposed for Gaussian noise environments produces superior performance in impulsive noise channels as well. Gaussian kernel of MZEP-CME has a strong effect of becoming insensitive to the large differences between the power of impulse-infected outputs and the constant modulus value.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.63-66
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• 2006

음성인식시스템은 사람이 별도의 장비 없이 음성만으로 시스템의 사용이 가능한 편리한 장점을 지니고 있으나 여러 가지 기술적인 어려움과 실제 환경의 낮은 인식률로 폭넓게 사용되지 못한 상황이다. 그 중 배경잡음은 음성인식의 인식률을 저하시키는 원인으로 지적 받고 있다. 이러한 잡음환경에 있는 ASR(Automatic Speech Recognition)의 성능 향상을 위해 외측억제 기능 이 추가된 파라미터를 제안한다. ASR 에서 널리 사용되는 파라미터인 MFCC을 본 논문에서 제안한 파라미터와 HMM를 이용하여 인식률을 비교하여 성능을 비교하였다. 실험결과를 통해 제안된 파라미터의 사용을 통해 잡음환경에 있는 ASR의 성능 향상을 확인할 수 있었다.

• Journal of Digital Convergence
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• v.11 no.6
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• pp.169-174
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• 2013

Speech recognition, the problem of performance degradation is the difference between the model training and recognition environments. Silence features normalized using the method as a way to reduce the inconsistency of such an environment. Silence features normalized way of existing in the low signal-to-noise ratio. Increase the energy level of the silence interval for voice and non-voice classification accuracy due to the falling. There is a problem in the recognition performance is degraded. This paper proposed a robust speech detection method in noisy environments using a silence feature normalization and voice energy maximize. In the high signal-to-noise ratio for the proposed method was used to maximize the characteristics receive less characterized the effects of noise by the voice energy. Cepstral feature distribution of voice / non-voice characteristics in the low signal-to-noise ratio and improves the recognition performance. Result of the recognition experiment, recognition performance improved compared to the conventional method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.7 no.3
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• pp.230-238
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• 1996

In this paper, the performance improvement of BFSK signal by using diversity reception and coding technique in Rayleigh fading and impulsive noise environments has been evaluated and compared with that in Gaussian noise environment. It is found that as the CNR increases, BFSK signal performance shows an error floor regardless of impulsive noise effects in Rayleigh fading environment. Also diversity reception technique can improve the error performance not only in a Gaussian noise environment but also in a fading and impulsive noise environment. When diversity reception and coding techniques are used together in impulsive noise and Rayleigh fading environment, the improvement of error performance becomes about 11[dB] in terms of CNR as compared with that of only coding technique is applied.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.185-188
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• 1998

실제의 음성 인식 및 음성 통신 등의 음성 처리 시스템에서는 음성 신호를 손상시키는 배경 잡음 신호의 존재로 그 성능이 많이 저하된다. 특히 차량 내와 같은 잡음이 극심한 상황에서는 전처리 부분에서 이러한 잡음을 제거시켜 주어야한다. 본 연구는 자동차 내의 배경 잡음에 의해 손상된 음성에서 배경 잡음을 주파수 차감에 의하여 제거시킨다. 특히 음성 정보의 손실이 적은 잡음 추정 방법으로 가중치를 이용하여 잡음을 추정하는 가중치 주파수 차감법을 이용하였다. 이러한 가중치 주파수 차감법 사용의 전제 조건은 잡음의 변화가 완만한 경우에 적당하다. 그러나 실제적인 상황에서 배경잡음신호의 변화가 큰 경우가 존재한다. 이러한 이유에서 본 연구는 잡은 추정시 잡음 추정값을 이용하여 추정 잡음 경계값을 적응적으로 변화하는 차감법을 제안한다. 이러한 방법은 추정된 잡음 신호의 변화율을 이용하여 경계값을 상황에 따라 적응적으로 변화시키는 방법이다. 모의 실험에 의하여 고정적인 경계값을 갖는 가중치 주파수 차감법에 비해 제안한 적응적 경계값을 갖는 가중치 주파수 차감법의 출력 SNR이 증가함을 확인하였고, 음성 인식 시스템에 정용한 인식 실험에서도 성능이 향상됨을 확인하였다.

• 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 IEEK Conference
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• 2005.11a
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• pp.465-468
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• 2005

심전도 신호의 잡음 신호는 일정한 주파수대역에 존재하지 않고 측정자의 신체 및 환경조건에 따라서 잡음의 종류와 정도가 다르다. 따라서 기존의 고정 주파수 특성을 갖고 있는 필터로는 효율적인 잡음 제거가 불가능하다. 그래서 본 논문에서는 상황인식을 통해 잡음의 형태를 파악하여 적응적으로 필터를 재구성하는 적응적 잡음제거기술을 제안한다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.1
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• pp.23-28
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• 2013

In this paper, background noise reduction method using dual microphone is proposed in mobile environment. Each Signal, reference and primary, would be replaced by microphone input signals, which were measured by reference and primary microphones, and then, noise reduction was performed using FDAF. After then, residual and background noise would be estimated and reduced by MMSE-LSA. For consistent noise reduction performance, result of VAD that could be caculated by PLD between two microphones was used.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.8
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• pp.1139-1145
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• 2002

Wavelet transformed data can filter signal with each frequency band, because it includes detail information about original signal. Therefore, in this paper, important two noises were removed by wavelet. About AWGN environment UDWT(undecimated discrete wavelet transform), applying hard-threshold, was used and about impulse noise environment, it can be possible to recognize edge of original signal as well as superior denoising effect by using two methods, denoising by threshold and slope of signal by wavelet. SNR was used as a judgemental criterion of a denoising effect and Blocks and DTMF(dual tone multi frequency) were used as a test signal.