• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• MALSORI
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• no.62
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• pp.113-131
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• 2007

It is reported that the segmental feature based recognition system shows better results than conventional feature based system in the previous studies. On the other hand, the various studies of combining neural network and hidden Markov models within a single system are done with expectations that it may potentially combine the advantages of both systems. With the influence of these studies, tandem approach was presented to use neural network as the classifier and hidden Markov models as the decoder. In this paper, we applied the trend information of segmental features to tandem architecture and used posterior probabilities, which are the output of neural network, as inputs of recognition system. The experiments are performed on Auroral database to examine the potentiality of the trend feature based tandem architecture. From the results, the proposed system outperforms on very low SNR environments. Consequently, we argue that the trend information on tandem architecture can be additionally used for traditional MFCC features.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.43-51
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.164-167
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• 1995

RPE 음성부호화기에서 합성 필터로 인한 구동벡터 양자화잡음의 증폭효과를 분석하고 regular pulse 시퀀스의 양자화로 인한 성능감쇄를 줄이기 위해 pyramid vector 양자화방식을 도입하였다. 제안된 방식의 성능평가는 구동시퀀스 양자화를 위해 adaptive PCM을 이용하는 GSM 표준 RPE 방식과의 객관적 및 주관적 성능비교를 통해 수행하였다.T JDSMDQLRY 결과 제안된 방식은 대략 1dB의 SNR 및 segmental SNR 값 증가를 가져왔고, 또한 비공식 청취시험결과 명료도의 증가를 느낄 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1E
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• pp.46-55
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• 1995

In this paper, we present a new wavelet transform-based CELP coder, called band selection wavelet transform CELP (BS-WTCELP) operated at 4.8 kbps. The proposed algorithm uses a band selection scheme of frequency bands of wavelet transform and selective vector quantization (VQ). The band selection and selective VQ structure is implemented by using a classified VQ structure. The proposed algorithm has about 0.5-1.0 dB improvement in segmental SNR compared with the conventional CELP that uses the random codebook search, while is has significantly reduced computational and storage complexity. Many experimental results have shown that the proposed algorithm is more suitable for most real-applications than the conventional CELP and wavelet transform CELP.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.5
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• pp.811-816
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• 2019

This paper first proposes a technique to separate clean speech signals and mixed noise signals by using an independent vector analysis algorithm of frequency band for 4 channel speech source signals with a noise. An improved output speech signal from the proposed independent vector analysis algorithm is obtained by using the cross-correlation between the signal outputs from the frequency domain delay-sum beamforming and the output signals separated from the proposed independent vector analysis algorithm. In the experiments, the proposed algorithm improves the maximum SNRs of 10.90dB and the segmental SNRs of 10.02dB compared with the frequency domain delay-sum beamforming algorithm for the input mixed noise speeches with 0dB and -5dB SNRs including white noise, respectively. Therefore, it can be seen from this experiment and consideration that the speech quality of this proposed algorithm is improved compared to the frequency domain delay-sum beamforming algorithm.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.291-294
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• 2004

본 논문에서는 잡음의 변화(variance)가 없는 정적인 통계적 특성(Stationary Static Characteristic)을 갖는 환경에서 잡음 추정을 통해 지각 필터의 성능을 개선하는 알고리즘을 제안한다. 제안된 잡음 추정 알고리즘은 입력되는 잡음에 열화 된 신호의 묵음 구간에서 추정된 잡음을 이용하여 입력되는 잡음의 SNR을 추정 후, 대역 별로 smoothing 상수 값으로 잡음 에너지를 제어하여 첨가된 잡음을 추정함으로써 초기 추정 잡음 보다 가까운 추정 잡음을 얻을 수 있게 된다. 이는 신호를 열화 시킨 잡음을 보다 정확한 추정을 제공함으로써, 지각 필터의 응답을 개선할 수 있고 더불어 잡음에 의해 열화 된 신호의 음질을 개선할 수 있다. 또한 저 대역에 영향을 미치는 잡음인 경우 다른 방법들과는 달리 음질의 개선이 뚜렷하다. 기존의 방식과 비교를 위해 다양한 신호 대 잡음 비(signal-to-noise ratio, SNR)에서 열화 된 오디오 신호를 입력으로 사용하였다. 입력 SNR이 5dB, 10dB, 15dB와 20dB의 각각의 경우에 대하여 SSNR(Segmental SNR)과 잡음 대 마스킹 비(Noise-to-mask ratio, NMR), 음질 테스트를 수행한 결과, 청감 테스트(Mean Opinion Score, MOS Test) 결과의 향상과 음질개선의 개선을 확인할 수 있었다.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.43-46
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• 2004

본 논문에서는 최근에 제안된 음성신호 개선을 위한 적응 웨이블릿 패킷 기반의 스펙트럼 차감법을 이용하여 다양한 측면에서의 성능평가 결과를 제시한다. 사용된 음성신호 개선 방식은 적응 웨이블릿 패킷의 불균등 주파수 해상도와 높은 에너지 집중도로 인해 발생하는 극대, 극소값의 영향을 피하기 위해 기하평균을 이용하는 스펙트럼 추정법을 사용하였다. 다양한 측면의 성능평가를 위해 주관적 평가 척도인 MOS 와 높은 상관도를 갖는 것으로 알려진 log likelihood ratio, log area ratio, segmental SNR, weighted spectral slope 등을 평가 척도로 사용하였다. Fourier 기저를 사용한 방식과의 비교에서 적응 웨이블릿 패킷 방식은 SegSNR 과 음성의 명료도를 비교적 잘 반영하는 weighted spectral slope 측면에서 우수한 성능을 보였다.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.6
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• pp.1120-1125
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• 2010

In the speech system, for example hearing aid as well as speech communication, speech quality is degraded by environmental noise. In this study, to enhance the speech quality which is degraded by environmental speech, we proposed an algorithm to reduce the noise and reinforce the speech. The minima controlled recursive averaging (MCRA) algorithm is used to estimate the noise spectrum and spectral weighting factor is used to reduce the noise. And partial masking effect which is one of the human hearing properties is introduced to reinforce the speech. Then we compared the waveform, spectrogram, Perceptual Evaluation of Speech Quality (PESQ) and segmental Signal to Noise Ratio (segSNR) between original speech, noisy speech, noise reduced speech and enhanced speech by proposed method. As a result, enhanced speech by proposed method is reinforced in high frequency which is degraded by noise, and PESQ, segSNR is enhanced. It means that the speech quality is enhanced.