• ETRI Journal
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• v.36 no.5
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• pp.772-782
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• 2014

Two-microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time-frequency (T-F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T-F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T-F units with hit rates above 85%. It outperforms previous solutions in terms of signal-to-noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.169-174
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• 2003

A novel subspace speech enhancement using subband whitening filter is proposed. Previous subspace speech enhancement method either assumes additive white noise or uses whitening filter as a pre-processing for colored noise. The proposed method tries to minimize the signal distortion while reducing residual noise by processing the signal using subband whitening filter. By incorporating the notion of subband whitening filter, spectral resolution in Karhunen-Loeve(KL) domain is improved with the negligible additional computational load. The proposed method outperforms both the subspace method suggested by Ephraim and the spectral subtraction suggested by Boll in terms of segmental signal-to-noise ratio (SNRseg) and perceptual evaluation of speech quality (PESQ).

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.6
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• pp.959-967
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• 1999

In this paper, we propose a Linear Prediction Method(LPM) in conjunction with signal enhancement for solving the direction-of-arrival estimation problem of multiple incoherent plane waves incident on a uniform linear array. The basic idea of signal enhancement is that of finding the covariance matrix of given rank that lies closest to a given estimated matrix in Frobenius norm sense. It is well known that LPM has a high-resolution performance in general applications, while it provides a lower statistical performance in lower SNR environment. To solve this problem, the LPM combined with signal enhancement approach is herein proposed. Simulation results are illustrated to demonstrate the better performance of the proposed method than conventional LPM.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.386-393
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• 2008

This paper proposes a new approach for speech enhancement in highly nonstationary noisy environments. The spectral subtraction (SS) is a well known technique for speech enhancement in stationary noisy environments. However, in real world, noise is mostly nonstationary. The proposed method uses an auto control parameter for an adaptive threshold to work well in highly nonstationary noisy environments. Especially, the auto control parameter is affected by a linear function associated with an a posteriori signal to noise ratio (SNR) according to the increase or the decrease of the noise level. The proposed algorithm is combined with spectral subtraction (SS) using a hangover scheme (HO) for speech enhancement. The performances of the proposed method are evaluated ITU-T P.835 signal distortion (SIG) and the segment signal to-noise ratio (SNR) in various and highly nonstationary noisy environments and is superior to that of conventional spectral subtraction (SS) using a hangover (HO) and SS using a minimum statistics (MS) methods.

• Phonetics and Speech Sciences
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• v.4 no.4
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• pp.87-92
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• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

• Journal of Broadcast Engineering
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• v.8 no.1
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• pp.11-18
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• 2003

In this paper, we introduce a novel signal quality enhancement algorithm with a perceptual interference analysis and perceptual convolutional noise suppression. A perceptual convolutional noise is reflected in the audible disturbance that can still be recognized after the additional noise suppression and tonality change which is caused by the noise energy excitation. The enhancement system is organized with a perceptual additional noise suppression part and a perceptual convolutional noise suppression part. Experimental results show that these two parts have an equivalent quality enhancement performance.

• Archives of Pharmacal Research
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• v.28 no.4
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• pp.413-420
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• 2005

We previously demonstrated the ability of ginseng saponins (active ingredients of Panax ginseng) to enhance $Ca^{2+}$-activated $Cl^{-}$ current. The mechanism for this ginseng saponin-induced enhancement was proposed to be the release of $Ca^{2+}$ from $IP_{3}-sensitive$ intracellular stores through the activation of PTX-insensitive $G\alpha_{q/11}$ proteins and PLC pathway. Recent studies have shown that calmodulin (CaM) regulates $IP_{3}$ receptor-mediated $Ca^{2+}$ release in both $Ca^{2+}-dependent$ and -independent manner. In the present study, we have investigated the effects of CaM on ginseng saponin-induced $Ca^{2+}$-activated $Cl^{-}$ current responses in Xenopus oocytes. Intraoocyte injection of CaM inhibited ginseng saponin-induced $Ca^{2+}$-activated $Cl^{-}$ current enhancement, whereas co-injection of calmidazolium, a CaM antagonist, with CaM blocked CaM action. The inhibitory effect of CaM on ginseng saponin-induced $Ca^{2+}$-activated $Cl^{-}$ current enhancement was dose- and time-dependent, with an $IC_{50} of 14.9\pm3.5 {\mu}M$. The inhibitory effect of CaM on saponin's activity was maximal after 6 h of intraoocyte injection of CaM, and after 48 h the activity of saponin recovered to control level. The half-recovery time was calculated to be $16.7\pm4.3 h$. Intraoocyte injection of CaM inhibited $Ca^{2+}$-induced $Ca^{2+}$-activated $Cl^{-}$ current enhancement and also attenuated $IP_{3}$-induced $Ca^{2+}$-activated $Cl^{-}$ current enhancement. $Ca^{2+}$/CaM kinase II inhibitor did not inhibit CaM-caused attenuation of ginseng saponin-induced $Ca^{2+}$-activated $Cl^{-}$ current enhancement. These results suggest that CaM regulates ginseng saponin effect on $Ca^{2+}$-activated $Cl^{-}$ current enhancement via $Ca^{2+}$-independent manner.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.181-184
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• 2008

The paper describes a Layered Modulation using the SVC signals and studies the properties of the modulation with respect to several parameters by the computer simulation. The SVC signals will include a base layer signal and an enhancement signal, and the base layer signal is the more important one in its channel robustness. The parameters will include a carrier frequency, a bandwidth, power level, modulation type and code rate. We analyze the demodulating and decoding process of the Layered Modulation system through several scatter plots. And then we discuss the affect of the layer signal power difference to the BER performance, which also proves the base layer signal is more important than the enhancement layer signal.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.687-693
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• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3E
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• pp.87-96
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• 2007

In this paper we propose a new noise reduction algorithm for stationary and nonstationary noisy environments. Our algorithm classifies the speech and noise signal contributions in time-frequency bins, and is not based on a spectral algorithm or a minimum statistics approach. It relies on calculating the ratio of the standard deviation of the noisy power spectrum in time-frequency bins to its normalized time-frequency average. We show that good quality can be achieved for enhancement speech signal by choosing appropriate values for ${\delta}_t\;and\;{\delta}_f$. The proposed method greatly reduces the noise while providing enhanced speech with lower residual noise and somewhat higher mean opinion score (MOS), background intrusiveness (BAK) and signal distortion (SIG) scores than conventional methods.