• International Journal of Fuzzy Logic and Intelligent Systems
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• v.9 no.3
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• pp.172-177
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• 2009

In this paper, a new morphology-based homomorphic filtering technique is presented to enhance features in medical images. The homomorphic filtering is performed based on the morphological sub-bands, in which an image is morphologically decomposed. An evolutionary design is carried to find an optimal gain and structuring element of each sub-band. As a search algorithm, Differential Evolution scheme is utilized. Simulations show that the proposed filter improves the contrast of the interest feature in medical images.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.1
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• pp.68-79
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• 1992

In the high speed analog system satellite communication system, video signal processing and optical fiber interface circuits, GaAs high gain operational amplifier is advantageous due to obtain a high gain because of its low transconductance and other drawbacks, such as low frequency dispersion and process variation. Therefore in this paper, a circuit techniques for improving the voltage gain for GaAs MESFET amplifier is presented. Also, various types of existing current mirror and current mirror proposed are compared.To obtain the high differential gain, bootstrap gain enhancement technique is used and common mode feedback is employed in differential amplifier.The simulation results show that gain is higher than that of basic amplifier about 18.6dB, and stability and frequency performance of differential amplifier are much improved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.8
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• pp.867-880
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• 1992

In this paper, a circuit design techniques for Improving the voltage gain of the GaAs MESFET single amplifier is presented. Also, various types of existing current mirror and proposed current mirror of new configuration are compared. To obtain the high differential mode gain and low common mode gain, bootstrap gain enhancement technique Is used and common mode feedback Is employed In the design of differential amplifier. The simulation results show that designed differential amplifier has differential gain of 57.66dB, unity gain frequency of 23.25GHz. Also, differential amplifier using common mode feedback with alternative negative current mirror has CMRR of 83.S8dB, stew rate of 3500 V /\ulcorners.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.9
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• pp.165-170
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• 2014

Speech enhancement techniques that remove surrounding noise are stressed to preprocessor of speech recognition. Among the various speech enhancement techniques, Codebook-based Speech Enhancement (CBSE) operates efficiently in non-stationary noise environments. But, CBSE has some problems that inaccurate gains can be estimated if mismatch occur between input noisy signal and trained speech/noise codevectors. In this paper, the Normalized Weighting Factor (NWF) is calculated by long-term noise estimation algorithm based on Signal-to-Noise Ratio, compensated to the conventional inaccurate gains. The proposed CBSE shows better performance than conventional CBSE.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.5
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• pp.35-43
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• 2014

In this paper, we propose a performance enhancement method of the target tracking system for communication signal using the monopulse and the ${\alpha}{\beta}$ filter to keep the connection of the communication system between the airplane and the ground. We suggest the minimum distance measurement method for tracking error angle of the monopulse signal instead of the generally used method of MR(Monopulse Ratio) curve, and the ${\alpha}{\beta}$ filter with variable gain for enhancement of the tracking accuracy and the probability of re-tracking the monopulse signal under the disconnection of link. We show the performance enhancement of the proposed method of monopulse system using the measured MR Curve results of the prototype system. And also, the comparison of simulation results between the ${\alpha}{\beta}$ filter with variable gain and the ${\alpha}{\beta}$ filter with fixed gain shows the performance enhancement of the proposed ${\alpha}{\beta}$ filter. Using the proposed methods, we expect the enhanced performance of the existing target tracking system for communication signal only by changing the algorithm without hardware changes.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.36D no.3
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• pp.75-84
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• 1999

Systematic studies for the effect of the linewidth enhancement factor, the confinement factor, the internal loss and the cavity length on the single mode gain difference and the frequency detuning are performed for $\lambda$/4 phase shifted DFB lasers above threshold. The above threshold characteristics are mainly determined by the linewidth enhancement factor, not by the confinement factor or the parameter defined by the product of the linewidth enhancement factor and the confinement factor. The normalized internal loss defined by the product of the internal loss and the cavity length mainly determines the above threshold characteristics compared to that of the internal loss or the cavity length alone. The effect of the cavity length on threshold characteristics is larger than that of the internal loss in the case of the same normalized internal loss. The above threshold characteristics of quantum well lasers are more resistant to the variations of the confinement factor and the normalized internal loss than those of bulk lasers due to the small linewidth enhancement factor.

• The Journal of the Acoustical Society of Korea
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• v.27 no.2
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• pp.75-79
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• 2008

In this paper, we propose a novel speech enhancement technique using global soft decision (GSD) based on the probabilistic outputs of support vector machine (SVM). Generally, speech enhancement algorithms applied soft decision gain modification and noise power estimation have bettor performance than those employing hard decision. Especially, global speech absence probability (GSAP), which is known as an effective measure of the speech absence in each frame, has been adopted to SD-based speech enhancement methods. For this reason, we introduce a new GSAP estimated from the probabilistic output of SVM using sigmoid function. The performance of the proposed algorithm is evaluated by the PESQ and MOS test under various noise environments and yields better results compared with the conventional GSD scheme.

• Journal of IKEEE
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• v.13 no.4
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• pp.116-124
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• 2009

Recently the CPU performance of modem chipsets or multimedia processors of mobile phone is as high as notebook PC. That is why mobile phone has been emerged as a leading ICON on the convergence of consumer electronics. The various applications of mobile phone such as DMB, digital camera, video telephony and internet full browsing are servicing to consumers. To meet all the demands the image quality has been increasingly important. Mobile phone is a portable device which is widely using in both the indoor and outside environments, so it is needed to be overcome to deteriorate image quality depending on environmental light source. Furthermore touch window is popular on the mobile display panel and it makes contrast loss because of low transmittance of ITO film. This paper presents the image enhancement algorithm to be embedded on image enhancement SoC. In contrast enhancement, we propose Clipped histogram stretching method to make it adaptive with the input images, while S-shape curve and gain/offset method for the static application And CIELCh color space is used to sunlight readability enhancement by controlling the lightness and chroma components which is depended on the sensing value of light sensor. Finally the performance of proposed algorithm is evaluated by using histogram, RGB pixel distribution, entropy and dynamic range of resultant images. We expect that the proposed algorithm is suitable for image enhancement of embedded SoC system which is applicable for the small-sized mobile display.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.784-788
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• 2008

In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.