• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.529-536
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• 2003

In this Paper, a new DCT HLMS DFE(Discrete Cosine Transform Hierarchical Least Mean Square Decision Feedback Equalizer) algorithm is proposed to improve the convergence speed and MSE(Mean Square Error) performance of a receive channel equalizer in ATSC(Advanced Television System Committee) 8VSB(Vestigial Side Band) digital terrestrial TV system. The proposed algorithm reduces the eigenvalue range of input data autocorrelation by transforming LMS (Least Mean Square) DFE into the subfilter of hierarchical structure. Moreover, the use of DCT and power estimation algorithm makes it possible to reduce the eigenvalue deviation of input data which results from distortion and delay of the receive signal in the miulti-path environment. Simulation results show that proposed DCT HLMS DFE has SNR improvement of approximately 3.8dB, 5dB and 2dB as compared to LMS DFE when the equalized symbol error rate is 0.2 in ATTC defined digital terrestrial TV broadcasting channels A, B and F, respectively.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.36C no.9
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• pp.13-19
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• 1999

This paper describes a design of an 8-bit 100KSPS 1mW CMOS A/D Converter. Using a novel systematic offset cancellation technique, we reduce the systematic offset voltage of operational amplifiers. Further, a new Gain amplifier is proposed. The proposed A/D Converter is fabricated with a $0.6{\mu}m$ single-poly triple-metal n-well CMOS technology. INL and DNL is within ${\pm}1LSB$, and SNR is about 43dB at the sampling frequency of 100KHz. The power consumption is $980{\mu}W$ at +3V power supply.

• Korean Journal of Digital Imaging in Medicine
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• v.17 no.1
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• pp.33-41
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• 2015

This study attempts to examine the clinical usefulness of High b-value DWI (diffusion weighted imaging) for brain tumors with an edema. Subjects were seven patients selected from 65 patients who received an MRI scan for suspected encephalopathy and confirmed diagnosis at our hospital from February to July 2015 (male: 7, average age : 66 years old). As test equipment, 3.0T MR System (ACHIEVA Release, Philips, Best, The Netherlands) and 8Channel SENSE Head Coill were used. DWI checks on the use of the variable TR 5460ms, TE 132ms, Slice Thickness 4mm, gap 1mm, Slice number 29 is, 3D T1WI is TR 8.4ms, TE 3.9ms, matrix size $240{\times}240$, Slice can set 180 piecesIt was. b value of 0, 1,000, 2,000 s/mm2 with DWI acquisition and 3D T1WI enhancement five minutes after the Slice Thickness 3mm, gap 0mm to reconstruct the upper face axis (MPR TRA CE) was. As for the experiment, in b-value 1,000 and 2,000 images, SNR and the lesion at the lesion site and CNR in the normal site opposite to the lesion are measured. WW(window width) and WL(window level) are made equal in MRICro software, and the volume of the lesion is measured from each of b-value and MPR TRA CE image. Using SPSS ver. 1.8.0.0 Mann Whitney-test was analyzed for SNR and CNR, while Kruskal-Wallis test was analyzed for volume.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.11 no.1
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• pp.8-17
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• 2000

The BER performance and the capacity variation of a microcell due to power control error(PCE) is analyzed on service types(data and voice, respectively) for the reverse link of an overlaid cell CDMA system. The procedure of analysis is followed as: First, we calculate BER performance according to PCE. Next, we find the minimum SNR for voice service, BER=TEX>$10^{-3}$, and data service, BER=TEX>$10^{-5}$. Then, according to the calculated SNR, we find the maximum capacity of a microcell and macrocell and the capacity of a microcell where interference is considered is found and analyzed with that in perfect power control. We get to the results as follows. The BER performance in 1 dB PCE is similar to that in perfect power control, however, with a increase in PCE, the BER performance is largely degraded. In terms of capacity, it is shown that if the PCE is equal or less than 2 dB, the effect of the PCE on voice service is more than that on data service, but if the PCE is equal or more than 3 dB, effect of the PCE on data service is more than that on voice. Besides, if the PCE is equal or less than 2 dB, both PCE and interference should be considered to calculate the capacity of a microcell, but if the PCE is equal or more than 3 dB, interference can be negligible since the effect of PCE is much stronger than that of interference. Therefore, the microcell should be located where $R_d$, the ratio of a microcell to a macrocell radius, is equal to 0.1, and d, the ratio of the distance between a microcell and a macrocell to the macrocell radius, is equal or more than 0.5, in order to obtain a appropriate microcell capacity against interference. If $\sigma$ is adjusted to less than 2 dB, we may get equal or more than 70% of the maximum microcell capacity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10C
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• pp.1378-1386
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• 2004

In this paper, we propose an OFDM scheme to increase the BER performance. The proposed OFDM scheme makes the baseband OFDM signal by using the oversampled OFDM signal values which are obtained by inserting N(=2$\^$k/) zeros and carrying out 2N point IFFT. In the demodulation part, the sampling operation for A/D conversion is carried out with the 2 times high sampling speed. 2 N point FFT is carried out for the data demodulation. In this paper, we show, with the mathematical method and the computer simulation, that the SNR of the proposed OFDM scheme is 3 dB higher than that of the conventional OFDM in the same AWGN channel conditions given.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.4
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• pp.895-902
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• 2015

This paper presents a high-speed sensing and noise cancellation technique for large touch screens, which is called FDCS (Frequency Division Concurrent Sensing). Most conventional touch screen detection methods apply excitation pulses sequentially and analyze the sensing signals sequentially, and so are often unacceptably slow for large touch screens. The proposed technique applies sinusoidal signals of orthogonal frequencies simultaneously to all drive lines, and analyzes the signals from each sense line in frequency domain. Its parallel driving allows high speed detection even for a very large touch screens. It enhances the sensing SNR (Signal to Noise Ratio) by introducing a frequency domain noise filtering scheme. We also propose a pre-distortion equalizer, which compensates the drive signals using the inverse transfer function of touch screen panel to further enhance the sensing SNR. Experimental results with a 23" large touch screen show that the proposed technique enhances the frame scan rate by 273% and an SNR by 43dB compared with a conventional scheme.

• Journal of IKEEE
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• v.17 no.4
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• pp.511-516
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• 2013

This paper presents a SDM using the FDPA technique. The FDPA technique is the added feedback path which is the delayed path of DAC output. The designed SDM increases the SNR by adding the delayed digital feedback path. The proposed SDM is easily implemented by eliminating the analog feedback path. Through the MATLAB modeling, the optimized coefficients are obtained to design the SDM. The designed SDM has a power consumption of $220{\mu}W$ and SNR(signal to noise ratio) of 81dB at the signal-bandwidth of 20KHz and sampling frequency of 2.56MHz. The SDM is designed using the $0.18{\mu}m$ standard CMOS process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8A
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• pp.689-697
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• 2012

In this paper, we propose base layer repetition and HARQ schemes for improving the reliability and the performances of H.264 SVC video transmission in a wireless channel, and investigate its performances. The proposed method may solve the problems of transmission delay as well as the scarcity of wireless resources, since the proposed scheme was applied for only base layer, not for enhancement layer. The numerical results show that the proposed scheme can enhance the BER performance of $1.5{\times}10^{-5}$ and the FER of $1.2{\times}10^{-3}$, when SNR=3.4dB. Also, it was confirmed through the resultant images that the proposed method can improve the SVC performance in the wireless link.

• Journal of Biomedical Engineering Research
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• v.26 no.3
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• pp.177-184
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• 2005

In this paper, we proposed a new hearing aid algorithm to improve SNR(signal to noise ratio) of noisy speech signal and speech perception. The proposed hearing aid algorithm is a multi-band loudness compensation based independent component analysis (ICA). The proposed algorithm was compared with a conventional spectral subtraction algorithm on behind-the-ear type hearing aid. The proposed algorithm successfully separated a target speech signal from background noise and from a mixture of the speech signals. The algorithms were compared each other by means of SNR. The average improvement of SNR by ICA based algorithm was 16.64dB, whereas spectral subtraction algorithm was 8.67dB. From the clinical tests, we concluded that our proposed algorithm would help hearing aid user to hear clearly a target speech in noisy conditions.