• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.9
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• pp.155-163
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• 2012

The geometrical and electrical errors of array sensors can severely degrade the performance of array sensor system. Various calibration techniques are developed to alleviate this problem. In this paper, two different calibration methods with respect to location, gain and phase of array sensors are presented. One method applies the first-order Taylor series expansion to approximate the true steering vector from the nominal values of array sensors. Then a set of equations is formed by using the null characteristics of the MUSIC spectrum to estimate errors of location, gain and phase of array sensors. Another method estimates these errors based on the data covariance matrix of pilot sources. From the simulations, it is demonstrated that two calibration algorithms calibrated an array system successfully. In addition to that, Fistas and Manikas's algorithm is more robust against noise than Ng and Lie's one when SNR is from 10dB to 50dB.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.8
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• pp.710-718
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• 1991

In this pape, an analysis approach for the performance of serial search burst-format direct sequence spread-spectrum code acquisiion system using matched filter is presented. And an acquistion system using analog matched filter is analyzed by this approach. And special consideration is focused on the effect of the code autocorelation sideobes. The basic idea of thi approach is the constuction of state tansition digram at the time of random packet arrival and the calculation of the packet loss probability, which is deveoped in terms of the following parameters: the false alarm probability. the etection probability, and the probability of receiver blocking obtained from the blocked customers cleae queeing system model. The analysis results reveal that the performance of the acquisition system is degraded by the effet of the code autocorrelation sidelobes above a certain value of SN/chip.

• The Proceedings of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.10 no.2
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• pp.63-70
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• 1996

In this paper, we propose the low sldelobe digital doppler filter bank design algorithm which is practically applicable to coherent radar equipment in complex air-defense system. Using the proposed algorithm, we can obtain a digital doppler filter bank of lower peak sidelobe level and higher clutter suppression capability. In this method, it is possible to achieve higher target-detectability without loss of Target SNR. The proposed algorithms are verified by simulations and experiments.

• Journal of Korea Multimedia Society
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• v.11 no.10
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• pp.1325-1337
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• 2008

The difference of environments between training and recognition is the major reason of degradation of speech recognition. To solve this mismatch of environments, various noise processing methods have been studied. Among them, ERN(log-Energy dynamic Range Normalization) and SEN(Silence Energy Normalization) for normalization of log energy features show better performance than others. However, these methods have a problem that they can hardly achieve normalization for the relatively higher values of log energy features and the environmental mismatch caused by this problem becomes bigger especially in low SNR environments. To solve these problems, we propose applying ARMA filter as post-processing for smoothing log energy features by calculating the moving average in auto-regression scheme. From the recognition results conducted on Aurora 2.0 DB, the proposed method shows improved recognition results comparing with conventional methods.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.8
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• pp.65-73
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• 2007

This paper presents a low power CMOS continuous-time $\Sigma\Delta$ (sigma-delta) modulator for UMTS receivers. The loop filter of the continuous-time $\Sigma\Delta$ modulator consists of an active-RC filter which performs high linearity characteristics and has a simple tuning circuit for low power operating system The architecture of this modulator is the $3^{rd}-order$ 4-bit single loop configuration with a 24 of OSR (Oversampling Ratio) to increase the power efficiency. The modulator includes a half delay feedback path to compensate the excess loop delay. The experimental results of the modulator are 71dB, 65dB and 74dB of the peak SNR, peak SMR and dynamic range, respectively. The continuous-time $\Sigma\Delta$ modulator is fabricated in a 0.18-um 1P4M CMOS standard process and dissipates 15mW for a single supply voltage of 1.8V.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.9-14
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• 2013

In this paper, we propose a low complexity bilateral search SIC for OFDM in fast time-varying channels. Due to the possibility of error propagation in SIC, symbol detection ordering within the block of symbols has a significant effect on the overall performance. In this paper, the first symbol to be detected is determined based on CSEP values, and then the next symbol to be detected is selected according to the updated CSEP while bilaterally searching from the boundary of the detected symbol group. Through computer simulations, we show that the proposed method has performance improvements with almost the same computation complexity over the conventional methods in the high SNR region. It has a performance approaching the MFB, known as the performance upper bound, within 2dB at the BER of $10^{-5}$.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.12
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• pp.93-102
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• 1998

In noisy environments, human speech productions are influenced by noises(Lombard effect), and speech signals are contaminated. These distortions dramatically reduce the performance of speech recognition systems. This paper proposes a method of the Lombard effect compensation and noise suppression in order to improve speech recognition performance in noise environments. To estimate the intensity of the Lombard effect which is a nonlinear distortion depending on the ambient noise levels, speakers, and phonetic units, we formulate the measure of the Lombard effect level based on the acoustic speech signal, and the measure is used to compensate the Lombard effect. The distortions of speech under noisy environments are cancelled out as follows. First, spectral subtraction and band-pass filtering are used to cancel out noise. Second, energy nomalization is proposed to cancel out the variation of vocal intensity by the Lombard effect. Finally, the Lombard effect level controls the transform which converts Lombard speech cepstrum to clean speech cepstrum. The proposed method was validated on 50 korean word recognition. Average recognition rates were 82.6%, 95.7%, 97.6% with the proposed method, while 46.3%, 75.5%, 87.4% without any compensation at SNR 0, 10, 20 dB, respectively.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.46 no.3
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• pp.51-59
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• 2009

This paper presents the design of continuous-time sigma-delta modulator for acoustic sensor. The feedforward structure without summing block is used to reduce power consumption of sigma-delta modulator. A high-linearity active-RC filter is used to improve resolution of sigma-delta modulator. Excess loop delay problem in conventional continuous-time sigma-delta modulators is solved by our proposed architecture. A low power, high resolution fourth-order continuous-time sigma-delta modulator with 1-bit quantization was realized in a 0.13-${\mu}m$ 1-Poly 8-metal CMOS technology, with a core area of $0.58\;mm^2$. Simulation results show that the modulator achieves 91.3-dB SNR over a 25-kHz signal bandwidth with an oversampling ratio of 64, while dissipating $290{\mu}W$ from a 3.3-V supply.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.166-173
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• 2004

Satellite communication system uses a high non-linear HPA(high power amplifiers) in the earth station and satellite transponder. Therefore, it is important to consider the nonlinear effect of HPA on the communication system. In this paper, we find the variation of power spectrum density by nonlinearity HPA and the change of harmonic component according to IBO (input back-off). When the BPSK is used for satellite communication system, we analyze BER performance including the external co-channel interference (CCI) and the adjacent channel interference (ACI) resulting from the HPA nonlinearity. BER degrades as ACI magnitude grows up when the uplink SNR, uplink SIR (signal to co-channel interference power ratio) and downlink SIR are constant at some level. In case there is only non-linear HPA in the satellite, it is shown that BER considerably depends on the ACI magnitude ACI. When there are two non-linear HPAs in the both earth station and satellite, much BER degradation results from the CCI and ACI.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6A
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• pp.614-623
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• 2008

A very low complexity QRD-M algorithm based on adaptive search area is proposed for MIMO systems. The conventional QRD-M scheme extends each survivor paths to all constellation symbols at each layer and selects M paths of minimum path metrics. We found that performance will not be degraded even if we adaptively restrict the survivor path extension only to the neighboring points of temporary detection symbol according to the channel condition at each layer. By employing this feature, we propose a new QRD-M algorithm achieving the near MLD performance with a reduced complexity. We employ the channel gain ratio among the layers as a channel condition indicator, which does not require SNR estimation. The simulation results show that the proposed scheme effectively achieves near MLD performance while maintaining the overall average computation complexity much smaller than the conventional QRD-M algorithm.