• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.512-515
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• 2010

SIGINT(SIGnal INTelligence) includes several parameters intercepted by measurement and analysis of the RF(Radio frequency) signal from free space. One of the important parameters is frequency information. Expecially, in order to perform instantaneous frequency measurement of Radar and Missile seeker's RF signals, we use dedicated RF modules as a DFD(Digital Frequency Discriminator) to provide frequency information by measurement of the relative phase difference between signals via intended RF delay lines. It must measure and provide realtime based frequency information on short pulsed RF signal up to 100 nSec or less. This document proposes Ultra wideband DFD consisted of a RF input section of Wideband 4 channel RF delay line and correlator, a digital processing section to measure and provide frequency information from I/Q signal, and a frequency calibration section. Also, it will show design suitability based on test results measured under test condition of very short input pulse signals.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.10
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• pp.1275-1281
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• 2019

The automatic modulation recognition of a radio signal is a major task of an intelligent receiver, with various civilian and military applications. In this paper, we propose a method to recognize the modulation of radio signals in wireless communication based on the deep neural network. We classify the modulation pattern of radio signal by using the LSTM model, which can catch the long-term pattern for the sequential data as the input data of the deep neural network. The amplitude and phase of the modulated signal, the in-phase carrier, and the quadrature-phase carrier are used as input data in the LSTM model. In order to verify the performance of the proposed learning method, we use a large dataset for training and test, including the ten types of modulation signal under various signal-to-noise ratios.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.6
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• pp.1103-1108
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• 2023

In this paper, we propose a new algorithm for attenuating the background noises in acoustic signal. This algorithm improves the noise attenuation performance by using the FNN(: Full-connected Neural Network) deep learning algorithm instead of the existing adaptive filter after wavelet transform. After wavelet transforming the input signal for each short-time period, noise is removed from a single input audio signal containing noise by using a 1024-1024-512-neuron FNN deep learning model. This transforms the time-domain voice signal into the time-frequency domain so that the noise characteristics are well expressed, and effectively predicts voice in a noisy environment through supervised learning using the conversion parameter of the pure voice signal for the conversion parameter. In order to verify the performance of the noise reduction system proposed in this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed. As a result of the experiment, the proposed deep learning algorithm improved Mean Square Error (MSE) by 30% compared to the case of using the existing adaptive filter and by 20% compared to the case of using the STFT(: Short-Time Fourier Transform) transform effect was obtained.

• Transactions on Semiconductor Engineering
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• v.2 no.3
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• pp.1-5
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• 2024

This paper presents a crystal-less reference clock recovery (CR) frequency synthesizer with compensation designed for Bluetooth Low Energy (BLE) Smart-tag applications, operating at frequencies of 32, 72, and 80MHz. In contrast to conventional frequency synthesizers, the proposed design eliminates the need for external components. Using a single-ended antenna to receive a minimal input power of -36dBm at a 2.4GHz signal, the CR synthesizes frequencies by processing the RF signal received through a Low Noise Amplifier ( L N A ) . This approach allows the system to generate a reference clock without relying on a crystal. The received signal is amplified by the LNA and then input to a 16-bit ACC (Automatic Clock Compensation) circuit. The ACC compares the frequency of the received signal with the oscillator output signal, using the synthesis of a 32MHz reference clock through a frequency compensation method. The oscillator is constructed using a Ring Oscillator (RO) with a Frequency Divider, offering three different frequencies (32/72/80MHz) for various system components. The proposed frequency synthesizer is implemented using a 55-nm CMOS process.

• Progress in Superconductivity
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• v.9 no.1
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• pp.23-28
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• 2007

In order to have a superconductive connection between the wire-wound pickup coil and input coil, typically Nb terminal blocks with screw holes are used. Since this connection structure occupies large volume, large stray pickup area can be generated which can pickup external noise fields. Thus, SQUID and connection block are shielded inside a superconducting tube, and this SQUID module is located at some distance from the distal coil of the gradiometer to minimize the distortion or imbalance of uniform background field due to the superconducting module. To operate this conventional SQUID module, we need a higher liquid He level, resulting in shorter refill interval. To make the fabrication of gradiometers simpler and refill interval longer, we developed a novel method of connecting the pickup coil into the input coil. Gradiometer coil wound of 0.125-mm diameter NbTi wires were glued close to the input coil pads of SQUID. The superconductive connection was made using an ultrasonic bonding of annealed 0.025-mm diameter Nb wires, bonded directly on the surface of NbTi wires where insulation layer was stripped out. The reliability of the superconductive bonding was good enough to sustain several thermal cycling. The stray pickup area due to this connection structure is about $0.1\;mm^2$, much smaller than the typical stray pickup area using the conventional screw block method. By using this compact connection structure, the position of the SQUID sensor is only about 20-30 mm from the distal coil of the gradiometer. Based on this compact module, we fabricated a magnetocardiography system having 61 first-order axial gradiometers, and measured MCG signals. The gradiometers have a coil diameter of 20 mm, and the baseline is 70 mm. The 61 axial gradiometer bobbins were distributed in a hexagonal lattice structure with a sensor interval of 26 mm, measuring $dB_z/dz$ component of magnetocardiography signals.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.50-61
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• 1995

A time scale modification method yielding rate-modified speech while conserving the characteristic of speech was implemented in real-time using a goneral purpose digital signal processor. Time scale modification changed pronunciation speed only, producing a time difference between the input signal and the modified signal, making it impossible to implement it in real-time. In this thesis, a system was implemented to remove the time difference between the input and modified signals. Speech signals slowed down or speeded up by a physical time scale modification method, such as adjusting the motor speed of the cassett tape recorder, was used as the input signal. Physical modification that controled only the inter speed of the cassette tape player distorted the pitch period of the original speech. In this study, a real-time system was implemented so that the pitch-distorted speech was reconstructed back to the original by fractional sampling pitch shifting using an FIR filter, and this signal was time scale modified to match the cassette tape recorder motor speed using SOLA time-scale medification. In experiments using speech signals medifiedby the proposed method, results obtained using a 16-bit resolution ADSP2101 processor and using computer simulations employing floating point operations showed about the same average frame signal-to-noise ratio of about 20 dB.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.4
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• pp.162-167
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• 2015

This paper describes the reason and the problem solving for the double-layer pattern of a Daejeon correlator operated in Korea-Japan Correlation Center. When the electric power of an input signal in the correlator is charged small enough to be buried in the noise, it is hard to see a signal with a specific pattern in the input signal, but when the electric power is large, a specific one is reported to be seen. By comparing data from observation with one from software correlator, it was confirmed from the analysis using the AIPS software that the amplitude gain of a source signal was affected about 3%. Therefore, in order to solve the problem of double-layer patterns, we found that a problem in the memory management module responsible for both the data input and the data serialization of the correlator is a cause for the double-layer pattern detected periodically. In other words, while data is serialized and read repeatedly in the memory area assigned to serialize the data from the serialization module, redundant last data is generated and an overlap for the memory allocation is occurred. Therefore, by modifying the program of the FPGA memory sections on serialization module to correct the problem, we confirmed that double-layer pattern is disappeared and correlation results are normally acquired.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.27 no.4
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• pp.537-543
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• 2003

Transient vibration generated by developing localized fault in gear can be used as indicators in gear fault detection. This vibration signal suffers from the background noise such as gear meshing frequency and its harmonics and broadband noise. Thus in order to extract the information about the only gear fault from the raw vibration signal measured on the gearbox this signal is processed to reduce the background noise with many kinds of signal-processing tools. However, these signal-processing tools are often very complex and time waste. Thus. in this paper. we propose a novel approach detecting the damage of gearbox and analyzing its pattern using the raw vibration signal. In order to do this, the residual signal. which consists of the sideband components of the gear meshing frequent) and its harmonics frequencies, is extracted from the raw signal by the power spectral density (PSD) to obtain the information about the fault and is used as the input data of the artificial neural network (ANN) for analysis of the pattern of gear fault. This novel approach has been very successfully applied to the damage analysis of a laboratory gearbox.

• Journal of Drive and Control
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• v.18 no.1
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• pp.17-23
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• 2021

The flow signal or spool position signal is used to determine the dynamic characteristics of directional control valves. Alternatively, the signal of spool position or flow can be replaced with the velocity of a low friction, low inertia actuator. In this study, the frequency response of the servo valve equipped with a spool position transducer is measured with a metering cylinder. The input signal, spool displacement, load pressure, and velocity of the metering cylinder are measured, and the theoretical results from the transfer function analysis are verified. The superposition rule for magnitude ratio and phase angle was found to be always applicable among any signal type, and it was found that the load pressure signal is not appropriate for use as the signal for measuring the frequency response of a servo valve. It was confirmed that the frequency response of a servo valve using metering cylinder was similar to the results from a spool displacement signal. The metering cylinder used for measuring the frequency response of a servo valve should be designed to have sufficiently greater bandwidth frequency than the bandwidth frequency of the servo valve.