• Progress in Superconductivity and Cryogenics
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• v.16 no.4
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• pp.53-56
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• 2014

A numerical method of magnetic field calculation for the air-core solenoid is presented in this paper. In application of the Biot-Savart law, the magnetic field induced from the source current can be obtained by a double integration ormula. The numerical method named composite Simpson's rule for the integration is applied to the program and the adaptive quadrature method is used to adjust the step size in the calculation according to the precision we need. When the target point is in the solenoid and the intergrand's denominator may be zeroin the process of calculation, the method sill can provide an appropriate result. We have developed a program which calculates the magnetic field with at least 1ppm precision and named it as rzBI() to implement this method. The method has been used in the design of an MRI magnet, and the result show it is very flexible and convenient.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.5
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• pp.810-816
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• 2001

This paper proposes a Truncated Type-II Hybrid ARQ scheme using an adaptive modulation system to achieve high throughput data transmission systems for mobile communication systems. In this paper, the adaptive modulation system analyzed in Nakagami (m-distribution) fading channel environment. The adaptive modulation system controls the modulation level and symbol rate according to the Nakagami fading parameter(m). When the received $E_bN_0$ is high or the Nakagami fading parameter m is high, the propose system selects higher modulation level and higher symbol rate to increase throughput. On the other hand, this system selects lower modulation level and lower symbol rate to prevent throughput performance degradation when the received $E_bN_0$ is low. The modulation method have been adopted QPSK(Quadrature Phase Shift Keying), 16 QAM(Quadrature Amplitude Modulation), 64 QAM, 256 QAM. Therefore, Adaptive Modulation Systems with Truncated Type-II Hybrid ARQ Scheme is proper for mobile and radio for mobile and radio data communication system that require high reliability and delay-limited applications.

• Journal of Advanced Navigation Technology
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• v.13 no.4
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• pp.482-489
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• 2009

This paper proposes a CW(Continuous-Wave) bio-radar applying a new adaptive noise canceller based on ALE(Adaptive Line Enhancer) which can remove the Gaussian noise and system noise. Recently the research works on this CW bio-radar which can be used to detect heartbeat and respiration are advanced by the university and research facility. Although the researches describe CW bio-radar not only is vulnerable for the Gaussian noise but also has a disadvantage of decreasing the heart-rate accuracy due to the noise, the researches do not demonstrate the effective method for removing the noise component in a baseband signal. In this paper, a CW bio-radar applying the new adaptive noise canceller based on ALE which can remove the noise component is proposed. This paper compares and analyzes the performance for increasing the heart-rate accuracy according to removing the Gaussian noise and system noise in the baseband signal through the quadrature receiver which can alleviate the demodulation sensitivity to target position.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.8
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• pp.2055-2062
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• 1996

This paper presents a method of compensating for the gain and phase imbalance of quadrature modulator and demodulator. The gain and phase imbalance are compensated using the received power after the compensation pattern was transmitted at tranceiver. The proposed method is less vulnerable to changes in the transmission channel than the conventional method because compensation is made possible within the tranceiver system, and even the change in phase mblance in accordance with frequency can be compensated utilizing the adaptive algorithm. According to numerical results, a normalized eye opening and a bit error rate are improved by 1.8dB and 3.8dB, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.5
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• pp.893-898
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• 2003

This paper proposes a Truncated Type-II Hybrid ARQ scheme and coding techniques using an adaptive modulation system to achieve high throughput data transmission systems for wireless multimedia communication systems. In this paper, the adaptive modulation system analyzed in Nakagami (m-distribution) fading channel environment. The adaptive modulation system controls the modulation level and symbol rate according to the Nakagami fading parameter(m). When the received Eb/No is high or the Nakagami fading parameter m is high, the propose system selects higher modulation level and higher symbol rate to increase throughput. On the other hand, this system selects lower modulation level and lower symbol rate to prevent throughput performance degradation when the received Eb/No is low. The modulation method have been adopted QPSK(Quadrature Phase Shift Keying), 16QAM(Quadrature Amplitude Modulation), 64QAM, 256QAM. Therefore, adaptive modulation systems with truncated type-II hybrid ARQ scheme is proper for wireless multimedia communication system that require high reliability and delay-limited applications.

• Journal of Electrical Engineering and Technology
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• v.13 no.1
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• pp.468-475
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• 2018

In medical ultrasound imaging, frequency-dependent attenuation downshifts and reduces a center frequency and a frequency bandwidth of received echo signals, respectively. This causes considerable errors in quadrature demodulation (QDM), result in lowering signal-to-noise ratio (SNR) and contrast resolution (CR). To address this problem, adaptive dynamic QDM (ADQDM) that estimates center frequencies along depth was introduced. However, the ADQDM often fails when imaging regions contain hypoechoic regions. In this paper, we introduce a valid region-based ADQDM (VR-ADQDM) method to reject the misestimated center frequencies to further improve SNR and CR. The valid regions are regions where the center frequency decreases monotonically along depth. In addition, as a low-pass filter of QDM, Gaussian wavelet based dynamic filtering was adopted. From the phantom experiments, average SNR improvements of the ADQDM and the VR-ADQDM over the traditional QDM were 1.22 and 5.27 dB, respectively, and the corresponding maximum SNR improvements were 2.56 and 10.58 dB. The contrast resolution of the VR-ADQDM was also improved by 0.68 compared to that of the ADQDM. Similar results were obtained from in vivo experiments. These results indicate that the proposed method would offer promises for imaging technically-difficult patients due to its capability in improving SNR and CR.

• Journal of Communications and Networks
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• v.15 no.4
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• pp.398-406
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• 2013

In this paper, we investigate two types of in-phase and quadrature-phase (IQ) data transfer methods for cloud multiple-input multiple-output (MIMO) network operation. They are termed "after-precoding" and "before-precoding". We formulate a cloud massive MIMO operation problem that aims at selecting the best IQ data transfer method and transmission strategy (beamforming technique, the number of concurrently receiving users, the number of used antennas for transmission) to maximize the ergodic sum-rate under a limited capacity of the digital unit-radio unit link. Based on our proposed solution, the optimal numbers of users and antennas are simultaneously chosen. Numerical results confirm that the sum-rate gain is greater when adaptive "after/before-precoding" method is available than when only conventional "after-precoding" IQ-data transfer is available.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1082-1091
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• 2000

In this paper, the efficient equalization method for OFDM(Orthogonal Frequency Division Multiflexing) System using the QAM(Quadrature Amplitude Modulation) in multipath fading channel is proposed in order to faster and more efficiently equalize the received signals that are sent over real channel. In generally, the one-tap linear equalizers have been used in the frequency-domain as the existing equalization method for OFDM system. In this technique, if characteristics of the channel are changed fast, the one-tap linear equalizers cannot compensate for the distortion due to time variant multipath channels. Therefore, in this paper, we use one-tap non-linear equalizers instead of using one-tap linear equalizers in the frequency-domain, and also use the linear equalizer in the time-domain to compensate the rapid performance reduction at the low SNR(Signal-to-Noise Ratio) that is the disadvantage of the non-linear equalizer. In the frequency-domain, when QAM signals, consisting of in-phase components and quadrature (out-phase) components, are sent over the complex channel, the only in-phase and quadrature components of signals distorted by the multipath fading are changed the same as signals distorted by the noise. So the cross components are canceled in the frequency-domain equalizer. The time-domain equalizer and the adaptive algorithm that has lower-error probability and fast convergence speed are applied to compensate for the error that is caused by canceling the cross components in the frequency-domain equalizer. In the time-domain, To compensate for the performance of frequency-domain equalizer the time-domain equalizes the distorted signals at a frame by using the Gold-code as a training sequence in the receiver after the Gold-codes are inserted into the guard signal in the transmitter. By using the proposed equalization method, we can achieve faster and more efficient equalization method that has the reduced computational complexity and improved performance.

• ETRI Journal
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• v.37 no.1
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• pp.43-53
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• 2015

In this paper, a selective coefficient updating (SCU) approach at each branch of the per-tone equalization (PTEQ) structure has been applied for insufficient cyclic prefix (CP) length. Because of the high number of adaptive filters and their complex adaption process in the PTEQ structure, SCU has been proposed. Using this method leads to a reduction in the computational complexity, while the performance remains almost unchanged. Moreover, the use of set-membership filtering with variable step size is proposed for a sufficient CP case to increase convergence speed and decrease the average number of calculations. Simulation results show that despite the aforementioned algorithms having similar performance in comparison with conventional algorithms, they are able to reduce the number of calculations necessary. In addition, compensation of both the channel effect and the transmitter/receiver in-phase/quadrature-phase imbalances are achievable by these algorithms.