• Transactions of the Korean Society of Automotive Engineers
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• v.22 no.2
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• pp.107-114
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• 2014

This paper applies a compensated low pass filter (LPF) to current measurements for permanent magnet synchronous motor (PMSM) drives. The noise limits the bandwidth of current controllers and has more adverse influences on control performances under the light load condition because of the low signal-to-noise ratio. In order to eliminate the noise sensitivity, this paper proposes a digital LPF with a compensator of gain attenuation and phase delay which are unacceptable in current information for PMSM drives. Characteristics of the proposed LPF are analyzed in comparison with the general LPFs. The compensated LPF is basically designed by the orthogonal property of the measured currents in the ${\alpha}{\beta}$ stationary reference frame. In addition, an implementation issue of the proposed method is discussed. Experimental results using the proposed method show improvements of the current control performance from two perspectives, rapid step responses and reductions of harmonic distortion.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.263-268
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• 2000

In this paper, a new indirect feedback active noise control (ANC) scheme based on the fundamental frequency estimation is proposed for systems with a harmonic noise. When reference signals necessary for feedforward ANC configuration is difficult to obtain, the conventional ANC algorithms for multi-tonal noise do not measure the reference signals but generate them with the estimated frequencies. However, the beating phenomena, in which certain frequency components of the noise vanish intermittently, may make the adaptive frequency estimation difficult. The confusion in the estimated frequencies due to the beating phenomena makes the generated reference signals worthless. The proposed algorithm consists of two parts. The first part is a reference generator using the fundamental frequency estimation and the second one is the conventional feedforward control. We propose the fundamental frequency estimation algorithm using decision rules, which is insensitive to the beating phenomena. In addition, the proposed fundamental frequency estimation algorithm has good tracking capability and lower variance of frequency estimation error than that of the conventional cascade ANF method. We are also able to control all interested modes of the noise, even which cannot be estimated by the conventional frequency estimation method because of the poor SIN ratio. We verify the performance of the proposed ANC method through simulations for the measured cabin noise of a passenger ship and the measured time-varying engine booming noise of a passenger vehicle.

• Journal of KSNVE
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• v.10 no.6
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• pp.1059-1066
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• 2000

In this paper. a new indirect feedback active noise control (ANC) scheme barred on the fundamental frequency estimation is proposed for systems with a harmonic noise. When reference signals necessary for feedforward ANC configuration are difficult to obtain, the conventional ANC algorithms for multi-tonal noise do not measure the reference signals but generate them with the estimated frequencies.$^{(4)}$ However, the beating phenomena, in which certain frequency components of the noise vanish intermittently, may make the adaptive frequency estimation difficult. The confusion in the estimated frequencies due to the beating phenomena makes the generated reference signals worthless. The proposed algorithm consists of two parts. The first part is a reference generator using the fundamental frequency estimation and the second one is the conventional feedforward control. We propose the fundamental frequency estimation algorithm using decision rules. which is insensitive to the beating phenomena. In addition, the proposed fundamental frequency estimation algorithm has good tracking capability and lower variance of frequency estimation error than that of the conventional cascade ANF method.$^{(4)}$ We are also able to control all interested modes of the noise, even which cannot be estimated by the conventional frequency estimation method because of the poor S/N ratio. We verify the performance of the proposed ANC method through simulations for the measured cabin noise of a passenger ship and the measured time-varying engine booming noise of a passenger vehicle.

• Proceedings of the KIEE Conference
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• 1995.07a
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• pp.171-173
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• 1995

Slot field harmonics exist in air gap due to inevitable slot constructure of induction motors. They give rise to noise by the electromagnetic vibration and mechanical pulsation. We calculate the slot field harmonics for varying the ratio of slot width vs teeth pitch using the carter's coefficent. We computate the flux density in air gap by FEM(Finite Element Method) and analyze it in frequency domain using DFT(Discrete Fourier Transform). We develop the new algorithm mixing FEM with DFT.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.10a
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• pp.842-849
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• 2009

The objective of this study is to investigate design parameters of a tuned liquid column damper(TLCD), which is affected by various excitation amplitudes, through shaking table test. Design parameters of a TLCD are examined based on the equivalent tuned mass damper(TMD) model of a TLCD, in which the nonlinear damping of a TLCD is transposed to equivalent viscous damping. Shaking table test is carried out for a TLCD specimen subjected to harmonic waves with various amplitudes. Transfer functions are ratios of liquid displacement of TLCD and control force produced by a TLCD, respectively, with respect to the acceleration excited by a shaking table. They are derived based on the equivalent TMD model of a TLCD. Then, the variation of design parameters according to the excitation amplitude is examined by comparing analytical transfer functions with experimental ones. Finally, the dissipation energy due to the damping of a TLCD, which is experimentally observed from the shaking table test, is examined according to the excitation amplitude. Comparisons between test results and analytical transfer functions showed that natural frequencies of TLCD and the ratio of the liquid mass in a horizontal column to the total liquid mass does not depend on the excitation amplitude, while the damping ratio of a TLCD increases with larger excitation amplitudes.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.11
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• pp.1167-1176
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• 2009

The objective of this study is to investigate design parameters of a tuned liquid column damper(TLCD), which is affected by various excitation amplitude, through shaking table test. Design parameters of a TLCD are examined based on the equivalent tuned mass damper(TMD) model of a TLCD, in which the nonlinear damping of a TLCD is transposed to equivalent viscous damping. Shaking table test is carried out for a TLCD specimen subjected to harmonic waves with various amplitude. Transfer functions are ratios of liquid displacement of TLCD and control force produced by a TLCD, respectively, with respect to the acceleration excited by a shaking table. They are derived based on the equivalent TMD model of a TLCD. Then, the variation of design parameters according to the excitation amplitude is examined by comparing analytical transfer functions with experimental ones. Finally, the dissipation energy due to the damping of a TLCD, which is experimentally observed from the shaking table test, is examined according to the excitation amplitude. Comparisons between test results and analytical transfer functions showed that natural frequencies of TLCD and the ratio of the liquid mass in a horizontal column to the total liquid mass do not depend on the excitation amplitude, while the damping ratio of a TLCD increases with larger excitation amplitude.

• Journal of the Computational Structural Engineering Institute of Korea
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• v.21 no.2
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• pp.135-143
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• 2008

In this study, it was investigated for dynamic nonlinear characteristics using dynamic data obtained by shaking table test. The design of Tuned Liquid Damper(TLD) has limitation to plan based on Tuned Mass Damper(TMD) analogy and linear wave theory. Also, while there are many studies regarding properties of TLD under harmonic load, there are not estimated for dynamic non-linear characteristics of TLD under the load that is not governed by particular frequency like a white noise. This paper investigated dynamic non-linear characteristics of TLD varied with load amplitude using a white noise and suggested equations that can estimate damping ratio, natural frequency ratio and effective mass ratio of TLD.

• Phonetics and Speech Sciences
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• v.9 no.3
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• pp.85-91
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• 2017

The present study aimed to investigate the changes of acoustic parameters of the aspirated voice in stroke patients. The eighty-eight subjects diagnosed with cerebro-vascular accident were divided into 32 penetration/aspiration (P/A) and 56 Non-P/A groups according to the videofluroscopic swallowing study (VFSS) results, and 26 control subjects participated. All subjects preformed VFSS and vowel /a/ was recorded three times pre- and post VFSS. Since the variation in the acoustic parameters within a single phonation has been observed, we proposed a delta formula for the acoustic parameters which can reflect the temporal changes of the each parameter in an utterance. We measured from the voice data eight acoustic parameters: fundamental frequency (F0), standard deviation of F0 (F0_SD), Jitter, relative average perturbation (RAP), Shimmer, amplitude perturbation quotient (APQ), harmonic to noise ration (HNR), noise to harmonic ratio (NHR). Then we found parameters which show the meaningful biggest temporal change in an utterance using the suggested delta parameter. Among them, the deltas of shimmer and APQ were significantly different pre- and post VFSS. These deltas of the P/A and the control group were increased after VFSS, while those of the Non-P/A group was descended. The variation patterns of the P/A and the control group were similar but the change width of the P/A group was larger. The large variations in an aspirated phonation of the P/A group are thought to be caused by irregular changes in air resistance due to residual food on the vocal cords.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.7
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• pp.3370-3392
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• 2017

Next Generation Beyond 4G/5G systems will rely on the deployment of small cells over conventional macrocells for achieving high spectral efficiency and improved coverage performance, especially for indoor and hotspot environments. In such heterogeneous networks, the expected performance gains can only be derived with the use of efficient interference coordination schemes, such as Fractional Frequency Reuse (FFR), which is very attractive for its simplicity and effectiveness. In this work, femtocells are deployed according to a spatial Poisson Point Process (PPP) over hexagonally shaped, 6-sector macro base stations (MeNBs) in an uncoordinated manner, operating in hybrid mode. A newly introduced intermediary region prevents cross-tier, cross-boundary interference and improves user equipment (UE) performance at the boundary of cell center and cell edge. With tools of stochastic geometry, an analytical framework for the signal-to-interference-plus-noise-ratio (SINR) distribution is developed to evaluate the performance of all UEs in different spatial locations, with consideration to both co-tier and cross-tier interference. Using the SINR distribution framework, average network throughput per tier is derived together with a newly proposed harmonic mean, which ensures fairness in resource allocation amongst all UEs. Finally, the FFR network parameters are optimized for maximizing average network throughput, and the harmonic mean using a fair resource assignment constraint. Numerical results verify the proposed analytical framework, and provide insights into design trade-offs between maximizing throughput and user fairness by appropriately adjusting the spatial partitioning thresholds, the spectrum allocation factor, and the femtocell density.