• Transactions of the Korean Society of Mechanical Engineers A
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• v.29 no.12 s.243
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• pp.1597-1604
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• 2005

The identification of a moving noise source is important in reducing the source power of the transport systems such as airplanes or high speed trains. However, the direct measurement using a microphone running with noise source is usually difficult due to wind noise, white the source motion distorts the frequency characteristics of the pass-by sound measured at a fixed point. In this study the relationship between the spectra of the source and the pass-by sound signal is analyzed for an acoustic source moving at a constant velocity. Spectrum of the sound signal measured at a fixed point has an integral relationship with the source spectrum. Nevertheless direct conversion of the measured spectrum to the source spectrum is ill-posed due to the singularity of the integral kernel. Alternatively a differential equation approach is proposed, where the source characteristics can be recovered by solving a differential equation relating the source signal to the distorted measurement in time domain. The parameters such as the source speed and the time origin, required beforehand, are also determined only from the frequency-phase relationship using an auxiliary measurement. With the help of the regularization method, the source signal is successfully recovered. The effects of the parameter errors to the estimated frequency characteristics of the source are investigated through numerical simulations.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.240-242
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• 1996

In this paper, we proposed a method of a hearing aid suitable for the sensorineural hearing impaired. Generally as the sensorineural hearing impaired have narrow audible ranges between threshold and discomfortable level, the speech spectrum may easily go beyond their audible range. Therefore speech spectrum must be optimally amplified and compressed into the impaired's audible range. The level and frequency of input speech signal are varied continuously. So we have to make compensation input signal for frequency-gain loss of the impaired, specially in the frequency band which includes much information. The input sigaal is divided into short time block and spectrum within the block is calculated. The frequency-gain characteristic is determined using the calculated spectrum. The number of frequency band and the target gain which will be added input signal are estimated. The input signal within the block is processed by a single digital filter with the calculated frequency-gain characteristics. From the results of monosyllabic speech tests to evaluate the performance of the proposed algorithm, the scores of test were improved.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.10
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• pp.4808-4824
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• 2016

Signal individual reconnaissance and identification is an extremely important research topic in non-cooperative domains such as electronic countermeasures and intelligence reconnaissance. Facing the characteristics of the complexity and changeability of current communication environment, how to realize radiation source signal individual identification under the low SNR conditions is an emphasis of research. A novel emitter individual identification method combined bi-spectrum analysis with wavelet feature is presented in this paper. It makes a feature fusion of bi-spectrum slice characteristics and energy variance characteristics of the secondary wavelet transform coefficient to identify MFSK signals under the low SNR (signal-to-noise ratios) environment. Theoretical analyses and computer simulation results show that the proposed algorithm has good recognition performance with the ability to suppress noise and interference, and reaches the recognition rate of more than 90% when the SNR is -6dB.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.22 no.6
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• pp.725-733
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• 1998

The pressure fluctuation on the surface of a submerged body has been recognized as a dominant noise source. There have been many studies concerning the flow induced noise on a flat plate. However, the noise over an axisymmetric body has not been well reported. This paper addresses the way in which we have investigated the mechanism of noise generation due to an axisymmetric body. The associated experiments and signal processing methods are introduced. A 3-dimensional axisymmetric body whose length and diameter were 2 m and 10.4 cm, was prepared as a test specimen. The wall pressure on the surface of the body was measured in a large scale low noise wind tunnel at KIMM(Korea Institute of Machinery and Metals). To measure the wall pressure, we used two microphone arrays which were tangential and normal to the flow. Based on the measured signal, frequency-wavenumber spectrum which explains the structure of turbulence noise, was estimated. Tangential to the flow, there exists convective ridge at a relatively higher wavenumber region; this can cause spatial aliasing. To circumvent this problem, the cross spectrum was interpolated. The interpolation has been performed by unwrapping the phase and smoothing the cross spectrum. The phase unwrapping was done based on the Corcos model; the phase of cross spectrum decreases linearly with the distance between microphones. Aforementioned signal processings are possible by employing the experimental results that the estimated wavenumber spectrum quite resembles the Corcos model. We try to modify the Corcos model which is applicable to the flat plate, by altering the magnitude of cross spectrum to fit the experimental data more accurately. We proposed that this wavenumber spectrum model is suitable for the 3-dimensional axisymmetric body. Normal to the flow, there exists a little correlation between signals of different microphones. The circumferential wavenumber spectrum contains uniform power along the wavenumbers.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.273-288
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• 2010

In this paper, we propose a cooperative multi-relay scheme for a secondary system to achieve spectrum access along with a primary system. In the primary network, a primary transmitter (PT) transmits the primary signal to a primary receiver (PR). In the secondary network, N secondary transmitter-receiver pairs (ST-SR) selected by a centralized control unit (CCU) are ready to assist the primary network. In particular, in the first time slot, PT broadcasts the primary signal to PR, which is also received by STs and SRs. At STs, the primary signal is regenerated and linearly combined with the secondary signal by assigning fractions of the available power to the primary and secondary signals respectively. The combined signal is then broadcasted by STs in a predetermined order. In order to achieve diversity gain, STs, SRs and PT will combine received replicas of the primary signal, using selection combining technique (SC). We derive the exact outage probability for the primary network as well as the secondary network. The simulation results are presented to verify the theoretical analyses.

• Speech Sciences
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• v.10 no.2
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• pp.85-95
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• 2003

In speech signal processing, it is very important to detect the pitch exactly in speech recognition, synthesis and analysis. If we exactly pitch detect in speech signal, in the analysis, we can use the pitch to obtain properly the vocal tract parameter. It can be used to easily change or to maintain the naturalness and intelligibility of quality in speech synthesis and to eliminate the personality for speaker-independence in speech recognition. In this paper, we proposed a new pitch detection algorithm. First, positive center clipping is process by using the incline of speech in order to emphasize pitch period with a glottal component of removed vocal tract characteristic in time domain. And rough formant envelope is computed through peak-fitting spectrum of original speech signal infrequence domain. Using the roughed formant envelope, obtain the smoothed formant envelope through calculate the linear interpolation. As well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. Inverse fast fourier transform (IFFT) compute this flattened harmonics. After all, we obtain Residual signal which is removed vocal tract element. The performance was compared with LPC and Cepstrum, ACF. Owing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.1
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• pp.33-39
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• 2018

An additive noise reduction algorithm for a mass spectrum analyzer is proposed. From the measured ion signal, we first used an estimated threshold from the mode of the measured signal to eliminate background noises with the white Gaussian characteristics. Also, a signal block corresponding to each mass index is constructed to perform a second order curve fitting and a linear approximation to signal block. In this process, the effective signal block composed of only the ion signal can be reconstructed by removing the impulsive noises and the sample signals which are insufficient to be viewed as normal ion signals. By performing curve fitting on the effective signal block, the noise-free mass spectrum can be obtained. To evaluate the performance of the proposed method, a simulation was performed using the signals acquired from the development equipment. Simulation results show the validity of the threshold setting from the mode and the superiority of the proposed curve fitting and linear approximation based noise canceling algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.3
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• pp.515-523
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• 1997

A high resolution algorithm is presented for resolving multiple coherent spread spectrum signals that are incident on an equispaced linear array. Unlike the conventional noise-eigenvector based methods, this algorithm makes use of the signal eigenvectors of the array spectral density matrix that are associates with eigenvalues that are larger than the sensor noise level. Simulation results are shown to demonstate the high performance of the proposed approach in comparison with MUSIC in which coherent signal subspace method (CSM) is employed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.5
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• pp.2015-2028
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• 2016

In this paper, we propose a sum transmission rate maximization based cooperative spectrum sharing protocol with quality-of-service (QoS) support for both of the primary and secondary systems, which exploits the situation when the primary system experiences a weak channel. The secondary transmitter ST_b which provides the best performance for the primary and secondary systems is selected to forward the primary signal. Specifically, ST_b helps the primary system achieve the target rate by using a fraction of its power to forward the primary signal. As a reward, it can gain spectrum access by using the remaining power to transmit its own signal. We study the secondary user selection and optimal power allocation such that the sum transmission rate of primary and secondary systems is maximized, while the QoS of both primary and secondary systems can be guaranteed. Simulation results demonstrate the efficiency of the proposed spectrum sharing protocol and its benefit to both primary and secondary systems.

• Journal of Communications and Networks
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• v.16 no.2
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• pp.140-145
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• 2014

In this paper, we propose an anti-interference cooperative spectrum sharing strategy for cognitive system, in which a secondary system can operate on the same spectrum of a primary system. Specifically, the primary system leases a fraction of its transmission time to the secondary system in exchange for cooperation to achieve the target rate. To gain access to the spectrum of the primary system, the secondary system needs to allocate a fraction of bandwidth to help forward the primary signal. As a reward, the secondary system can use the remaining bandwidth to transmit its own signal. The secondary system uses different bandwidth to transmit the primary and its own signal. Thus, there will be no interference felt at primary and secondary systems. We study the joint optimization of time and bandwidth allocation such that the transmission rate of the secondary system is maximized, while guaranteeing the primary system, as a higher priority, to achieve its target transmission rate. Numerical results show that the secondary system can gain significant improvement with the proposed strategy.