• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.245-250
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• 2009

In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.717-726
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• 2003

This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.11-20
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• 1997

The passive noise control techniques used until now cancel the noise in terms of the characteristics of materials, which increase the mass and the dimension and have a limit that is effective only to attenuate the high frequency components of the noise. But the active noise control techniques developed in recent years have merits that they decrease the mass and the dimension and are effective to attenuating the low frequency noises. In this paper, the real-time active noise control system attenuating the engine booming noise in a car using the digital signal processing(DSP) techniques in terms of the principle of active noise control. The multiple-error filtered-x LMS(Least-Mean Square) algorithm is used as the adaptive algorithm for active noise control and is implemented using the DSP processor Motorola DSP56001 as a controller. According to the result that the experiments are performed for the engine as the RPM changes in a car, the noise attenuating performances are achieved in an overall car interior and is verified to be 20 dB higher for pure-tone and globally, 15 dB.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.208-215
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• 2016

To control and improve a combustion behavior of an engine, various studies for the in-cylinder pressure have been consistently carried out. In this paper, the level of the combustion noise for a diesel engine is estimated from the in-cylinder pressure and defined as the combustion noise index. The combustion noise index is calculated from the FFT(Fast Fourier Transform) of the in-cylinder pressure and its validity is verified. The control system based on the combustion noise index is developed and implemented in a vehicle. A number of injection parameters are controlled to meet the desired combustion noise index, and the combustion noise of a vehicle is improved up to 4.0 dB(A) in the specified frequency band.

• The Journal of the Acoustical Society of Korea
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• v.39 no.3
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• pp.163-169
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• 2020

The asynchronous bistatic sonar needs to estimate direct blast arrival time at a receiver to localize targets, and therefore the direct blast arrival time estimation error could be added to target localization error in comparison with synchronous system. Direct blast especially appears as several peaks at the matched filter output by multipath, thus we compared the first peak detection technique and the maximum peak detection technique of those peaks for direct blast arrival time estimation through sea trial data. The test was performed in a shallow sea with bistatic sonar made up of spatially separated source and line array sensors. Line array sensors obtained the target signal which is generated from the echo repeater. As a result, the first peak detection technique is superior to maximum peak detection technique in direct blast arrival time estimation error. The result of this analysis will be used for further research of target tracking in the asynchronous bistatic sonar.

• The Journal of the Acoustical Society of Korea
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• v.8 no.1
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• pp.5-12
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• 1989

This paper is concerned with the characteristics on the statistical parameters of vibration signal from bearing with changing its operating conditions as well as the spreading of faults. The rms, Kurtosis, crest factor, probability of exceedance and probability density function have been chose as the statistical parameters. To characterize of each, vibration signals have been recorded from four ball tester at different loads, operation speeds and time. The values of the statistical parameters for each frequency band have been calculated after A/D conversion and digital filtering of the recorded signals. It has been found that unlike rms values the statistical parameters such as Kurtosis etc. are almost unchanging with the change of the operating conditions such as load and speed. This suggests that the statistical parameters may be used for determining the development of faults independent of the operating conditions. In fact, the statistical parameters deviate considerably from their respective normal values when the faults developed under load conditions in the samples, conforming the suggestion.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.53-59
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• 1989

A new proposed method can separate heart sounds and lung sounds by the realization of adaptive noise canceler using adaptive lattice Wiener filter in contrast to adaptive transversal LMS filter and high pass filter as before. Lung sounds and ECG signal are detected for this purpose, and especially the second heart sounds are reduced by finding T wave location with a T wave seeking algorithm. As a result, for heart sounds reduction It was found that adaptive transversal LMS filter required 100-200's orders, 75-100's orders In adaptive transversal MLMS filter, and only 10-20's orders in adaptive lattice Wiener filter. Adaptive filtering technique has shown greater accuracy than high pass filtering without loss of low frequency component.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.5-13
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• 2001

It is well known that there are two experimental errors on acoustic decay measurements. ,One is due to the influence of the band pass filter the other one is that of an averaging device. In this paper the influence of the filter is investigated in detail. To minimize the influence of the filter, the product of the filter bandwidth B (3dB bandwidth) and the reverberation time T/sub 60/ of the room under test should be at least 16. Moreover, if the initial part of an acoustic decay curve is important, the strong requirement, i. e. BT/sub 60/〉64, must be satisfied. In this paper, the wavelet filter bank instead of the band pass filter bank is applied to obtain an acoustic decay curve. As a result, the influence of filter is reduced and then the value of BT/sub 60/ required for obtaining an acceptable decay curve becomes at least 4. The strong requirement for the initial part of a decay curve is also replaced by the BT/sub 60/〉16 instead of BT/sub 60/〉64.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.87-93
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• 2001

As a new type of a linear decorrelating receiver, the Pseudo-Decorrelator was presented for asynchronous code division multiple access systems by the author. In this paper, the concept of the Pseudo-Decorrelator is extended to derive a receiver for WCDMA uplink systems over an additive white Gaussian noise channel. Starting with the analysis of the multiple access components of the decision statistics, a non-square cross-correlation matrix for each bit is obtained. This cross-correlation matrix is then inverted, and the inverted matrix is applied to the decision statistics obtained from a conventional receiver. In this receiver, the detection process can be started after the first three consecutive bits are received. Simulation results are presented for K-user systems over an additive white Gaussian noise channel under the circumstances in which synchronization errors, including time delay errors and carrier phase errors exist. It is shown that the proposed receiver performs better than a conventional receiver and parallel interference canceller.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2
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• pp.76-85
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• 1994

A speech enhancement algorithm which attenuates nonstationary colored noise is presented In this paper. The algorithm consists of a stationary Kalman filter and the simple speech/nonspeech detector. While the conventional enhancement systems are focused on a stationary and/or white background noise, this study Is focused on the mort realistic nonstationary and nonwhite noise. An AR model-based vector Kalman filter is used as a noise suppression system and a short-time energy threshold logic is used as a speech/nonspeech classifier. For Kalman filtering. noise coefficients are estimated in the nonspeech frame, and speech coefficients are estimated by applying the EM iteration algorithm. Simulation results using the car noise are presented based on the signal-to-noise ratio and informal listening tests. According to the experimental results, background noises in the nonspeech frames are eliminated almost completely, while some distortions are noticed in the speech frames. The distortion becomes severer as the SNR is reduced to 0dB and -5dB. Intelligibility, however, is not degraded significantly.