• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.16-24
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• 2000

In this paper the performance of a closed-loop power control scheme using variable step size decision method for DS/CDMA based-low earth orbit(LEO) mobile satellite systems in which the long round trip delay is a dominant performance degradation factor is evaluated. Because there are fundamental differences in the characteristics between the LEO mobile satellite channel and terrestrial mobile channel, such as long round trip delay and different elevation angle, these factors are considered in channel modeling based on the European Space Agency(ESA) measurement data. Since the round trip delay (from the mobile terminal to the gateway station via satellite) is typically 10∼20ms in low altitude satellite channels, closed-loop power control is much less effective than it is on a terrestrial channel. Thus, the adaptive power control scheme using a variable step size control is essential for overcoming the long round trip delay and fading due to the elevation angle. It is shown that the standard deviation of signal to interference ratio(SIR) adopting a variable step size closed-loop power control scheme is much less than that of a fixed step size closed-loop power control. Furthermore, we have driven the conclusion that the measurement interval of power control commands is optimal choice when it is twice the round trip delay.

• Journal of the Korean Society for Railway
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• v.20 no.4
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• pp.440-447
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• 2017

Spectrum level for the acoustic roughness of wheels and rail surface should be periodically maintained under the limitation of ISO to reduce rolling noise of railway vehicles. Thus, in maintaining railway track, displacement sensor-based measuring devices are broadly used to measure the surface roughness and to perform spectral analysis. However, these measuring devices cause unexpected measuring errors since the displacement sensors are fixed at moving platforms and the main frame produces pitching motion during measurement. To increase the accuracy of the measured values, this paper has investigated the effects of design variables such as wheel base, additional wheels, and elastic deformation of wheels on the surface roughness and acoustic roughness spectrum.

• Journal of Advanced Navigation Technology
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• v.16 no.6
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• pp.1092-1100
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• 2012

In this paper, it has designed and developed the integrated aural cueing system(ACS) system of the reconfigurable tactical Flight Training System(RTT) for the 6 rotorcraft such as UH-1H, UH-60, AH-1H, 500MD, BO-100, and CH-47. RTT is an evolving alternative instructional training system to provide the ability to rehearsal and collectively train, through networked simulators in a unit-collective and combined arms simulated battlefield environment. ACS handles the volume, pitch and repetition of the digitally stored sounds based on commands it receives from the Host server. This paper explained and implemented the conceptual and detail design the ACS system. In order to evaluating the performance of the ACS system, we made the monitoring system for interworking the virtual Host and the ACS system. As the result, it was confirmed the good performance.

• The Journal of the Acoustical Society of Korea
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• v.13 no.4
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• pp.77-85
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• 1994

In this study an adaptive algorithm with optimum convergence factor for steepest descent method is proposed, which controls automatically the filter order to take the appropriate level. So far, fixed order filters have been used when adaptive filter is employed according to the priori knowledge or experience in various adaptive signal processing applications. But, it is so difficult to know the filter order needed in real implementations that high order filters have to be performed. As a result, redundant calculations are increased in the case of high order filters. The proposed variable length optimum convergence factor (VLOCF) algorithm takes the appropriated filter order within the given one so that the redundant calculation is decreased to get the enhancement of convergence speed and smaller convergence error during the steady state. The proposed algorithm is evaluated to prove the validity by computer simulation for system Identification.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.76-82
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• 1994

The characteristics of bottom sediment may be able to vary within a few meters of depth in shallow water. Since bottom attenuation coefficient as well as sound velocity in the bottom layer is determined by the composition and characteristics of sediment itself, it is reasonable to assume that the bottom attenuation coefficient is accordingly variable with depth. In this study, we use a parabolic equation scheme to examine the effects of depth-varying compressional wave attenuation on acoustic wave propagation in the low frequency ranging from 100 to 805 Hz. The sea floor under consideration is sandy bottom where the water and the sediment depths are 40 meters and 10 meters, respectively. Depending on the assumption that attenuation coefficient is constant or depth-varying, the propagation loss difference is as large as 10dB within 15 km. The predicted propagation loss is very much comparable to the measured one when we employ a depth-varying attenuation coefficient.

• The Journal of Engineering Research
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• v.2 no.1
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• pp.57-62
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• 1997

Two psychoacoustic models used in MPEG-1 employ different masking patterns, different masking indexes, and different computational procedures. As a result, Model 1 is inferior to Model 2 due to its worst case approach in computing the SMR even though it determines tonality and masking levels accurately. In this study, we investigate the performances of psychoacoustic models when we modify the MPEG-1 audio coder for variable bitrates. Simulation results show that Model 2 has a gain of 30 kbps in the dual channel mode and 20 kbps in the joint stereo mode. It is generally known that the joint stereo mode has a gain in bitrate compare to the dual channel mode. For signals with frequent attacks, this gain becomes larger in Model 1 than in Model 2. This is due to the fact that Model 1 uses the worst case approach in computing the SMR to reduce pre-echo

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

• The Journal of the Acoustical Society of Korea
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• v.27 no.4
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• pp.200-206
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• 2008

This paper presents a method of quantization to extract quantization parameters using the straight-line and DCT (Discrete Cosine Transform) for two splited frequency bands. As the number of harmonic is variable frame to frame, harmonics in low frequency band is oversampled to fix the dimension and straight-lines present a spectral envelope, then the discontinuous points of straight-lines in low frequency is sent to quantizer. Thus, extraction of quantization parameters using the straight-line provides a fixed dimension. Harmonics in high frequency use variable DCT to obtain quantization parameters and this paper proposes a method of quantization combining the straight-line with DCT. The measurement (If proposed method of quantization uses spectral distortion (SD) for spectral magnitudes. As a result, The proposed method of quantization improved 0.3dB in term of SD better than HVXC.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.223-230
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• 2011

Korean bell is hung with some air gap between the bell bottom and the ground. In addition, it has a peculiar acoustic element, so called resonance cavity below the bell. A proper design of the air gap and cavity size dramatically amplifies the bell sound by resonance effect. Bell interior cavity, air gap and resonance cavity consist of an acoustic cavity system. When the acoustic cavity frequency coincides with the natural frequency of the bell body, the frequency component is significantly amplified. On the Sacred Bell of the Great King Seongdeok, this study proposes a resonance condition of the cavity system considering air gap effect for the first time. With the exact dimension of the bell, boundary element analysis is performed using SYSNOISE. Finally, this study reveals how the temperature in season influences the resonance condition and proposes a concept of variable type resonance cavity. By using the variable type resonance cavity, the cavity size is controlled on site and exact resonance is available regardless of temperature difference in season.

• Phonetics and Speech Sciences
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• v.11 no.3
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• pp.39-47
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• 2019

We propose a method to improve the performance of spontaneous speech recognizers by extending their phone set using speech data. In the proposed method, we first extract variable-length phoneme-level segments from broadcast speech signals, and convert them to fixed-length latent vectors using an long short-term memory (LSTM) classifier. We then cluster acoustically similar latent vectors and build a new phone set by choosing the number of clusters with the lowest Davies-Bouldin index. We also update the lexicon of the speech recognizer by choosing the pronunciation sequence of each word with the highest conditional probability. In order to analyze the acoustic characteristics of the new phone set, we visualize its spectral patterns and segment duration. Through speech recognition experiments using a larger training data set than our own previous work, we confirm that the new phone set yields better performance than the conventional phoneme-based and grapheme-based units in both spontaneous speech recognition and read speech recognition.