• The Journal of the Acoustical Society of Korea
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• v.10 no.1
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• pp.20-29
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• 1991

A fundamental assumption in conventional linear predictive coding (LPC) analysis procedure is that the input to an all-pole vocal tract filter is white process. In the case of periodic inputs, however, a pitch bias error is introduced into the conventional LP coefficient. Multi-pulse (MP) LP analysis can reduce this bias, provided that an estimate of the excitation is available. Since the prediction error of conventional LP analysis can be modeled as the sum of an MP excitation sequence and a random noise sequence, we can view extracting MP sequences from the prediction error as a classical detection and estimation problem. In this paper, we propose an algorithm in which the locations and amplitudes of the MP sequences are first obtained by applying a likelihood ratio test (LRT) to the prediction error, and LP coefficients free of pitch bias are then obtained from the MP sequences. To verify the performance enhancement, we iterate the above procedure with adaptive threshold at each step.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.340-343
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• 2019

In this paper, a problem of mechanical resonance between traction motor's rigid body mode and traction motor's excitation force is introduced, and a bogie design variable affecting the control of resonance response is reviewed numerically. To solve the resonance problem in rotating machinery with variable rotational speeds, resonance frequency should be out of rotational machine's operation range or dynamic stiffness of structures should be increased for resonance response enough to be low. In general, operation range of a traction motor is from 0 r/min to 4800 r/min. It is not possible that all bogie modes are more than 80 Hz. Therefore, it is very important to find design factor affecting resonance response of traction motor's rigid body modes. It is found that key design variable is the gab between transom pipes from finite element analysis. The larger gab is, the higher resonance response when resonance between traction motor's excitation force and traction motor's rigid body mode is happened.

• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.148-153
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• 2010

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. A Speech signal consists of Voice and Unvoiced Band in The MBE excitation model. And the energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced. We use the IMBE vocoder for the Voice or Unvoice band of segmented speech signal. Continuously we calculate the segmented SNR using that information and the energy of each band. And we estimate the SNR of continuous speech signal.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.497-502
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• 2005

In mobile phones of multimedia era, microspeakers of high quality sound are essential parts to generate human voice in speaker phone and MP3 song player. In this paper, two types of microspeakers, outer permanent magnet (PM) and combined PM type, are analyzed using electromagnetic, mechanical, acoustical and their coupling analysis. For performance comparison, voice coil diameter is chosen as a design parameter to change excitation position and magnet volume for both types. For combined PM type, sound pressure level (SPL) is improved due to increased PM volume compared to outer PM type. Also, with the decreased voice coil diameter for combined PM type, the 1st resonant mode of the diaphragm is more efficiently excited due to concentrative excitation, resulting in lower and broader frequency range. Therefore, it can be said that the combined PM type microspeakers are more advantageous for high performance microspeaker which are essential for multimedia era.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.275-280
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• 1999

Deep-water sonar transducers of FFR (Free Flooded Ring) type have been designed using a coupled FE-BEM. The proposed sonar transducers are composed of piezoelectric ceramic tubes and structural steel materials for simple fabrication. In order to have an omnidirectional beam pattern around 30 kHz, a conic steel is placed below a piezoelectric tube or a steel disc is placed between two piezoelectric tubes. The dynamics of the sonar transducer is modelled in three dimensions and is analyzed with external electrical excitation conditions. Various results are available such as directivity patterns and transmitting voltage responses. The most optimal structure and dimensions of the steel material were calculated, so that the beam patterns of the sonar transducers had +/- 3 dB omnidirectivity at 30 kHz.

• The Journal of the Acoustical Society of Korea
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• v.33 no.4
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• pp.238-247
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• 2014

In this paper, rail vibration and its sound radiation are investigated, then the rail noise measurement by using microphone array is explored theoretically. A concrete slab track for domestic high speed trains is modeled as a Timoshenko beam on elastic support, regarding the moving of the excitation force on the rail. From the radiation characteristics of rail noise generated by a line source, the effect of moving load on sound radiation is obtained. Also it is found that the beam angle of the microphone array is a prominent factor to measure the rail noise level reliably because the rail noise propagates as a plane wave. In this investigation, a proper beam angle for the rail noise measurement by microphone array is suggested.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2006.11a
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• pp.506-509
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• 2006

Shaped Sound Focusing is defined as the generation of acoustically bright shape in space using multiple sources. The acoustically bright shape is a spatially focused region with relatively high acoustic potential energy level. In view of the energy transfer, acoustical focusing is essential because acoustic energy is very small to use other type of energy. Practically, focused sound shape control not a point is meaningful because there are so many needs to enlarge the focal region especially in clinical uses and others. If focused sound shape can be controlled, it offers various kinds of solutions for clinical uses and others because a regional focusing is essentially needed to reduce a treatment time and enhance the performance of transducers. For making the shaped-sound field, control variables, such as a number of sources, excitation frequency, source positioning, etc., should be taken according to geometrical sound shape. To verify these relations between them, wavenumber domain matching method is suggested because wavenumber spectrum can provide the information of control variables of sources. In this paper, the procedures of shaped sound focusing using wavenumber domain matching and relations between control variables and geometrical sound shape are covered in case of an acoustical ring.

• The Journal of the Acoustical Society of Korea
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• v.9 no.3
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• pp.24-30
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• 1990

In this paper, meander line type EMAT(Electro-Magnetic Acoustic Transducer) has been designed and fabricated with effective properties for detecting flaw existing within one wavelength in depth, and its characteristics have been analyzed. For the purpose of getting effective dynamic and static magnetic intensity, the coil has been arrayed using wire with interval of 0.75 mm and width of 0.65 mm and permanent magnets with 1500 Gauss have been constructed respectively. The center frequency and fractional bandwidth of the fabricated EMAT was 2 MHz and 36% respectively and its impulse response has been measured by non-contacting technique(the distance between the conducting media and the coil was 0.15mm). In the measuring results, it has been shown that Insertion Loss(IL) was 45.46dB and it was good agreement with theoretical result.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.3-12
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• 2001

It is known that the roughness is one of the most important metrics in assessing the sound quality. In this study, a new roughness model is suggested by combing the previous auditory filter model and several signal processing methods for the enhancement of calculation efficiency and accuracy. For testing the usefulness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements. Also, it is found that the previous models have limitations to search frequency components mainly contributed to overall roughness. By modifying the correlation criteria of the present model, the revised model for the proper estimation of roughness-contributed components is embedded.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.99-103
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• 1996

Most waveform coding techniques attempt to reduce redundancy of speech signal while preserving the shape of the waveform. In speech synthesis, wavefrom coding methods are used to the synthesis by rule for high quality speech. However, it is difficult to apply the waveform coding to the synthesis by rule because the parameters of the wavefrom coding cannot be classified as either the excitation or the vocal tract parameters. The proposed method shows little spectrum distortion of 2.7% or less for 50% pitch changes. It also achieves smooth connection of wavefrom magnitudes among the frames by compensating the phase in time domain.