• The Journal of the Acoustical Society of Korea
• /
• v.13 no.6
• /
• pp.45-49
• /
• 1994

In this paper, photoacoustic image processing system was constructed by using 2W CW $CO_{2}$ laser of $10.6{\mu}m$ wavelength and PZT 5A acoustic transducer. Stainless steel of 5mm thickness was used as a sample in experiment. Three line cracks of $50{\mu}m$ in each width and depth were made by using plasma on the surface of the sample. Also, each gap among their lines was $200{\mu}m$ and $300{\mu}m$ in width. In the scan range of $2.2mm\times2mm$ including surface defects, a good image of $50{\mu}m$ resolution had been shown when modulation frequency of CW laser was 100Hz.

• The Journal of the Acoustical Society of Korea
• /
• v.12 no.5
• /
• pp.38-46
• /
• 1993

스피커는 입력이 전기신호이고 출력이 소리인 전기 음향 장치로서, 고유의 전달함수를 갖는 하나의 시스템으로 생각할 수 있다. 일한 스피커 시스템의 전달함수는 여러 가지 원인으로 인해 주파수 영역에서 평탄한 특성을 가지지 못하며 선형위상의 조건을 만족시키기가 힘들다. 음색을 변화시키는 스피커의 전달함수를 선형이라고 가정할 때 스피커의 앞단에서 스피커의 선형 전달함수의 역에 해당하는 선형 필터를 추가함으로쏘, 음색의 왜곡은 보정할 수 있다. 본 연구에서는 이러한 스피커의 선형왜곡을 보상하기 위한 선형필터로서 FIR 필터를 사용하였으나, 2장에 제시되어 있는 기존의 알고리듬과는달리 귀의 청감특성을 이용하여 가청주파수 대역에 대해 몇 개의 대역으로 나눈 후, 각 대역마다 보상 할 주파수 해상도를 달리하는 새로운 알고리듬에 대해 기술하였다. 이러한 다해상도 선형왜곡 보상시스템은 각 대역마다 다른 해상도로 보상한다는점에서는 필터뱅크 보상 시스템과 유사하지만, 대역통과필터가 아닌 저역통과필터만을 이용하여 설계의 복잡서응ㄹ 줄였으며 각 대역마다 독립적으로 보상하는 필터뱅크 보상시스템과는 달리 1차 대역에서 보상하지 ath한 오차를 2차, 3차에서 보상하는 새로운 기법을 이용하였다. 본 알고리듬을 적용한 모의실험을 통하여 성능을 평가한 결과 가청음대역에서 ±6dB의 왜곡을 보인 스피커가 실효 tap수를 272로 했을 때 ±0.5dB의 리플을 보이는 양호한 결과를 얻었다.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.5
• /
• pp.110-115
• /
• 2001

In this paper, we propose a beamformer adequate for the nested away that is generally used for multiple frequency band signal processing. The nonisotropic beam pattern of channel in this array causes two problems: the bearing-estimate error of mainlobe and the difference between design and output in sidelobe level. By separating the time delay among channel signals and the time delay among sensor signals in channel, we can remove the effects of the nonisotropic beam pattern of channel in the beamformer output. Through this process, a method to correct simultaneously these problems is proposed.

• Journal of Biomedical Engineering Research
• /
• v.32 no.3
• /
• pp.264-269
• /
• 2011

To increase therapeutic efficiency and biological safety, it is important to precision control of acoustic output for therapeutic ultrasound equipment. In this paper, the electro-acoustic radiation conductance, one of electroacoustic characteristics of therapeutic ultrasound equipment, was measured by the radiation force balance method according to IEC 61161 standards and the acoustic output was estimated using the electro-acoustic radiation conductance. The estimation of acoustic output was conducted to continuous wave mode and pulse wave mode of duty cycle between 20% and 80%. The differences between prediction values and measurement results are within 5% of measurement uncertainty, which is a reasonably good agreement. The results show that acoustic output controlled by electro-acoustic radiation conductance was found to be an effective method.

• Korean Journal of Optics and Photonics
• /
• v.14 no.6
• /
• pp.657-662
• /
• 2003

Optimum Hamming apodized acousto-optic 2${\times}$2 add/drop multiplexers, using SiO$_2$ film loaded acoustic waveguide with angular offset to the Ti:LiNbO$_3$ waveguide, were fabricated. The four-port device consists of two input waveguides, a polarization beam splitter (PBS), two polarization conversion/acoustooptic tuning waveguide sections, a second PBS and two output waveguides. Insertion loss <7.1 ㏈ has been obtained and side-lobe of -19 ㏈ for -32 ㎽ RF driving power has been realized. Add/drop performance has been confirmed and a linear tuning rate of 8.1 nm/MHz and a 3 ㏈ spectral width of -1.5 nm were demonstrated.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.1
• /
• pp.24-29
• /
• 2004

In ocean acoustic tomography (OAT), the pulse compression techniques using M-sequences are employed in the many studies for investigating the ocean structures. M-sequences can provide the good time and Doppler resolution in the process of demodulation using matched-filter. The signal-to-noise (SNR) performance at the output of receiver may be improved by manipulating received signal, i. e. coherently averaging. The processing time can be significantly reduced by using fast hadarmard transform (FHT) or fast Fourier transform (FFT). In this paper, we estimate the multipath arrival structures and delay times using the East Korean Sea experiment data and explore the compensation method for the detrimental effects on performance due to sampling rate error. We also analyze the characteristics of the ocean acoustic channels through scattering function, delay power profile, and time dispersions.

• Journal of KIISE:Software and Applications
• /
• v.28 no.10
• /
• pp.699-705
• /
• 2001

As it became possible to generate 3D sound on a PC environment due to the advances in computing performance and digital signal processing technology, 3D sound technology gains its focus in the multimedia area Specifically a two-channel based 3D sound technology is being studied by many researchers because of its space efficiency and economical structure. While the positional sound effect is simple in its implementation, the moving sound effect has many problems to be resolved as there are only discrete measured point of HRTF database. In this paper, we propose the method of generating smooth moving sound in a two-channel based 3D sound technique with respect to generating smooth trajectory, and the interpolation method of discrete measured HRTF data. We perform the tests in the PC environment and prove the utility of the proposed method.

• The Journal of the Acoustical Society of Korea
• /
• v.7 no.2
• /
• pp.20-25
• /
• 1988

This study is to explain the characteristic of excess attenuation on the ground through the outdoors experiment about noise propagation and the reduced model experiment of acoustic. The outdoors experiment on the attenuation of noise propagation was tried with the small engine that had large acoustic output, and then it was conformed that there was relationship between the excess attenuation calculated by measurement from distance attenuation and Log(D/(Hs+Hr)). As a result, it was found that the attenuation of noise propogation depended upon the direction of the wind and frequency and was regressed in a straight line. And the numerical values of excess attenuation on the ground could be calculated by regarding Log(D/(Hs+Hr)) as a parameter with an airing resistance $\sigma$. It was found that when the mean square error between the excess attenuation calculated by measurement and the value calculated by a fomula $L=-20Log\mid1+(r_1/r_2)Qexp(ik, \bigtriangleup r)\mid$ about optional $\sigma$ was least, the optimal decision of u was made. As the characteristic of model is the model experiment on a reduced scale of 1 to 40, It was conformed that it corresponds enough with the measurement value with measuring the distance attenuation in the large anecoic chamber.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.16 no.3
• /
• pp.401-406
• /
• 2021

Recently, mean-teacher models based on convolutional recurrent neural networks are popularly used in audio event detection. The mean-teacher model is an architecture that consists of two parallel CRNNs and it is possible to train them effectively on the weakly-labelled and unlabeled audio data by using the consistency learning metric at the output of the two neural networks. In this study, we tried to improve the performance of the mean-teacher model by using additional derivative features of the log-mel spectrum. In the audio event detection experiments using the training and test data from the Task 4 of the DCASE 2018/2019 Challenges, we could obtain maximally a 8.1% relative decrease in the ER(Error Rate) in the mean-teacher model using proposed derivative features.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2021.05a
• /
• pp.584-588
• /
• 2021

The impedance of the speaker is likely to be recognized as a fixed value. However, speaker impedance continues to vary with frequency variation, especially larger in resonant frequency region. The sound pressure level of loudspeakers is determined by the current flowing throughout the coil that consists loudspeakers. If loudspeakers are driven by voltage, sound pressure level of the loudspeaker is distorted by the variation of loudspeaker impedance. Current-drive of loudspeakers can solve this problem, but distortion of sound pressure level occurs in low frequencies due to resonance. The distortion can degrade the sound quality of the sound system. So to solve this problem, In this paper, we propose a resonance compensation circuit using DSP. we simulates audio systems using an equivalent model of loudspeakers to verify distortion of sound pressure level due to impedance variation and propose a circuit to compensate it. The proposed circuit is configured using a state variable filter and it can adjust the center frequency and output, so it will be used various sound systems.