• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.9
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• pp.724-730
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• 2014

It is essential to control noise and vibration in various industrial fields. In the automobile industry, various plastics have been developed and replaced metallic materials in order to reduce mass and vibration effectively. In this study, we measured and analyzed the Young's moduli and the loss factors of Acrylonitrile butadiene styrene(ABS) and Polypropylene(PP). In order to solve the fundamental error to determine the two quantities, a loudspeaker was used instead of conventional electromagnetic devices to generate bending motion to the specimens and a laser vibrometer was also used in detection of vibration signal of the specimen. The measured Young's moduli and loss factors of the ABS specimen were nearly constant as the temperature($-10{\sim}60^{\circ}C$) was increased. The loss factor of PP specimen showed peak value at $20^{\circ}C$ and it means that there is glass transition for the PP specimen. Young's modulus of PP specimen was linearly decreased as the temperature was increased.

• The Journal of the Acoustical Society of Korea
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• v.10 no.4
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• pp.5-11
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• 1991

This paper suggests a segmentation method of speech signal into phonemic units. The suggested segmentation system is speaker-independent and performed without anyprior information of speech signal. In segmentation process, we first divide input speech signal into purevoiced region and not pure voiced speech regions. After then we apply the second algorithm which segments each region into the detailed phonemic units by using the voiced detection parameters, i.e., the time variation of 0th LPC cepstrum coefficient parameter and the ZCR parameter. Types of speech, used to prove the availability of segmentation algorithm suggested in this paper, are the vocabulary composed of isolated words and continuous words. According to the experiments, the successful segmentation rate for 507 phonemic units involved in the total vocabulary is 91.7%.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.585-591
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• 2003

In this paper, fiber optic sensor using Fabry-Perot interferometer which had benefit of minimize and light-weight was used. The sensor head has 1cm in length, total length of fiber is 9.5 chi and the sensor supported at both ends, simply. To analyze the acoustic characteristic non-directional speaker is used as a sound source. Acoustic applied in lateral direction and detected two signals were compared each other. Below 1㎑ fiber optic sensor has more sensitive than microphone, but in 2㎑ fiber optic sensor has less sensitive than microphone. This characteristic varies to the supporting system of fiber optic sensor. It was confirmed that the Fabry-Perot interferometric sensor detected acoustic signal, effectively. This kind of sensor can be applied to the structural health monitoring field of intellectual structure.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.68-75
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• 1997

Spectral subtraction is widely-used preprocessing technique for speech recognition in additive noise environments, but it requires a good estimate of the noise power spectrum. In this paper, we employ the histogram technique for the estimation of noise spectrum. This technique has advantages over other noise estimation methods in that it does not requires speech/non-speech detection and can estimate slowly-varying noise spectra. According to the speaker-independent isolated word recognition in both colored Gaussian and car noise environments under various SNR conditions. Histogram-technique-based spectral subtraction method yields superier performance to the one with conventional noise estimation method using the spectral average of initial frames during non-speech period.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.7
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• pp.626-633
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• 2012

In this paper, sound pressure sensitivity of the fiber optic acoustic sensor according to sensor direction and mandrel material were investigated experimentally. Three different directions were selected as stand, lay, and hole. Hollow cylinder type mandrel dimension is 30 mm in outer diameter, 45 mm in length, and 2 mm in thickness, and about 50 m optical fibers were wounded on the surface of the mandrel. Non-directional sound speaker was used as a sound source. Sagnac interferometer and single mode fiber, a laser with 1,550 nm in wavelength, $2{\times}2$ coupler were used. Based on the experimental results, lay direction's sensitivity is the highest in the frequency range of 2 kHz~4 kHz. 'PTFE+carbon' material is more sensitive than PTFE in the frequency range of 5 kHz~20 kHz. Sound pressure detection sensitivity depends on the mandrel direction and material under certain frequency.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4
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• pp.14-22
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• 1998

본 연구에서는 Texas Instruments 사의 저가형 부동소수점 디지털 신호 처리기 (Digital Singnal Processor, DSP)인 TMS320C32를 이용하여 실시간 화자종속 음성인식 하 드웨어 모듈(VR32)을 개발하였다. 하드웨어 모듈의 구성은 40MHz의 TMS320C32 DSP, 14bit 코덱인 TLC32044(또는 8bit μ-law PCM 코덱), EPROM과 SRAM 등의 메모리와 호 스트 인터페이스를 위한 로직 회로로 이루어졌다. 뿐만 아니라 이 하드웨어 모듈을 PC사에 서 평가해보기 위한 PC 인터페이스용 보드 및 소프트웨어도 개발하였다. 음성인식 알고리 즘의 구성은 에너지와 ZCR을 기반으로 한 끝점검출(Endpoint Detection) 침 10차 가중 LPC 켑스터럼(Weighted LPC Cepstrum) 분석이 실시간으로 이루어지며 이후 Dynamic Time Warping(DTW)를 통하여 최고 유사 단어를 결정하고 다시 검증과정을 거쳐 최종 인식을 수행한다. 끝점검출의 경우 적응 문턱값(Adaptive threshold)을 이용하여 잡음에 강인한 끝 점검출이 가능하며 DTW 알고리즘의 경우 C 및 어셈블리를 이용한 최적화를 통하여 계산 속도를 대폭 개선하였다. 현재 인식률은 일반 사무실 환경에서 통상 단축다이얼 용도로 사 용할 수 있는 30 단어에 대하여 95% 이상으로 매우 높은 편이며, 특히 배경음악이나 자동 차 소음과 같은 잡음환경에서도 잘 동작한다.

• Journal of the Korean Society of Environmental Restoration Technology
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• v.25 no.2
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• pp.85-92
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• 2022

The amphibian breeding habitat is confirmed by mating call. In some cases, the researcher directly identifies the amphibian individual, but in order to designate the habitat, it is necessary to map the mating call region of the amphibian population. Until now, it has been a popular methodology for researchers to hear mating calls and outline their breeding habitats. To improve this subjective methodology, we developed a technique for mapping mating call regions using Unmanned Aerial Vehicle (UAV). The technology uses a UAV, fitted with a sound recorder to record ground mating calls as it flies over an amphibian habitat. The core technology is to synchronize the recorded sound pressure with the flight log of the UAV and predict the sound pressure in a two-dimensional plane with probability density. For a demonstration study of this technology, artificial mating call was generated by a potable speaker on the ground and recorded by a UAV. Then, the recorded sound data was processed with an algorithm developed by us to map mating calls. As a result of the study, the correlation coefficient between the artificial mating call on the ground and the mating call map measured by the UAV was R=0.77. This correlation coefficient proves that our UAV recording system is sufficiently capable of detecting amphibian mating call regions.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.51-56
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• 2000

Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.4
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• pp.289-297
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• 2010

In this paper, A method to prevent the external acoustic noise from being misrecognized as the speech recognition object is presented in the speech activity detection process for the speech recognition. Also this paper confirmed besides the acoustic energy to the lip movement image signals. First of all, the successive images are obtained through the image camera for personal computer and the lip movement whether or not is discriminated. The next, the lip movement image signal data is stored in the shared memory and shares with the speech recognition process. In the mean time, the acoustic energy whether or not by the utterance of a speaker is verified by confirming data stored in the shared memory in the speech activity detection process which is the preprocess phase of the speech recognition. Finally, as a experimental result of linking the speech recognition processor and the image processor, it is confirmed to be normal progression to the output of the speech recognition result if face to the image camera and speak. On the other hand, it is confirmed not to the output the result of the speech recognition if does not face to the image camera and speak. Also, the initial feature values under off-line are replaced by them. Similarly, the initial template image captured while off-line is replaced with a template image captured under on-line, so the discrimination of the lip movement image tracking is raised. An image processing test bed was implemented to confirm the lip movement image tracking process visually and to analyze the related parameters on a real-time basis. As a result of linking the speech and image processing system, the interworking rate shows 99.3% in the various illumination environments.

• KIPS Transactions on Computer and Communication Systems
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• v.4 no.9
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• pp.321-326
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• 2015

In this paper, we proposed an algorithm which could transmit reliable data between smart devices by using inaudible high frequency of audible frequency range and cyclic redundancy check method. The proposed method uses 18 kHz~22 kHz as high frequency which inner speaker of smart device can make a sound in audible frequency range (20 Hz~22 kHz). To increase transmission quantity of data, we send mixed various frequencies at high frequency range 1 (18.0 kHz~21.2 kHz). At the same time, to increase accuracy of transmission data, we send some mixed frequencies at high frequency range 2 (21.2 kHz~22.0 kHz) as checksum. We did experiments about data transmission between smart devices by using the proposed method to confirm data transmission speed and accuracy of the proposed method. From the experiments, we showed that the proposed method could transmit 32 bits data in 235 ms, the transmission success rate was 99.47%, and error detection by using cyclic redundancy check was 0.53%. Therefore, the proposed method will be a useful for wireless transmission technology between smart devices.