• The Journal of the Acoustical Society of Korea
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• v.33 no.3
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• pp.184-190
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• 2014

Although it is ideal that the sound with the range of 20 ~ 20,000 Hz is produced by the loudspeaker system composing of only one loudspeaker unit, it is almost impossible now. Therefore, the audible frequency range is now produced by the loudspeaker system, which is consisted of more than two loudspeaker units. The multi-way loudspeaker system certainly requires a crossover filter, which is divided the audible frequency into low and high frequency ranges. The crossover frequency of filter usually is in the range of 1 ~ 4 kHz, the frequency range can largely affect to the sound articulation and quality. It is an example that the phase and amplitude at the crossover frequency can be different from each other, due to the variously reasons. Conversely, they can not produce the smooth frequency response due to the different distance between the two loudspeaker units and electrical properties. As a result, the sound articulation and quality can be degraded. Therefore, the phase and amplitude response at the crossover frequency has to be deeply considered, in order to exactly match the woofer and tweeter. In this study, it is proposed the methods which are the flat frequency and phase response to be obtained by adjusting of the delay time between loudspeaker units, and the choice of filter to be considered the roll-off properties of each unit. It is achieved the frequency response with ${\pm}1$ dB, and the sound articulation is also improved.

• The Journal of the Acoustical Society of Korea
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• v.31 no.8
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• pp.507-513
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• 2012

The room modes were showed as the peaks and dips on the low frequency response of the loudspeaker, and were one of the biggest obstacles to reproduce the sound. In order to reduce the occurrence of resonance, equalizing is performed at one position, and the frequency response to be equalized is valid only at that position. Performing the equalization would improve the flatness of the frequency response a little, but it cannot eliminate the ringing. Another method is that it is located the speaker at the specific position where the room resonances were not frequently happened. However, there may be the practical limitation that you may not be able to install the speaker at the position to be wanted. One of the practical solutions to decrease the production of resonance in room is the use of bass trap. In this paper, the practical limit for the room tuning using an equalizer and the room optimization will be discussed. The use of bass traps to solve the resonance problem that is always happened in the room is also proposed.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.159-166
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• 2005

This thesis deals with a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. First, the cross-talk cancellation method that eliminates cross-talks on the path from left speaker to right ear and from right speaker to left ear is explained. Then the application and replaying method using the cross-talk cancellation explained here is introduced. The acoustical model for cross-talk cancellation is the free field model This model minimizes distortion of sound. Many experts also make studies on this model. I used the bark scale sound quality compensation based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1199-1202
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• 1998

4채널 입력으로부터 입력되는 오디오를 압축,복원,저장, 전송하는 ㅅ스템을 설계한다. 이러한 시스템은 보안 시스템 중에서 특정 센서로부터 alarm 신호를 디지털 데이터로 변환한 후, 압축시켜 저장하고 동시에 압축된 오디오 데이터를 비디오 데이터와 통합하여 하나의 스트림으로 만들어 통신망으로 보내주는 시스템에 적용된다. 이러한 시스템의 구조를 간단히 설명하면 아날로그 음성 신호를 디지털 음성 data로 변환하기 위해 OKI사의 MSM 7570L-91이라는 ADPCM codec을 사용하였고 ADPCMcodec을 거쳐 나온 ADPCM 데이터를 64Mbyte SDRAM에 저장하였다가 FIFO를 거쳐서 통신망으로 전송을 한다. 복원은 SDRAM에 저장된 ADPCM 데이터를 MSM 7570L-01을 거쳐 아날로그 신호로 변환한 후 엠프를 거쳐 스피커로 출력을 하게 된다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.1
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• pp.199-205
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• 2015

Indoor positioning technology has been developed very actively for the smart phone handheld by most users. Especially, many TDOA positioning systems using acoustic signal have been studied, and it estimates the smart phone position by measuring the distance between the smart phone speaker and the microphones which is installed to receive the acoustic signal from the smart phone, and by calculating the hyperbolic equations. But there are always errors for the distance measurements, and furthermore the microphone installation error produces huge position estimation error. In this paper, the position estimation error due to the position error of acoustic sensor in the 3 dimensional TDOA positioning system, is analyzed by PDOP simulation and experiment.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.728-730
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• 2002

사무실 환경을 재현하는 챔버를 제작하고 사무 공간에서 발생 가능한 소음을 재현하는 음향시스템을 구축하였다. 소음원 확보를 위해 다양한 공간 조건하에서 발생하는 실제음을 녹취하였고, 이를 소음원 제시용 CD로 제작하였다. 사무실 환경 챔버 내의 다채널 스피커 음향 시스템을 구축한 후, 챔버의 공간 조건과 소음원의 컨벌루션을 통해 실제 사무실 발생음을 재현시켰다. 소음 환경 안에는 시간, 주파수 및 잔향 성분을 포함하였다.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.43 no.10 s.352
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• pp.68-73
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• 2006

This, paper proposes a cosimulation methodology that results in an efficient SoC design as well as fast verification by integrating HDL, SystemC, and algorithm-level abstraction using the design tools Active-HDL and Matlab's Simulink. To demonstrate the proposed design methodology, we implemented the design technique on a serial connection multi-channel speaker system. We have demonstrated the proposed cosimulation method utilizing an ARM processor based SoC Master board with the AMBA bus interface and a Xilinx Vertex4 FPGA. The proposed method has the advantage of simultaneous simulation verification of both software and hardware parts in high levels of abstraction mixed with some performance critical parts in more concrete RTL codes. This allows relatively fast and easy design of a speaker connection system which typically requires significant amount of data processing for verification.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.509-517
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• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.

• Journal of the Institute of Convergence Signal Processing
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• v.23 no.3
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• pp.137-142
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• 2022

Every year, the number of wild animals appearing in human settlements increases, resulting in increased damage to property and human life. In particular, the damage is more severe when wild animals appear on highways or farmhouses. To solve this problem, ecological pathways and guide fences are being installed on highways. In addition, in order to solve the problem in farms, horn repelling using sensors, installing a net, and repelling by smell of excrement are being used. However, these methods are expensive and their effectiveness is not high. In this paper, we used YOLO (You Only Look Once), an AI-based image analysis method, to analyze harmful animals in real time to reduce malfunctions, and high-brightness LEDs and ultrasonic frequency speakers were used as extermination devices. The speaker outputs an audible frequency that only animals can hear, increasing the efficiency to only exterminate wild animals. The proposed system is designed using a general-purpose board so that it can be installed economically, and the detection performance is higher than that of the devices using the existing sensor.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.297-297
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• 2010

이 논문에서는 음향 임피던스를 제어하는 능동형 소음 제어 방법을 연구하였다. 음향 임피던스 제어는 기존의 적응 제어 알고리즘 대신 임피던스의 크기 및 위상 최적화를 통하여 닫힌 관의 끝단에 음파의 반사를 최소화 하는 임피던스를 계산하고 제어함으로써 소음을 저감시킨다. 그리고 음향 임피던스 제어 방법을 끝단이 개방되어 있고, 중앙에 스피커가 있는 관에 적용시킨다. 이 경우에는 개방된 음장에서 음향 전달이 최소화하기 위한 임피던스를 계산하고, 음향 임피던스 제어를 통한 소음 저감을 확인한다. 이를 위해 스피커, 마이크로폰, 진동측정용 레이저 바이브로미터, 그리고 능동 소음 제어 하드웨어로 구성된 시스템을 구축하고 검증하였다.