• Journal of the Korean Society for Precision Engineering
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• v.21 no.5
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• pp.91-98
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• 2004

Human uses level difference and time difference to get space information. Therefore this paper shows that method to presume direction of sound source by time difference and to mark presumed position. The position means direction from geometrical center of sensors to the sound source. To get the time difference of microphones input level, we will be explained about arrangement of microphones which used for the sensor to take the sound signal. It is included distance among the 3 microphones and distance between microphones and sound source. Secondly, input signals are transmitted to CPU througth digital process. CPU is used to DSP(Digital Signal Processor) for manage the signal by real time. Finally, the position of sound source is perceived by an explained algorithm in this paper.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.530-533
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• 2009

This paper proposes a method of recognizing that a person doesn't put on a safety cap using image processing method in DSP(Digital Signal Processor). It processes inputted images by image input devices that equipped in a industrial settings. If the method recognizes a person that doesn't put on a safety cap, a system transfers relevant recognition result to a supervisor and takes proper measures. If an accident happens and someone doesn't put on a safety cap, additional casualities could be. Proposed method can nip additional casualties in the bud. To recognize that a person don't put on a safety cap, images are processed by object abstraction, removal of noise, decision of a thing or a person, abstraction of a head part in a image, recognizing whether a man puts on a safety cap using HSV color space or not, and so on. Image input and image process are processed by DSP. And C language-based codes are optimized by an eignefunction(Intrinsics) for speed improvement of algorithms.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.2
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• pp.139-144
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• 2011

The modern digital logic technology does not yet satisfy the speed requirements of real-time DSP circuits due to synchronized operation of multiplication and accumulation. This operation degrades DSP performance. Therefore, the double-base number system (DBNS) has emerged in DSP system as an alternative methodology because of fast multiplication and hardware simplicity. In this paper, authors propose a novel multiplication architecture. One operand is an output of a flash analog-to-digital converter (ADC) in DBNS format, while the other operand is a coefficient in the IEEE standard floating-point number format. The DBNS digital output from ADC is produced through a new double base number encoder (DBNE). The multiplied output is in the format of the IEEE standard floating-point number (FPNS). The proposed circuits process multiplication and conversion together. Compared to a typical multiplier that uses the FPNS, the proposed multiplier also consumes 45% less gates, and 44% faster than the FPNS multiplier on Spartan-3 FPGA board. The design is verified with FIR filter applications.

• Journal of Korea Multimedia Society
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• v.13 no.9
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• pp.1322-1328
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• 2010

Recently, keyword spotting system is greatly in the limelight as UI(User Interface) technology of ubiquitous home network system. Keyword spotting system is vulnerable to non-stationary noises such as TV, radio, dialogue. Especially, speech recognition rate goes down drastically under the embedded DSP(Digital Signal Processor) environments because it is relatively low in the computational capability to process input speech in real-time. In this paper, we propose a new keyword spotting system using the call-command method, which is consisted of small number of recognition networks. We select the call-command such as 'narae', 'home manager' and compose the small network as a token which is consisted of silence with the noise and call commands to carry the real-time recognition continuously for input speeches.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.227-232
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• 2015

Nowadays, there are many innovative products in the pro-audio market thanks to advanced IT technology, DSP is very important technology to process high quality audio signal in SR(Sound Reinforcement) system. Digital audio technology that converged with IT technology can give new user-experience. In this paper, we present a new digital crossover system for active speakers using DSP and network technology. The prototype of crossover module consists of various audio process module such as filters, delay, phase controls and also it provides user to remote monitoring and remote control features by internet connection.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• Proceedings of the Korean Society of Computer Information Conference
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• 2016.01a
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• pp.321-322
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• 2016

본 논문에서는 수배전반에서 부하 설비 또는 외부의 영향을 확인할 수 있는 장치 및 보호설비가 설치되어 있으나 자체 사고를 신속히 검출하고 판단할 수 있는 기술의 확보를 위하여 해상도가 낮은 저가의 열화상 센서에 고속 DSP(Digital Signal Process)를 사용하여 영상 처리 기법인 이차원 보간법 기술을 이용하여 아크플래시에 의해 발생되는 열 특성을 검출하고 검출된 데이터를 전송하여 전기화재사고를 미연에 방지할 수 있는 장치 개발을 위한 기초 특성 연구를 수행하였다.

• The Proceedings of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.8 no.2
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• pp.41-45
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• 1994

In this paper, it is implemented that Sub- MRA PWM techinque which is applied to MRA PWM technique using the Digital Signal Processor. Unstable element of analog is reduced for Sub - MRA PWM technique by digital signal pressing. And harmonic is analized by simulation to verify that. It is afford the process induction motor control with real time by minimizing the delay time of digital system. Time delay which is a defect of digital control can by minimized using fast caculation. Therefore, real time control is implemented in the induction motor

• Transactions of the Korean Society of Machine Tool Engineers
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• v.11 no.5
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• pp.65-72
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• 2002

In this study, actuator, sensor, guide, power transmission element and control method are considered for ultra-precision positioning apparatus. Through previous process, single plane X-Y stage with ultra-precision positioning is manufactured. Global stage for the purpose of materialization with robust system, is combined by using AC servo motor and ball screw and rolling guide. And ultra-precision positioning system is developed by micro stage with elastic hinge type and piezo element. global servo and micro servo for the purpose of materialization positioning accuracy with nm(nanometer) are controlled simultaneously by using incremental encoder and laser interferometer as displacement measurement sensor. Through previous process, ultra-precision positioning system(100mm stroke and $\pm$ l0nm positioning accuracy) with single plane X-Y stage are materialized.

• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.187-190
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• 2000

In this paper, implementation of speech Recognizer system, Separated from Personal computer. By using DSP, this intends to extend the voice recognizing, limited into PC because of amount of data and calculations. For this performance The thesis uses the real time End point detector and organizes no additional device between human and the system, characteristic vector are that detects End point and voice from absolute energy and ZCR, that uses 12 difference Cepstrum from LPC, that uses the method to compensate the process of pattern separating and pre-calculated standard pattern limitation.