• Title/Summary/Keyword: DSP(digital signal processing)

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Evaluation of Resolution Improvement Ability of a DSP Technique for Filter-Array-Based Spectrometers

  • Oliver, J.;Lee, Woong-Bi;Park, Sang-Jun;Lee, Heung-No
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.38C no.6
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    • pp.497-502
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    • 2013
  • In this paper, we aim to evaluate the performance of the digital signal processing (DSP) algorithm used in [8] in order to improve the resolution of spectrometers with fixed number of low-cost, non-ideal filters. In such spectrometers, the resolution is limited by the number of filters. We aim to demonstrate via new experiments that the resolution improvement by six times over the conventional limit is possible by using the DSP algorithm as claimed by [8].

Development Based on Signal Processing Platform for Automotive UWB Radar System (차량용 UWB 레이다를 위한 DSP 기반의 신호처리 모듈 플랫폼 개발)

  • Ju, Yeonghwan;Kim, Sang-Dong;Lee, Jong-Hun
    • IEMEK Journal of Embedded Systems and Applications
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    • v.6 no.5
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    • pp.319-325
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    • 2011
  • Recently, collision avoidance systems are under development to reduce the traffic accidents and driver comfort for automotive radar. Pulse radar can detect their range and velocities of moving vehicles using range gate and FFT(Fast Fourier Transform) of the doppler frequency. We designed the real time DSP(Digital Signal Processing) based automotive UWB(Ultra Wideband) radar, and implemented DSP to detect the range and velocity within 100ms for real time system of the automotive UWB radar. We also measured the range and velocity of a moving vehicle using designed automotive UWB radar in a real road environment.

An Implementation and Verification of Performance Monitor for Parallel Signal Processing System (병렬신호처리시스템을 위한 성능 모니터의 구현 및 검증)

  • Lee Won-Joo;Kim Hyo-Nam
    • Journal of the Korea Society of Computer and Information
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    • v.10 no.5 s.37
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    • pp.313-322
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    • 2005
  • In this paper, we implement and verify performance monitor for parallel signal processing system, using DSP Starter Kit(DSK) of which the basic Processor is TMS302C6711 chip. The key ideas of this performance monitor is, using Real Time Data Exchange(RTDX) for the Purpose of real-time data transfer and function of DSP/BIOS, the ability to measure the Performance measure like DSP workload, memory usage, and bridge traffic. In the simulation, FFT, 2D FFT, Matrix Multiplication, and Fir Filter, which are widely used DSP algorithms, have been employed. Using performance monitor and Code Composer Studio from Texas Instrument(Tl) , the result has been recorded according to different frequencies, data sizes, and buffer sizes for a single wave file. The accuracy of our performance monitor has been verified by comparing those recorded results.

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Signal Processing Module Development for specialized Underwater Acoustic Communication Modem (특성화된 수중음향통신모뎀을 위한 신호처리 모듈 구현)

  • Sun, Hee-Gab;Park, Young-Joon;Kim, Young-Kil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.4
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    • pp.810-816
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    • 2011
  • The existing underwater acoustic communications modem is developing based on commercial products, so it is possible on functionality and performance of underwater acoustic communications modem implementation. But the production of development system is required to single platform system implementation at the system's size and cost. So this paper designs and implements the signal processing module as an important role to single hardware platform production of underwater acoustic communication modem by using TigerSHARC DSP processor.

Implementation of the Audio CODEC for Digital Audio Broadcasting Service (디지털 오디오 방송 서비스를 위한 오디오 코덱의 구현)

  • 장대영;홍진우
    • Journal of Broadcast Engineering
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    • v.6 no.1
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    • pp.66-71
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    • 2001
  • This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

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Implementation of the AAC Audio CODEC for Digital Audio Broadcasting (디지털 오디오 방송을 위한 AAC 오디오 코덱 구현)

  • 장대영;홍진우
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2000.11b
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    • pp.43-48
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    • 2000
  • This paper introduces MPEG-2 AAC codec system fur digital audio broadcasting. This system consists of encoder and decoder, and this system provides MPEG-2 system multiplexing and demultiplexing functions. Four DSPs are adopted fur encoder and three DSPs fur decoder. Each DSP processes system control, I/O control, and audio signal processing, multiplexing and demultiplexing. This paper also discusses about some near future estimations related to DAB system and services. And at the end of this paper describes about future development plans.

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A Study on Embedded DSP Implementation of Keyword-Spotting System using Call-Command (호출 명령어 방식 핵심어 검출 시스템의 임베디드 DSP 구현에 관한 연구)

  • Song, Ki-Chang;Kang, Chul-Ho
    • 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.

Recognizing that a person doesn't put on a safety cap using DSP. (DSP(Digital signal proccesor)를 이용한 산업현장에서의 안전모 미착용 인식 기술)

  • Lee, Yong-Woog;Song, Kang-Suk;Jeong, Moo-Il;Lim, Chul-Hoo;Moon, Sung-Mo
    • 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.

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A bitwidth optimization algorithm for efficient hardware sharing (효율적인 하드웨어 공유를 위한 단어길이 최적화 알고리듬)

  • 최정일;전홍신;이정주;김문수;황선영
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.22 no.3
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    • pp.454-468
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    • 1997
  • This paper presents a bitwidth optimization algorithm for efficient hardware sharing in digital signal processing system. The proposed algorithm determines the fixed-point representation for each signal through bitwidth optimization to generate the hardware requiring less area. To reduce the operator area, the algorithm partitions the abstract operations in the design description into several groups, such that the operations in the same group can share an operator. The partitioning result are fed to a high-level synthesis system to generate the pipelined fixed-point datapaths. The proposed algorithm has been implemented in SODAS-DSP an automatic synthesis system for fixed-point DSP hardware. Accepting the models of DSP algorithms in schematics, the system automatically generates the fixed-point datapath and controller satisfying the design constraints in area, speed, and SNR(Signal-to-Noise Ratio). Experimental results show that the efficiency of the proposed algorithm by generates the area-efficient DSP hardwares satisfying performance constraints.

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A Study of Real-Time Implementation of Audio/Data Processor for Digital/Analog Dual mode Mobile Phone (디지탈/아날로그 겸용 이동통신 단말기를 위한 오디오/데이타 프로세서의 실시간 구현에 관한 연구)

  • Byun, Kyung-Jin;Kim, Jong-Jae;Han, Ki-Chun;Yoo, Hah-Young;Cha, Jin-Jong;Kim, Kyung-Su
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.2
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    • pp.80-88
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    • 1997
  • In this paper, the implementation of audio/data processor using ETRI DSP to support analog mode in digital/analog dual mode mobile phone is presented. Audio/data processor performs the wideband data processing, audio signal processing, demodulation function, and data rate conversion when it is operated in analog mode. These functions are programmed in assembly language, and then loaded to ETRI DSP together with vocoder program for the digital mode operation. This is a very efficient implementation of the dual mode cellular phone ASIC since the vocoder for the digital mode and audio/data processor for the analog mode are programmed together in the same hardware.

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