• Title/Summary/Keyword: VLSI DSP

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Implementation of Recursive DSP Algorithms Based on an Optimal Multiprocessor Scheduler (최적 멀티프로세서 스케줄러를 이용한 재귀 DSP 알고리듬의 구현)

  • Kim Hyeong-Kyo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.10 no.2
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    • pp.228-234
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    • 2006
  • This paper describes a systematic process which can generate a complete circuit specification efficiently for a given recursive DSP algorithm based on an optimal multiprocessor scheduler. The process is composed of two states: scheduling and circuit synthesis. The scheduling part accepts a fully specified flow graph(FSFG) as an input, and generates an optimal synchronous multiprocessor schedule. Then the circuit synthesis part translates the modified schedule into a complete circuit diagram including a control specification. The circuit diagram can be applied to a silicon compiler for VLSI layout generation. This paper illustrates the whole process with an example of a second order Gray-Market lattice filter.

Design of the DSP for the FM Sound Synthesis (FM 합성방식을 이용한 악기음 합성용 DSP 설계)

  • Kwon, Min-Do;Jang, Ho-Keun;Kim, Jae-Yong;Park, Ju-Sung;Kim, Hyung-Soon;Yun, Pyung-Woo;Baek, Kwang-Ryul;Im, Chang-Hun
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.5
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    • pp.63-73
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    • 1995
  • The conventional acoustic sounds can be synthesized by Frequency Modulation which includes the variation of frequency, amplitude, and modulation index. In this paper the number of variable synthesis parameters are limited to easily implement the existing two carrier FM algorithm by hardware. The DSP(Digital Signal Processor), which is able to carry out the modified algorithm and synthesize 16 sounds at a time, is designed with $0.8{\mu}m$ standard sells. The DSP which can synthesize 2 sounds at a time is implemented by ASIC emulator to examine the sound quality of the designed DSP. Through the objective and subjective estimation, it is confirmed that the sounds of many instruments from the implemented DSP are very closed to their real sound. Finally the designed DSP is layouted and simulated by VLSI desgn tool. According to the simulation, the designed DSP has the sufficiently fast speed for synthesizing 16 sounds at a time.

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A Study on Cycle Based Simulator of a 32 bit floating point DSP (32비트 부동소수점 DSP의 Cycle Based Simulator에 관한 연구)

  • 우종식;양해용;안철홍;박주성
    • Journal of the Korean Institute of Telematics and Electronics C
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    • v.35C no.11
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    • pp.31-38
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    • 1998
  • This paper deals with CBS(Cycle Base Simulator) design of a 32 bit floating point DSP(Digital Signal Processor). The CBS has been developed for TMS320C30 compatible DSP and will be used to confirm the architecture, functions of sub-blocks, and control signals of the chip before the detailed logic design starts with VHDL. The outputs from CBS are used as important references at gate level design step because they give us control signals, output values of important blocks, values from internal buses and registers at each pipeline step, which are not available from the commercial simulator of DSP. In addition to core functions, it has various interfaces for efficient execution and convenient result display, CBS is verified through comparison with results from the commercial simulator for many application algorithms and its simulation speed is as fast as several tenth of that of logic simulation with VHDL. CBS in this work is for a specific DSP, but the concept may be applicable to other VLSI design.

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VLSI Design of MPEG-2 AAC Audio Decoder (MPEG-2 AAC 오디오 복호화기의 VLSI 설계)

  • 방경호;김준석;정남훈;이근섭;박영철;윤대희
    • Proceedings of the IEEK Conference
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    • 2000.09a
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    • pp.247-250
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    • 2000
  • 본 논문에서는 MPEG-2 AAC 오디오 복호화 시스템을 효율적으로 설계하고 구현하였다. 구현된 시스템은 2채널의 메인 프로필 MPEG-2 AAC 비트열을 실시간으로 복호화하고, 32, 44.1, 48kHz의 표본화 주파수를 지원하여, 표준안에서 제안하는 툴 중 커플링 채널을 제외한 모든 툴을 지원한다. 설계된 시스템은 허프만 복호화와 예측 과정을 수행하는 두개의 독립된 모듈과 Programmable DSP 코어의 혼합 구조(hybrid architecture)로 최적화된 구조를 갖는다.

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VLSI Design of MPEG-2 AAC Decoder (VLSI를 이용한 MPEG-2 AAC 복호화기 설계)

  • 이근섭;정남훈;방경호;윤대희
    • Proceedings of the IEEK Conference
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    • 1999.06a
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    • pp.1099-1102
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    • 1999
  • This paper presents a real-time MPEG-2 AAC decoding system, which can decode 2-channel main profile MPEG-2 AAC bitstream. The proposed system supports all decoding tools except for coupling channel tool, and provides sampling rates of 32, 44.1, 48 KHz. The system consists of a simple programmable DSP core and two hardwired logic modules that perform Huffman decoding and prediction for real-time implementation.

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Signal Processing in Medical Ultrasound B-mode Imaging (의료용 초음파 B-모드 영상을 위한 신호처리)

  • Song, Tai-Kyong
    • Journal of the Korean Society for Nondestructive Testing
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    • v.20 no.6
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    • pp.521-537
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    • 2000
  • Ultrasonic imaging is the most widely used modality among modern imaging device for medical diagnosis and the system performance has been improved dramatically since early 90's due to the rapid advances in DSP performance and VLSI technology that made it possible to employ more sophisticated algorithms. This paper describes "main stream" digital signal processing functions along with the associated implementation considerations in modern medical ultrasound imaging systems. Topics covered include signal processing methods for resolution improvement, ultrasound imaging system architectures, roles and necessity of the applications of DSP and VLSI technology in the development of the medical ultrasound imaging systems, and array signal processing techniques for ultrasound focusing.

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VLSI Architecture of a Recursive LMS Filter Based on a Cyclo-static Scheduler (Cyclo-static 스케줄러를 이용한 재귀형 LMS Filter의 VLSI 구조)

  • Kim, Hyeong-Kyo
    • Journal of the Institute of Convergence Signal Processing
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    • v.8 no.1
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    • pp.73-77
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    • 2007
  • In this paper, we propose a VLSI architecture of an LMS filter based on a Cyclo-static scheduler for fast computation of LMS filteing algorithm which is widely used in adptive filtering area. This process is composed of two steps: scheduling and circuit synthesis. The scheduling step accepts a fully specified flow graph(FSFG) as an input, and generates an optimal Cyclo-static schedule in the sense of the sampling rate, the number of processors, and the input-output delay. Then the generated schedule is transformed so that the number of communication edges between the processors. The circuit synthesis part translates the modified schedule into a complete circuit diagram by performing resource allocations. The VLSI layout generation can be performed easily by an existing silicon compiler.

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Evaluation of Bit-Pipelined Array Circuits for Datapath DSP Applications

  • Israsena, Pasin
    • Proceedings of the IEEK Conference
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    • 2002.07b
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    • pp.1280-1283
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    • 2002
  • This paper discusses issues in VLSI design and implementation of high performance datapath circuits. Of particular concern will he various types of multiplier and adder, which are fundamental to DSP operations. Performance comparison will be provided in terms of sampling speed, layout area, and in particular, power consumption, with techniques that may be applied to reduce power dissipation also suggested. As an example, a low power, high performance recursive filter achieved through bit-level pipelining technique is illustrated

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Design and Implementation of Modulator Channel Card and VLSI Chip for a Wideband CDMA Wireless Local Loop System (광대역 CDMA WLL 시스템을 위한 변조기 채널 카드 및 VLSI 칩 설계 및 구현)

  • 이재호;강석봉;조경록
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.10A
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    • pp.1571-1578
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    • 1999
  • In this paper, we present the Modulator Channel Card and VLSI chip for the Radio Transceiver Unit (RTU) of direct sequence code division multiple access (DS-CDMA) Wireless Local Loop (WLL) System. The Modulator Channel Card is designed and implemented using ASIC's, FPGA's and DSP's. The ASIC, compliance with Common Air Interface specification proposed by ETRI, has 40K gates which is designed to operate at 32MHz, and is fabricated using $0.6\mu\textrm{m}$ CMOS process. The ASIC carries out for I- or Q- phase data channel signal processing at a time, where each data channel processing consists of channel coding, block interleaving, scrambling, Walsh modulation, Pseudo-Noise (PN) spreading, and baseband filtering. The Modulator Channel Card has been integrated as a part of RTU of WLL system and is confirmed that it meets all functional and performance requirements.

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High-performance Digital Hearing Aid Processor Chip with Nonlinear Multiband Loudness Correction (비선형 다중채널 Loudness 교정을 위한 고성능 보청기 칩)

  • Park, Young-Cheol;Kim, Dong-Wook;Kim, Won-Ky;Park, Sang-Il
    • Proceedings of the KOSOMBE Conference
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    • v.1997 no.05
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    • pp.342-344
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    • 1997
  • Owing to technical advances in very large-scale integrated circuits (VLSI), high-speed digital signal processing (DSP) chips become fast enough to allow for real-time implementation of hearing aid algorithms in units small enough to be wearable. In this paper, we present a digital hearing aid processor (DHAP) chip built around a general-purpose 16-bit DSP core. The designed DHAP performs a nonlinear loudness correction of 8 octave frequency bands based on audiometric measurements. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the has a low power feature and $5.410\times5.720mm^2$ dimensions that fit for wearable devices.

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