• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1161-1164
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• 1998

We designed a DSP core simulator with C language, that is able to simulate 5-stage pipelined DSP core, named YS-DSP. It can emulate all 5 stage pipelines in the DSP core. It can also emulate memory access, exception processing, and DSP parallel processing. Each pipeline stage is implemented by combination of one or more functions to process parts of each stage. After modeling and validating the simulator, we can use it to verify and to complement the DSP core HDL model and to enhance its performance.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.137-140
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• 1995

In the paper, CDMA's vocoder algorithm, QCELP, was analyzed. And, 16-bit programmable DSP core for QCELP was designed. When it is used two MACs in DSP, we can implement low-power DSP and estimate decrease of parameter computation speed. Also, we implemented in FIFO memory using register file to increase the access time of the data. This DSP was designed using logic synthesis tool, COMPASS, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4E
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• pp.137-141
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• 2006

This paper presents HE-AAC decoder optimization on TMS320C55x fixed-point DSP core using a DSP-C like FFR code, which provides fast and flexible porting to a DSP core. Our optimization efforts are focused on methodologies that include general optimization methods of FFR code suitable for general DSP or RISC platform in high-level language and software optimization methods in assembly language level. The implementation result requires 48 MIPS and 135 Kbytes memory space to decode 48 Kbps stereo using real Korean S-DMB data.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.962-964
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• 2014

신호처리 시스템에서 FFT는 많이 사용되고 있으며, 고속화를 위하여 많은 연구가 진행되어 왔다. FFT은 통신, 영상처리, 레이더 등 많은 영역에서 직접 또는 변형되어 많이 활용되고 있으나 실시간 처리 속도 한계와 가격의 문제로 FFT 길이가 제한되는 경우가 많다. 본 연구에서는 TI사의 고속 DSP인 8 core의 TMS320C6678에 OpenMP 병렬처리 기법으로 FFT를 구현한 결과를 제시한다. 속도 개선을 위한 다양한 병렬처리 방안에 대하여 단일 FFT의 길이별 성능과 다중 FFT를 처리하기 위한 방안을 제안하였다. 이러한 OpenMP기반의 FFT는 DSP간 hyperlink 연결로 다수의 DSP로 병렬처리로 성능 개선이 가능하며, 본 연구에서는 16 core로 확장하여 그 성능이 30% 내외 개선되는 것을 보였다. 본 연구 결과는 초 고속 신호처리가 요구되는 의료영상, 초고해상도 영상처리, 고정밀 레이더 등에 활용이 가능할 것이다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.135-138
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• 2000

본 논문에서는 DSP Group사의 16 비트 고정 소수점 DSP인 OaKDSP Core를 사용하여 유럽의 이동통신에서 표준으로 사용되고 있는 음성 부호화기 알고리즘인 GSM-EFR (Global System for Mobile communications -Enhanced Full Rate) 을 실시간으로 구현하였다. 구현된GSM-EFR 음성 부호화기의 계산량은 약 24 MIPS가 소요되며, 7.06K 워드의 코드 메모리와 12.19K 워드의 데이터 메모리를 사용하였다. 구현된 음성 부호화기는 ETSI에서 제공하는 시험 벡터 샘플을 모두 통과하였으며, 객관적 평가 툴을 이용하여 지각 평가를 수행한 결과, 32kbps ADPCM과 비슷한 음질을 보였다. 본 논문에서 실시간으로 구현된 GSM-EFR 음성 부호화기는 IMT2000 비동기 방식의 음성 부호화기 표준인 GSM-AMR의 최상위 전송률 모드로서. 앞으로 IMT-2000 비동기식 단말기용 모뎀 ASIC에 탑재할 GSM-AMR 음성부호화기의 구현을 위한 기본 구조로 이용될 예정이다.

• Journal of the Korea Society of Computer and Information
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• v.7 no.4
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• pp.89-93
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• 2002

This paper describes an ALU core which is suitable for 32-bit DSP This ALU operates in 32-bit data and occupies the third stage, execution, among 5 stage pipeline structure. The supplied functions of the ALU are arithmetic operations, logical operations, shifting, and so on. For the implementation of this ALU core, each functional block is described by HDL. And the functional verification of the ALU core is performed through HDL simulation. This ALU is designed to use the 32-bit DSP.

• IEMEK Journal of Embedded Systems and Applications
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• v.17 no.3
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• pp.185-190
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• 2022

We propose a high speed data communication method for motor drive systems with fast control cycle in order to collect state variables of motor control without degrading control performance. Ethernet is chosen for communication device, and multi-core DSP architecture is exploited for communication processing load distribution. The communication program including network protocol stack and motor control program are assigned to two separate cores, and data between two cores are exchanged using interrupt-based inter-process communication mechanism, which enables to achieve a high-speed communication performance without degrading the motor control performance. The performance of developed communication method is demonstrated by real experiments using TCP, UDP and Raw Socket protocols in an experimental setup consisting of TI's TMS320F28388D motor control card and MS Windows PC.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.467-476
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• 1997

This paper presents a designing of a digital hearing aid processor (DHAP) chip being operated by a dedicated DSP core. The DHAP for hearing aid devices must be feasible within a size and power consumption required. Furthermore, it should be able to compensate for wide range of hearing losses and allow sufficient flexibility for the algorithm development. In this paper, a programmable 16-bit fixed-point DSP core is employed thor the designing of the DHAP. The designed DHAP performs a nonlinear loudness correction of 8 frequency bands based on audiometric measurements of impaired subjects. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the chip has low-power feature and $5, 500\times5000$$\mu$$m^2$ dimensions that fit for wearable hearing aids.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.190-195
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• 2005

The hearing aid chip described in this paper is an analog & digital mixed system. The design focuses on the$\mu$DSP core. This $\mu$DSP core includes internal time delays to two inputs from front and rear microphones. The paper consists of two parts; one is the composure and signal processing algorithm of digital hearing aids and the other is Verilog HDL codes for$\mu$DSP cores. All digital modules in the design were coded and synthesized by Verilog HDL codes which were verified by Mentor Graphics and Synopsis semiconductor chip design tools.