• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.14-22
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• 1998

본 연구에서는 Texas instruments 사의 저가형 부동소수점 디지털 신호 처리기인 TMS320C32를 이용하여 실시간 화자종속 음성인식 하드웨어 모듈을 개발하였다. 하드웨어 모듈의구성은 40MHz 의 TMS320C32, 14bit 코덱인 TLC32044, EPROM 과 SRAM 등의 메모리와 호스트 인터페이스를 위한 로직회로로 이루어져 있다. 뿐만 아니라 이 하드웨어 모듈을 PC 상에서 평가해보기 위한 PC 인터페이스용 보드 및 소프트웨어도 개발하였다. 음성인식 알고리즘은 C 및 어셈블리를 이용한 최적화를 통하여 계산속도를 대폭 개선하였다. 현재 인식률은 일반 사무실 환경에서 30단어에 대하여 95% 이상으로 매우 높은 편이며, 특히 배경음악이나 자동차 소음과 같은 잡음환경에서도 잘 동작한다.

• Speech Sciences
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• v.14 no.4
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• pp.157-167
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• 2007

In this paper, we implemented a single-chip speech recognizer using the TMS320C2000 DSPs. For this implementation, we had developed very small-sized speaker-dependent recognition engine based on dynamic time warping, which is especially suited for embedded systems where the system resources are severely limited. We carried out some optimizations including speed optimization by programming time-critical functions in assembly language, and code size optimization and effective memory allocation. For the TMS320F2801 DSP which has 12Kbyte SRAM and 32Kbyte flash ROM, the recognizer developed can recognize 10 commands. For the TMS320F2808 DSP which has 36Kbyte SRAM and 128Kbyte flash ROM, it has additional capability of outputting the speech sound corresponding to the recognition result. The speech sounds for response, which are captured when the user trains commands, are encoded using ADPCM and saved on flash ROM. The single-chip recognizer needs few parts except for a DSP itself and an OP amp for amplifying microphone output and anti-aliasing. Therefore, this recognizer may play a similar role to dedicated speech recognition chips.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4
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• pp.14-22
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• 1998

본 연구에서는 Texas Instruments 사의 저가형 부동소수점 디지털 신호 처리기 (Digital Singnal Processor, DSP)인 TMS320C32를 이용하여 실시간 화자종속 음성인식 하 드웨어 모듈(VR32)을 개발하였다. 하드웨어 모듈의 구성은 40MHz의 TMS320C32 DSP, 14bit 코덱인 TLC32044(또는 8bit μ-law PCM 코덱), EPROM과 SRAM 등의 메모리와 호 스트 인터페이스를 위한 로직 회로로 이루어졌다. 뿐만 아니라 이 하드웨어 모듈을 PC사에 서 평가해보기 위한 PC 인터페이스용 보드 및 소프트웨어도 개발하였다. 음성인식 알고리 즘의 구성은 에너지와 ZCR을 기반으로 한 끝점검출(Endpoint Detection) 침 10차 가중 LPC 켑스터럼(Weighted LPC Cepstrum) 분석이 실시간으로 이루어지며 이후 Dynamic Time Warping(DTW)를 통하여 최고 유사 단어를 결정하고 다시 검증과정을 거쳐 최종 인식을 수행한다. 끝점검출의 경우 적응 문턱값(Adaptive threshold)을 이용하여 잡음에 강인한 끝 점검출이 가능하며 DTW 알고리즘의 경우 C 및 어셈블리를 이용한 최적화를 통하여 계산 속도를 대폭 개선하였다. 현재 인식률은 일반 사무실 환경에서 통상 단축다이얼 용도로 사 용할 수 있는 30 단어에 대하여 95% 이상으로 매우 높은 편이며, 특히 배경음악이나 자동 차 소음과 같은 잡음환경에서도 잘 동작한다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.19-23
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• 2001

DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

• Journal of Industrial Technology
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• v.21 no.B
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• pp.107-116
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• 2001

The speech includes various kinds of information : language information, speaker's information, affectivity, hygienic condition, utterance environment etc. when a person communicates with others. All technologies to utilize in real life processing this speech are called the speech technology. The speech technology contains speaker's information that among them and it includes a speech which is known as a speaker recognition. DTW(Dynamic Time Warping) is the speaker recognition technology that seeks the pattern of standard speech signal and the similarity degree in an inputted speech signal using dynamic programming. ln this study, using TMS320C32 DSP processor, we are to embody this DTW and to construct a security system.

• Proceedings of the KIEE Conference
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• 2009.07a
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• pp.2193_2194
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• 2009

A digital over current relay(OCR) is realized by using TMS320C32 microprocessor. Fourier Transform is used to obtain the phasor of a current signal and a 2nd low pass filter is adopted to prevent aliasing error. H/W test shows almost same results with those of the S/W test. It could be possible that confirm simularity between H/W and S/W test in this paper.

• Proceedings of the KIEE Conference
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• 2003.07d
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• pp.2218-2220
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• 2003

현대 산업현장에서 80%이상 사용되는 PID(proportional integral derivative) 제어기는 제어성, 적응성, 제어이득 조정 등의 특정이 있어나 제어 대상에 대한 PID 제어 계수를 경험적 방법인 수동으로 동조해야하는 문제점이 있다. 이를 개선하기 위해 본 논문에서 PID제어기의 최적 자동동조를 위한 릴레이 동조 방법을 제안한다. 기존의 한계감도법과 과도응답 법으로 초기계수를 결정하는 방법보다 유연성과 적용성이 높고, 이를 마이크로프로세서(DSP : TMS-320C32)에 적용하여 소프트웨어적으로 릴레이의 기능이 이루어지도록 설계했다. 이는 Ziegler-Nichols 계수조정법이 갖는 적용대상의 제약성을 극복한 방법이며, 릴레이에 의해 출력을 강제 진동시키고 출력의 진폭과 주기를 이용하여 PID 계수를 조정하고, 또 상대안정성의 척도인 위상여유를 고려하므로 시스템의 상대안정성과 견실성을 향상시킬 수 있음을 확인하였다.

• Journal of the Semiconductor & Display Technology
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• v.3 no.1
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• pp.21-26
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• 2004

Recently with the development of power switching device and DSP which has peripheral devices to control AC servo system, the servo technology has met a new development opportunity. Those things make it possible to reduce the time of developing a AC servo system. Fixed point DSP such as TMS320F240x, and TMS320F28x series have a disadvantage in calculating floating number where TMS320C32 or TMS320C31 are floating point DSP. However they usually become a complex hardware system to implement the AC servo system and it increases the cost. In this study, a DSP based AC servo system with a 3-phase PMSM is proposed. The newly produced DSP TMX320F28l2-version C which has the performance of fast speed, 150MIPS, and a rich peripheral interface such as a 12bit high speed AD converter, QEP(Quadrature Encoder Pulse) circuit, PDPINT(Power Drive Protect Interrupt), SVPWM module and dead time module are used. This paper presents a method to overcome fixed point calculating using scaling all parameters. Also space vector pulse width modulation (SVPWM) using off-set voltage and a digital PI control are implemented to the servo system.

• Speech Sciences
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• v.8 no.2
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• pp.73-81
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• 2001

In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

• Proceedings of the KIEE Conference
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• 2004.07e
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• pp.120-122
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• 2004

This paper describes the digital relay simulator by use of DSP for digital relay test. The simulator software has EMTP simulation data file conversion. user define simulation data generation. data analysis engine. etc. The simulator hardware design uses 32bit floating point DSP architecture to achivev flexibility and high speed operation.