• 음성과학
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• 제11권3호
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• pp.143-156
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• 2004

In this paper, we implemented a vocabulary-independent speech recognizer using the TMS320VC33 DSP. For this implementation, we had developed very small-sized recognition engine based on diphone sub-word unit, which is especially suited for embedded applications where the system resources are severely limited. The recognition accuracy of the developed recognizer with 1 mixture per state and 4 states per diphone is 94.5% when tested on frequently-used 2000 words set. The design of the hardware was focused on minimal use of parts, which results in reduced material cost. The finally developed hardware only includes a DSP, 512 Kword flash ROM and a voice codec. In porting the recognition engine to the DSP, we introduced several methods of using data and program memory efficiently and developed the versatile software protocol for host interface. Finally, we also made an evaluation board for testing the developed hardware recognition module.

• 한국산업융합학회 논문집
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• 제3권1호
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• pp.9-15
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• 2000

This paper presents a implementation of digital motion control system for induction motor vector drives using the 16bit DSP TMS320F240. The DSP controller enable enhanced real time algorithm and cost-effective design of intelligent controllers for motors which can be yield enhanced operation, fewer system components, lower system cost, increased efficiency and high performance. The system presented are speed and current sensing, sine look-up table and generated SVPWM by fully integrated control software. The developed system in a implementation are shown a good speed response and motion control characteristic results, and high performance features in general purposed 2.2[kW] machine. The system can be adapted variform motor drive system.

• 전력전자학회논문지
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• 제17권3호
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• pp.259-265
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• 2012

This paper introduces a method for design and implementation of a current controller for boost converter operating in continuous conduction mode (CCM) using a digital signal processor (DSP). A Proportional-Integral (PI) type current controller outputs an average voltage command for inductor, used in the input side of the boost converter, and the duty-ratio of PWM (pulse width modulation) signal for switching device is directly calculated from the average voltage command. The gains of the PI current controller are selected such that the current response characteristics are the same as those of a first-order low-pass filter. The proposed current control scheme is implemented using a DSP based on fixed-point math operations and an experimental study has been performed to validate the proposed method.

• 대한전기학회논문지:전기기기및에너지변환시스템부문B
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• 제52권12호
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• pp.608-615
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• 2003

This paper describes an implementation of a DSP (Digital Signal Processor) controlled random carrier frequency modulation for the PWM inverter for acoustic noise reduction of induction motors. Real-time generation of the random variable and RPWM(Random PWM) along with the speed control was achieved by DSP TMS320C31. The experimental results show that the voltage and current harmonics are spread to a wide band area and the power spectrum of the acoustic switching noise was spread to create a more appealing, less annoying sound. Also, the speed response of the implemented method and the conventional method is nearly similar to each other from the viewpoint of the v/f constant control.

• 전력전자학회:학술대회논문집
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• 전력전자학회 1999년도 전력전자학술대회 논문집
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• pp.377-381
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• 1999

This paper presents a implementation of digital motion control system for induction motor vector drives using the 16bit DSP TMS320F240. The DSP controller enable enhanced real time algorithm and cost-effective design of intelligent controllers for induction motors which can be yield enhanced operation, fewer system components, lower system cost, increased efficiency and high performance. The system presented are speed and current sensing, sine look-up table and generated SVPWM by fully integrated control software. The developed system in a implementation are shown a good speed response characteristic results and high performance features. The system can be adapted variform motor drive system.

• 대한음성학회지:말소리
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• 제65호
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• pp.143-152
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• 2008

This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

• 음성과학
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• 제6권
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• pp.219-233
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• 1999

This paper presents a real-time implementation of a speaker independent speech recognition system based on a discrete hidden markov model(DHMM). This system is developed for a car navigation system to design on-chip VLSI system of speech recognition which is used by fixed point Oak DSP core of DSP GROUP LTD. We analyze recognition procedure with C language to implement fixed point real-time algorithms. Based on the analyses, we improve the algorithms which are possible to operate in real-time, and can verify the recognition result at the same time as speech ends, by processing all recognition routines within a frame. A car noise is the colored noise concentrated heavily on the low frequency segment under 400 Hz. For the noise robust processing, the high pass filtering and the liftering on the distance measure of feature vectors are applied to the recognition system. Recognition experiments on the twelve isolated command words were performed. The recognition rates of the baseline recognizer were 98.68% in a stopping situation and 80.7% in a running situation. Using the noise processing methods, the recognition rates were enhanced to 89.04% in a running situation.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.257-258
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• 2006

In this paper, we address several issues in the real time implementation of MPEG-1 Layer II decoder on a fixed-point digital signal processor (DSP), especially TMS320C6416. There is a trade-off between processing speed and the size of program/data memory for the optimal implementation. In a view of the speed optimization, we first convert the floating point operations into fixed point ones with little degradation in audio quality, and then the look-up tables used for the inverse quantization of the audio codec are forced to be located into the internal memory of the DSP. And then, window functions and filter coefficients in the decoder are precalculated and stored as constant, which makes the decoder faster even larger memory size is required. It is shown from the real-time experiments that the fixed-point implementation enables us to make the decoder with a sampling rate of 48 kHz operate with 3 times faster than real-time on TMS320C6416 at a clock rate of 600 MHz.

• 한국통신학회논문지
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• 제16권1호
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• pp.85-100
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• 1991

디지틀 이동 통신 시스템 검토에 있어서 고음질 저전송 속도의 음성 부호화기 연구는 가용 주파수 대역의 제한을 극복하여 통신 서비스를 증대시키기 위한 필수 사항의 하나이다. 본 논문에서는 디지특 이동 통신용 13 kbps RPE LTP 음성 부호화기의 구현에 관하여 다루었다. 하나의 DSP칩을 이용하여 양방향 통신방식으로 실시간 구현(DSP 로딩율 약 75%)이 가능함을 나타냈으며, 또한 H/W 구현을 위한 고정소수점 시뮬레이션 및 채널코딩과의 연계를 고려한 각 전송 파리미티의 비트 중요도 분석 결과를 제시하였다.

• 한국음향학회지
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• 제23권1호
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• pp.17-23
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• 2004

일반적으로 무선 이동 채널은 협대역 스펙트럼을 갖는 복소 랜덤 프로세스로 모델링된다. 본 논문에서는 TMS320C6414 DSP칩과 XC2VP30 FPGA칩을 사용한 페이딩 신호의 실시간발생에 대해 기술한다. 여기서 제시된 시뮬레이터는 플랫 페이딩 및 주파수 선택성 페이딩 채널의 일반적인 모델로 그 대상으로 하였으며 여러 다양한 채널 왜곡 현상을 모사할 수 있도록 설계되었다. 시뮬레이터는 다음 세가지 기본적인 채널 특성을 포함한다; 레일리 페이딩, log-normal shadowing, 주파수 선택성 페이딩. DSP의 연산부담을 최소로 하면서 이동 채널의 통계 특성을 재현하기 위해 multi-rate신호처리 기법이 사용되었다. 채널 파라미터 값을 변화시키면서 많은 실험을 행하였고, 시뮬레이터에서 실시간 출력된 채널 포락선들의 각 통계 특성이 이론치와 일치됨을 확인했다.