• Journal of Digital Contents Society
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• v.8 no.3
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• pp.341-346
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• 2007

The purpose of this paper is to propose the design of VADSM(Value-Added Digital Speaker Module) which tunes up the speaker unit by measuring the speaker's frequency responses and controlling EQ band. This module can reduce audible distortions at particular frequency band and improve some flatness in the speaker's frequency response. VADSM is composed of DSP AMP and speaker unit. When a speaker transforms electrical signal to sound, the magnitude response at some frequencies are more or less than normal level. So, DSP AMP can be used to adjust those magnitudes up or down by controlling its EQ bands.

• 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.

• Journal of Sensor Science and Technology
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• v.24 no.3
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• pp.169-174
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• 2015

In this study, we developed a system to detect the various defects in the metallic objects using the phenomenon that the defects cause the changes of the natural resonant frequencies. Our system consists of a FFT Amp, an Auto Impact Hammer, a Hammer controller and a PC. Auto Impact Hammer creates vibrations in the metallic objects when tapped on the surface. These vibrational signals are converted to the voltage signals by an acceleration sensor attached to the metallic part surface. These analog voltage signals were fed into an ADC (analog-digital converter) and an FFT (fast fourier transform) conversion in the FFT Amp to obtain the digital data in the frequency domain. Labview graphical program was used to process the digital data from th FFT amp to display the spectrum. We compared those spectra with the standard spectrum to find the shifts in the resonant frequencies of the metal parts, and thus detecting the defects. We used PCB's acceleration sensor and TI's TMS320F28335 DSP (digital signal processor) to obtain the resolution of 2.93 Hz and to analyze the frequencies up to 44 kHz.

• The Transactions of the Korean Institute of Power Electronics
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• v.9 no.6
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• pp.584-591
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• 2004

The conventional analog-based firing circuit can be implemented by comparing a linearly decreasing periodic sawtooth waveform synchronized to the ac supply, with a control signal corresponding to the desired converter delay angle. This circuit requires a large number of passive components (resistance and capacitor) and careful adjustment of the synchronization circuity. In this paper a novel firing circuit is proposed for thyristor switch. The proposed circuit is implemented by using digital components(FPGA, A/D, and DSP etc.) on the basis of the analog cosine method.

• Journal of Institute of Control, Robotics and Systems
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• v.7 no.2
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• pp.102-108
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• 2001

In this paper, an active noise controller for a communication headset was designed. In a communication headset, there exist information signals such as voices from the end for the communication line as well as also, undesirable noises with are induced by external noise sources such as engine noises. Therefore, it is necessary to reduce the external noises for clear hearing of the communication signals. This problem was solved by robust H(sub)$\infty$ controller to reduce noise and a compensator for information signals The designed controller was implemented using TMS320C31 DSP Op-amp, and several experiments were performed to verify its performance. The results showed that the controller reduces the undesirable noises sufficiently, while communication signals are not reduced.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.20 no.12
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• pp.1128-1138
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• 2010

One of the typical problems in the precise vibration is resonance characteristics at low frequency disturbance due to a heavy mass. An electro-magnetic(EM) air spring is a kind of vibration control unit and active isolator. The EM air spring in this study aims at removing the low frequency resonance for semiconductor manufacturing. The mechanical and electronic parts in the active isolator are designed to operate under a weight of 2.5 tons. The EM spring is floated using air pressure in a pneumatic elastic chamber and actuated by EM levitation force. The actuator consists of a EM coil and a permanent magnetic plate which are installed inside of the chamber. An air mount was constructed for the experiment with a stone surface plate, 4 active air springs, 4 gap sensors, a DSP controller, and a multi-channel power amp. A PD control method and operating logic was applied to the DSP. Simulation using 1/4 model was carried out and compared with the experiments. The time duration and maximum peak at resonance frequency can be reduced sharply by the proposed system. The results show that the active system can avoid the resonance caused by the natural frequency of the passive system.

• Journal of the Korean Solar Energy Society
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• v.33 no.6
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• pp.92-97
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• 2013

This paper proposes DC offset current compensation method of transformerless fuel cell/PV PCS. DC offset current is generated by the unbalanced internal resistance of the switching devices in full bridge topology. The other cause is the sensitivity of the current sensor, which is lower than DSP in resolution. If power converter system has these causes, the AC output current in the inverter will generate the DC offset. In case of transformerless grid-connected inverter system, DC offset current is fatal to grid-side, which results in saturating grid side transformer. Several simulation results show the difficulties of detecting DC offset current. Detecting DC offset current method consists of the differential amplifiers and PWM is compensated by the output of the Op amp circuit with integrator controller. PSIM simulation verifies that the proposed method is simpler and more effective than using low resolution current sensor alone.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.4
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• pp.569-574
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• 2007

최근 많은 광대역 유무선 통신 응용분야에서 OFDM(Orthogonal Frequency Division Multiplexing) 방식을 표준기술로 채택하고 있다. OFDM 방식의 고속 무선 데이터 통신을 위한 FFT 프로세서는 일반적으로 DSP(Digital Signal Processing)로 구현되었으나, 큰 전력 소비를 필요로 한다. 따라서, OFDM 통신방식의 단점인 전력문제를 보완하기 위해서 전류모드 FFT LSI가 제안되었고, 저소비전력 전류모드 FFT LSI를 동작시키기 위해서는 전류모드를 전압모드로 바꾸는 VIC(Voltage to Current Converter) 그리고 다시 전류모드를 전압모드로 바꾸어 주는 IVC(Current to Voltage Converter)가 필요하다. 그러나, OP-AMP로 구현되는 종래의 IVC는 회로규모가 크고, 전력소비가 크며, LSI 내에 크고 정확한 높은 저항을 필요로 한다. 또한 전류모드신호처리에서 많이 이용되는 Current Mirror 회로 등의 출력단자로부터 전류신호를 입력받은 경우, 입력단자간의 전위차가 발생하며, DC offset 전류가 발생하는 등의 문제점을 갖는다. 따라서 본 연구에서는 저전력 동작이 가능하고, 향후, single chip 응용이 가능한 IVC를 $0.35{\mu}m$ 공정에서 설계함으로서, $0.35{\mu}m$ 공정에서의 전류모드 FFT LSI의 전압모드 출력이 가능해졌다 설계된 IVC는 FFT LSI의 출력이 디지털신호로 환산한 ${\pm}1$인 점을 감안하여, 전류모드 FFT LSI의 출력이 $13.65{\mu}A$ 이상일 때에 3.0V의 전압을 출력하고, FFT LSI의 출력이 $0.15{\mu}A$ 이하일 때에 0.5V 이하의 전압을 출력하도록 하였으며, IVC의 총 소비전력은 약 1.65mV이하로 평가되었다.

• Journal of Biomedical Engineering Research
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• v.19 no.1
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• pp.59-68
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• 1998

Digital hearing aids offer many advantages over conventional analog hearing aids. With the advent of high speed digital signal processing chips, new digital techniques have been introduced to digital hearing aids. In addition, the evaluation of new ideas in hearing aids is necessarily accompanied by intensive subject-based clinical tests which requires much time and cost. In this paper, we present an objective method to evaluate and predict the performance of hearing aid systems without the help of such subject-based tests. In the hearing impairment simulation(HIS) algorithm, a sensorineural hearing impairment medel is established from auditory test data of the impaired subject being simulated. Also, the nonlinear behavior of the loudness recruitment is defined using hearing loss functions generated from the measurements. To transform the natural input sound into the impaired one, a frequency sampling filter is designed. The filter is continuously refreshed with the level-dependent frequency response function provided by the impairment model. To assess the performance, the HIS algorithm was implemented in real-time using a floating-point DSP. Signals processed with the real-time system were presented to normal subjects and their auditory data modified by the system was measured. The sensorineural hearing impairment was simulated and tested. The threshold of hearing and the speech discrimination tests exhibited the efficiency of the system in its use for the hearing impairment simulation. Using the HIS system we evaluated three typical hearing aid algorithms.