• JSTS:Journal of Semiconductor Technology and Science
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• v.14 no.1
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• pp.109-116
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• 2014

This paper presents a high-efficiency digital class-D audio amplifier using a composite interpolation filter for portable audio devices. The proposed audio amplifier is composed of an interpolation filter, a delta-sigma modulator, and a class-D output stage. To reduce power consumption, the designed interpolation filter has an optimized composite structure that uses a direct-form symmetric and Lagrange FIR filters. Compared to the filters with homogeneous structures, the hardware cost and complexity are reduced by about half by the optimization. The coefficients of the digital delta-sigma modulator are also optimized for low power consumption. The class-D output stage has gate driver circuits to reduce shoot-through current. The implemented class-D audio amplifier exhibited a high efficiency of 87.8 % with an output power of 57 mW at a load impedance of $16{\Omega}$ and a power supply voltage of 1.8 V. An outstanding signal-to-noise ratio of 90 dB and a total harmonic distortion plus noise of 0.03 % are achieved for a single-tone input signal with a frequency of 1 kHz.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.35D no.10
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• pp.39-50
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• 1998

Wide-band sharp-transition filters are widely used in applications such as wireless CODEC design or medical systems. Since these filters suffer from large sensitivity and roundoff noise, large word-length is required for the VLSI implementation, which increases the hardware size and the power consumption of the chip. In this paper, a low-power implementation technique for digital filters with wide-band sharp-transition characteristics is proposed using CPL (Complementary Pass-Transistor Logic), LWDF (Lattice Wave Digital Filter) and a modified DIFIR (Decomposed & Interpolated FIR) algorithm. To reduce the short-circuit current component in CPL circuits due to threshold voltage reduction through the pass transistor, three different approaches can be used: cross-coupled PMOS latch, PMOS body biasing and weak PMOS latch. Of the three, the cross-coupled PMOS latch approach is the most realistic solution when the noise margin as well as the energy-delay product is considered. To optimize CPL transistor size with insight, the empirical formulas for the delay and energy consumption in the basic structure of CPL circuits were derived from the simulation results. In addition, the filter coefficients are encoded using CSD (Canonic Signed Digit) format and optimized by a coefficient quantization program. The hardware cost is minimized further by a modified DIFIR algorithm. Simulation result shows that the proposed method can achieve about 38% reductions in power consumption compared with the conventional method.

• Transactions of the Korean Society of Machine Tool Engineers
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• v.14 no.2
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• pp.96-103
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• 2005

The early detection of the bearing defects in rotating machinery is very important since the critical failure of bearing causes a machinery shutdown. However it is not easy to detect the vibration signal caused by the initial defects of bearing because of the high level of random noise. A signal processing technique, called the adaptive line enhancer(ALE) as one of adaptive filter, is used in this study. This technique is to eliminate random noise with little a prior knowledge of the noise and signal characteristics. Also we propose the optimal methods fir selecting the three main ALE parameters such as correlation length filter order and adaptation constant. Vibration signals f3r three abnormal bearings, including inner and outer raceways and ball defects, were acquired by Anderon(angular derivative of radius on) meter. The experimental results showed that ALE is very useful f3r detecting the bearing defective signals masked by random noise.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.549-560
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• 1989

In this paper, the method of determining the initial weighting vlaues fuctioning as a filter under the Directional Constrained Minimization of Power(DCMP) algorithm is presented. By analyzing the sideband beamformer with the Finite Impulse Response (FIR) filter concepts, the constraints of any desired directions are obtained and the initial weighing values with fast adaptation time are formulated from those constraints. By applying this proposed initial weighting values to the DCMP and the spatial averaging processor, the interference of a desired direction and the coherent noises are eliminated at the same time. The improvement of this method compared with the existing algorithm is confirmed by computer simulation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.8
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• pp.1805-1812
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• 2011

Bandpass sampling technique has an advantage that it uses lower sampling frequency than Nyquist criterion. But special care is required in choosing sampling frequency to avoid self-image overlapping in the first Nyquist region. Recently, the second-order BPS techniques which can suppress possible self-image by using an additional ADC and by employing digital signal processing have been proposed. This paper addresses a complex BPS based SDR front-end. Unlike general second-order BPS, it needs simple FIR filter to compensate delay in the second ADC. We show a method to find proper sampling frequencies to down convert RF signals selected by tunable RF filter operating in arbitrary frequency range.

• ETRI Journal
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• v.32 no.1
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• pp.33-43
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• 2010

The increasing demand for non-sinusoidal currents affects the quality of distribution networks. Harmonic detection is a crucial step in the cancellation of those components by active power filters. In this paper, the discrete cosine transform (DCT) is compared with different implementations based on Fourier transforms, demonstrating their equivalences and the advantages provided by the former. We demonstrate that the phase error in the presence of grid frequency deviations and the transient length are reduced by half in comparison to the discrete Fourier transform. A novel algorithm is developed to provide frequency adaptation to the DCT, taking advantage of its good features. The window width is adjusted in real time according to the actual value of the grid fundamental frequency by means of a phase-locked loop. A technique based on dithering is employed to overcome the limitation caused by the truncation of the window number of samples, so the frequency resolution is enhanced. The theoretical approach is verified by simulated and experimental results.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.1
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• pp.97-104
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• 1999

In this study we show a telemedicine system using a DSP, which gives fast and exact medical data such as the ECG signal of the external emergency patient and enables the patient to get temporary treatments under the direction of a doctor in the hospital. This transceiver, which is able to treat the real time transmission of dynamic medical data captured by the measuring instrument and bidirectional communication of voice signal, is implemented using DSB-SC as modulation and demodulation technique and digital filter of each terminal are implemented as FIR filter. The system designed with DSP in this study is very small and compact and it can k, furthermore, to support additional biomedical signals just by renewing the software.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6A
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• pp.341-350
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• 2003

In this paper, we evaluate the performance of a moving average (MA) channel estimation filter when space-time block coding transmit diversity (STBC-TD) is applied to the wideband direct sequence code division multiple access (WCDMA) forward link. And we present the infinite impulse response (IIR) filter scheme that can reduce the required memory buffer and the channel estimation delay time. This paper also compares the performance between MA filter scheme and IIR filter scheme in various Rayleigh fading channel environments through the bit error rate (BER) and the frame error rate (FER). Extensive computer simulation results show that transmission with STBC-TD provides a significant gain in performance over no transmit diversity technique, particularly at pedestrian speeds. If STBC-TD technique is employed in the channel estimator based on MA filter, it provides considerable performance gains against Rayleigh fading and reduces the optimum filter tap number. Consequently, the channel estimation delay time and the complexity of the receiver are reduced. In addition, the channel estimator based on IIR filter has the advantages such as little memory requirement and no delay time compared to the MA scheme. However, IIR filter coefficients is very sensitive to the mobile speed change and it exerts a serious influence upon the performance. For that reason, it is important to set uP the optimum IIR filter coefficients.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.22 no.5
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• pp.845-851
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• 2013

In this paper, a robust optimization approach is applied to the two-axes stage using a piezoelectric actuator for precise motion tracking. Robust control is based on LQG/LTR (linear quadratic Gaussian control with loop transfer recovery) control. Further, an LMI (linear matrix inequality) is used to find the optimal parameter in the loop transfer recovery step, instead of a trial and error method. A decoupler in the shape of FIR filter is added to reduce the coupling effect between the motions of the two axes, and hence, the feedback control loop is designed independently for each axis motion. The experimental result shows that the proposed control scheme can be applied effectively for motion control of the two-axes stage.