• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.10
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• pp.1598-1605
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• 1993

In this paper, the performance degradation caused by the band pass filter and the data sampling timing error which are generally considered ideal for the simplicity was analyzed. The well known intersymbol interference theory can be applicable to calculate only the upper bound of a system performance. Therefore to obtain the average error probability, a method average intersymbol interference was proposed. This method can be applicable to obtain average error probablity with a computer simulation and with a measured value in test lab. easily.

• 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
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• v.17 no.4
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• pp.11-17
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• 1980

This paper describes realization of digital filter using $\mu$p controler and discusses measured characteristics of this filter, The idea of P. L. implementation[1] is used in realization, and in this system we utilize a DMA and arithmatic control program of $\mu$p. In this way, we can get more flexible capability than the basic PL method, and higher speed than a filter using general purpose $\mu$p in hardware, Furthermore, we get a 15 KHz sampling frequency(fs) as speed limit in real time processing, and know that this limitation is restricted by execution time (58$\mu$ sec) of DMA control statement. As for filter charateristics, maximum stop band frequencies (fsp) are 1665 Hz, 1445 Hz in Butterworth and Tchebichef approximation, respectively, when fs = 14 KHz. Measured errors between the design specification and the actual result are 0.2dB, 0.1 dB in pass band (when cufoff frequency is 500Hz),-1.1dB (when fsp is 1000Hz), 0.4dB(when fsp is 750 Hz) in stop band frequency of Butterworth and Tchebichef, respectively.

• The Journal of the Acoustical Society of Korea
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• v.42 no.5
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• pp.403-411
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• 2023

Functional ultrasound imaging, such as elasticity imaging and micro-blood flow Doppler imaging, enhances diagnostic capability by providing useful mechanical and functional information about tissues. However, the implementation of functional ultrasound imaging poses limitations such as the storage of vast amounts of data in Radio Frequency (RF) data acquisition and processing. In this paper, we propose a sub-Nyquist approach that reduces the amount of acquired axial samples for efficient shear-wave elasticity imaging. The proposed method acquires data at a sampling rate one-third lower than the conventional Nyquist sampling rate and tracks shear-wave signals through RF signals reconstructed using band-pass filtering-based interpolation. In this approach, the RF signal is assumed to have a fractional bandwidth of 67 %. To validate the approach, we reconstruct the shear-wave velocity images using shear-wave tracking data obtained by conventional and proposed approaches, and compare the group velocity, contrast-to-noise ratio, and structural similarity index measurement. We qualitatively and quantitatively demonstrate the potential of sub-Nyquist sampling-based shear-wave elasticity imaging, indicating that our approach could be practically useful in three-dimensional shear-wave elasticity imaging, where a massive amount of ultrasound data is required.

• Journal of IKEEE
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• v.21 no.4
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• pp.331-337
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• 2017

In this paper, we present the design of a 20MHz bandwidth 3rd-order continuous-time low-pass sigma-delta modulator with low-noise and low-power consumption. The bandwidth of the system is sufficient to accommodate LTE and other wireless network standards. The 3rd-order low-pass filter with feed-forward architecture achieves the low-power consumption as well as the low complexity. The system uses 3bit flash quantizer to provide fast data conversion. The current-steering DAC achieves low-power and improved sensitivity without additional circuitries. Cross-coupled transistors are adopted to reduce the current glitches. The proposed system achieves a peak SNDR of 65.9dB with 20MHz bandwidth and power consumption of 32.65mW. The in-band IM3 is simulated to be 69dBc with 600mVp-p two tone input tones. The circuit is designed in a 0.18-um CMOS technology and is driven by 500MHz sampling rate signal.

• Journal of Power Electronics
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• v.18 no.5
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• pp.1555-1566
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• 2018

Proportional capacitor-current-feedback active damping (AD) is a common damping method for the resonance of LCL-type grid-connected inverters. Proportional capacitor-current-feedback AD behaves as a virtual resistor in parallel with the capacitor. However, the existence of delay in the actual control system causes impedance in the virtual resistor. Impedance is manifested as negative resistance when the resonance frequency exceeds one-sixth of the sampling frequency ($f_s/6$). As a result, the damping effect disappears. To extend the system damping region, this study proposes a virtual resistor-inductor-capacitor (RLC) AD method. The method is implemented by feeding the filter capacitor current passing through a band-pass filter, which functions as a virtual RLC in parallel with the filter capacitor to achieve positive resistance in a wide resonance frequency range. A combination of Nyquist theory and system close-loop pole-zero diagrams is used for damping parameter design to obtain optimal damping parameters. An experiment is performed with a 10 kW grid-connected inverter. The effectiveness of the proposed AD method and the system's robustness against grid impedance variation are demonstrated.

• JSTS:Journal of Semiconductor Technology and Science
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• v.17 no.1
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• pp.129-140
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• 2017

In this paper, a $4{\times}32$-channel neural recording system capable of acquiring neural signals is introduced. Four 32-channel neural recording ICs, complex programmable logic devices (CPLDs), a micro controller unit (MCU) with USB interface, and a PC are used. Each neural recording IC, implemented in $0.18{\mu}m$ CMOS technology, includes 32 channels of analog front-ends (AFEs), a 32-to-1 analog multiplexer, and an analog-to-digital converter (ADC). The mid-band gain of the AFE is adjustable in four steps, and have a tunable bandwidth. The AFE has a mid-band gain of 54.5 dB to 65.7 dB and a bandwidth of 35.3 Hz to 5.8 kHz. The high-pass cutoff frequency of the AFE varies from 18.6 Hz to 154.7 Hz. The input-referred noise (IRN) of the AFE is $10.2{\mu}V_{rms}$. A high-resolution, low-power ADC with a high conversion speed achieves a signal-to-noise and distortion ratio (SNDR) of 50.63 dB and a spurious-free dynamic range (SFDR) of 63.88 dB, at a sampling-rate of 2.5 MS/s. The effectiveness of our neural recording system is validated in in-vivo recording of the primary somatosensory cortex of a rat.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.62-70
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• 2000

In this study, speech recognition for Korean digit is performed using filter bank which is programmed discrete HMM and DTW. Spectral analysis reveals speech signal features which are mainly due to the shape of the vocal tract. And spectral feature of speech are generally obtained as the exit of filter banks, which properly integrated a spectrum at defined frequency ranges. A set of 8 band pass filters is generally used since it simulates human ear processing. And defined frequency ranges are 320-330, 450-460, 640-650, 840-850, 900-1000, 1100-1200, 2000-2100, 3900-4000Hz and then sampled at 8kHz of sampling rate. Frame width is 20ms and period is 10ms. Accordingly, we found that the recognition rate of DTW is better than HMM for Korean digit speech in the experimental result. Recognition accuracy of Korean digit speech using filter bank is 93.3% for the 24th BPF, 89.1% for the 16th BPF and 88.9% for the 8th BPF of hardware realization of voice dialing system.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11C
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• pp.1085-1097
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• 2009

We present a structure and design method of the D8PSK modem compatible with the VDL mode-2 standard and performance test results of the developed modem. In VDL mode-2, the raised cosine filter is used only in the transmitter and a general low pass filter is used in the receiver. Consequently, we can not achieve ISI reduction but can have better spectrum characteristics. Although there is 1~2 dB performance degradation with an un-matched filter compared to that with a matched filter, it is more important to minimize adjacent channel interference in narrow band communications. The transmit signal is generated digitally to avoid the problems(I/Q imbalance and DC offset etc.) of analog modulators. In addition the digital down converter using digital IF sampling technique is adopted for the receiver. This paper contains the overall configuration, design method and simulation results based in part on the previously proposed structures and algorithms. It is confirmed that the modem transmits and receives messages successfully at a speed of max. 870 km/h over ranges of up to 310 km through the ground and in-flight communication tests.