• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• Proceedings of the IEEK Conference
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• 2000.11b
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• pp.293-296
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• 2000

In this paper, we designed a 192-tap echo canceller for a modem using VHDL. To evaluate errors, we used adaptive algorithm. We adopted pipeline technique and realized delay-taps with RAMs. We simulated this design using Altera MAX+PLUS II.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.8
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• pp.2033-2042
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• 1996

In this papre, we describe a recently implemented echo canceller for digital cellular communication of Code Division Multiple Access(CDMA) that features time sharing of digital signal processor(DSP) over four channels in one DSP to reduce per channel costs. In the Public Land Mobile Network(PLMN), it is important to cancel the echo reflected from the Public Switched Telephone Network(PSTN) side. In case of digital mobile system, the round-trip delay of the echo is in excess of about 180 milliseconds due to frame-by-frame voice coding. It is necessary to cancel the echo in PLMN. We have developed a multi-channel echo canceller tht operates with Time Switch Module in a Mobile Switching Center(MSC). The general echo canceller needs PCM trunk interface circuits and the tone detection and disabling circuits, but the multi-channel echo canceller linked with Time Switch Module does not need them. Therefore we could develop the effective and economical echo canceller.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• Journal of Institute of Control, Robotics and Systems
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• v.5 no.7
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• pp.800-807
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• 1999

A Periodic disturbance canceller is proposed to compensate for the periodic disturbance due to cutting process in a CNC machining center. The periodic disturbance canceller estimates the Periodic disturbance without phase delay. This is achieved by using linear phase low-pass filter and frequency response reciprocal filter of plant at the frequency of the periodic disturbance. This method is implemented in the position control system of the CNC machining center with general disturbance compensators in order to compensate for both the frictional force and the periodic disturbance. The experimental results are described to show its effectiveness.

• 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.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.8 s.99
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• pp.825-834
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• 2005

In this paper, a new signal canceller that input signals are equally group-delayed and cancelled each other is proposed and feedforward linearizing power amplifier that adopt the proposed signal cancellers is fabricated. Although the conventional signal canceller can't matches the phase and the group delay time of input signals simultaneously, the proposed signal canceller matches those simultaneously. Simultaneous matching of the phase and the group delay time can makes wideband signal cancellation. The main signal cancellation loop of the fabricated feedforward amplifier with the proposed signal cancellers cancel input signal more than 26.3 dB and the intermodulation distortion signal cancellation loop cancel more than 15.2 dB for 200 MHz bandwidth. And the proposed feedforward power amplifier improves C/I ratio by 20.8 dB with two tones at 2,115 MHz, 2,165 MHz, respectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.2
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• pp.128-136
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• 2002

In this paper, we developed an acoustic echo canceller in real-time using TeakLite DSP Core, which will be placed in the vocoder chip of a mobile handset. Considering the limited computational capacity given to the acoustic echo canceller in a vocoder chip, we employed a FIR-type adaptive filter using a conventional NLMS algorithm. To begin with, we designed and implemented an acoustic echo canceller with floating-point format C-source code, and then converted it into fixed-point format through integer simulation. Then we programmed and optimized it in the assembler level to make it run ill real-time. After optimization procedure, the implemented echo canceller has approximately 624 words of program memory and 811 words of data memory. With 8 KHz sampling rate and 256 filter taps in the echo canceller that corresponds to 32 msec of echo delay, it requires 14.12 MIPS of computational capacity. For coverage of 16 msec echo delay, i.e., 128 filter taps, 9 MIPS is requited.

• Journal of Korea Multimedia Society
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• v.19 no.2
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• pp.189-199
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• 2016

Many implantable hearing aids are being developed as alternatives to conventional hearing aids which has inconveniences for use and social stigma that make hearing-impaired people avoid to wear it. Particularly, the fully-implantable middle ear hearing devices (F-IMEHD) are being actively studied for mixed or sensorineural hearing impaired people. In development of F-IMEHD, the most difficult problem is improving the performance of implantable microphone. Recently, Cho et al. have studied the tympanic membrane installed microphone which has better sensitivity and is easier to operate on patient than the microphone implanted under the skin. But, it may cause howling problem due to the feedback signal via oval window and ossicle chain from the transducer on round window in the middle ear cavity, therefore, a feedback canceller is necessary. In this paper, we designed NLMS (normalized least mean square) adaptive feedback canceller for F-IMEHD with tympanic membrane installed microphone and a transducer implemented at round window, and computer simulation was performed to verify its operation. The designed adaptive feedback canceller has a delay filter, a 64 point FIR fixed filter and a 8-tap adaptive FIR filter. Computer simulation of the feedback path is modeled by using the data obtained through human cadaver experiment.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.