• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.1
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• pp.66-69
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• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• The KIPS Transactions:PartB
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• v.11B no.3
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• pp.345-352
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• 2004

This paper deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. 1 used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound qualify assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the tort chamber which satisfies the ITU-R specifications. 1 uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4∼0.5 point Increased then before.

• Transactions of the Korean Society of Automotive Engineers
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• v.1 no.3
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• pp.55-62
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• 1993

Autoregressive moving average(ARMA) model which is a time domain parametric modeling method is implemented for modeling and reproducing characteristics of exhaust noise of an automobile in various RPM range. Experiments have been carried out using 9 set of exhaust noise signals measured at 1,000-3,000 RPM range. Characteristics of sampled signals were estimated using ARMA modeling and Akaike's FPE(final prediction error) criterion to define exact model structure and for model validation. The digital filter consisted of the esitmated ARMA(70,1) model parameters was programed to reproduce exhaust noise. The spectral analysis of reproduced noise is very close to original. The results show that our approaching technique for reproducing acoustic characteristics is valid and feasible to apply in the field of noise quality control.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.4
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• pp.645-651
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• 2009

In this paper, we present bit error rate(BER) performance of an adaptive decision feedback equalizer(DFE) using experimental data. The experiment was performed at the shore of Geoje in November 2007. The BER of the adaptive DFE whose tap weight is updated by RLS is described with change of feedforward filter length, feedback filter length, training sequence length, and delay, which shows that the uncoded average BER is $4{\times}10^2\;and\;1.5{\times}10^{-2}$ with transmission range of 9.7km and 4km, respectively. The BER of the adaptive DFE can be lower than 10-3 by a forward error correction code and therefore the adaptive DFE may be a good candidate for a high speed AUV communications since the volume and weight of the underwater acoustic modem should be small because of the restricted space and power in the battery-operated AUV.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.8
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• pp.1765-1772
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• 2015

In the underwater acoustic communication channels, multipath reflection become the cause of obstacle. Generally, equalizer has been applied to overcome these problems. In this paper, the method was proposed to optimize tap-length of decision feedback equalizer using genetic algorithm. After inputting feed-forward filter length and feed-back filter length as genetic information of the genetic algorithm, it optimize tap-length using BER(bit error rate) calculation in accordance with object function. The object function consist of decision feedback equalizer and BER calculation. For the purpose of BER calculation in the object function, the method was proposed to optimize the tap-length of decision feedback equalizer with genetic algorithm using preamble signals. As a result of experiments, the optimized BER is 0.0355 for signals which were received through a 25m receiver and which were applied to calculate BER merely using preamble signals in object function. When all data were used to calculate BER in object function, the optimized BER is 0.0215.

• Transactions of the Korean Society of Mechanical Engineers
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• v.15 no.3
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• pp.809-823
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• 1991

본 연구에서는 Wiener 필터링 이론에 의하여 소음원의 입력신호에 대한 최적 한 예측을 할 수 있는 최적예측기(optimal predictor)로써 부가적인 음을 발생시키고 입력신호 및 출력신호 간의 차인 오차를 최소화시키도록 하는 적응신호처리방법에 대 해 설명하고 이러한 적응 신호처리 방법을 이용한 능동 소음 제어 방법을 제시하였다. 이와 아울러 제어계의 환경 변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해 야 하는 적응 소음 제어계에서, 음향궤환과 함께 필히 고려해야하는 부가적인 전달함 수-모델과 스피커를 포함하는 보조경로 및 오차미이크로폰을 포함하는 오차경로의 전 달함수의 영향을 고려한 능동소음제어에 대해 연구하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.7
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• pp.1321-1325
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• 2001

In this article, piezoelectric films and their application for film bulk acoustic resonator (FBAR) devices are presented. The FBAR is composed of piezoelectric film sandwiched between top and bottom electrodes and an acoustic reflector of SiO2/W stacked multilayers. Various FBAR devices were fabricated and evaluated through simulation and measurement. The insertion loss, return loss and Q-factor were observed to be reasonably high and good. The FBAR technology seems very promising particularly for RF band filter application.

• The Journal of the Acoustical Society of Korea
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• v.31 no.8
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• pp.532-541
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• 2012

An underwater transient signal is distinguished from an ambient noise. Database for the underwater transient signal is required since the underwater transient signal shows various characteristics depending on acoustic features. In the paper, hence, sound mask-filter was applied to extract the transient signals which exist temporally and locally in the ocean. The standard signal was chosen and cross-correlated with the raw signal. A mask-filter for a transient signal was obtained using the threshold which was decided by the maximum likelihood method in the envelope of the cross-correlated signal. Using the sound mask-filter, the transient signal of a sea catfish {Galeichthys felis (Linnaeus)} was extracted from the underwater ambient noise. Similarly, the man-made signal was added into the noise and it was extracted by the same method. We also have demonstrated the significance of the transient signal through comparing the extracted signals depending on the standard signal. In the results, the proposed method, sound mask-filtering, could be utilized as a database construction of the transient signals in underwater noise. Particularly, this study would be useful to extract the wanted signal from arbitrary signals.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.2
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• pp.250-254
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• 2003

A film bulk acoustic resonator (FBAR) device for 2 GHz radio frequency (RF) bandpass filter application is presented. This FBAR device consists of an aluminum nitride (AlN) film sandwiched between top(Al) and bottom(Au) electrodes and an acoustic multilayer reflector of a silicon dioxide/tungsten (SiO2/W). The A/N film deposited using a RF sputtering was observed to have small columnar grains with a strongly preferred orientation towards c axis. In addition to a high quality factor (4300), a large return loss of 37.19 dB was obtained.