• The Magazine of the IEIE
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• v.19 no.6
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• pp.55-60
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• 1992

본 논문은 ISDN U-interface 회로 중에서 digital 부분의 설계에 대하여 기술하였다. 이 회로는 MMS43 code와 echo cancellation 방식을 사용하여 구현되었다. 회로 구성상 interface부분과 DSP부분으로 나누어 설계하였으며 gate-array ASIC을 이용하여 chip을 제작하였다. 공정은 1um CMOS 기술을 사용하였다.

• Speech Sciences
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• v.8 no.1
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• pp.159-165
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• 2001

In this paper, we make an experiment on speech recognition systems with barging-in and non-barging-in utterances. Barging-in capability, with which we can say voice commands while voice announcement is coming out, is one of the important elements for practical speech recognition systems. Barging-in capability can be realized by echo cancellation techniques based on the LMS (least-mean-square) algorithm. We use three kinds of telephone interface boards with barging-in capability, which are respectively made by Dialogic Company, Natural MicroSystems Company and Korea Telecom. Speech database was made using these three kinds of boards. We make a comparative recognition experiment with this speech database.

• Proceedings of the Korean Society of Computer Information Conference
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• 2012.07a
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• pp.345-346
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• 2012

다양한 모바일 디바이스에서는 스피커와 마이크로폰의 자체에서 발생하는 반향(Echo)에 대한 문제점이 제시되어 녹음할 때 생기는 에코를 제거하기 위한 알고리즘 연구의 필요성이 증대되고 있다. 따라서 본 논문은 적응 신호 처리에 기반한 음향 반향 제어 알고리즘을 안드로이드 플랫폼 기반의 JNI를 이용한 설계 방법을 제안한다. 적응 반향 제거 애플리케이션은 일반적인 LMS 적응 필터를 적용하여 사운드에서 반향을 제거하고 FIR 필터를 사용하여 사운드의 웨이브폼에 FIR 필터를 적용한다. 안드로이드 플랫폼 기반의 JNI(Java Native Interface)를 이용한 설계는 기존에 사용되는 소스 코드를 재활용하고 프로그래밍을 쉽게하도록 도와준다.

• Journal of Advanced Navigation Technology
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• v.6 no.4
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• pp.306-311
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• 2002

In this paper, we have proposed the Multi-Carrier (MC) CDMA system by adopting multi-interference cancellation techniques connecting an adaptive array antenna and a CCI canceller in cascade form. And, we have analyzed the BER performance of system in AWGN channel where MUI is a major interference degrading the performance of the proposed MC CDMA system. Also, we have evaluated the degree of performance improvement in the proposed MC CDMA system. As the result of analysis, a CCI canceller is better efficient than an adaptive array antenna in the performance improvement and that the performance improvement becomes large as the number of users increases when multi-interference cancellation techniques are used in cascade form. Also, we found that performance of $10^{-5}$ in a point of view BER could be achieved when $E_b/N_o$ is more than 10 dB.

• The Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.272-279
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• 2016

In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.12 no.1
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• pp.71-81
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• 1987

To fuly utilize the availability of the ISDN, it is very important to develop the ISDN user terminals which can provide various user services. In this paper, the basic concept of the ISDN user terminal is briefly studied and, based on this study, a stimulus type digital telephone which can be connected directly to the ISDN is designed. This digital telephone can provide the basic voice service and has general functions such as voice encoding / decoding(PCM is used), user-network signalling, digital tone supply, channel selection, key pad and atatus display. Echo cancellation method is adopted for the digital subscirber loop transmission and user-network signalling is implemented according to the LAPD(Link Access Procedure on D-channel) protocol recommended by the CCITT's recommendations I.440-I.441. The validity of designed S/W and H/W functions are verified by testing them with ISDN circuit switching emulator described in the first part of this paper.

• Journal of Power Electronics
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• v.14 no.3
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• pp.444-457
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• 2014

This paper presents the design and realization of a digital PV simulator with a Push-Pull Forward (PPF) circuit based on the principle of modular hardware and configurable software. A PPF circuit is chosen as the main circuit to restrain the magnetic biasing of the core for a DC-DC converter and to reduce the spike of the turn-off voltage across every switch. Control and I/O interface based on a personal computer (PC) and multifunction data acquisition card, can conveniently achieve the data acquisition and configuration of the control algorithm and interface due to the abundant software resources of computers. In addition, the control program developed in Matlab/Simulink can conveniently construct and adjust both the models and parameters. It can also run in real-time under the external mode of Simulink by loading the modules of the Real-Time Windows Target. The mathematic models of the Push-Pull Forward circuit and the digital PV simulator are established in this paper by the state-space averaging method. The pole-zero cancellation technique is employed and then its controller parameters are systematically designed based on the performance analysis of the root loci of the closed current loop with $k_i$ and $R_L$ as variables. A fuzzy PI controller based on the Takagi-Sugeno fuzzy model is applied to regulate the controller parameters self-adaptively according to the change of $R_L$ and the operating point of the PV simulator to match the controller parameters with $R_L$. The stationary and dynamic performances of the PV simulator are tested by experiments, and the experimental results show that the PV simulator has the merits of a wide effective working range, high steady-state accuracy and good dynamic performances.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.945-951
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• 2007

Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.3
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• pp.193-203
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• 1988

In this paper, an echo canceller(EC) using decision feedback at the ISDN U-interface is presented and its performance based on the stochastic iteration algorithm is analyzed, and compared with the other conventional EC. The steady state mean-squared error(MSE) by the analytical resutls on the decision feedback-EC turns out to be smaller than that of the other linear EC. The performance of the ECs with the same convergence factor are almost the same regardless of different channel characteristics.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.2
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• pp.280-286
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• 2013

In this paper, a noise reduction system is proposed for the speech interface in intelligent TV applications. To reduce TV speaker sound which are very serious noises degrading recognition performance, a noise reduction algorithm utilizing the direct TV sound as the reference noise input is implemented. In the proposed algorithm, transfer functions are estimated to compensate for the difference between the direct TV sound and that recorded with the microphone installed on the TV frame. Then, the noise power spectrum in the received signal is calculated to perform Wiener filter-based noise cancellation. Additionally, a postprocessing step is applied to reduce remaining noises. Experimental results show that the proposed algorithm shows 88% recognition rate for isolated Korean words at 5 dB input SNR.