• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.425-433
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• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.57-63
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• 1997

This paper proposes a new stereophonic acoustic echo canceller to preven the performance degradation at the instant of far-end talker's movement or change in transmission room environment in stereophonic teleconferencing system. In stereophonic acoustic echo canceller, the filter coefficients of the adaptive filters for echo cancellation do not have unique solutions and not converge to their optimum values. Therefore, the change of the far-end transmission room environment leads to degradation of ERLE(Echo Return Loss Enhanceement). Moreover, their computational complexity increases as the number of the filter coefficient increases, and their adaptive filters converge very slowly. So the real-time implementation is very difficult. To overcome these problems we propose a pre-processor consisting of an adaptive filter for making pseudo stereophonic signal, and it results in improved performance at the instant of environment change at far-end side and reduction of the total complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3A
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• pp.327-334
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• 2008

Conventional on-channel-repeaters (OCRs) have a crucial problem that the power of a re-transmitted signal is highly limited by a feedback signal due to antenna coupling. The power limitation problem in OCRs has been solved by incorporating a demodulation-type feedback canceller which eliminates unwanted feedback signals by estimating a feedback channel. In applying the demodulation-type feedback canceller to T-DMB repeaters, there is a troublesome problem of unfrequent known pilot symbols, resulting in poor convergence performance of channel estimation. To solve this problem and enhance the accuracy of estimation, we propose a partitioning method of the Phase Reference Symbol (PRS) transformed in time domain. Since filter coefficients are updated every one partitioned subgroup, the number of updates is increased by the number of partitioned subgroups and thus the convergence speed is enhanced. The improved performance of feedback-channel estimation is directly connected with the feedback-cancellation performance. Simulation result shows that the feedback canceller incorporating the proposed partitioning method has a good performance in terms of residual feedback power.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.37-40
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• 2001

본 논문에서는 수렴속도 향상을 위해 시간영역의 적응알고리즘을 직교변환인 아다말(Hadamard)변환을 이용하여 적응알고리즘을 변환영역에서 수행하며, 변환영역에서 수렴성능 향상을 위해 가변스텝 사이즈를 갖는 적응알고리즘을 제안한다. 제안한 알고리즘을 일반적인 부엽제거기(Generalized Sidelobe Canceller GSC)에 적용하여 컴퓨터 시뮬레이션을 하였으며, 각 알고리즘들의 계산량, 수렴 성능을 이용하여 각각 비교, 분석하여 제안한 알고리즘의 성능이 우수함을 입증하였다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.9
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• pp.1444-1452
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• 1989

This paper describes the implementation of a real time fetal ECG monitoring system in which an adaptive multi-channel noise canceller is realized using the Texas Instruments TMS32020 progrmmmable ditital signal processor. An ECG signal from the electrode placed on the mother's abdomen and three ECGs from those on the chest are applied as the desired signal and the referened inputs, respectively, of the multi-channel filter. The coefficients of the filter are updated using the LMS algorithm such that the output of the multi-channel filter copies the maternal ECG embedded in the abdominal ECG. The enhanced fetal ECG is obtained by subtracting the filter output from the abdominal ECG, and the difference signal is recorded. Both off-line and on-line experimental results are presented to verify the effectiveness of the parameters for the digital signal processing algorithms and the prototype system.

• JSTS:Journal of Semiconductor Technology and Science
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• v.12 no.3
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• pp.278-285
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• 2012

A digital readout IC for capacitive sensors is presented. Digital capacitance readout circuits suffer from static capacitance of sensors, especially single-ended sensors, and require large passive elements to cancel such DC offset signal. For this reason, to maximize a dynamic range with a small die area, the proposed circuit features digital filters having a coarse and fine compensation steps. Moreover, by employing switched-capacitor circuit for the front-end, correlated double sampling (CDS) technique can be adopted to minimize low-frequency device noise. The proposed circuit targeted 8-kHz signal bandwidth and oversampling ratio (OSR) of 64, thus a $3^{rd}$-order ${\Delta}{\Sigma}$ modulator operating at 1 MH was used for pulse-density-modulated (PDM) output. The proposed IC was designed in a 0.18-${\mu}m$ CMOS mixed-mode process, and occupied $0.86{\times}1.33mm^2$. The measurement results shows suppressed DC power under about -30 dBFS with minimized device flicker noise.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.2
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• pp.61-70
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• 1984

This paper examines the array antenna theory, basic relations between the array size (aperture) and its beamwidth and resultant patterns. This paper also provides array antenna system design criteria, mainly maximizing the signal-to-noise ratio (SNR) and its corresponding optimum array structure and weight functions. Explicit new expressions for array performance are also illustrated in terms of the array output SNR. An example is provided for a 37-element planar array to explicitly illustrate the beam-forming and nulling operations of the array. Fundamentals of sidelobe canceller (SLC) systems have been discussed along with a derivation of new SLC equations for optimum weights.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.291-297
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• 2004

In this paper, a new structure for the AEC(Acoustic Echo Canceller) is proposed in which echo signal components that can be created in mobile communications is effectively eliminated. Block Data Flow Architecture is a parallel architecture that achieves high performance, high efficiency, high throughput, and almost linear speed up. The proposed architecture employs AEC and is implemented using the TMS320C6711 for real-time applications. The proposed AEC shows improved performance by eliminating echoes at 55ms delay path. Since the proposed AEC can also be implemented in Firmware, it is believed to effectively work on various types of echoes if it is applied on CDMA mobile devices. The TMS320C6711 shows much better performance comparing to previous DSPs. For experimental verifications, filtering operation using adaptive algorithm is performed on TMS320C6711 board and error signals resulted from computations are monitored on PC, and then the performance of the implemented AEC is verified through ERLE computation. According the results of simulation, good characteristic of 100dB are shown after 500 sampling data.

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.179-182
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• 2002

The development of noise robust speech processing systems is becoming increasingly important as speech technology is currently widely applied in real world applications. Recently, to resolve such a noise problem, adaptive noise canceller(ANC) is frequently used, which is based upon adaptive filters. The adaptive recursive filters perform better than adaptive non-recursive filters due to the added poles, but the stability may be severely threatened. But these problems of adaptive recursive filters was solved by ACHARF algorithm. This paper presents a method which combines speaker verification system with ANC(Adaptive Noise Canceller) using the ACHARF algorithm. In the front-end stage, ANC is adopted to suppress the additive noise imposed on the speech signal. The results show that the performance of speaker verification system becomes better than before.

• Proceedings of the Korea Multimedia Society Conference
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• 2001.11a
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• pp.46-50
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• 2001

Adaptive signal processing is used to acoustic echo canceller. adaptive noise canceller and adaptive algorithm among adaptive algorithms is mainly used because the structure is simple and computa LMS algorithm has trade-off between the converge speed and the steady state error. In this paper, step-size of adaptive algorithm is varied with orthogonality Principles of optimal filter to get fasts though small steady state error. Time varying step-size is determined proportional to the maximum vector of LMS algorithm. As results of simulations, the adaptive algorithm with proposed time-v compared with conventional ones.