• 전자공학회논문지 IE
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• v.44 no.4
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• pp.1-10
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• 2007

In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

• 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 Society of Digital Industry and Information Management
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• v.10 no.1
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• pp.113-120
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• 2014

In this paper, I present a new echo canceller using the adaptive cascade system identification (CSI) method, which a system response is divided into several responses so that each response is adaptively estimated and combined. Echo cancellation is required for a dual-duplex DSL, in order to allow each individual loop to operate in a full duplex fashion. Echo cancellation was one of the most difficult aspects of DSL design, requiring high linearity and total echo return loss in excess of 70 dB. Especially, for a fickle response, if the response is estimated by an adaptive filter, the filter needs more taps and the performance is decreased. But the response is divided into several responses, the computation complexities are decreased and the performance is increased. For the stage constant n, which represents the number of stages, if the response is not divided (n=1), the computation complexity of multiply is $2N^2$. And if the response is divided into two responses (n=2), the computation complexity of multiply is $2N^2$. Also, if n=3, the computation complexity is ${\frac{2}{3}}N^2$. Therefore, it is known that the computation complexity is decreased as n is increased. Finally, this proposed method is verified through simulation of echo canceller for digital subscriber line (DSL) application.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.15-20
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• 2003

For some applications such as teleconference or telecommunication systems using a distant-talking hands-free microphone, the near-end speech signals to be transmitted is disturbed by an ambient noise and by an echo which is due to the coupling between the microphone and the loudspeaker. Furthermore, the environmental noise including channel distortion or additive noise is assumed to affect the original input speech. In the present paper, a new approach using echo canceller and maximum a posteriori(MAP) estimation is introduced to improve the accuracy of hands-free speech recognition. In this approach, it was shown that the proposed system was effective for hands-free speech recognition in ambient noise environment including echo. The experimental results also showed that the combination system between echo canceller and MAP environmental adaptation technique were well adapted to echo and noise environment.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.9
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• pp.65-71
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• 2004

This paper describes a new adaptive algorithm which can reduce the required computation quantities in the adaptive filter. The proposed S-LMS algorithm uses only the signs of the normalized input signal rather than the input signals when coefficients of the filter are adapted. By doing so, there is no need for the multiplications and divisions which are mostly responsible for the computation quantities. To analyze the convergence characteristics of the proposed algorithm, the condition and speed of the convergence are derived mathematically. Also, we simulate an echo canceller adopting this algorithm and compare the performances of convergence for this algorithm with the ones for the other algorithm. As the results of simulations, it is proved that the echo canceller adopting this algorithm shows almost the same performances of convergence as the echo canceller adopting the SIA algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.876-883
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• 1996

Adaptive FIR filters are widely used in the echo canceller. But, most of practical systems have the transfer function composed of poles and zeros. In that case, adaptive IIR filters may be more efficient rather than FIR fiters. In this paper, a recently developed C-HARF algorithm is used to implement an adaptive IIR echo canceller. The proposed convergence of the algorithm make it attractive for this application. Extensive computer simulations show that C-HARF algorithm performs better than the NLMS algorithm after convergence, although C-HARF algorithm converges more slowly.

• MALSORI
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• no.62
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• pp.85-96
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• 2007

In this paper, we carried out recognition experiments for noisy speech having various levels of car noise and output of an audio system using the speech interface. The speech interface consists of three parts: pre-processing, acoustic echo canceller, post-processing. First, a high pass filter is employed as a pre-processing part to remove some engine noises. Then, an echo canceller implemented by using an FIR-type filter with an NLMS adaptive algorithm is used to remove the music or speech coming from the audio system in a car. As a last part, the MMSE-STSA based speech enhancement method is applied to the out of the echo canceller to remove the residual noise further. For recognition experiments, we generated test signals by adding music to the car noisy speech from Aurora 2 database. The HTK-based continuous HMM system is constructed for a recognition system. Experimental results show that the proposed speech interface is very promising for robust speech recognition in a noisy car environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.3
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• pp.355-359
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• 1993

This paper presents a derivation of the optimum convergence constant to yield the most rapid convergence under a desired mean-square error (MSE) for echo canceller using the LMS algorithm. For white input data, the optimum convergence constant is a simple closed-form function of the number of filter taps, the input signal variance, the initial MSE, and the desired MSE. This characteristic makes it easily designed in many practical applications. Computer simulations are also employed to show the correctness and effectiveness of the derived results.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.3
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• pp.10-17
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• 1995

this thesis proposes a new echo canceller that can be used in a fulll-duplex digital subscriber loopmodem. The modem suffers from nonlinear distortion such as transmitted pulse asymmetry, saturation in transformers, and nonlinearity of data converters. The proposed nonlinear echo canceller can compensate the nolinear distortion by using a nonlinear digital filter based on canonical pieceewise-linear (CPWL) function. Numerical results based on computer simulation are given in this paper. It is shown that the convergence characteristics depend on the initial values of weights of linear filters with absoluters and that the nonlinearity in digital-to-analog(D/A) converter can be compensated by a relatively small number linear filters with absoluters. It is also shown that the proposed algorithm has a faster convergence rate in comparison with Voterra algorithm.