• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.79-84
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• 2004

In this paper, the AEC(acoustic echo canceller) is designed and implemented using VHDL(VHSIC hardware description language). The designed Echo Canceller employs the pipeline and the master-slave structure, and is realized with FPGA. As an adaptive algorithm, the Normalized LMS algorithm is used. For the coefficient adjustment, the Stochastic Iteration Algorithm(SIA) which uses only current residual values is used and the number of registers are evidently reduced and convergence speed is also much improved comparing to existing methods by using EAB of FPGA for FIR filter structure of transceiver. The designed Echo Canceller is verified with the test board implemented for this paper. From the timing simulation echo signals at about 1500 sampling data are converged and ERLE is improved by about 42-dB.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2E
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• pp.14-20
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• 1997

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

• 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.

• 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.20 no.5
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• pp.28-34
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• 2001

In stereo acoustic echo canceller scheme, coefficients of adaptive filter converge very slowly or misconverge to real acoustic echo path in receiving room. This is due to cross-correlation in stereo signals. In this paper, a new preprocess algorithm is proposed to improve the performance of stereo AEC(acoustic echo canceller) without computational burden. The proposed algorithm reduces cross-correlation using Gram-Schmidt orthogonality principles and nonlinear filtering. Computer simulations demonstrate that this algorithm performs well compared to conventional ones. When the acoustic path of transmitting room is changed, stereo AEC using proposed algorithm is well performed.

• Speech Sciences
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• v.9 no.3
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• pp.17-24
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• 2002

The goal of this research is the real-time implementation of an AEC (Acoustic Echo Canceller) using the floating-point digital signal processor of TMS320C31. We employ an FIR-type adaptive filter with the conventional NLMS (Normalized Least Mean Square) algorithm for the adaptation of filter coefficients. We program and optimize the system in the assembler level to make it run in real-time. With 8 kHz sampling rate, the implemented AEC requires $46\;\mu$sec and $77\;\mu$sec computational time per sample for 128-and 256-tap filter, respectively. It corresponds to 37% and 62% of maximum computational ability of TMS320C31 DSP.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.42-46
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• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.261-268
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• 2003

Conventional acoustic echo cancellers using ES algorithm have simple structure and fast convergence speed compared with those using NLMS algorithm, but they are very weak to external noise because ES algorithm updates the adaptive filter taps based on average energy reduction rate of room impulse response in specific acoustical condition. To solve this problem, in this paper, a new update algorithm for acoustic echo canceller with stepsize matrix generator is proposed. A set of stepsizes is determined based on residual error energy which is estimated by two moving average operators, and applied to the echo canceller in matrix from, resulting in improved convergence speed. Simulations in various noise condition show that the proposed algorithm improves the robustness of acoustic echo canceller to external noise.

• 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 the Acoustical Society of Korea
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• v.21 no.3
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• pp.264-273
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• 2002

Adaptive filters cannot exactly estimate the echo path of the receiving room because of the cross-correlation of stereo signals. In this paper, a new pre-processing method reducing the cross-correlation without degradation of stereophony is proposed to enhance the performance of stereophonic acoustic echo canceller. To reduce the cross-correlation, absolutes of two orthogonal signals derived from each channel signals are added to original channel signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, the computation of pre-processing can be reduced. As results of simulations, it is shown that the performance of stereo acoustic echo canceller with the proposed pre-processing method is better than that of conventional ones.