• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.2
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• pp.95-102
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• 1996

This paper describes the realization of a rela-time adaptive acoustic echo canceller, which adopts a microprogramming method, for removing acoustical echoes in speakerphone systems using th eADSP-2101 microprocessor with a pipeline and modified harvard architecture. We apply the LMS (least mean square) algorithm to estimate the coefficients of a transversal FIR filter. For the acustic adaptive echo canceller, we propose a parallel operation programming to imrove algorithm execution speed and apply a nonlinear quantization to reduce the quantization error caused by large dynamic range of voice signal.

• 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.

• 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.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.

• The Journal of the Acoustical Society of Korea
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• v.34 no.3
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• pp.177-183
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• 2015

This paper proposes a robust nonlinear acoustic echo canceller (NAEC) which is effective for modeling the nonlinearity of a speaker module and the long acoustic echo path within a speech communication environment. The proposed NAEC utilizes a sigmoid pre-processor for modeling the speaker nonlinearity and a partitioned block frequnecy-domain adaptive filter for identifying the acoustic echo path with small delay. Simulation results confirmed that the proposed algorithm achieves excellent performance with much lower computational complexity than the previous NAEC.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.3-11
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• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP 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.

• 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.