Double-Talk-Robust Prediction Error Identification Algorithms for Acoustic Echo Cancellation

The performance of an acoustic echo canceller may be severely degraded by the presence of a near-end signal. In such a double-talk situation, the variance of the echo path estimate typically increases, resulting in slow convergence or even divergence of the adaptive filter. This problem is usually tackled by equipping the echo canceller with a double-talk detector that freezes adaptation during near-end activity. Nevertheless, there is a need for more robust adaptive algorithms since the adaptive filter's convergence may be affected considerably in the time interval needed to detect double-talk. Moreover, in some applications, near-end noise may be continuously present and then the use of a double-talk detector becomes futile. Robustness to double-talk may be established by taking into account the near-end signal characteristics, which are, however, unknown and time varying. In this paper, we show how concurrent estimation of the echo path and an autoregressive near-end signal model can be performed using prediction error (PE) identification techniques. We develop a general recursive prediction error (RPE) identification algorithm and compare it to three existing algorithms from adaptive feedback cancellation. The potential benefit of the algorithms in a double-talk situation is illustrated by means of computer simulations. It appears that especially in the stochastic gradient case a huge improvement in convergence behavior can be obtained     

子带仿射投影及子带双端检测算法的回声消除系统

文中提出了一种新的多相位子带自适应回声消除系统。在子带内进行自适应滤波对建模长度比较长的脉冲响应特别有效,同时由于仿射投影算法具有预白化的作用,它同样也具有改善滤波器收敛性能的功能。该系统集中了多相子带自适应滤波和仿射投影算法的优点,结合了子带内的双端检测算法,使得系统在临界采样的情况下能进行稳定有效的工作。实验表明：该系统对于语音信号和强相关信号都表现出了良好的性能。     

Frequency-domain adaptive filtering with applications to acoustic echo cancellation

In this paper an algorithm is presented for adaptive filtering in the frequency-domain with application to acoustic echo cancellation. This algorithm, called generalized multi-delay filter(GMDFα, is derived from the frequency-domain implementation of the time-domain block least mean square algorithm. Two different implementations are introduced, one based on the discrete Fourier transform(dft) and one based on the discrete Hartley transform(DHT); some results on fixed-point implementation of the algorithm are provided which are compared to results obtained from floating point implementation. Finally, the application of thegmdfα algorithm to acoustic echo cancellation, in hand-free telephone systems, is detailed. Some control strategies are presented; in particular a novel double-talk detector based on a spectral dissimilarity measure is introduced ; also, a twin-filter structure which significantly enhances the echo rejection is derived.     

VoIP回声消除器设计及算法研究

本文提出了一种与线性预测编解码器相结合的新声学回声消除器,由去相关可变步长的NLMS自适应算法和基于回声路径失配方差的双端通话检测算法所组成.Matlab仿真结果表明,与Gordy所提出的回声消除算法相比,本文提出的算法在双端通话和回声路径改变时判别更准确,收敛速度更快;在收敛状态时,ERLE值平均提高了15dB,失调误差平均降低了10dB,具备更好的回声消除性能.     

基于相关函数的递推最小二乘算法及其在回波消除中的应用

本文给出一种新的类似于RLS（recursive least squares)算法的递推最小二乘算法，该算法直接对输入信号的相关函数进行处理而不是对输入信号本身进行处理，理论分析表明了该算法的收敛性。该算法应用于回波消除问题中，克服了常规自适应滤波算法在出现双方对讲的情况下需停止调节自适应滤波器系数这一不足。计算机模拟仿真表明该算法在双方对讲的情况下有良好的收敛性能。     

一种自适应电话会议回波抵消系统

文中给出一种基于去相关最小均方(DLMS)算法和迭代最大长度序列相关(IMLC)算法的电话会议回声抵消系统。鉴于DLMS算法在远端会话期间具有好的工作性能，而IMLC算法在双端会话期间具有良好的工作效果，这种新的回声抵消系统在远端会话期间用DLMS算法估计回声路径，而在双端会话期间用IMLC算法估计回声路径。计算机仿真表明，这种新的回声抵消系统在远端会话和双端会话情况下均能提供较好的回声路径估计。     

无双端会话检测回声抵消系统

提出一种无双端会话检测自适应回声抵消系统,这种系统突破了当前自适应回声抵消器必须进行双端会话检测的限制。仿真结果表明这种无双端会话检测回声抵消器在所有工作模式下均能取得理想的回声抵消效果。     

Robust multichannel least mean square-type algorithms with fast decaying transient for blind identification of acoustic channels

The multichannel least mean square (MCLMS) is an attractive and effective algorithm for blind channel identification in the noise-free case. Some recent studies show that the performance of the MCLMS algorithm significantly deteriorates in a noisy environment, that is, the blind MCLMS solution does not remain collinear with the channel vector. Therefore the authors propose non-conventional technique that helps the MCLMS algorithm converge to a novel steady-state solution that is a weighted combination of all the eigenvectors, with the weight profile inversely proportional to the eigenvalues. The improved performance of the proposed solution is verified both analytically and numerically. The algorithm is then optimised by introducing an adaptive step size that ensures fast decay of the transient response, giving stability as well as rapid convergence to the final solution. The authors then apply the proposed technique to different variants of the MCLMS algorithm, including frequency-domain implementations, to achieve a noise-robust performance. Computer simulations are presented that show improved performance of the proposed algorithms for blind identification of both acoustic and random channels with noise.     

一种基于DSP的音频实时处理系统

声学回声消除器一直是视频会议系统不可缺少的组件。将回声消除算法结合噪音消除和静音检测算法等,提出一种改进的实时音频处理系统方法,并在TMS320C6713B上实现,能够有效改善噪音、双工检测、非线性回声等导致自适应滤波器发散的问题。该系统在保证正常双工通话的同时,对非线性回声的抑制有着明显的改善效果。     

不需双端检测的声回波对消算法研究

声回波对消中双端对讲情况下的近端话音对自适应算法有很大影响。为避免双端话音检测,在滤波型LMS算法基础上,用远端信号和误差输出信号的和代替远端信号去激励预测误差滤波器,降低近端话音的影响。另为进一步提高算法抗近端干扰的能力,做了变步长的改进,首先将步长反比于输出信号预测误差的短时功率,其次将步长正比于预测误差的互相关系数。实验表明,文中提出的两算法在近端话音出现时表现出较好的性能,其中第二种有更好的稳态失调。     

A Robust ICA-Based Adaptive Filter Algorithm for System Identification

This paper proposes a new adaptive filter algorithm for system identification using independent component analysis. The additive noise is considered as an independent component to be separated from the noisy observation and is simultaneously estimated online. The proposed algorithm is derived by minimizing the mutual information between the estimated additive noise and the input signal. The local convergence conditions are also derived. The proposed algorithm can be directly applied to the acoustic echo canceller without any double-talk detector. Some simulations have been carried out to illustrate its effectiveness for synthetic and real speech signals.      

非凸投影自适应Hammerstein滤波算法

该文研究了Hammerstein系统参数辨识和非线性系统预测问题,提出一种基于非凸投影的自适应滤波算法。论文将问题归结为具有非凸可行域的约束优化问题,并建立了基于交替方向乘子法(ADMM)和递归最小二乘相结合的算法框架。在该算法框架下,非凸约束优化问题的全局最优解可通过岭回归和欧几里得(Euclid)投影循环计算得到。将提出的算法分别应用于Hammerstein系统的参数辨识、非线性未知系统预测以及非线性声学回声消除,并进行仿真实验,结果显示所提算法具有较好的收敛性和稳定性,能够得到较准确的辨识和预测效果。     

一种新型的空间声回波消除器

空间声回波消除因回波通道冲激响应长达几百毫秒一直是信号处理领域的一个难题.本文设计了空间声回波消除器,提出了自正交轮流受限分段批处理频域LMS(PBFLMS)算法,可以达到收敛性能、失调性能、计算复杂度以及延时等性能之间的折衷,并结合空间声回波消除器的具体应用,改善了该算法。最后将空间声回波消除器在ATD-C30开发仿真系统中实现,得到了比较满意的结果.     

A Nonlinear Acoustic Echo Canceller Using Sigmoid Transform in Conjunction With RLS Algorithm

Nonlinearity of amplifiers and/or loudspeakers gives rise to nonlinear echo in acoustic systems, which seriously degrades the performance of speech and audio communications. Many nonlinear acoustic echo cancellation (AEC) methods have been proposed. In this paper, a simple yet efficient nonlinear echo cancellation scheme is presented by using an adaptable sigmoid function in conjunction with a conventional transversal adaptive filter. The new scheme uses the least mean square (LMS) algorithm to update the parameters of sigmoid function and the recursive least square (RLS) algorithm to determine the coefficient vector of the transversal filter. The proposed AEC is proved to be convergent under some mild assumptions. Computer simulations show that the proposed scheme gives a superior echo cancellation performance over the well known Volterra filter approach when the echo path suffers from the saturation-type nonlinear distortion. More importantly, the new AEC has a much lower computational complexity than the Volterra-filter-based method.      

A multirate acoustic echo canceler structure

A new subband echo canceler (SBEC) structure is proposed to reduce the transmission delay introduced by conventional SBEC structures, without distorting the near-end signal. The proposed structure is based on computing two output errors, one for using during single-talk and the other one for using during double-talk periods. With the SBEC structure we propose a double-talk detector with a subband configuration which allows a fast and accurate detection of double-talk periods, enabling the SBEC algorithm to track changes in the echo path impulse response when the near-end signal is absent. Computer simulations using actual speech signals, and subjective evaluation tests are given to show the convergence performance, tracking and double-talk detection ability, of the proposed scheme     

Alternating fixed-point algorithm for stereophonic acoustic echocancellation

A novel algorithm for use in stereophonic acoustic echo cancellation is introduced. The alternating fixed-point structure of the algorithm descends each of the channels alternatively and avoids nonuniqueness in the solutions by employing inter- and intra-channel orthogonal projections. For a noisy environment, the algorithm is shown to provide improved misalignment performance and echo cancellation over leading competitors at a low computational cost. The inherent noise robustness is uncovered using H^∞ estimation techniques and noise-error bounds are derived     

Modulated Lapped Transform Domain Acoustic Echo Canceller with Double Talk Detector

A complete acoustic echo cancellation system with double talk detection capability is presented in this paper. The proposed system includes a new acoustic echo canceller (AEC) based on the modulated lapped transform (MLT) domain adaptive structure and a robust two-stage double talk detector (DTD) to cope with MLT domain AEC. The proposed AEC achieves better signal decorrelation via orthogonal MLT of size 2N× N rather than N× N square orthogonal transform such as DCT, DFT, etc. Both the input signal and the desired response are modulated lapped transformed in order to reduce the adaptation error between them so that the signal adaptation is purely operated in MLT domain. As a complementary of this, a two-stage DTD is developed to stabilize the operation of the AEC. The proposed DTD has robust algorithm structure and it allows faster switching according to the talker state change.Several simulation results with a synthetic and real speech are presented to demonstrate the performance of the proposed AEC and DTD. The proposed MLT based AEC proven to be very useful for the echo cancellation applications requiring high convergence speed and good echo attenuation. It can achieves faster convergence rate by more than twice over those of traditional DCT based AEC with an additional advantage of 10–15 dB ERLE improvement. On the other hand, a proposed two-stage DTD is shown to react quickly to both the onset and the end of the double-talk with reasonable high accuracy.     

一种条件自适应滤波算法及其在回声消除中的应用

对回声消除问题进行分析后,提出了一种条件自适应滤波算法。新算法并不要求自适应滤波器达到最优值,而是在满足一定条件时才进行系数更新。这种算法可以用在回声消除领域,很好地解决信道变化和双端发声的区分问题。文章还对这种条件自适应滤波算法的收敛性能做了严格分析,并且给出了和分析结果相吻合的计算机仿真结果。     

Numerical Solution for Fractional-Order Differential Systems with Time-Domain and Frequency-Domain Methods

For a general nonlinear fractional-order differential equation, the numerical solution is a good way to approximate the trajectory of such systems. In this paper, a novel algorithm for numerical solution of fractional-order differential equations based on the definition of Grunwald-Letnikov is presented. The results of numerical solution by using the novel method and the frequency-domain method are compared, and the limitations of frequency-domain method are discussed.     

Acoustic echo cancellor with sub-band noise cancelling

An acoustic echo canceller with sub-band noise cancelling that employs a cascade configuration is proposed. The adaptation control adopted to match the occurrence of intermittent speech/echo and continuous room noise using the NLMS algorithm is very effective in echo and noise cancellation. Hardware is implemented and its performance evaluated through experiments. The noise cancellation significantly enhances overall echo-cancellation performance.<>