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.     

Nonlinear Acoustic Echo Canceller to Combat Sigmoid-Type Nonlinearities Under Noisy Environment

This paper presents a nonlinear-acoustic-echo-cancellation (NAEC) technique to tackle sigmoid-type nonlinearities under noisy environment. The nonlinear echo in acoustic systems is inevitable due to the inherent nonlinear characteristics of amplifiers and/or loudspeakers, which deteriorates the quality of speech as well as audio signal reception. Here, the sigmoid-type nonlinearity is modelled by incorporating two control parameters, which determine the shaping- and clipping-parameter values of the saturation curve at a particular room temperature. These control parameters are adjusted by utilizing the variable-step-size (VSS) least-mean-square (LMS) algorithm to enhance the convergence rate and tracking capability of presented NAEC. Furthermore, the impulse response of a room (indoor channel) in the acoustic echo path is modelled as a tap-delay-line finite-impulse-response filter, whose tap-coefficients are estimated by utilizing a modified recursive-least-squares (RLS) algorithm (involving the noise statistics) at the different values of signal-to-noise-ratio (SNR), when correlated as well as uncorrelated input signals are processed. Simulation results demonstrate the efficiency and efficacy of above mentioned adaptive NAEC technique using the VSS-LMS and modified RLS algorithms in terms of the high convergence rate as well as high value of echo-return-loss-enhancement (ERLE) factor. Both the elevating value of shaping-parameter (i.e., increasing nonlinearity level) and the alleviating value of SNR adversely affect the performance of all NAECs. However, the VSS-LMS and modified RLS algorithm based presented adaptive NAEC outperforms the traditional VSS-LMS and normalized-least-mean-square (NLMS) algorithm based NAEC under similar conditions.

     

与VoIP声码器结合的回声消除器

NLMS算法是回声消除器中最常用的算法之一,然而语音信号的强相关性使NLMS(归一化最小均方)算法的收敛速度变慢。鉴于VoIP(网络电话)常用的低速率声码器大多基于LP(线性预测)编码,给出了一种与声码器结合的回声消除器,通过利用从解码端获得的LP系数和激励信号,省去了对去相关滤波器系数的计算,并且不再需要专门的前置滤波器为远端信号去相关,同时又改善了回声消除器性能。仿真结果表现出更高的收敛速度和回声抵消量。     

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DSP的声回声抵消器的设计

回声抵消器是视频会议系统和电话会议系统中一种必不可少的设备。在讨论回声抵消器原理和算法的基础上,给出了基于DSP的回声抵消器的设计方案。在设计中采用了变步长LMS算法,利用TI公司的DSP TMS320VC5402作为处理单元实现了回声抵消。     

区间自适应算法与回声抵消

本文提出了一种新的自适应算法-"区间自适应算法",给出了"区间自适应参数约束条件",并对该"约束条件"进行了严格的数学证明."区间自适应算法"大大减少了自适应滤波器的计算量.本文还论述了"区间自适应算法"在回声抵消器理论与实践中的应用.     

A Cascadable VLSI Echo Canceller

A new single-chip echo canceller has been developed. In addition to offering improved performance over an earlier single-chip echo canceller [14], the new canceller has many desirable features not available before. The most important of these features is that multiples of the chip can be connected in cascade to synthesize very long impulse responses. This paper describes the new features implemented on the chip and characterizes its performance.     

An Efficient RLS Data-Driven Echo Canceller for Fast Initialization of Full-Duplex Data Transmission

Computationally efficient recursive-least-squares (RLS) procedures are presented specifically for the adaptive adjustment of the data-driven echo cancellers (DDEC's) that are used in voiceband fullduplex data transmission. The methods are shown to yield very short learning times for the DDEC, while they also simultaneously reduce computational requirements to below those required for other leastsquare procedures, such as those recently proposed by Salz (1983). The new methods can be used with any training sequence over any number of iterations, unlike any of the previous fast-Converging methods. The methods are based upon the fast transversal filter (FTF) RLS adaptive filtering algorithms that were independently introduced by the authors of this paper; however, several special features of the DDEC are introduced and exploited to further reduce computation to the levels that would be required for slower-converging stochastic-gradient solutions. Several tradeoffs between computation, memory, learning time, and performance are also illuminated for the new initialization methods.     

一种实用的自适应回波抵消器

本文提出了一种实用、完整的自适应回波抵消器,可用于实现高速调制解调器的二一双工传输。实验表明,该回波抵消器具有回波抑制能力强、线路环境适应性好等特点。     

一种改进的与声码器结合的声学回声消除器

提出了一种改进的与线性预测声码器结合的声学回声消除器,通过增加一个自适应辅助滤波器,避免了对信号路径中所有信号在任何时刻都需经去相关、恢复相关过程所带来的信号质量降低、失调误差增加等负面影响。计算机仿真结果表明,该结构明显提高了已有算法的回声抵消量。     

NXP平台回声消除算法原理及应用

随着无线通信技术的迅猛发展,以手机为代表的无线通信终端已经成为大众应用最为广泛的通信工具。这其中语音通话功能是非常重要的用途,通话质量的优劣直接影响着用户感受。而通话回声问题是最难解决的部分,因为回声产生途径多样、软件处理算法复杂、回声调试参数众多,因此给研发工程师调试带来很大困难。文章介绍NXP平台回声消除算法的原理及其应用,旨在帮助工程师较快理解此平台回声消除算法机理并运用介绍的思路进行软件音频调试系统的开发和应用,从而达到改善无线通信终端回声效果的目的。     

LMS算法和RLS算法在水声信道通信系统中应用的比较

为了减小水声通信系统中存在的由于多径传播效应引起的符号间干扰,可在系统中使用线性均衡器。首先介绍了LMS算法和RLS算法基本原理,利用Matlab仿真软件,给出了两种算法在水声信道通信系统中的自适应均衡的仿真系统,并对这两种算法对于均衡器的影响进行了比较,最后对比较的结果进行了分析。     

LMS算法和RLS算法在水声信道通信系统中应用的比较

为了减小水声通信系统中存在的由于多径传播效应引起的符号间干扰,可在系统中使用线性均衡器。首先介绍了LMS算法和RLS算法基本原理,利用Matlab仿真软件,给出了两种算法在水声信道通信系统中的自适应均衡的仿真系统,并对这两种算法对于均衡器的影响进行了比较,最后对比较的结果进行了分析。     

A Co-Design Methodology for Telecommunication Systems: A Case Study of an Acoustic Echo Canceller

The acoustic echo cancellation with large adaptive filters is a computationally intensive problem and needs real time cost effective solution. To deal with these challenges, designers have increasingly turned to mixed Hardware/Software (HW/SW) implementation of echo canceller algorithms. This paper presents a co-design methodology and environment for both hardware and software modules. We describe how High Level Synthesis (HLS) tools like GAUT and SYNDEX can be efficiently used for rapid prototyping of heterogeneous architecture based on DSP TMS320C40 and ASIC. The HW/SW interface synthesis task is especially discussed since it constitutes a key issue of the whole design. As an illustration, we present a mixed implementation of the GMDF alpha algorithm, an adaptive filter well suited to acoustic echo cancellation, on both ASIC and TMS320C40 DSP.     

一种改进的NLMS算法在声回波抵消中的应用

收敛速度和残余均方误差是衡量最小均方算法性能的重要指标。在声回波抵消算法中,为了寻求收敛速度快和计算量小的自适应算法,在归一化最小均方误差算法基础上,把当前时刻以前的误差引入归一化收敛因子中得到一种新算法,可以减小信号样本波动对权重带来的影响。该算法比传统的归一化最小均方算法收敛性能更好,稳态失调也比其小。计算机仿真结果表明,新算法在自适应回波抵消中的综合性能要优于传统的归一化最小均方误差算法。     

基于小波变换的RLS波束成形算法研究

为了解决递归最小二乘算法(RLS)在较低信噪比(SNR)、遗忘因子较小的环境中,对噪声敏感、收敛时参数估计误差大的问题,引入小波变换去噪思想,提出了基于小波变换的RLS波束成形算法。该算法利用小波变换软阈值法进行信号去噪,再采用RLS算法进行波束成形。最后对实验进行仿真,仿真结果表明,与传统的RLS算法相比,该算法具有较小的稳态误差和较快的跟踪速度和收敛速度,并且波束成形效果好。     

卫星通信中快速非线性信道RLS均衡算法

幅度相位联合调制(Amplitude Phase Shift Keying, APSK)信号经过卫星信道时容易引起非线性失真,产生码间串扰。为此,本文将Wiener均衡应用于APSK卫星通信中,并提出了一种快速非线性信道递归最小二乘(Nonlinear Channel Recursive Least Squares, NCRLS)均衡算法,在传统NCRLS算法的基础上,建立了遗忘因子与步长因子的关系,推出了变遗忘因子的非线性信道递归最小二乘(Variable Forgetting Factor Nonlinear Channel Recursive Least Squares, VFFNCRLS)均衡算法,有效地提高了收敛速度。仿真结果表明,VFFNCRLS算法有效地纠正了非线性失真信号的星座扭曲和频谱再生,解决了卫星通信系统中有记忆非线性信道失真的问题,相比传统算法,具有较快的收敛速度和较好的稳定性。      

基于非线性预处理的声回波对消

考虑到扬声器的非线性特性,将实际房间声学系统模拟为一个无记忆非线性系统与一个动态线性系统相级联,并分别采用多项式函数和限幅函数模拟扬声器系统的非线性特性,通过在声回波对消的归一化最小均方算法中引入相应的非线性预处理,改善房间声回波对消的效果.计算机仿真证明,考虑了扬声器非线性影响并在算法中进行相应非线性预处理后的房间声回波对消比未经非线性预处理的声回波对消有更好的对消效果.     

基于小波变换的声学回波消除系统

声学回波消除技术一直是语音通信领域的研究热点。在声学回波消除系统中,通过估计回波路径中的固定时延区域来提高自适应滤波算法的收敛速度。提出了一种基于小波变换的固定时延估计算法以及基于小波变换的声学回波消除系统,解决传统时延估计算法在声学回波消除系统中估计误差大、抗干扰能力弱的问题。测试结果表明,算法稳健性、有效性等指标明显优于传统时延估计算法,基于小波变换的声学回波消除系统具有良好的消回波性能。     

An Adaptive-Step Sign Algorithm for Fast Convergence of a Data Echo Canceller

This paper proposes an echo cancellation algorithm that can be used for full-duplex digital data transmission over existing twisted-pair cables. The proposed algorithm is a form of the "sign algorithm." A new adaptation method is introduced to change the canceller step size according to the residual echo level by using the correlation between the echo and the error signal. The adaptation method can increase the echo canceller convergence speed by an order of magnitude over the conventional sign algorithm. An echo canceller utilizing the proposed algorithm is suitable for VLSI implementation, because it requires no multibits analog-to-digital converter. Theoretical analysis and parameter optimization are carried out. The rapid convergence property of the algorithm is verified by computer simulation as well as experiments.