一种基于小波变换的自适应滤波新方案

本文基于Ｌ＾２（Ｒ）函数的尺度函数表示法提出一种适用于声回波对消应用的自适应系统辨识的新方案。该方案可以利用小波变换的抽取特性，降低自适应迭代的次数且保持小波准正交变换的优点。计算机模拟证实了上述论断的正确性。     

基于DVB-T标准的数字电视回波对消技术的研究

以数字电视地面广播国家标准DVB-T为基础,提出了一种基于FPGA和DSP实现回波对消的方案.其原理是:首先该系统不发送数字电视信号,通过发射CAZAC序列进行信道初估计.由于信道在实际中会发生一定的变化,所以还需要用自适应算法进行信道特性的跟踪,其中DSP运用自适应算法保证对消信号接近回波信号.其次FPGA系统获得了回波信道的参数并生成逼近回波信号的对消信号.最后,将接收信号与对消信号相减完成对消.提出了对回波信道的初估计并利用变换域LMS算法进行信道跟踪的一种方案.最后仿真结果表明,该方法具有良好的回波对消性能.     

一种基于小波变换的自适应滤波新方案

本文基于L2(R)函数的尺度函数表示法提出一种适用于声回波对消应用的自适应系统辩识的新方案。该方案可以利用小波变换的抽取特性,降低自适应迭代的次数且保持小波准正交变换的优点。计算机模拟证实了上述论断的正确性。     

基于群延迟和模糊函数的LFMICW对消分析

针对线性调频中断连续波(LFMICW)信号的隐身问题,根据LFMICW回波信号中目标信息处理方法,提出利用信号和匹配滤波器的群延迟特性设计LFMICW对消信号.定义了对消误差因子,并分析了信噪比为6dB且该因子不同取值时单目标的对消结果,利用模糊函数分析对消后的目标合成回波和未对消的干扰目标回波信号的模糊特性,并给出了...     

自适应声回波对消

本文介绍了人们对高质量免提话音通信产品的需求及国内国际市场相关产品的现状，论述了声回波对消技术是改善免提话音通信质量的关键及其在国内市场的应用前景。     

一种改进的外辐射源雷达直达波对消算法

针对外辐射源雷达的直达波对消技术,传统的变步长对消算法依据参考信号或对消后瞬时误差进行步长调整,对于剩余的目标回波等信号也产生了抑制效果,同时影响了直达波信号的对消效果,为此,提出了一种基于对消信号与直达波的相关性的Sigmoid函数变步长最小均方(Variable Step Size LMS based on Sigmoid function,SVSLMS)算法,根据对消后的信号与直达波的相关性进行步长调整。理论分析与仿真结果表明,改进的SVSLMS算法对直达波对消有更快的收敛速度,同时对目标回波信号的影响更小。     

多窗谱估计法估计相干函数的双端语音检测

双端语音检测是声回波对消系统的重要组成部分。鉴于目前采用的基于相关的双端语音检测运算量较大,文中采用基于相干的双端语音检测方法,利用多窗谱估计法估计相干函数。该谱估计法能够利用较短的数据段获得可接受的频率分辨率而估计方差较小。软件仿真结果表明,基于相干的双端语音检测与基于相关的双端语音检测相比较,前者能够在不严重增大漏报概率的情况下大大提高运算效率。     

回波抵消中级联神经网络滤波器的研究

针对回波通道同时具有线性和非线性特征以及目前的声回波抵消器主要是基于对回波通道的线性假设,提出可用一个级联的神经网络滤波器来作为一个非线性AEC中的滤波器结构.它由一个前馈的神经网络滤波器构成,即由一系列线性FIR滤波器串联而成;并采用了不同的自适应算法来估算滤波器的线性和非线性.实验结果表明,这种自适应级联神经网络滤波器能收敛于一个比线性自适应FIR滤波器更高的回波损失增益.     

一种基于二阶盲信号分离的多路声回波抑制模型

本文提出了一个基于二阶盲信号分离的多路声回波抑制模型.该模型回避了多路声回波对消中因声回波源信号间的强互相关性所致的固有的解的非唯一性问题,而是充分利用了这种互相性来去除声回波.模型仅添加一个辅助麦克风并巧妙置位即可对各路麦克风信号中的多路声回波进行有效的分离和抑制.为了实时处理的目的,提出了一个计算复杂度低且收敛稳健的二阶频域盲信号分离算法来检验该模型.实验结果充分确认了提出模型的有效性.     

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。     

Acoustic Echo Cancellation Using a Pseudocoherence Function in the Presence of Memoryless Nonlinearity

Acoustic echo cancellation (AEC) is critical for telecommunication applications involving two or more locations such as teleconferencing. It is also challenging because of loudspeaker's nonlinearity, real-time implementation requirement, and multipath effects of indoor environments. This paper addresses the nonlinear AEC problem. We use a Hammerstein model to describe the memoryless nonlinearity of loudspeaker concatenated with a linear room impulse response. We propose a method using a pseudo magnitude squared coherence (MSC) function to identify the nonlinearity in the Hammerstein system and develop an on-line AEC algorithm. Our method identifies nonlinearity without knowing the linear block in the Hammerstein system, which guarantees the stability of the algorithm and leads to a faster convergence rate. Moreover, several alternative criteria based on the MSC function are also proposed for nonlinearity identification. Effectiveness of the proposed algorithms is demonstrated through computer simulations.      

Nonlinear loudspeaker compensation for hands free acoustic echocancellation

Loudspeaker nonlinearity at high volumes limits the achievable echo cancellation performance in linear acoustic echo cancellers. A new nonlinear adaptive filter for improving the echo cancellation performance at high volumes for hands free telephones is proposed. Experimental measurements show that an echo cancellation improvement of >8 dB can be obtained at high volumes as compared to a linear adaptive filter     

Adaptive Sampling Rate Correction for Acoustic Echo Control in Voice-Over-IP

Hands-free terminals for speech communication employ adaptive filters to reduce echoes resulting from the acoustic coupling between loudspeaker and microphone. When using a personal computer with commercial audio hardware for teleconferencing, a sampling frequency offset between the loudspeaker output D/A converter and the microphone input A/D converter often occurs. In this case, state-of-the-art echo cancellation algorithms fail to track the correct room impulse response. In this paper, we present a novel least mean square (LMS-type) adaptive algorithm to estimate the frequency offset and resynchronize the signals using arbitrary sampling rate conversion. In conjunction with a normalized LMS-type adaptive filter for room impulse response tracking, the proposed system widely removes the deteriorating effects of a frequency offset up to several Hz and restores the functionality of echo cancellation.     

Acoustic noise and echo cancelling with microphone array

A novel method of performing acoustic echo cancelling using microphone arrays is presented. The method employs a digital self-calibrating microphone system. The calibration process is a simple indirect on-site calibration that adapts to the particulars of the acoustic environment and the electronic equipment in use. Primarily intended for handsfree telephones in automobiles, the method simultaneously suppresses the handsfree loudspeaker and car noise. The system also continuously takes into account disturbances such as fan noise. Examples from an extensive evaluation in a car are also included. Typical performance results demonstrate 20-dB echo cancellation and 10-dB noise reduction simultaneously     

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.      

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。     

Acoustic echo cancellation for stereophonic systems derived from pairwise panning of monophonic speech

An algorithm is introduced that performs stereophonic acoustic echo cancellation (SAEC) for systems using pairwise panning of a single monophonic source to provide the effect of spatialisation. The technique exploits the inherent high correlation between the loudspeaker signals, unlike other general SAEC techniques, which try to utilise any small uncorrelated features in the signals. The algorithm maintains a single aggregate echo path estimate that is updated using normalised least mean square (NLMS) and the knowledge of any change in the spatialisation. Consequently, it achieves a computational complexity that is of the same order as a single channel NLMS algorithm.     

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。     

多波束声发射系统的实现

提出了基于能量对比原理的多波束形成算法,设计并实现了多方向声频发射系统.数值仿真和实验结果表明,该系统可实现两个不同方向的可听声波束传播,在800Hz以上,目标区域的测量声压级比其他方向高10dB以上.     

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。