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。     

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.     

Modelling of acoustic transfer functions for echo cancellers

ARMA modelling (infinite impulse response filters having both poles and zeros), rather than MA modelling, of acoustic transfer functions is potentially capable of reducing the number of coefficients required to achieve a given level of echo cancellation. The amount of reduction obtainable, however, has received little attention in the literature. To arrive at quantitative figures for the complexity reduction possible, equation error and output error ARMA and MA modelling of acoustic impulse responses are compared experimentally. It is shown, for both fullband and subband implementations, that the potential for reducing the number of coefficients required is small     

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。     

偏差补偿符号子带自适应滤波器

自适应滤波器在系统辨识、回声消除、信道均衡等领域获得了广泛应用.符号子带自适应滤波器（Sign Subband Adaptive Filter,SSAF）具有较强的抗脉冲干扰能力,但当输入信号受到噪声干扰时,其对未知系统系数向量的估计会产生偏差.为了解决上述问题,本文基于无偏估计准则,提出了一种偏差补偿符号子带自适应滤波器（Bias-Compensated Sign Subband Adaptive Filter,BC-SSAF）.为了解决定步长自适应滤波器需要在收敛速度和稳态失调之间进行折中的问题,本文采用随机梯度法来更新正则化参数,提出了变正则化参数偏差补偿符号子带自适应滤波器（Variable Regularization Bias-Compensated Sign Subband Adaptive Filter,VR-BC-SSAF）.仿真结果验证了BC-SSAF和VR-BC-SSAF性能的优越性.     

解卷积主动声呐目标回波高分辨时延估计技术

为提高浅海复杂海洋环境下的目标回波时延估计精度,增强主动声呐系统对目标的探测能力,该文基于稀疏表示理论和解卷积思想,提出一种高分辨目标回波时延估计技术.首先,引入Toeplitz算子,将发射信号的不同时延结果构造成时延字典矩阵,时延估计值存在于所求解的稀疏向量中.其次,利用交替方向乘子算法(ADMM)优化框架,求解全局...     

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.      

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.     

Acoustic echo cancellation: do IIR models offer better modelingcapabilities than their FIR counterparts?

The adequateness of IIR models for acoustic echo cancellation is a long-standing question, and the answers found in the literature are conflicting. We use the results from rational Hankel norm and least-squares approximation, and we recall a test that provides a priori performance levels for FIR and IIR models. We apply this test to the measured acoustic impulse responses. Upon comparing the performance levels of FIR and IIR models with the same number of free parameters, we do not observe any significant gain from the use of IIR models. We attribute this phenomenon to the shape of the energy spectra of the acoustic impulse responses so tested, which possess many strong and sharp peaks. Faithful modeling of these peaks requires many parameters, irrespective of the type of the model     

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.     

Design and discrete re-optimisation of all-pass based powersymmetric IIR filters

Tools are presented which enable the practitioner to efficiently design all-pass based, highly selective low-pass power symmetric-infinite impulse response (PS-IIR) filters which are well suited for sub-band decomposition in applications such as multirate acoustic echo cancellation (MAEC)     

Proportionate NSAF algorithms with sparseness-measured for acoustic echo cancellation

In acoustic echo cancellation (AEC), the sparseness of impulse responses can vary over time or/and context. For such scenario, the proportionate normalized subband adaptive filter (PNSAF) and μ-law (MPNSAF) algorithms suffer from performance deterioration. To this end, we propose their sparseness-measured versions by incorporating the estimated sparseness into the PNSAF and MPNSAF algorithms, respectively, which can adapt to the sparseness variation of impulse responses. In addition, based on the energy conservation argument, we provide a unified formula to predict the steady-state mean-square performance of any PNSAF algorithm, which is also supported by simulations. Simulation results in AEC have shown that the proposed algorithms not only exhibit faster convergence rate than their competitors in sparse, quasi-sparse and dispersive environments, but also are robust to the variation in the sparseness of impulse responses.     

基于附加信号回波抵消的FPGA实现

针对基于附加信号回波抵消在硬件设计中出现的迭代误差累积导致信道估计不准确和耗费大量FPGA资源的问题,对算法进行了改进。从主径开始估计回波信道的方法,提高了信道估计精度和减少了FPGA资源消耗。然后,在FPGA平台上用硬件语言Verilog HDL对此回波抵消系统加以实现。仿真结果表明此设计在回波抵消方面具有良好的效果。     

Cascaded power symmetric IIR filter banks and continuityconstrained adaptive algorithms for acoustic echo cancellation insubbands

Teleconferencing systems and hands-free mobile terminals use acoustic echo cancellation (AEC) for high-quality full-duplex speech communication. The problem of aliasing in subband AEC is addressed. Filter banks with implicit notch filtering are derived from cascaded power symmetric-infinite impulse response (CFS-IIR) filters. It is shown that adaptive filters used with these filter banks must be coupled via continuity constraints to reduce the aliasing in the residual echo. A continuity constrained NLMS algorithm is therefore proposed and evaluated     

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.<>     

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。     

Design and performance of adaptive systems based on structured stochastic optimization strategies

The theory and design of linear adaptive filters based on FIR filter structures is well developed and widely applied in practice. However, the same is not true for more general classes of adaptive systems such as linear infinite impulse response adaptive filters (MR) and nonlinear adaptive systems. This situation results because both linear IIR structures and nonlinear structures tend to produce multi-modal error surfaces for which stochastic gradient optimization strategies may fail to reach the global minimum. After briefly discussing the state of the art in linear adaptive filtering, the attention of this paper is turned to MR and nonlinear adaptive systems for potential use in echo cancellation, channel equalization, acoustic channel modeling, nonlinear prediction, and nonlinear system identification. Structured stochastic optimization algorithms that are effective on multimodal error surfaces are then introduced, with particular attention to the particle swarm optimization (PSO) technique. The PSO algorithm is demonstrated on some representative IIR and nonlinear filter structures, and both performance and computational complexity are analyzed for these types of nonlinear systems.     

基于对数域同态滤波的时延估计算法研究

针对水声信道中窄带信号的多途时延估计问题,本文在对复倒谱时延估计方法进行研究的基础上,提出了一种基于对数域同态滤波的时延估计算法.结合复倒谱与同态滤波思想,将接收的窄带信号首先变换到对数域,然后与本地存储的信号进行谱减法,再对相减后的信号进行滤波以消除残余的信号与噪声成分,最后将其恢复到时域以获取多途时延估计.与传统的匹配滤波/相关处理以及复倒谱分析方法相比,本文算法具有时延估计精度高、噪声抑制能力较强等特点.仿真与湖试数据处理结果证明了该方法的有效性.     

单频网中继站自适应回波消除器的设计

以数字电视地面广播国家标准为背景,提出一种在单频网通信系统中,通过专门电路对中继站回波干扰进行消除的方法.该方法通过对TDS-OFDM信号中的PN序列进行分析和计算,获取对回波干扰信道的估计,然后利用信道估计的结果生成一路反馈信号,用于逼近回波干扰信号,最后从接收信号中除去该反馈信号即实现回波消除.其中信道估计的过程是迭代的、自适应的.实验结果表明,该方法具有良好的回波消除性能.     

稀疏正则化在双基地ISAR距离成像中的应用

双基地逆合成孔径(ISAR)成像中,一维距离像的分辨率因双基地角的存在而下降,为解决此问题,研究了双基地ISAR的一维距离成像算法。在分析复基带回波稀疏特性的基础上,将一维距离像的生成问题转化为噪声环境下信号的稀疏分解问题。利用可任意调节的时延单元构建冗余基,基于L曲线准则估计正则参数,最后利用推广的正则化FOCUSS算法估计复基带回波的稀疏分解系数,并据此生成目标的一维距离像。一维距离像分辨率取决于构建稀疏分解冗余基的延时单元,据此解决了双基地角对距离分辨率的限制问题,仿真实验验证了算法的有效性。