新型CWDM器件的设计与制作

提出了一种新型结构的粗波分复用(CWDM)器件,该器件可实现8路波长、间隔为20nm的光信号的解复用。器件级联8个多层介质膜滤波器,两个由8个自聚焦透镜和8根光纤组成的阵列置于多层介质膜滤波器的对面以接收8路解复用后的光信号。     

阵列天线收发隔离技术

阵列天线在通信领域应用非常广泛,常规的收发隔离技术只是针对一路信号进行回波信号的抑制,假如套用常规方案对阵列天线的每一路输入进行一次自适应抵消将会大量消耗硬件资源,在实际操作中是不可行的,提出一种运用并行延时方案对16路输入的阵列天线进行回波信号抑制工作.利用这种并列式的结构可以有效的减少硬件消耗.     

基于谱优势与非线性变换混合的立体声声学回声消除方法

立体声声学回声控制系统中,两通道信号间的强相关性导致自适应算法的解不唯一,滤波器失调较大。为了解决此问题,并尽可能地保留语音质量,该文基于心理声学上的谱优势效应提出一种新型的混合立体声声学回声消除方法。根据谱优势效应,在3个低次谐频处注入能量较弱的正弦信号,以减弱低频相关性。同时对非线性变换法进行改进,以用于中高频去相关处理。与传统方法的多方面性能测试对比结果表明:所提方法能有效改善失调性能并提高收敛速度,且具有较小的语音失真。     

感觉加权回波消除器的研究

传统的自适应回波消除算法都是基于客观优化准则,而没有考虑回波消除的主观质量。本文提出在回波消除器中采用误差频率加权自适应滤波器结构,以充分利用人耳的听觉特性,提高回波消除的主观质量。客观测试和主观测试的仿真结果验证了新算法的有效性。     

基于自适应均衡的超声波穿墙通信回波消除

超声波通信在船舶、大型密闭容器等电磁屏蔽环境中具有特别的优势。但超声波信道的多径效应导致的码间干扰限制了超声波通信速率,因此需要消除回波信号来降低系统的码间干扰。超声波信道通常是时变的,需要系统能够自适应地消除回波。文章利用基于LMS的自适应均衡器辨识出信道回波传递函数,并利用回波的软消除方法实现回波消除。采用Xilinx System Generator搭建系统模型,通过初始化指令自动生成250阶的横向滤波器结构。最后通过仿真验证了方法的可行性。     

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。     

一种基于格型正交化的二阶Volterra自适应滤波算法

针对 Volterra 自适应滤波器输入信号相关性或附加的非线性畸变的增强使自适应滤波器性能下降的问题,本文提出基于格型正交化的二阶 Volterra 自适应滤波算法.先对输入信号进行格型预处理,得到互相正交的后向预测误差信号;然后将其作为自适应滤波器的输入,从而大大降低了一次项、平方项和交叉乘积项信号各项之间的耦合,改善了自适应算法的收敛性能.有源噪声对消的仿真结果表明,在输入噪声强相关和附加较强非线性畸变时本算法仍具有较好的消噪性能.     

一种MIMO中继系统的自适应自干扰消除方法

为节省频率资源,全双工中继一般采用相同的频率接收和发射信号。由于收发天线之间无法充分隔离,接收天线容易受到自身发射天线的回波干扰。针对宽带全双工多输入多输出( MIMO)中继的自干扰问题,提出了一种基于梯度下降自适应算法的自干扰消除方法。该方法利用中继反馈的已知信号进行自干扰信道估计,并产生一个对自干扰信号的估计信号,从而在接收端将干扰抑制。仿真结果表明,该方法在自适应滤波器的跟踪性能、收敛分布和不同 MIMO 配置下的均方误差( MSE)性能等方面均取得良好效果。     

基于横向滤波器理论的小型超宽带差分带通滤波器

 本文提出了一种基于横向滤波器理论的小型超宽带差分滤波器结构.由于输入/输出端口之间具有两条不同电长度的传输路径,使得差模信号的通带两端具有两个传输零点并且具有良好的谐波抑制特性.四条1/4波长的短路线被用来改善差模信号的通带特性.另外,共模信号可以很容易在整个频段上得到抑制.加工结构模型的测试结果表明:差模信号的相对带宽为102%,带内的回波损耗大于15dB.实验测试结果与理论仿真结果吻合较好.     

虚拟多参考输入自适应同址干扰抵消算法

针对同一空间内相邻发射机对接收机产生的同址干扰,将接收信号过采样后虚拟出多路参考信号并与两级自适应滤波器相结合,构成了一种虚拟多参考输入自适应同址干扰抵消算法。通过在接收纯干扰信号阶段调整第一阶滤波器的权系数,接收信号中包含有用信号时调整第二阶滤波器的系数,实现消除同址干扰恢复有用信号的功能。仿真结果表明,所提同址干扰抵消算法与传统算法相比,在简化了耦合装置的同时,也具有更好的同址干扰抑制效果。     

VoIP通信中的回声消除方法研究

介绍了语音通信声学回声产生模型和自适应AEC回声消除算法原理,分析了AEC应用于VoIP语音通信中存在的问题,设计了一种基于短时能量的非线性回声消除方法,在NGN网络的VoIP通信中,使用该方法实现了极高的回声抑制比。测试结果表明该方法的消回声效果、算法稳定性和实现复杂度等指标明显优于自适应AEC算法,适合于嵌入式VoIP通信终端设备的开发。     

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.     

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.      

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.     

Noise robust alternating fixed-point algorithm for stereophonicacoustic echo cancellation

A novel algorithm specifically for use in stereophonic acoustic echo cancellation (SAEC) environments is introduced. It is based on an alternating fixed-point (FP) structure. Analysis provides bounds to ensure that the algorithm has the form of a contraction mapping (CM). Simulation results show improved performance over algorithms with similar computational complexity in the presence of noise     

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     

一种利用人耳听觉特性的双声道回波抵消改进算法

本文提出了一种利用人耳听觉特性的双声道回波抵消改进算法.在该系统中,利用人耳的听觉特性给输入信号加上一个小的扰动来改善输入信号的自相关特性和条件数,从而达到改善系统收敛特性的作用.本文讨论了在双声道回波抵消系统中遇到的输入信号强相关从而引起系统性能恶化的问题,提出了采用输入信号加扰来改善系统性能的算法.并且利用人耳听觉特性使这种扰动信号隐藏在语音信号中,使其对语音信号的干扰最小.从本文的仿真试验来看,该方法对双声道回波抵消系统的性能有一定的改善.     

结合房间声学特点的子带自适应滤波声学回音抵消算法

子带自适应滤波算法是处理长阶声学回声抵消问题的重要方法之一。结合房间声学的特点,对各子带采用不同长度的FIR滤波器进行滤波,将有限的资源进行更合理的分配,节省了资源和计算量,并在一定程度上提高了收敛速度。     

A Signal Decorrelation PNLMS Algorithm for Double-Talk Acoustic Echo Cancellation

Acoustic echo cancellation (AEC) in voiced communication systems is used to eliminate the echo which corrupts the speech signal and reduces the efficiency of signal transmission. Usually, the performance of AEC system based on the adaptive filtering degrades seriously in the presence of speech issued from the near-end speaker (double-talk). In typical AEC scenarios, double-talk detector (DTD) must be added to AEC for improving speech quality. One of the main problems in AEC with DTD is that the DTD errors can result in either large residual echo or distorting the near-end input speech. Considering the strong correlation property of speech signals, this paper presents a novel proportionate decorrelation normalized least-mean-square (PDNLMS) adaptive AEC without DTD for echo cancellation as an interesting alternative to the typical AEC with DTDs. Unlike traditional AEC with a DTD, the proposed PDNLMS uses the difference of near-end speech as the residual error to update adaptive echo channel filter during the periods of double-talk, which can efficiently reduce the double-talk influence on the AEC adaptation process. The experimental results show that not only the proposed PDNLMS without DTD illustrate better stability and faster convergence rate, but it is also of a lower steady-state misalignment and better residual signal than current methods with DTDs at a lower computational cost.     

The complex subband decomposition and its application to the decimation of large adaptive filtering problems

We show that a near perfect reconstruction (NPR) M-channel filterbank with a diagonal system inserted between the analysis and synthesis filterbanks may be used to decompose a finite impulse response (FIR) system of order L into M complex subband components, each of order L/K, where K is the downsampling rate. This decomposition is at the expense of using complex arithmetic for the subband processing. The theory surrounding the proposed filterbank structure leads to a new understanding of subbanded adaptive filtering implementations. It also leads naturally to a delayless subbanded adaptive filter scheme. Using conditions on the analysis and synthesis filters, the formulas for the subband components and their respective properties are developed. Simulation results for an acoustic echo cancellation (AEC) example are given to support the developed theory.