感觉以回波消除器的研究

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。     

通信中的自适应回波抵消及其相关技术

现在,自适应回波抵消越来越多地被用来改善通信质量。本文介绍、比较了自适应回波低消的关键技术——回波抵消算法和双方同时通话检测,并对自适应回波抵消芯片中采用的其它相关技术作了介绍。     

一种改进的自适应滤波算法及其在回波消除中的应用

该文改进了一种自适应滤波算法,并对新算法进行了收敛性分析。作为应用,将新算法应用于混沌调制通信系统在回波信道中的回波消除问题。仿真结果表明,改进后的算法比原算法具有更好的收敛速度和收敛精度;在低信噪比下,新算法更为稳定。     

基于DLMS算法的自适应滤波器的VLSI实现

阐述自适应滤波器在通信领域的自动均衡、回波消除、天线阵波束形成，以及其他有关领域信号处理的参数识别、噪声消除、谱估计的应用。基于实时性和能量消耗已经成为信号处理中最重要的技术指标，探讨DLMS自适应滤波器的算法顺序执行规则，提高电路的并行性。     

通信中的自适应回波抵消及其相关技术

现在,自适应回波抵消越来越多地被用来改善通信质量.本文介绍、比较了自适应回波抵消的关键技术--回波抵消算法和双方同时通话检测,并对自适应回波抵消芯片中采用的其它相关技术作了介绍.     

移动通信中的回波抵消算法分析

移动通信领域中巳广泛采用自适应回波抵消技术，自适应回波抵消算法是该技术的核心，算法的优劣直接影响自适应回波抵消效果，同时它对降低硬件复杂度和产品成本也有特殊意义。文章对自适应回波抵消的NLMS算法、双端讲话（DT）的检测算法及后处理的NLP算法进行了分析和讨论。     

基于梯度向量投影的自适应滤波算法的改进及其在多路回波消除中的应用

该文首先对Lim(2000)的基于梯度向量正交投影的算法(OGA)进行了分析和改进,在此基础上获得了一种新的自适应滤波算法(MOGA)。新算法使用时变遗忘因子对误差进行指数加权平均来估计均方误差,并使用该因子改变自适应迭代过程中滤波器系数向量的更新方向.然后将改进后的新算法扩展成两路回波消除算法用于多路回波的消除中,获得了良好的效果。仿真结果表明, MOGA不仅对时变或时不变系统具有很好的跟踪能力,克服了Lim(2000)所提算法收敛性不佳甚至有时发散的缺陷,而且应用于多路回波消除时具有计算量小,收敛速度快和精度高等特点,其收敛速度和精度优于J.Benesty(1996)和G.Sankaran(1999)的相应结果。     

一种基于多视点视频的低复杂度自适应环路滤波算法

自适应环路滤波(ALF)是面向高清的多视点视频编码(MVC)中的一项新的滤波技术,虽然能提高视频压缩的主观质量和客观质量,但是其复杂度过高阻碍了其实时应用。本文提出了一种低复杂度的ALF算法,充分利用了视点间相关性和层间相关性以减少亮度和色度分量的分割次数,并且能自适应跳过大部分帧的块控制(DBC)处理过程。实验结果表明,所提出的算法能在保证视频主观质量和客观质量基本不变的情况下减少61%左右的ALF处理时间。因此,所提出的方法在编码效率和复杂度上做到了很好的折中处理,并且能很好地改善ALF的性能。     

一种新颖的自适应图像压缩算法

数字图像压缩技术是信息高速公路的关键技术之一。本文提出了一种新颖的自适应图像压缩算法，该算法能在保证图像客观和主观质量不受损害的前提下，根据图像的具体特征，合理地选择相应的处理方法。算法的基础是基于块匹配法（ＢＭＡ）和块递归法（ＢＲＡ）的自适应搜索方法、基于边缘检测的自适应运动补偿技术以及基于边缘复杂度的自适应量化算法。在前人研究成果的基础上，我们提出了自己的自适应法则和实现方法，使运算数据量更有效地压缩。本算法有较强的适应性。可以有效地应用于Ｈ．２６１建议、ＪＰＥＧ算法、ＭＰＥＧ算法和ＨＤＴＶ等方面而无需作重大改动。当然其较大的数据压缩比使其特别适用于低码速率Ｈ．２６１建议。     

Adaptive Reference Echo Cancellation

The AREC (adaptive reference echo cancellation) algorithm is presented for an echo canceler used in full-duplex two-wire digital transmission on digital subscriber loops. The AREC algorithm incorporates a decision-directed estimation of and compensation for the far-end signal which is a source of interference to the conventional echo canceler adaptation algorithm. The AREC algorithm thus offers much faster convergence and shorter coefficient wordlengths than the conventional algorithm. Analysis and simulation of the performance and convergence of both AREC and conventional echo canceler adaptation algorithms are carried out. Included in the analysis is the effect of receiver delay and coefficient wordlength requirements. A simple and robust startup procedure is proposed and investigated by simulation.     

Local echo canceler with optimal input for true full-duplex speechscrambling system

The purpose of this paper is to focus on the local echo-canceling problem of full-duplex scrambled speech communications over a two-wire telephone network when the scrambling transformation is located between the handset and body of a telephone. Such a design makes possible very efficient protection against electromagnetic compromising emanation, which in turn substantially enhances the overall security of a protected communication. We propose a new adaptive FIR filter algorithm for local echo cancellation in such applications. The proposed algorithm differs from the conventional one by the construction of input signals in an optimal way using the D-optimal experiment design. In this way, at each step, we generate a new sample of the D-optimal pilot sequence for the filter parameter estimation. Consequently, the adaptation of the local echo canceler is defined as an initialization process in the first phase of each protected telephone call. The advantage in using the proposed adaptive FIR echo canceler is demonstrated through simulation results     

A 64-tap CMOS echo canceller/decision feedback equalizer for 2B1QHDSL transceivers

A 60-MHz 64-tap adaptive finite-impulse-response (FIR) filter chip was fabricated in 1.2-μm CMOS. It can implement either an echo canceler or a decision feedback equalizer for 2B1Q high bit rate digital subscriber line (HDSL) transceivers. The 4.3×4.3 mm², 30000 transistor chip is a completely self-contained adaptive filter which incorporates the least mean square (LMS) algorithm for coefficient updating. The device can be cascaded to implement very long filter lengths, which are often required in high bit rate transceivers. At a 60-MHz clock rate, the echo canceler/decision feedback equalizer chip can accommodate symbol rates in excess of 800 kbaud     

Echo Canceler with Two Echo Path Models

An adaptive echo canceler with two echo path models is proposed to overcome the false adaptation problem for double-talking. The echo canceler possesses two separate echo path models (EPMs), one (background EPM) for adaptively identifying echo path transfer characteristics and the other (foreground EPM) for synthesizing an echo replica to cancel out echo. The parameter values of the foreground EPM are refreshed by those of the background EPM, according to a transfer control logic, when the logic determines that the background EPM is giving a better approximation of echo path transfer characteristics than the foreground EPM. Completely digital hardware implementation is described. Using the hardware, it is shown that virtually complete double-talking protection is actually realizable by the new method.     

Acoustic echo-canceler using the FBAF algorithm

An acoustic echo-canceler for teleconferencing systems is realized based on the frequency bin adaptive filtering (FBAF) algorithm. In the FBAF algorithm, each frequency bin does an independent adaptive filtering, so that parallel processing can be used to increase the throughput of the system. Hardware size can be reduced to about 25% of the FIR time domain adaptive filter (TDAF) requirement. The realized echo canceler allows a comfortable conversation with only 8 ms of delay. The hardware prototype contains 12 VSP chips and one DSP chip, An ERLE (echo return loss enhancement) of 30 dB was achieved using this prototype hardware for an echo reverberation path with 260 ms delay. An efficient method for normalizing the convergence factor of the FBAF algorithm with a correlated input signal is given that speeds up the convergence rate. The performance is shown by computer simulation     

A New Digital Echo Canceler for Two-Wire Full-Duplex Data Transmission

A new approach to echo canceling for two-wire fullduplex data transmission is proposed. The canceling signal is directly synthesized from the binary data, using a transversal filter approach, and the usual multiplications are replaced by additions and subtractions, thus allowing efficient operation of a large number of taps as required for the canceling of distant echoes. As a specific application, a system processing one sample per baud is discussed where timing signals at both communicating stations are assumed to be synchronized. A stochastic adjustment gradient-type algorithm is used for both training and adaptive tracking of the canceler. It is shown that convergence does not depend on intersymbol interference, timing phase, carrier phase, or the energy ratio of the local to the received signal, but is a function only of the number of taps. Convergence time is proportional to that number, and the optimum step size for fastest convergence is equal to the reciprocal of the number of taps. The residual fluctuation noise is proportional to that part of the mean-square (MS) error which cannot be reduced by the canceler and is a simple function of the product of the tap signal and the step size. The predicted convergence properties are verified by simulation results. Finally, it is shown how such an echo canceler might be used to allow two-wire full-duplex transmission for data rates as high as 4800 bit/s.     

A new double-talk detection algorithm based on the orthogonalitytheorem

The algorithm not only prevents the echo canceler from being disturbed by double talking but also tracks the echo path variations. Although the algorithm requires more computation and storage than conventional algorithms, excellent double-talk interference protection performance and echo path tracking have been obtained     

LMS coupled adaptive prediction and system identification: astatistical model and transient mean analysis

The LMS algorithm has been successfully used in many system identification problems. However, when the input data covariance matrix is ill-conditioned, the algorithm converges slowly. To overcome the slow convergence, an adaptive structure is studied, which incorporates an LMS adaptive predictor (prewhitener) prior to the LMS algorithm for system identification (canceler). Since the prewhitener is also adaptive, the input to the LMS canceler is nonstationary, even when the input is stationary. Because of the coupling and the nonstationarity of LMS canceler input, analysis of the performance of the two adaptations is extremely difficult. A simple theoretical model of the coupled adaptations is presented and analyzed. First and second moment analysis indicates that the adaptive predictor significantly speeds up the LMS canceler as compared to a system without prewhitening and enlarges the stability domain of the canceler (larger allowable μ). Monte-Carlo simulations are presented which are in good agreement with the predictions of the mathematical model     

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     

Echo cancellation in the telephone network

Most long-distance telephone connections generate echoes, which must be heavily attenuated in order to obtain satisfactory transmission quality. Voice-actuated switches (echo suppressors) are widely used to eliminate echoes but have an unfortunate tendency also to cut out part of the desired signal from the other end of the line. Because the distortion caused by echo suppressors is particularly noticeable on satellite-routed connections, the advent of telephone communication via satellite, including the recent introduction of satellite circuits into the U.S. domestic network, has motivated the search for a better way to eliminate echoes. The answer may be the echo canceler, an adaptive filter which selectively eliminates echoes. Advanced echo canceler designs have been undergoing field trials in recent years. This article explains why echo cancelers are advantageous and how they work.