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.     

The Effects of Large Interference on the Tracking Capability of Digitally Implemented Echo Cancellers

Adaptive mean-square-tapped-delay-line echo cancellers for voice applications are conventionally designed to stop adjustment during periods of "double-talking", i.e., when a large informationbearing signal is present along with the echo signal to be cancelled. Continuous adaption is, however, desirable in full-duplex, two-wire data transmission where the periods of double-talking are so long that the echo channel may vary. We presume that the tap weights of an echo canceller have converged during a training period free of double talking, and address the problem of subsequent echo-canceller tap adjustment via the estimated-gradient algorithm in the presence of double talking. In the estimated-gradient algorithm the tap increment should be proportional to the product of the residual echo and the tap voltage. However, when double talking occurs the residual echo can only be estimated. For an idealized double-talking model, it is demonstrated, from infinite-precision considerations, that use of the memoryless maximum-likelihood estimate of the residual echo is nearly equivalent to abrupt reduction of the step size of the adjustment algorithm when double-talking begins, and could provide an automatic mechanism for recognizing double-talking. Unfortunately, the response of a digitally implemented canceller to a sharply reduced step size can be a deterioration in performance. In fact, the use of an exceedingly small step size during periods of doubletalking may lead to a cancellation error considerably larger than that predicted by coefficient precision. It is demonstrated how averaging the estimated gradient can significantly decrease the mean-squared tap error during periods of double talking. To a first approximation, the tap-weight error can be reduced by a factor proportional to the averaging interval, with an equivalent decrease in tracking capability.     

Echo cancellation and applications

Practical echo cancellation techniques, in particular, those used in telecommunications, are reviewed. The various situations in which echoes are generated are examined. Echo path modeling techniques and adaptive algorithms for coefficient control are reviewed. Current international standardization activities are discussed, and echo canceler implementation considerations are set forth. These include echo cancelers for telephone circuits, echo cancelers for full-duplex data transmission over voice channels, acoustic echo cancelers, and echo cancelers for ISDN digital loop transmission     

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.     

A New Digital Echo Canceler for Two-Wire Subscriber Lines

A new type of digital echo canceler for two-wire digital transmission is presented. The new principle involves very simple signal processing and is thus an interesting alternative for digital transmission on subscriber lines. The principle is compared with other echo cancellation techniques, and it is shown how choice of line code, equalization, and carrier recovery are affected by the new echo canceler. A theoretical analysis of the principle is given, taking into account finite accuracy, jitter, noise, and correlated data streams. The echo canceler can be used for line attenuation up to 40 dB. At 80 kbits/s this corresponds to at least 7 km 0.6 mm cable and is sufficient to cover more than 99 percent of the existing Norwegian subscriber lines.     

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。     

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.     

Asynchronous and Timing Jitter Insensitive Data Echo Cancellation

An approach to the implementation of asynchronous and timing jitter insensitive data echo cancellation is described. This approach introduces a small amount of jitter in the transmitted data signal, or alternatively in the received signal sampling, and uses a simple digital phase-locked loop together with the storage of two sets of echo canceler coefficients. The effect of derived timing jitter on the echo cancellation accuracy is completely eliminated for a loop timed transceiver (as in a digital subscriber loop network termination transceiver), and is easily reduced to negligible levels for a nonloop timed transceiver (as in a digital subscriber loop line card transceiver or a voiceband data modem). In the case of a voiceband data modem, this approach is one method to achieve asynchronous echo cancellation without the need to recover and resample a continuous-time far-end data signal.     

Fast initialization of Nyquist echo cancelers using circular convolution technique

For full-duplex high-speed data transmission over the two-wire line using the same frequency band, it is required to sufficiently suppress the echo. The use of a conventional adaptation method may take a long time to train the echo canceler. Fast training can be achieved by initializing the coefficients of the echo canceler with an estimate of the impulse response of the echo path. We propose a method for fast initialization of the echo canceler by using a circular convolution technique. The proposed method enables the use of real-valued training signals instead of complex-valued ones, resulting in significant reduction of the initialization time as well as the implementation complexity. Finally, the performance of the proposed method is analyzed and verified by computer simulation.     

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     

A new configuration for echo canceller adaptable during double talkperiods

A three-port echo canceler (EC) configuration is proposed which observes the signal of the near-end side on a two-wire circuit in addition to the four-wire circuit signals. Embedding these signals on hybrid ports into a three-dimensional autoregressive process, echo path and innovations of near- and far-end speeches can be estimated through a three-channel lattice filter. The new configuration is then able to track echo path time variance, even during double talk (DT), and requires no changeover at either the beginning or end of DT, thus eliminating the need for DT detection. Two echo synthesizers utilizing inverse lattice and the echo path estimate possess guaranteed stability without the need for testing     

基于频域逐级回归的声学回声控制

传统声学回声控制算法一般采用基于随机梯度法更新的频域分块自适应滤波(PBFDAF)方法,但在以语音为主要回声信号的室内混响环境中,由于回声路径不稳定,往往收敛速度较慢,难以实现足够的回声抑制。该文提出一种基于频域逐级回归的声学回声控制算法。通过逐级回归分析远端信号和麦克风信号之间的线性关系,可以在保持较小的偏差的同时实现收敛较快的系统估计。同时,由于逐级分析了两通道间的短时相干性,因而该算法无需像常见方法一样,额外进行基于通道间相干函数的残余回声抑制或双讲检测,从而保持系统的紧凑性。若进一步假定近端背景噪声准平稳,则可利用基于近端信号非平稳程度的自适应平滑因子,在实现系统估计快速收敛的同时确保其稳定性。实验表明,该方法在常见的近端环境噪声水平下,在收敛速度和稳态误差上相对传统方法有显著优势,非常适合应用在室内远讲模式下的声学回声控制中。     

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。     

Echo cancellation in DMT-receivers: circulant decomposition canceler

Asymmetric digital subscriber lines (ADSLs) employ discrete multitone modulation (DMT) as transmission format, where subcarriers are assigned to the up- and/or downstream transmission direction. To separate up- and downstream signals, the ADSL standard allows the use of echo cancellation resulting in improved bit rates, reach, and/or noise margins. In DMT-based modems, typically, the mixed time/frequency (MTF) domain echo canceling scheme, as proposed by Ho et al., is implemented. This technique estimates the echo filter in the frequency domain using the least mean square (LMS) algorithm with the transmitted echo symbols as update directions. Since not every tone of the transmitted echo signal will carry data, i.e., will be excited, the MTF adaptation process does not lead to a good estimate for the echo channel, unless extra power on unused echo tones is transmitted. However, transmitting extra power on such tones is often undesired. In this paper, we present an alternative echo canceling scheme referred to as the circulant decomposition canceler (CDC), which works without extra power requirements and with comparable complexity as the method of Ho et al. Similar to MTF echo canceling, the CDC scheme can easily be incorporated into a multirate environment with different transmit and receive rates and can also cheaply be combined with per-tone equalization and double talk cancellation to allow fast tracking and/or convergence in the presence of a far-end signal.     

一种地基雷达回波模拟器设计与FPGA实现

本文研究了地基雷达的工作方式及其回波信号的基本组成原理,分析了目标回波、噪声、背景杂波以及干扰的数学建模方式和硬件实现方法,并且在Virtex-5 FPGA平台上将雷达回波模拟器实现。测试表明,该模拟器工作正确稳定。     

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     

Nonintrusive measurements of the telephone channel

Algorithms for a system which performs in-service nonintrusive measurements of telephone line parameters such as echo path delay, echo attenuation, speech and noise levels are presented. It forms the theoretical background of a hardware implementation of the ITU-T P.561 recommendations for nonintrusive assessment of quality of a telephone network for voice transmissions. The type of input signal to the echo path, whether it is speech, fax, or data, has to be determined during the measurement procedure. In the system, the relevant parameters are estimated by means of frequency-domain block-processing algorithms and the estimation process is controlled by algorithms monitoring the signal flow. The complexity of the echo path estimation algorithm is determined and compared to a standard procedure based on the adaptive normalized least mean square (NMLS) algorithm. Simulations showing the performance of the system in single-and double-talk situations are presented. Examples of results from the hardware real-time implementation operating on an international connection are also shown     

基于TMS320DM642实现的一种低复杂度回声抵消器

以TMS320DM642定点DSP为平台,实现了一种低复杂度并适应双端会话情况的回声抵消器.在算法上.为提高双端会话的鲁棒性,采用了双回声路径模型(two echo path moolel,TEPM)作为基本框架;为降低计算的复杂度,采用了多延时分块频域自适应滤波器(multi-delay block frequencydomain adaptive filter,MDF)估计回声路径的冲击响应.在实现上,针对DSP的并行执行能力和指令集的单指令多数据(SIMD)特性对主要运算进行了线性汇编级的优化.测试表明,这种回声抵消器可以在0.45%CPU占用率下处理128 ms路径延时的组合声源信号(composite source signal,CSS)和实际声源信号的回声.     

Double-Talk-Robust Prediction Error Identification Algorithms for Acoustic Echo Cancellation

The performance of an acoustic echo canceller may be severely degraded by the presence of a near-end signal. In such a double-talk situation, the variance of the echo path estimate typically increases, resulting in slow convergence or even divergence of the adaptive filter. This problem is usually tackled by equipping the echo canceller with a double-talk detector that freezes adaptation during near-end activity. Nevertheless, there is a need for more robust adaptive algorithms since the adaptive filter's convergence may be affected considerably in the time interval needed to detect double-talk. Moreover, in some applications, near-end noise may be continuously present and then the use of a double-talk detector becomes futile. Robustness to double-talk may be established by taking into account the near-end signal characteristics, which are, however, unknown and time varying. In this paper, we show how concurrent estimation of the echo path and an autoregressive near-end signal model can be performed using prediction error (PE) identification techniques. We develop a general recursive prediction error (RPE) identification algorithm and compare it to three existing algorithms from adaptive feedback cancellation. The potential benefit of the algorithms in a double-talk situation is illustrated by means of computer simulations. It appears that especially in the stochastic gradient case a huge improvement in convergence behavior can be obtained