Silencing echoes on the telephone network

Echo of a trasmitted signal can occur in the telephone network and is sometimes a disturbing phenomenon. Various methods of echo control have been used for over 50 years but recent introduction of transmission delays in excess of 300 ms, over satellite circuits, has resulted in a general reexamination of suppression methods. This led to development of the echo canceller. Adaptive echo cancellation is accomplished by automatically synthesizing a replica of the echo path response, convolving the incoming signal with the replica and subtracting the resulting echo estimate from the echo path output. Transmission is not thereby interrupted in any way and two-way information transmission proceeds normally with little or no echo. Detailed mathematical formulations for the convergence properties of such devices allow understanding of the effects of various hardware configurations and input signal properties. Echo cancellers may be applied to the network in a number of ways and the development of a VLSI canceller chip promises inexpensive implementation of these applications. Worldwide tests have shown the effectiveness of cancellation methods for satellite transmission.     

一种改进的自适应回波抵消器的设计与实现

回波是广泛存在于各种通信信道中的噪声干扰.传统的回波抵消器只能滤除缓变的、线性的回波干扰,对一些非线性且较复杂的信道产生的回波,滤波效果并不理想.本文通过比较融合两种经典的自适应算法,LMS算法和RLS算法各自的优点,提出了一种改进的自适应回波抵消器的结构,该回波抵消器已通过MATLAB仿真实现.从仿真的结果来看,该回波抵消器在对信道的兼容性,以及鲁棒性均优于传统回波抵消器.     

Residual echo signal in critically sampled subband acoustic echocancellers based on IIR and FIR filter banks

The residual echo signal characteristics of critically sampled subband acoustic echo cancellers are analyzed. For finite impulse response (FIR) filter banks, the residual echo signal usually has a relatively broad spectral nature around the subband edges. The residual echo signal of power symmetric infinite impulse response (PS-IIR) filter banks, on the other hand, has very narrowband spectral components around the subband edges. These components can be efficiently removed with PS-IIR notch filters that integrate neatly into the filter banks without introducing perceptually noticeable degradation to the near-end speech. This solution has very low computational complexity and does not impinge on the system performance. Simulation studies with recordings from the cockpit of a car, based on a fast QR least-squares adaptive algorithm, demonstrate the potential of this approach for a practical AEC system     

一种改进的与声码器结合的声学回声消除器

提出了一种改进的与线性预测声码器结合的声学回声消除器,通过增加一个自适应辅助滤波器,避免了对信号路径中所有信号在任何时刻都需经去相关、恢复相关过程所带来的信号质量降低、失调误差增加等负面影响。计算机仿真结果表明,该结构明显提高了已有算法的回声抵消量。     

Cost-Effective Joint Echo-NEXT Canceller Designs for 10GBase-T Ethernet Systems Based on a Shortened Impulse Response Filter (SIRF) Scheme

Echo canceller plays an important role in the full-duplex communication system. Conventional implementations of echo cancellers are often the adaptive transversal filter architectures due to the simplicity and robustness of stability and convergence. However, the conventional echo cancellers suffer from high cost problem especially when the response time of the echo is long. In this paper, a new cost-efficient architecture of echo cancellers, targeting on 10GBase-T Ethernet System, is presented. The proposed scheme inherits the concept of channel shortening which is widely employed in DSL systems. A shortened impulse response filter is implemented at the receiver to shorten the impulse response of the echo signal. Hence, the overall cost of echo cancellers can be reduced. We generalize the channel shortening architecture to a joint multi-channel shortening scheme. The joint multi-channel shortening architecture can be applied to multiple-input multiple-output wireline communication systems to further reduce both the cost of echo and near-end crosstalk (NEXT) cancellers. We apply the proposed scheme to 10GBase-T Ethernet system. The simulation results show that the proposed echo and NEXT cancellers can save up to 35% hardware cost compared to the conventional transversal implementations.

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     

电话语音回声消除的研究

为了消除电话语音系统中的反射回声,本文通过对电话信道的研究,提出通过估算信道冲击响应,得到提示语音经过信道卷积后的结果,从而抑制回声的算法.与LMS自适应算法相比,它可获得很好的回声抑制和对原始信号的保真效果.回声抵消后,语音识别正确率和准确率分别提高约29%和60%.     

A Phase Adaptive Structure for Echo Cancellation

Echo cancellation as an effective technique for echo control on telephone circuits is burdened by the appearance of carrier frequency offsets and phase jitter in some echo paths. In this paper we present a digital phase-adaptive structure for cancellation which will automatically compensate for echo path frequency offsets as large as 0.001 of the sampling rate without any reduction in return/loss enhancement.     

An improved adaptive power line interference canceller for electrocardiography

Power line interference may severely corrupt a biomedical recording. Notch filters and adaptive cancellers have been suggested to suppress this interference. We propose an improved adaptive canceller for the reduction of the fundamental power line interference component and harmonics in electrocardiogram (ECG) recordings. The method tracks. the amplitude, phase, and frequency of all the interference components for power line frequency deviations up to about 4 Hz. A comparison is made between the performance of our method, former adaptive cancellers, and a narrow and a wide notch filter in suppressing the fundamental power line interference component. For this purpose a real ECG signal is corrupted by an artificial power line interference signal. The cleaned signal after applying all methods is compared with the original ECG signal. Our improved adaptive canceller shows a signal-to-power-line-interference ratio for the fundamental component up to 30 dB higher than that produced by the other methods. Moreover, our method is also effective for the suppression of the harmonics of the power line interference.     

Timing Jitter Effects on Digital Subscriber Loop Echo Cancellers: Part I--Analysis of the Effect

We analyze the effect of Subscriber-end timing recovery Circuit jitter on the performance of two types of adaptive echo cancellers that can be used for full-duplex digital transmission on two-wire subscriber loops. Under severe echo-to-far-end signal ratios, echo canceller performance is found to be quite sensitive to high-frequency jitter components. Satisfactory performance with respect to jitter requires that a narrow-band phase-locked loop, rather than a single-tuned high-Qfilter, be employed for timing recovery.     

基于多相滤波器组的非线性回声抵消

在免提通话系统和移动通信设备中，扬声器常常工作在较高的音量下，容易发生过载现象，从而产生明显的非线性声学回声，这在小微型扬声器中更加常见。常用的线性AEC(Acoustic Echo Cancellation)算法无法消除此类非线性回声，因此通话质量受到严重影响。非线性回声主要表现为额外的高频谐波分量，这些分量使得全带系统不再满足线性关系，而通常的AEC算法都是基于最小化全带误差推导而来，因此性能很容易受到非线性失真的影响。本文提出了一种基于多相滤波器组的子带AEC算法，把全带误差变成了各个子带的误差，因而把谐波失真成分变成了某些子带内的加性噪声，这使得谐波失真较小的那些子带依然能够正常收敛。通过仿真和实测实验，当出现非线性失真时，新方法的ERLE(Echo Return Loss Enhancement)明显高于经典的全带时域和频域方法，对于非线性失真明显的语音信号，ERLE提升约10 dB。      

Binary Passband Echo Canceller in a 4800 Bit/s Two-Wire Duplex Modem

This paper deals with the echo cancellation applied to highspeed (4800 bit//s or higher) voiceband data modems. Simple echo cancellers are presented for the whole range of data bit rates from 1200 bit//s to 9600 blt/s. Their structure full=y integrates the modulation process and especially exploits the various relationships existing between the modulation rate and the carrier frequency. The generation of the echo replica is performed by duplicating components of the line echo from simple coded binary symbols processed at the modulation rate only. The implementation of these binary passband echo cancellers can be made quite simple by properly selecting the coded binary symbols from the user's data. The application of these principles to a 4800 bit/s two-wire duplex modem is then highlighted. The essential characteristics of the modem echo canceller are outlined and its implementation with bit slice processors is described. Finally, test results on the switched telephone network and on satellite links are presented.     

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     

A data-driven multitone echo canceller

A data-driven echo canceller for full-duplex data transmission with multitone modulation is presented. This multitone echo canceller (MTEC) is not impaired by eigenvalue-spread problems that are inherent in signal-driven echo cancellers-it has numerical performance and cancellation range that equals, and in most cases exceeds, that of data-driven echo cancellers used in data transmission with baseband or quadrature amplitude modulation. The method makes use of frequency-domain updating, but time-domain implementation of the canceller. It introduces no delay into the received signal path and presents no special difficulties for interframe interpolation between near-end echo, far-end echo, or far-end data signal. The authors also investigate the fast initialization and the special case of far-end frequency offset in the echo signal     

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     

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     

Cascade configurations for recursive-like adaptive noise cancellation

Two novel recursive-like two-stage adaptive noise cancellers that circumvent the requirement, in prior art one-stage recursive-like cancellers, of incorporating a constraint on filter coefficients, are presented. These novel two-stage cancellers are based on a cascade configuration, rather than the prior art parallel configuration. For applications in which even a small amount of signal distortion is intolerable, a third novel two-stage noise canceller that guarantees distortion-free performance is presented. Finally, a multistage cascade configuration that has the potential for distortion-free, high-performance noise cancellation is presented.     

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.     

一种基于软判决的立体声回声抵消算法

提出了一种新的基于维纳滤波的频域算法来解决立体声回声抵消问题,该算法不需要对立体声信号预处理,从而最大程度地保证了近端语音质量。并且它具有很好的鲁棒性,很快的收敛速度和跟踪速度,因而具有一定的实用价值。引入了语音增强中的软判决方法来进一步提高算法的性能。新的算法在保证近端语音质量的同时达到了更好的回声压制效果。仿真实验证明了新算法的良好性能。