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     

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     

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.     

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。     

A flexible adaptive FIR filter VLSI IC

The architecture and features of the Motorola DSP56200 are described. The DSP56200 is an algorithm-specific cascadable digital signal processing peripheral designed to perform the computationally intensive tasks associated with finite impulse response (FIR) and adaptive FIR digital filtering applications. The DSP56200 is implemented in high-performance, low-power 1.5-μm HCMOS technology and is available in a 28-pin DIP package. The on-chip computation unit includes a 97.5-ns 24-bit×24-bit coefficient RAM, and a 256-bit×16-bit data RAM. Three modes of operation allow the part to be used as a single, dual, or single adaptive FIR filter, with up to 256 taps per chip. In the adaptive mode, the part performs the FIR filtering and least-mean-square (LMS) coefficient update operations for a single tap in 195 ns, permitting use of the part as a 19-kHz sampling rate, 256-tap adaptive FIR filter. A programmable DC tap, coefficient leakage, and adaptation coefficient parameters in the adaptive FIR mode allow the DSP56200 to be used in a wide variety of adaptive FIR filtering applications. The performance of the part in an echo canceler configuration is presented. Typical applications of the part are also described     

The Wheatstone bridge-based analog adaptive filter with application in echo cancellation

In this paper, a new analog adaptive filter is introduced with application in adaptive echo cancellation namely, the Wheatstone bridge-based analog adaptive filter (WAAF). It is proved the WAAF is a variable weight analog IIR filter. IIR filter weights vary with gate-source voltage control of a MOSFET transistor in triode region. The best balance point control of the WAAF is achieved using least mean square (LMS) algorithm. It is proved that analog LMS algorithm converges faster than digital LMS adaptive filter. The superiority of the proposed WAAF is observed in the designing process, computational cost, convergence speed and real time operation. Also, experimental results show ability of the proposed WAAF in the hybrid circuit of the telephone echo cancellation.     

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.     

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.     

Improvement in adaptive noise canceling using a nonlinear filter based on the pontryagin minimum principle

This paper introduces a new nonlinear filter that is used for adaptive noise canceling. The derivation and convergence properties of the filter are presented. The performance, as measured by the root mean square error between the signal and its estimate, is compared with that of the commonly used least mean square (LMS) algorithm. It is shown, through simulation, that the proposed nonlinear noise canceler has, on the average, better performance than the LMS canceler. The proposed adaptive noise canceler is based on the Pontryagin minimum principle and the method of invariant imbedding. The computational time for the proposed method is about 10% of that of the LMS, in the studied cases, which is a substantial improvement.     

Adaptive multichannel lattice-escalator filter structure: anapplication to generalized sidelobe canceler

An adaptive digital filter structure which can be used in multichannel noise-canceling applications is described. The proposed structure is obtained by creating a hybrid from the lattice and the escalator (Gram-Schmidt) structures, with the coefficients being updated using the least mean square (LMS) algorithm. As an application, the proposed adaptive multichannel digital filter is applied to implement an adaptive generalized sidelobe canceler     

VoIP回声消除器设计及算法研究

本文提出了一种与线性预测编解码器相结合的新声学回声消除器,由去相关可变步长的NLMS自适应算法和基于回声路径失配方差的双端通话检测算法所组成.Matlab仿真结果表明,与Gordy所提出的回声消除算法相比,本文提出的算法在双端通话和回声路径改变时判别更准确,收敛速度更快;在收敛状态时,ERLE值平均提高了15dB,失调误差平均降低了10dB,具备更好的回声消除性能.     

A stereo echo canceler with correct echo-path identification basedon an input-sliding technique

A new stereo echo canceler with correct echo-path identification based on an input-sliding technique is proposed. A time-varying filter located in one of the two channels periodically delays the input signal. By this input sliding, the correct echo-path identification is achieved. Aliasing components and audible clicks by input-sliding are made inaudible by selecting appropriate parameter values for the time-varying filter. Simulations with the NLMS algorithm and a white Gaussian signal confirm the correct echo-path identification. The subjective quality of the input signal with slides is 4.38 based on the ITU-R five-grade impairment scale. Experimental results based on an implementation by 32-bit floating-point digital signal processors show that ERLE is not degraded by talker changes in the remote room. The mean opinion score is as much as 0.55-point higher than the conventional stereo echo canceler for different round-trip delays     

Dual Sign Algorithm for Adaptive Filtering

A new algorithm, which is a variant of the sign algorithm, is proposed for the adaptive adjustment of an FIR digital filter with an aim of improving the original convergence characteristics, yet retaining the advantage of hardware simplicity. Based on a recently proposed theory for the sign algorithm, a practical design method is derived for the new algorithm, and it is shown by computer simulation that the new algorithm in fact performs significantly better than the original algorithm.     

用于改变图像大小的M带插值滤波器设计

改变图像尺寸是多速率系统的重要应用之一。提出了一种通过采样率转换改变图像尺寸的有效方法，给出了线性相位FIR插值滤波器和线性相位FIR抗混叠滤波器的设计算法。结合实例说明了方法的有效性。     

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.     

A fast convergence algorithm for sparse-tap adaptive FIR filters identifying an unknown number of dispersive regions

This paper proposes a fast convergence algorithm for sparse-tap adaptive finite impulse response (FIR) filters to identify an unknown number of multiple dispersive regions. Coefficient values and tap-positions of the adaptive filter are simultaneously controlled. A constrained region for new-tap positions is selected from equisize subgroups of all possible tap-positions, and it hops from one subgroup to another to cover multiple dispersive regions. The hopping order and the stay time for each subgroup are adaptively determined based on the absolute coefficient values. Simulation results with colored signals show that the proposed algorithm saves more than 80% in the convergence time over the full-tap NLMS and 50% over the STWQ. Tracking capability of the proposed algorithm exhibits its superior characteristics. These characteristics are confirmed by hardware evaluations with a telephone network simulator.     

Design of FIR filters with extra constraints using modified LMSalgorithm

An adaptive approach to the design of linear phase low-pass FIR filters with extra constraints on filter coefficients is presented. In this approach, the procedures using the LMS adaptive algorithm are modified to include the constraints on the filter coefficients. Numerical examples are presented and compared to the results obtained using least-square design in the frequency domain in which the filter design problem is transformed into an equivalent nonlinear optimisation problem     

Analytical model for the first and second moments of an adaptiveinterpolated FIR filter using the constrained filtered-X LMS algorithm

The authors present an analytical model for the mean weight behaviour and weight covariance matrix of an adaptive interpolated FIR filter using the LMS algorithm to adapt the filter weights. The particular structure of this adaptive filter determines that special analytical considerations must be used. First, the introduction of an interpolating block cascaded with the adaptive sparse filter requires that the input signal correlations must be considered. It is well known that such correlations are disregarded by the independence theory, which is the basis for the analysis of the LMS algorithm adapting FIR structures. Secondly a constrained analysis is used to deal mathematically with the sparse nature of the adaptive section. Experimental results demonstrate the effectiveness of the proposed analytical models as compared with the results obtained by classical analysis     

一种低复杂度的稀疏FIR陷波滤波器的设计方法

FIR陷波滤波器具有线性相位、精度高、稳定性好等诸多优势,然而当陷波性能要求较高时,通常需要较高的阶数,导致FIR陷波滤波器硬件实现复杂度大大提高。该文基于稀疏FIR滤波器设计算法和共同子式消除的思想,提出一种低复杂度的FIR陷波滤波器设计方法。该方法首先采用稀疏滤波器设计算法得到满足频域性能设计要求的FIR陷波原始滤波器系数,然后对其进行CSD编码,并分析CSD编码量化系数集中所有的2项子式和孤子的灵敏度,最后根据灵敏度的大小依次选择合理的2项子式或孤子直接合成滤波器系数集。仿真结果表明,新算法设计实现的FIR陷波滤波器比已有的低复杂度设计方法最多可减少51%的加法器,有效地降低了硬件实现复杂度,大大节省了硬件资源。     

Adaptive Echo Cancellation Based on a Multipath Model of Acoustic Channel

Usually Acoustic Echo Cancellers (AECs) are realized by adaptive Finite duration Impulse Response (FIR) filter having large number of coefficients and Least Mean Square (LMS) as an adaptive algorithm resulting in slow convergence speed and poor tracking performance of these adaptive filters. In this paper, we have proposed a Multiple Sub-filter (MSF) parallel structure based on multipath acoustic echo model using the basis that each sub-filter will compensate the echo contributed by each path of multipath acoustic channel. To realize the MSF, modified Generalized Autocorrelation-based Estimator (MAE) has been used to estimate time delay associated with each path while the order of each sub-filter has been estimated using Power Spectral Density (PSD) method. Accuracy Percentage (AP) performance measure has been used to characterize the performance of the estimator. Simulation results show that the performance of the MAE improves with the increase in SNR and/or decrease in number of multipath. Using these estimates MSF based AEC is constructed. The convergence performance of MSF-based AEC has been studied, via computer simulation, and compared with the conventional Single Long length adaptive Filter (SLF)-based canceller for different SNRs and number of multipath. The results of MSF have been found to be very encouraging in almost all of the various situations considered. Subsequently, the tracking behavior has also been studied with variation in the channel parameters of the multipath model. The proposed MSF can track variations in the channel parameters of the multipath model faster as compared to the conventional echo canceller.