Adaptive Echo Cancellation Research Articles

A family of variable step-size sparsity-aware SSAF algorithms with individual-weighting-factors under model-driven method

Recently, the sparsity-aware sign subband adaptive filter algorithm with individual-weighting-factors (S-IWF-SSAF) was devised. To accomplish performance enhancement, the variable parameter S-IWF-SSAF (VP-S-IWF-SSAF) algorithm was developed through optimizing the step-size and penalty factor, respectively. Different from the optimization scheme, we devise a family of variable step-size strategy S-IWF-SSAF (VSS-S-IWF-SSAF) algorithms based on the transient model of algorithms via minimizing the mean-square deviation (MSD) on each iteration with some reasonable and frequently adopted assumptions and Price's theorem. And in order to enhance the tracking capability, an effective reset mechanism is also incorporated into the proposed algorithms. It is worth mentioning that the presented algorithms could acquire lower computational requirements and exhibit higher steady-state estimation accuracy obviously and acceptable tracking characteristic in comparison to the VP-S-IWF-SSAF algorithm. In addition, the stable step-size range in the mean and mean square sense and steady-state performance are concluded. And the computational requirements are exhibited as well. Monte-Carlo simulations for system identification and adaptive echo cancellation applications certify the proposed algorithms acquire superior performance in contrast to other related algorithms within various system inputs under impulsive interference environments.

Performance Evaluation of Selected Adaptive Filters for Acoustic Echo Cancelation over Voice over Internet Protocol

Advancement of technology has increased the value of the telecommunications industry. One of the services with the rapid growth is Voice over Internet Protocol (VoIP). However, echo is a common disturbing effect in VoIP calls. Instead of suppressing the presence of echo in the VoIP communication channel, echo cancellation works better in full-duplex mode. This paper compares performances of some selected adaptive filters for cancelation of acoustic echo in VoIP channel. The framework for adaptive echo cancellation is built around finite impulse response and fourth-order infinite impulse response Chebyshev type II filters for the modeling of echo and acoustic environment, respectively. Selected adaptive algorithms for analysis are least mean square, recursive least square and frequency domain adaptive filter. The analysis is carried out in MATLAB 2018a environment. Echo return loss enhancement, error estimate and convergence rate are the performance indices employed for performance comparison of selected adaptive algorithms. It was found that the FDAF algorithm performed best based on computed error return loss enhancement and error estimate figures. In addition, FDAF converges faster than LMS and RLS with average time of 0.026431 s over 2048 filter lengths. Thus, FDAF shows a better promise over the two variants of time domain adaptive algorithm (LMS and RLS) analyzed in this paper.

Howling Detection and Suppression Based on Segmented Notch Filtering.

The existing adaptive echo cancellation based howling (typically in hearing aids) removal methods have several drawbacks such as insufficient attenuation of the howling component, slow response and nonlinear distortion. To solve these problems, we propose a segmented notch filtering based scheme. Specifically, firstly, it is proved that the attenuation value can reach dB at any detected howling frequency; secondly, the filter coefficients can be readily calculated by a closed-form formula, yielding a fast response to the sudden howling accident; thirdly, the closed-form formula of this filter is theoretically an even function, indicating that this filter possesses a linear transfer characteristic. In combination with proper segmentation and precisely removing these transient samples arising from FIR (Finite Impulsive Response) filtering, nonlinear distortion can be entirely avoided. Experimental results show that our proposed scheme can not only accurately estimate the howling frequency, but can also completely remove it, which yields a high-quality output waveform with a recovery SNR of about 22 dB. Therefore, the proposed segmented notching based scheme possesses vast potential for hearing aid development and other relevant applications.

Residue monitor enabled charge-mode adaptive echo-cancellation for simultaneous bidirectional signaling over on-chip interconnects

The full-duplex data communication over on-chip global interconnects suffers from large amounts of amplitude residue echo, leading to the closure of the vertical eye. This work proposes, a residue monitor enabled adaptive echo-cancellation scheme for full-duplex communication with power efficient charge-mode circuits. The proposed architecture employs charge-domain echo cancellation scheme for separating the outbound signal from the inbound signal by precisely controlling the capacitor bank output, thanks to the availability of residue monitor and high precision MOS switches and capacitors at sub-100 nm technology nodes. The residue monitor in the loop tracks the amount of amplitude residue and then sets the control bits, when the loop is locked. Further, this work proposes power efficient charge-mode circuits based replica generation stage, which does not contribute to additional power consumption like in the conventional echo-cancellation schemes. The proposed scheme is implemented at the architectural level with behavioural modeling in Verilog-AMS and also at the circuit level in 1.2 ​V, 65-nm CMOS Technology. The behavioural simulations shows that the residue monitor loop is getting locked while tracking the control voltage generated by the charge regulator for the target near-zero amplitude residue and accordingly control bits to the capacitor bank are set. Circuit level performance shows the functioning at a bidirectional data rate of 20 Gbps over a 1-mm on-chip global interconnect and the extracted received signal has 149 ​mV differential swing at a power consumption of 9.64 ​mW.

Bias-Compensated Sparsity-Aware NLMM Algorithms for Robust Adaptive Echo Cancellation

For adaptive echo cancellation in hands-free communication systems, a family of bias-compensated sparsity-aware normalized least mean M-estimate (NLMM) algorithms is proposed that are robust to both impulsive noise and noisy inputs. First, we define a new cost function that integrates the M-estimate function of the a posteriori error, the Riemannian distance between the current and previous weight vectors, and the $L_{0}$ norm of the weighted current weight vector. By minimizing the defined cost function, we develop the sparsity-aware NLMM algorithms to achieve robustness against the impulsive noise. Then, an improved cost function is developed and proved to be unbiased in the presence of input noise. This leads to the development of the bias-compensated sparsity-aware NLMM algorithms that can deal with noisy inputs. The stability and computational complexity of the proposed algorithms are analyzed as well. Experimental results for system identification and adaptive echo cancellation demonstrate that the proposed algorithms outperform the existing ones in terms of convergence rate and steady-state misalignment.

A 32-Gb/s Simultaneous Bidirectional Source-Synchronous Transceiver With Adaptive Echo Cancellation Techniques

Increased bandwidth per pin is necessary to support the growing data throughput in mobile systems. This article presents a low-power simultaneous bidirectional (SBD) transceiver employing a novel voltage-mode driver with a resistor-transconductance (R-gm) hybrid subtractor to enable concurrent transmission and reception of data on a single differential channel. A finite impulse response (FIR) filter-based adaptive echo cancellation system that cancels both near- and far-end echoes allows for the support of a wide range of channels, while a continuous-time linear equalizer (CTLE) efficiently compensates for channel loss. The quarter-rate source-synchronous transceiver utilizes a low-complexity phase interpolator-based clock and data alignment system to set the optimum data sampling point. Fabricated in 28-nm CMOS, the 32-Gb/s SBD transceiver achieves 1.83 mW/Gb/s and compensates 10.2-dB loss at a bit error rate (BER) < 10−12.

Affine Projection Versoria Algorithm for Robust Adaptive Echo Cancellation in Hands-Free Voice Communications

In vehicle hands-free communication systems and video teleconferencing systems, echoes are typically encountered. Adaptive filters are usually employed to remove the echoes in these applications. However, large outliers and highly correlated speech input signals are two key factors that limit the performance of the adaptive filters. In this paper, we propose an affine projection Versoria (APV) algorithm, which is obtained by maximizing the summation of Versoria-cost reusing with a constraint on the square of the L 2-norm of the filter weight vector difference. In this way, the features of the Versoria-cost maximization and data reusing are combined and, consequently, the proposed APV algorithm obtains robustness to the large outliers and accelerates the convergence rate for the correlated input signals. The complexity of the proposed APV algorithm is analyzed and then a fast recursive filtering technique is introduced to reduce its complexity. A stability analysis proves that the proposed APV algorithm is convergent. In addition, an analytical expression of the steady-state excess mean-square error for the proposed APV algorithm has been derived and simulated results are in agreement with the theoretical analysis result. Simulations on echo channel estimation and echo cancellation show that the proposed APV algorithm performs much better than the maximum Versoria criterion, affine projection sign, and affine projection algorithms.

A Full-Duplex Line Driver for Gigabit Ethernet With Rail-to-Rail Class-AB Output Stage in 28 nm CMOS

This paper details a duplex architecture coupled with a linear Class-AB push-pull output stage that maximizes the power efficiency of linear wideband drivers for high-speed transceivers. The duplex driver merges transmission, reception, and termination; eliminates hybrid and termination overhead; and enables adaptive echo cancellation and rail-to-rail full-duplex operation. Implemented in 28 nm CMOS, this GPHY driver passes the 1000BASE-T and 100BASE-TX compliance tests using a 2.5 V supply and meets the transmit specifications for the half-duplex 10BASE-T Ethernet using a 3.3 V supply.

A Robust Adaptive Acoustic Echo Cancellation (AEC) for Hands-free Communications using a Low Computational Cost Algorithm in Frequency Domain

This study presents a new algorithm for cancelling the acoustic echo, which is a major problem for hands- free communications. The proposed adaptive Acoustic Echo Canceller algorithm is designed and developed using a digital signal processing technique in frequency domain. The main scope of this study is to implement this module, benefiting the advantage of circular convolution properties and Fast Fourier Transform (FFT) with high computation speed in frequency domain rather than adaptive algorithms Normalized Least Mean Square (NLMS) and Recursive Least Square (RLS) in time domain with high complexity, also the simplicity of the implementation using SIMULINK programming. The results obtained at the simulation level prove the module behavior for cancellation of echo for hands free communications using adaptive algorithm frequency domain. Nevertheless, our algorithm shows more performances in terms of convergence and complexity. Our algorithm has been verified using the ERLE criteria to measure the attenuation of the echo signal at the output of an AEC; at this level we obtained the best values according to IUT-T recommendation G.168.

Gradient Based Adaptive Algorithm for Echo Cancellation from Recorded Echo Corrupted Speech

An offline single channel acoustic echo cancellation (AEC) scheme is proposed based on gradient based adaptive least mean squares (LMS) algorithm considering a major practical application of echo cancellation system for enhancing recorded echo corrupted speech data. The unavailability of a reference signal makes the problem of single channel adaptive echo cancellation to be extremely difficult to handle. Moreover, continuous feedback of the echo corrupted signal to the input microphone can significantly degrade the quality of the original speech signal and may even result in howling. In order to overcome these problems, in the proposed scheme, the delayed version of the echo corrupted speech signal is considered as a reference. An objective function is thus formulated and thereby a modified LMS update equation is derived, which is shown to converge to the optimum Wiener-Hopf solution. The performance of the proposed method is evaluated in terms of both subjective and objective measures via extensive experimentation on several real-life echo corrupted signals and very satisfactory performance is obtained.

부대역 비선형 Volterra 적응필터의 응용과 성능분석

본 논문에서는 부대역 신호를 사용한 병렬 적응 비선형 Volterra 필터를 소개하고, 이 시스템의 성능을 분석하는 것을 주요 내용으로 한다. 연구 결과 제시한 부대역 신호를 사용한 적응 Volterra 필터는 수렴성이 우수함을 입력신호의 상관함수의 eigenvalue 분포를 사용하여 해석적으로 도출되었으며, 이러한 응용이 최근에 보고된 적응 반향 제거기에서 유용하게 사용될 수 있음을 이론적으로 밝혔다. 또한 각 부대역 에서의 최적필터를 이론적으로 유도하였고 컴퓨터 모의실험으로 이를 검증하였다. In this paper, the subband nonlinear adaptive Volterra filters are introduced and its analysis are presented. From the eigenvalue analysis of the input correlation matrix, we show that the proposed subband adaptive Volterra filter has superior convergence performance as compared to the conventional one, which shows that the it can be useful for the recently proposed subband nonlinear adaptive echo canceller. Also, the optimum filter in each subband are introduced and verified from the computer simulations.

Performance Enhancement of Adaptive Acoustic Echo Canceller Using a New Time Varying Step Size LMS Algorithm (NVSSLMS)

This paper presents a new method for variable step size for LMS algorithm. The proposed algorithm is based on an absolute mean of estimation current and prior error vector. It is called New Varying Step Size LMS Algorithm (NVSSLMS). The main goal of this algorithm is performance enhancement of adaptive echo cancellation system. The proposed time varying step size method begins the learning process with high learning rate value ( µ MAX ) and then , it decays in an exponential profile to its minimum value (µMIN ) .The proposed algorithm is tested with real speech input signal and the result shows that it has fast convergence time, low level of misadjustment , and high Echo Return Loss Enhancement (ERLE) compared with LMS and other Variable Step Size LMS algorithm (VSSLMS) which proposed by R.H. Kwong and E.W. Johnston under similar conditions. The amount of ERLE enhancement using proposed algorithm compared with LMS and VSSLMS algorithms is about 10 dB and 8 dB respectively

Adaptive Echo Cancellation Combined with Delay Estimation

This paper proposed an echo canceller to reduce the computational complexity of the long delay echo cancellation. Two adaptive filters were used for constructing this echo canceller. One was designed as a delay estimator, which used down-sampled sub-band signals to estimate the echo delay time. The other was designed as a short-tap filter to subtract the echo from the recorded signal with delay compensation. Experimental results showed that the output signal-to-noise ratio from the proposed canceller with low complexity is similar to that from the conventional canceller based on a long-tap filter.

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.

Frequency Domain Echo Canceller for DMT-Based Systems

This letter addresses echo cancellation for digital subscriber line (DSL) communications with discrete multitone (DMT) modulation. The described adaptive echo canceller operates entirely in the frequency domain. The method allows estimation and subtraction of an echo signal impaired by intercarrier and intersymbol interference present when the impulse response of the echo channel is longer than the cyclic prefix. Implementation aspects for symmetric and asymmetric DMT-based systems are described together with ways to lower the computational complexity. Finally, computer simulation results are presented that validates the proposed method.

A New Variable Step-Size NLMS Algorithm and its Performance Evaluation in Echo Cancelling Applications

In this work we introduce a variable step-size normalized LMS algorithm for adaptive echo cancellation in a FIRstructure. In the proposed scheme, the step-size adjustment is controlled by using the square of the cross-correlationbetween the squared output error and the adaptive filter output. The proposed algorithm (that we call VSSSC aftervariable step size based on the squared cross-correlation) was evaluated using white noise and speech signals.Simulation results show that our proposal achieves better performance than similar algorithms in single and doubletalk. The proposed algorithm can be used in a number of applications such as in echo reduction for long-haul voicecommunications.

A New Echo Cancellation Algorithm without Double-Talk Detection Based on Correlation Function

In an adaptive echo canceller, the detection algorithm able to distinguish echo path change (EPC) from double-talk (DT) is vital to ensure that adaptive filter tap coefficients are updated in case of EPC and frozen during the DT period. The paper presents a new echo cancel algorithm, which can protect the adaptive filter performance during double-talk in acoustic echo cancellation of teleconference without setting a detector. A judgment value can be directly used in the iteration formula to control the iteration speed of the filter, which composed of the correlation of the far-end signal and near-end received signal, the pre-correlation of the error signal. The computer simulation results verify that the mentioned algorithm has the good double talk protection performance, and it is very useful and efficient in distinguishing EPC from DT but with less computational complexity contrast to the congener algorithm.

AUDIO QUALITY IMPROVEMENT OF VEHICULAR HANDS-FREE COMMUNICATION USING VARIABLE STEP-SIZE AFFINE-PROJECTION ALGORITHM

For a vehicular hands-free communication system, the sound quality of communication is usually degraded by noise which is known to be detrimental to system performance. In this paper, a novel adaptive filtering algorithm and an integrated system for acoustic echo and noise cancellation are presented. The proposed system includes adaptive noise cancellation, line enhancer, and echo cancellation which are based on a variable step-size affine-projection algorithm (VSS APA). The proposed VSS APA filtering algorithm is a combination of a variable step-size least-mean-square (VSS LMS) and an affine-projection algorithm (APA). The matrix of the APA allows more accurate, thorough input data and transforms the data into the structure of orthogonality, thus making the estimate of the weight vector faster and more accurate. To understand and verify the effectiveness of the proposed system, performance evaluation and comparison were conducted to compare the proposed algorithm and various traditional adaptive filtering algorithms in this application. The results demonstrated that the VSS APA has an effective performance and convergence in sound quality improvement of hands-free communication systems.

DTD-free nonlinear acoustic echo cancellation based on independent component analysis

An adaptive acoustic echo cancellation method based on independent component analysis, which can remove nonlinear echo components, is presented. Since this method takes into account the independence between a far-end signal and the echo-cancelled output to cancel acoustic echoes which are independent of near-end signals, it can continue to update AEC parameters even in double-talk situations, thereby improving the cancellation performance. Experiments show that the proposed method provides better performance than the conventional least-mean-square methods for both linear and nonlinear AEC systems without requiring separate double-talk detection (DTD).

Digital Adaptive Echo-Canceller for Room Acoustics Improvement

Digital Adaptive Echo-Canceller for Room Acoustics Improvement