Adaptive Echo Cancellation Research Articles

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.

Constrained Adaptive Echo Cancellation for Discrete Multitone Systems

In communication systems where full-duplex transmission is required, echo cancellers are deployed to cancel the interference of the transmitted signal at the collocated receiver. For systems using discrete multitone (DMT) modulation, echo cancellation is performed partially in the time and frequency domains to decrease the processing complexity. In this paper, echo cancellation for DMT systems is reformulated as a constrained optimization problem, where a cost function is minimized over an extended linear space. This extended space contains the weights of the finite-impulse-response (FIR) filter emulating the echo channel in the time and the frequency domains, while linear constraints are used to ensure the proper mapping between these two domains. Based on this proposed formulation, a new constrained adaptive echo cancellation structure for DMT-based digital subscriber lines (DSL) systems is proposed. The proposed formulation provides a unifying framework for different practical DSL systems (i.e., frame asynchronous and multirate), as well as additional flexibility in implementation by allowing the incorporation of supplementary constraints that can improve the performance of the system. As an illustrative example, we show how the robustness of the echo canceller can be improved in the presence of radio frequency interference by adding appropriate constraints on the extended linear space.

Low-Complexity Echo and NEXT Cancellers for High-Speed Ethernet Transceivers

Gigabit and multigigabit transceivers require very long adaptive filters for echo and near-end crosstalk (NEXT) cancellation. Implementation of these filters not only occupies large silicon area but also consumes significant power. These problems become even worse when the Tomlinson-Harashima precoding (THP) technique is used in applications such as 10-Gigabit Ethernet over Copper (10 GBASE-T) as the input to the echo and NEXT cancellers is no longer a simple PAM-M signal. To reduce the complexity of these cancellers, in this paper, a novel method based on word-length reduction technique is proposed. The proposed design is derived by replacing the original input to the echo and NEXT cancellers with a finite-level signal, which is the sum of the input to the TH precoder and a finite-level compensation signal. Then, this modified input signal is recoded to have shorter word-length compared with the original input. Hence, the overall complexity can be reduced by using the proposed method. To further reduce the complexity of these cancellers, an improved design is proposed by exploiting the property of the compensation signal. Compared with the traditional design, the proposed echo and NEXT cancellers have exact input and do not suffer from the quantization problem, and thus they are more suitable for VLSI implementation. The proposed method can also be applied to design adaptive echo and NEXT cancellers with little modification. The performance evaluation is performed by simulations to verify the proposed design. It is shown that, by applying the proposed method to a 10 GBASE-T Ethernet system, the hardware complexity of echo and NEXT cancellers can be reduced by about 10.82% without performance loss, compared with the traditional design.

An efficient method for combining adaptive echo and noise canceller in hands‐free systems

AbstractAcoustic interferences severely degrade the quality and the intelligibility of the desired speech signal, thus posing a severe problem for many speech applications. Several acoustic echo cancellation (AEC) techniques have been proposed with a view to solving this problem. There are, however, few reports of AEC methods working under real noisy conditions. In this paper, we investigate the maximum positive synergies of the combination of acoustic echo canceller with a new adaptive beamformer. The proposed system achieves both the AEC and noise reduction of speech in an actual environment with real noise sources. Since the AEC is located behind the fixed beamformer of the new adaptive beamformer only one AEC is required for an arbitrary number of array elements, and the AEC does not feel any repercussions from the new adaptive beamformer. The proposed system was implemented in a real environment using National Instruments NI‐PXI‐1042Q controller system and data acquisition card PXI‐4472. Experimental results show that the proposed system has successfully improved the performance of hands‐free systems. Copyright © 2008 John Wiley & Sons, Ltd.

Sub-band Implementation of Adaptive Nonlinear Filter for Adaptive Nonlinear Echo Cancellation

The adaptive Volterra filter has been successfully applied in nonlinear acoustic echo cancellation (AEC) systems and nonlinear line echo cancellation systems, but its applications are limited by its required computational complexity and slow convergence rate, especially for systems with long memory length. In this paper, we first apply a more general nonlinear filter, the function expansion nonlinear filter, in the acoustic echo cancellation - the Volterra filter can be regarded as special case of the function expansion nonlinear filter. Then by leveraging to a multi-channel configuration of the function expansion nonlinear filter and the sampling theory for nonlinear systems, we extend linear sub-band delay-less adaptive filter techniques to develop an efficient sub-band implementation of the adaptive function expansion nonlinear filter. The developed sub-band configuration of the adaptive nonlinear filter can greatly improve the convergence rate and reduce the computational complexity of nonlinear echo cancellers, which is shown by analyses and simulations.

Enhancing echo cancellation via estimation of delay

The advent of packetized audio transmission, such as voice over IP (VoIP), has resulted in challenging requirements for echo cancellation technology. One key aspect of this technology is the need to characterize, quickly and accurately, the echo paths in the transmission media. Echo paths consist of a constant time delay with no echo signal and active regions in which the echo signal is present. When an adaptive filter echo cancellation algorithm is used, its performance can be greatly increased, and its complexity can be reduced if it is only applied to the active regions. This requires an algorithm to estimate the constant delay and locate the active regions. Traditionally, delay estimation has been based on direct application of cross-correlation. This method has poor performance because the input signals are highly correlated and has a high implementation cost because many cross-correlation lags have to be computed for longer time delays. The delay estimation addressed in this paper has two major advantages over the traditional methods. The first is that it has improved performance because the input signals are processed to have less correlation. The second is that the implementation cost is significantly reduced because fewer cross-correlation lags are computed, and an efficient method to estimate lags is created.

Residual echo filter for enhanced acoustic echo control

This paper presents a method for efficient control of an acoustic echo control (AEC) system. The control is achieved with the help of so-called residual echo filter structure. The residual echo filter can be exploited to provide an accurate estimate of the norm of the system mismatch vector (sometimes also called as an echo coupling factor), which is an essential element for providing advanced control means for the adaptive echo cancellation filters. In addition to the adaptive filter a sophisticated AEC systems usually include echo suppression means to complete and ensure sufficient echo cancellation performance. The residual echo filter may also be utilized to estimate the power spectral density of the residual echo, which is needed to implement the suppression filter. The use of the residual echo filter structure overcomes some weaknesses in the ordinary AEC controlling means and provides enhanced overall AEC performance.

A Low-Complexity Adaptive Echo Canceller for xDSL Applications

A finite impulse response (FIR)-based adaptive filter structure is proposed for echo cancellation in xDSL applications. The proposed algorithm consists of an FIR filter, a cascaded interpolated FIR filter, and a tap-weight overlapping and nulling scheme. This filter requires low computational complexity and inherits the stable characteristics of the conventional FIR filter. Simulations show that the proposed echo canceller can effectively cancel the echo up to 73.4 dB [for a single-pair high-speed digital subscriber line (SHDSL) system]. About 55% complexity reduction can be achieved compared with a conventional FIR filter.

Local echo canceler with optimal input for true full-duplex speech scrambling 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.

Control of an adaptive echo canceller using a near-end speech detector

In the context of hands-free mobile telephony, acoustic echo cancellation (AEC) remains an important problem to deal with. This work addresses the study of an acoustic echo canceller controlled by a near-end speech detector. As a matter of fact, the echo canceller is an adaptive filter whose coefficients are disturbed by the presence of speech coming from the near-end speaker. Consequently, it is useful to stop the adaptation when this signal is detected to get a better estimation of the echo (far-end speaker). In this paper, a near-end speech detector based on the partial coherence – computed from the observations received by two microphones conditioned on the signal emitted by the loudspeaker – is investigated. To improve the performance of the AEC system, two approaches performing a saving of the AEC coefficients are proposed. The system is assessed through the power of the residual echo.

Acoustic echo cancellation using iterative-maximal-length correlation and double-talk detection

The conventional maximal-length-correlation (MLC) algorithm to estimate room impulse response for adaptive echo cancellation (AEC) is disturbed by both far-end and near-end speech. In this paper, a new iterative-maximal-length-correlation (IMLC) algorithm is proposed to reduce the far-end speech interference. To avoid the near-end interference, a new double-talk detection (DTD) method is proposed by tracking the squared coefficients errors of the AEC filter. This DTD method has well-separated detection margins among single-talk (ST), double-talk (DT), and echo path changes. Statistical analysis and computer simulations confirm that our proposed IMLC-DTD algorithm outperforms conventional methods.

Step-size control for acoustic echo cancellation filters – an overview

In this paper we present an overview about several approaches for controlling the step size for adaptive echo cancellation filters in hands-free telephones. First an optimal step size is derived. For the determination of this step size the power of a non-measurable signal has to be estimated. Detection and estimation methods for the determination of this power and for the determination of the optimal step size can be grouped into four classes. For each class, several principles, which differ in their reliability and in their complexity, are presented. Possibilities for combining elements from each class to create an entire step size control unit are also described. An outlook on detector combinations based on neuronal networks concludes this paper.

Gabor expansion for adaptive echo cancellation

A good echo cancellation algorithm should have a fast convergence rate, small steady-state residual echo, and less implementation cost. The normalized least mean square (NLMS) adaptive filtering algorithm may not achieve this goal. We show that using the Gabor expansion is a way to achieve this goal. For direct digital signal processing compatibility the Gabor expansion introduced in this paper is for discrete-time signals, although the Gabor expansion also can be used for continuous-time signals. The Gabor expansion can be defined as a discrete-time signal representation in the joint time-frequency domain of a weighted sum of the collection of functions (known as the synthesis functions). There are several design issues in the echo canceller based on the Gabor expansion: the design of the analysis functions for the far-end speech, the design of the analysis functions for the near-end signal containing the echo plus the near-end speech, the design of the adaptive filters in the subsignal path, and the design of the synthesis functions. All the adaptive filters are designed using identical NLMS adaptive filtering algorithms.

Hybrid LMS–LMF algorithm for adaptive echo cancellation

The coefficients of an echo canceller with a near-end section and a far-end section are usually updated with the same updating scheme, such as the LMS algorithm. A novel scheme is proposed for echo cancellation that is based on the minimisation of two different cost functions, i.e. one for the near-end section and a different one for the far-end section. The approach considered leads to a substantial improvement in performance over the LMS algorithm when it is applied to both sections of the echo canceller. The convergence properties of the algorithm are derived. The proposed scheme is also shown to be robust to noise variations. Simulation results confirm the superior performance of the new algorithm.

Telephone set in which an echo signal is effectively canceled in spite of being provided with a plurality of sound producing units

In a telephone set having a receiver (42) and a loudspeaker (39), a switch circuit (37) selects supply of a reception speech signal, transmitted from a counterpart section (60), to the receiver or the loudspeaker in response to a selection signal. A first adaptive echo canceller (43a) produces a pseudo echo signal with reference to the reception speech signal. A controller (44) controls operation of the first adaptive echo canceller in response to the selection signal to optimize the pseudo echo signal. A first digital adder (35a) carries out a particular calculation between the pseudo echo signal and a transmission speech signal, relating to the transmission sound supplied to a microphone (46). The adder produces a particular calculation result signal representative of a result of the particular calculation. In accordance with the particular calculation result signal, a transmission is carried out towards the counterpart section.

A quantitative convergence and tracking comparison of two adaptive echo-cancellation algorithms

The sign algorithm (SA) and the signed regressor algorithm (SRA) have become appealing in high data rate adaptive filtering applications such as speech processing and echo cancellation. The paper provides a quantitative comparison of the two algorithm convergence times for adaptive echo cancellation. The paper also compares the steady-state algorithm performances when the echo-path parameters are randomly time varying and the far-end signal is fading. The comparison is done for Gaussian near-end and far-end signals. An operating range is determined where the SA performs nearly as well as the SRA for a stationary echo path. It is also shown that the SA outperforms the SRA for a randomly time-varying echo path and a fading far-end signal.

Adaptive echo cancellation using least mean mixed-norm algorithm

A novel algorithm for echo cancellation is presented in this work. The algorithm consists of simultaneously applying the least mean square (LMS) algorithm to the near-end section of the echo canceller and the least mean fourth (LMF) algorithm to the far-end section. This new scheme results in a substantial performance improvement over the LMS algorithm and other algorithms.

A DSP-based lattice-pole-zero acoustic ECHO canceller

In this paper, a practical pole-zero lattice adaptive acoustic echo canceller (AEC) for hands-free telephone is proposed. The proposed algorithm consists in two parts: forward lattice and inverse lattice. Collectively, they are referred to as LATIN (Lattice and Inverse Lattice) configuration. While the forward lattice is responsible for acoustic echo cancellation, the inverse lattice is employed in the double-talk (DT) mode only as to undo the distortion of the near-end speech brought about by the forward lattice filter when suppressing the acoustic echo. Assuming M poles and M zeros for the proposed AEC, the complexity of the proposed algorithm is approximately twice the complexity of an M-tap FIR gradient lattice algorithm. Real-time experimentation conducted on a floating-point DSP chip and simulation results attest to the stability, fast convergence, and high acoustic echo suppression of the proposed pole-zero algorithm.

Stereo acoustic echo cancellation: A review

Adaptive echo cancelers have been used successfully for over a decade to combat echoes that arise on a telephone network due to mismatches in impedance at various points in the circuit. Echoes arise in teleconferencing too, due to coupling between the loudspeaker and microphone in each room. This problem is similar to the network echo problem, although the impulse responses are longer and can change more rapidly. When echoes arise in a stereophonic system, however, the problem changes character due to an inherent nonuniqueness in estimating the echo paths. In this talk we will review the current status of the problem of stereo echo cancellation. After a brief introduction to the general problem of echo control, the nonuniqueness mentioned above will be discussed and then several approaches will be described to get around this nonuniqueness. At present echo path responses can be matched with an error of less than −20 dB, without any loss of the stereophonic effect.

An algorithm-independent burst-free adaptive hybrid

This paper presents a structural solution to the bursting phenomena in adaptive hybrids. An algorithm-independent burst-free adaptive echo canceler is developed by introducing a stand-by adaptive filter in a single adaptive hybrid system. The purpose of this filter is to cancel out the pole(s) of the closed-loop characteristic equation, which, generally, lead to the bursting phenomena in adaptive systems with feedback. Several simulations examples are presented in this paper to demonstrate the superiority of the new configuration under the different conditions that would normally cause bursting, including practical situations where we have actual speech signals.