Impulse Response Of Echo Path Research Articles

Block‐sparsity‐aware LMS algorithm for network echo cancellation

Network echo path impulse response is single-block-sparse in nature. In order to obtain a single-block-sparse estimate of the unknown echo path, a new least mean squares (LMS) algorithm is proposed by introducing the penalty of single block sparsity, which is the difference between the mixed l 2 , 1 norm and l 2 norm of the uniformly partitioned filter tap-weight vector, into the original mean-square-error cost function. This is motivated by the fact that the difference between the mixed l 2 , 1 norm and l 2 norm of a vector is minimised only when there is at most one non-zero block in the vector. Numerical simulation results show that the proposed algorithm can effectively estimate and track the unknown echo path, outperforming existing block-sparsity-induced LMS algorithms.

The Dynamic LMS for Line Echo Cancellation

To the best of our knowledge, this study represents the proposal using the dynamic least mean square (DLMS) algorithm to reduce the computation load of LMS. Moreover, three regions of impulse response of line echo path are also proposed to analyze the redundant coefficients. Using the DLMS method, redundant coefficients can be detected and grouped, thereby automatically reducing computation. We employed line echo cancellation (LEC) to evaluate the performance of DLMS. The pure-delay and overlong regions of impulse response of line echo path are grouped and the associated computation load is reduced. The experimental results confirm the excellent performance of DLMS achieving a 35% savings in computation. Moreover, the quality echo return loss enhancement (ERLE) of DLMS also maintains at a level nearly equal to LMS.

Stereophonic Acoustic Echo Suppression Based on Wiener Filter in the Short-Time Fourier Transform Domain

An open-loop stereophonic acoustic echo suppression (SAES) method without preprocessing is presented for teleconferencing systems, where the Wiener filter in the short-time Fourier transform (STFT) domain is employed. Instead of identifying the echo path impulse responses with adaptive filters, the proposed algorithm estimates the echo spectra from the stereo signals using two weighting functions. The spectral modification technique originally proposed for noise reduction is adopted to remove the echo from the microphone signal. Moreover, a priori signal-to-echo ratio (SER) based Wiener filter is used as the gain function to achieve a trade-off between musical noise reduction and computational load for real-time operations. Computer simulation shows the effectiveness and the robustness of the proposed method in several different scenarios.

Selective partial update and set‐membership improved proportionate normalized subband adaptive filter algorithms

AbstractIn this article, the concept of proportionate adaptation is extended to the selective partial update (SPU) and set‐membership (SM) normalized subband adaptive filters (NSAFs), and three proportionate normalized subband adaptive filter algorithms are established. The proposed algorithms are the improved proportionate NSAF (IPNSAF), the SPU improved proportionate NSAF (SPU‐IPNSAF), and the SM‐IPNSAF which are suitable for sparse system identification. When the impulse response of the echo path is sparse, the IPNSAF algorithm has faster convergence than NSAF. The performance of IPNSAF is also suitable for dispersive impulse responses. In SPU‐IPNSAF, the filter coefficients are partially updated rather than the entire filter at every adaptation which reduces the computational complexity of IPNSAF. The SM‐IPNSAF exhibits good performance with significant reduction in the overall computational complexity compared with the ordinary IPNSAF. The simulation results show good performance of the proposed algorithms. Copyright © 2010 John Wiley & Sons, Ltd.

An improved partial Haar dual adaptive filter for rapid identification of a sparse echo channel

Recently, a coupled echo canceller was proposed that uses two short adaptive filters for sparse echo cancellation. The first filter operates in the partial Haar domain and is used to locate the channel's dispersive region; the second filter is then centered around this location to cancel the echo in the time domain. In this paper, we propose feasible solutions to improve the performance of this partial Haar dual adaptive filter (PHDAF) in practical applications. These include: (1) alleviating the dependence of the PHDAFs performance on the echo-path impulse response's bulk delay; (2) improving the tracking performance of the PHDAF in response to abrupt changes in the echo path; and (3) extending the original PHDAF structure to support the cancellation of multiple echoes. The proposed algorithmic solutions exploit the Haar transform's polyphase representation and make use of a novel peak tendency estimator (PTE) based on Dezert–Smarandache theory (DSmT). The improved PHDAF is evaluated in terms of its mean-square error (MSE) curves and its mean time to properly locate a dispersive region for different SNRs. Results show that enhanced performance can be obtained using the proposed solutions at a minimal increase in computational cost.

Echo Cancellation in DMT Modems With Frame-Asynchronous Operation

Echo cancellation in DMT modems has been long studied as a means to improve transmission performance. Most of the current echo-canceller structures make use of FIR filters that emulate the echo-path adaptively. In frame-asynchronous operation, the misalignment of the TX and RX DFT windows (combined with other modem and channel parameters like: the transmit tone-mask and the echo-path impulse response) determines the echo input statistics and can worsen the conditioning of the coefficient update formula. This effect is analyzed in the paper, within the context of two specific algorithms: the circulant decomposition canceller (CDC) and also the time-domain canceller (TD-EC). Self-orthogonalizing (S-O) update formulas, implemented through compensation matrices, and their respective low-rank approximations are proposed as a method to improve the convergence rate and, more generally, to render the echo-canceller performances less sensitive to the input statistics. The method introduced does not employ additional power on the unsent tones

A Low Delay and Fast Converging Improved Proportionate Algorithm for Sparse System Identification

A sparse system identification algorithm for network echo cancellation is presented. This new approach exploits both the fast convergence of the improved proportionate normalized least mean square (IPNLMS) algorithm and the efficient implementation of the multidelay adaptive filtering (MDF) algorithm inheriting the beneficial properties of both. The proposed IPMDF algorithm is evaluated using impulse responses with various degrees of sparseness. Simulation results are also presented for both speech and white Gaussian noise input sequences. It has been shown that the IPMDF algorithm outperforms the MDF and IPNLMS algorithms for both sparse and dispersive echo path impulse responses. Computational complexity of the proposed algorithm is also discussed.

Robust adaptive algorithm based on nonlinear error cost function for acoustic echo cancellation

Motivated by recent progress in the blind source separation (BSS) technique, a robust echo cancellation algorithm is investigated, which would inherently identify the echo path even during double-talk by separating the acoustic echo from the local speech. An adaptive filter has been introduced into acoustic echo cancellers to identify the acoustic echo path impulse response and generate the echo replica. However, most adaptive algorithms suffer from instability during double-talk. Although step-size control cooperating with a double-talk detector (DTD) is a promising approach to stop the adaptation temporarily during double-talk, it cannot handle the echo path change during double-talk. To overcome this problem, novel robust algorithms are derived by applying nonlinearity to the cost function of a conventional echo cancellation algorithm such as the normalized least mean squares algorithm (NLMS) or the affine projection algorithm (APA). Using the simulation results, there is a discussion about how the robustness of the derived algorithms depends on the choice of the nonlinear function and the original algorithm.

Suppressing acoustic echo in a spectral envelope space

Full-duplex hands-free telecommunication systems employ an acoustic echo canceler (AEC) to remove the undesired echoes that result from the coupling between a loudspeaker and a microphone. Traditionally, the removal is achieved by modeling the echo path impulse response with an adaptive finite impulse response (FIR) filter and subtracting an echo estimate from the microphone signal. It is not uncommon that an adaptive filter with a length of 50-300 ms needs to be considered, which makes an AEC highly computationally expensive. In this paper, we propose an echo suppression algorithm to eliminate the echo effect. Instead of identifying the echo path impulse response, the proposed method estimates the spectral envelope of the echo signal. The suppression is done by spectral modification-a technique originally proposed for noise reduction. It is shown that this new approach has several advantages over the traditional AEC. Properties of human auditory perception are considered, by estimating spectral envelopes according to the frequency selectivity of the auditory system, resulting in improved perceptual quality. A conventional AEC is often combined with a post-processor to reduce the residual echoes due to minor echo path changes. It is shown that the proposed algorithm is insensitive to such changes. Therefore, no post-processor is necessary. Furthermore, the new scheme is computationally much more efficient than a conventional AEC.

Partitioned residual echo power estimation for frequency‐domain acoustic echo cancellation and postfiltering

AbstractResidual echo arises in hands‐free telephony equipment due to insufficient echo canceler convergence, but can be suppressed using a postfilter. The residual echo power spectral density is the most crucial control parameter for both frequency‐domain acoustic echo cancellation and combined residual echo and noise postfiltering. In this contribution we present and compare residual echo power spectral estimation techniques. We introduce a new partitioned block‐adaptive estimation technique delivering considerably improved residual echo estimates in strongly reverberant and noisy acoustic environments. We show that the adaptation loop of the frequency‐domain adaptive filter (FDAF) can be used simultaneously for residual echo power estimation and tracking of the echo path impulse response. In this way, the FDAF and the postfilter concept supplement each other in a true synergy with low complexity. The resulting echo and noise control system proves to be robust in double talk situations as well.

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.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Multiple-Microphone Acoustic Echo Cancellation System with the Partial Adaptive Process

The acoustic echo canceler using an adaptive transversal filter and the least-mean square (LMS) algorithm is the most effective technique to reduce acoustic echoes in a hands-free telephone system. However, the requirement of a very high order filter for each microphone echo path results in difficulties in convergence speed and hardware implementation. In this paper, a partial adaptive process (PAP) which utilizes the exponential decay characteristics of the acoustic echo path impulse response is developed. Simulations are conducted to show that the performance of the PAP method could be acceptable in practical use. A combination FIR and IIR filter structure and a new time-weighted LMS (TW-LMS) algorithm to complement the PAP are also proposed. This new adaptation algorithm and filter structure further improve the performance of the PAP method in terms of echo reduction and computational savings.

An echo canceller with a block-implemented learning identification algorithm

Characteristics of an echo canceller with use of a block-implemented learning identification algorithm are examined through an approximate analysis and computer simulation experiments. Block digital signal processing, which involves the calculation of a block or finite set of system output from a block of input, is a promising technique to allow efficient use of parallel processors. The echo return loss enhancement, defined as a performance index of the echo canceller, is represented by an approximate equation as a function in the signal-to-noise ratio of a circuit and an acceleration coefficient for updating the echo-path impulse response estimates. This approximate analytical expression is observed to agree well with the results from simulation experiments.

A system identification algorithm using orthogonal functions

An adaptive filter (ADF) structure is proposed for applications in which large-order ADFs are required. It is based on modeling the impulse response of the system to be identified as a linear combination of a set of discrete Legendre orthogonal functions. The proposed adaptive filter structure has desirable stability features and a unimodal mean-square error surface as well as a modular structure that permits an easy increase of the filter order without changing the previous stages. Computer simulations in which the proposed structure is used to identify actual acoustic echo path impulse responses show that the Legendre ADF has better convergence performance than the transversal ADF when identifying systems with long impulse response.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

A Blockless Echo Suppressor

Practically all echo suppressors presently used in long-distance telephone circuits operate on the principle of cutting off the sending circuit when the receiving speech level is greater than the sending speech level, thereby suppressing echoes. Therefore, they generate a faulty speech problem due to the cutting off of speech in simultaneous talking, thus making speech unnatural. To correct this fault, a blockless echo suppressor which, unlike the usual echo suppressors, generates a signal equivalent to the echo to be canceled, has been developed. In the suppressor, a signa1 equivalent to an echo is produced from the convolution of a received speech signal and an echo-path impulse response which is stored in a memory in advance of the commencement of commercial service. It has been confirmed by subjective evaluation that an echo suppressor of this kind manufactured for trials has better talking quality than any other kind of typical echo suppressors now in service. This is because an echo is effectively suppressed more than 30 dB below the transmitted speech signal, and speech is not cut off in this suppressor.