We have discussed the application of high-order adaptive filters to the problem of acoustical echo cancellation with particular application to hands free telephone systems. We described a means to achieve robust performance. We further presented methods for reducing computational complexity that allow implementation in low-cost, fixed-point digital signal processors. Progress in technology will allow the use of more sophisticated algorithms at lower cost in the near future.

Acoustic echo control. An application of very-high-order adaptive filters

To find the position of an acoustic source in a room, the relative delay between two (or more) microphone signals for the direct sound must be determined. The generalized cross-correlation method is the most popular technique to do so and is well explained in a landmark paper by Knapp and Carter. In this paper, a new approach is proposed that is based on eigenvalue decomposition. Indeed, the eigenvector corresponding to the minimum eigenvalue of the covariance matrix of the microphone signals contains the impulse responses between the source and the microphone signals (and therefore all the information we need for time delay estimation). In experiments, the proposed algorithm performs well and is very accurate.

Adaptive eigenvalue decomposition algorithm for passive acoustic source localization

A solution is proposed to the long-standing problem of the numerical instability of fast recursive least squares transversal filter (FTF) algorithms with exponential weighting, an important class of algorithms for adaptive filtering. A framework for the analysis of the error propagation in FTF algorithms is first developed; within this framework, it is shown that the computationally most efficient 7N form is exponentially unstable. However, by introducing redundancy into this algorithm, feedback of numerical errors becomes possible; a judicious choice of the feedback gains then leads to a numerically stable FTF algorithm with a complexity of 8N multiplications and additions per time recursion. The results are presented for the complex multichannel joint-process filtering problem. >

Numerically stable fast transversal filters for recursive least squares adaptive filtering

Teleconferencing systems employ acoustic echo cancelers to reduce echoes that result from coupling between the loudspeaker and microphone. To enhance the sound realism, two-channel audio is necessary. However, in this case (stereophonic sound) the acoustic echo cancellation problem is more difficult to solve because of the necessity to uniquely identify two acoustic paths. We explain these problems in detail and give an interesting solution which is much better than previously known solutions. The basic idea is to introduce a small nonlinearity into each channel that has the effect of reducing the interchannel coherence while not being noticeable for speech due to self masking.

A better understanding and an improved solution to the specific problems of stereophonic acoustic echo cancellation

In speech communication systems, such as voice-controlled systems, hands-free mobile telephones, and hearing aids, the received microphone signals are degraded by room reverberation, background noise, and other interferences This signal degradation may lead to total unintelligibility of the speech and decreases the performance of automatic speech recognition systems In the context of this work reverberation is the process of multi-path propagation of an acoustic sound from its source to one or more microphones The received microphone signal generally consists of a direct sound, reflections that arrive shortly after the direct sound (commonly called early reverberation), and reflections that arrive after the early reverberation (commonly called late reverberation) Reverberant speech can be described as sounding distant with noticeable echo and colouration These detrimental perceptual effects are primarily caused by late reverberation, and generally increase with increasing distance between the source and microphone Conversely, early reverberations tend to improve the intelligibility of speech In combination with the direct sound it is sometimes referred to as the early speech component Reduction of the detrimental effects of reflections is evidently of considerable practical importance, and is the focus of this dissertation More specifically the dissertation deals with dereverberation techniques, ie, signal processing techniques to reduce the detrimental effects of reflections In the dissertation, novel single- and multimicrophone speech dereverberation algorithms are developed that aim at the suppression of late reverberation, ie, at estimation of the early speech component This is done via so-called spectral enhancement techniques that require a specific measure of the late reverberant signal This measure, called spectral variance, can be estimated directly from the received (possibly noisy) reverberant signal(s) using a statistical reverberation model and a limited amount of a priori knowledge about the acoustic channel(s) between the source and the microphone(s) In our work an existing single-channel statistical reverberation model serves as a starting point The model is characterized by one parameter that depends on the acoustic characteristics of the environment We show that the spectral variance estimator that is based on this model, can only be used when the source-microphone distance is larger than the so-called critical distance This is, crudely speaking, the distance where the direct sound power is equal to the total reflective power A generalization of the statistical reverberation model in which the direct sound is incorporated is developed This model requires one additional parameter that is related to the ratio between the direct sound energy and the sound energy of all reflections The generalized model is used to derive a novel spectral variance estimator When the novel estimator is used for dereverberation rather than the existing estimator, and the source-microphone distance is smaller than the critical distance, the dereverberation performance is significantly increased Single-microphone systems only exploit the temporal and spectral diversity of the received signal Reverberation, of course, also induces spatial diversity To additionally exploit this diversity, multiple microphones must be used, and their outputs must be combined by a suitable spatial processor such as the so-called delay and sum beamformer It is not a priori evident whether spectral enhancement is best done before or after the spatial processor For this reason we investigate both possibilities, as well as a merge of the spatial processor and the spectral enhancement technique An advantage of the latter option is that the spectral variance estimator can be further improved Our experiments show that the use of multiple microphones affords a significant improvement of the perceptual speech quality The applicability of the theory developed in this dissertation is demonstrated using a hands-free communication system Since hands-free systems are often used in a noisy and reverberant environment, the received microphone signal does not only contain the desired signal but also interferences such as room reverberation that is caused by the desired source, background noise, and a far-end echo signal that results from a sound that is produced by the loudspeaker Usually an acoustic echo canceller is used to cancel the far-end echo Additionally a post-processor is used to suppress background noise and residual echo, ie, echo which could not be cancelled by the echo canceller In this work a novel structure and post-processor for an acoustic echo canceller are developed The post-processor suppresses late reverberation caused by the desired source, residual echo, and background noise The late reverberation and late residual echo are estimated using the generalized statistical reverberation model Experimental results convincingly demonstrate the benefits of the proposed system for suppressing late reverberation, residual echo and background noise The proposed structure and post-processor have a low computational complexity, a highly modular structure, can be seamlessly integrated into existing hands-free communication systems, and affords a significant increase of the listening comfort and speech intelligibility

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Single- and multi-microphone speech dereverberation using spectral enhancement

It is likely that stereophonic (and more generally, multichannel) sound pick-up, transmission and diffusion will be implemented in future teleconference systems to provide the users with enhanced quality. Therefore, adequate solutions must be found to solve the problem of stereophonic acoustic echo which will occur in such systems. We explain in this paper the difference between the mono and two-channel systems and the behavior of the two-channel classical adaptive algorithms in comparison with the same algorithms in the mono-channel case. Also, we outline a new NLMS-like algorithm derived from the two-channel RLS algorithm as a first member of a family of improved two-channel adaptive filters.

Adaptive filtering algorithms for stereophonic acoustic echo cancellation

We focus on the problem of stereophonic acoustic echo cancellation for teleconference applications. To limit the well-known detrimental effect of the correlation between the loudspeaker input signals, we propose a new method which consists in adding to these signals random noise controlled by the auditory properties. We describe this method in some detail and we show that its complexity can be fairly low. We demonstrate experimentally that the improvement yielded by this method is higher than the one provided by a former method based on the use of non-linearity.

Using auditory properties to improve the behaviour of stereophonic acoustic echo cancellers

We consider an acoustic echo control system composed of a short conventional acoustic echo canceller combined with a post-filter in a teleconference context. The post-filter is implemented in an open-loop structure in the frequency domain, which provides good adaptive performance and flexibility for the choice of the post-filter length. Three post-filtering algorithms are compared in terms of residual echo attenuation and near-end speech distortion. The effect of the post-filter length is also examined. Our study confirms that the post-filtering approach provides high residual echo attenuation. Moreover, it appears that the distortion of the near-end speech can be controlled by choosing appropriately the post-filter length.

Comparison of three post-filtering algorithms for residual acoustic echo reduction

An effective method to stabilize fast RLS algorithms is proposed. It is based on the analysis of the propagation of the numerical errors according to a first-order linear model. Two variables are shown to be responsible for the numerical instability. The proposed method modifies the numerical properties of these variables, while preserving the theoretical form of the algorithms. This method is applied to the FTF (fast transversal filter) and fast Kalman algorithms. Experimental results in floating- and fixed-point arithmetic show the efficiency of the method. >

A new method to stabilize fast RLS algorithms based on a first-order of the propagation of numerical errors

We propose a new fast projection algorithm for stereophonic acoustic echo cancellation. This algorithm can be viewed as a generalization of the extended two-channel LMS algorithm which takes into account the correlation between the input signals. Moreover, this algorithm fits naturally with the framework of affine projection techniques extended to the two-channel case. Its computational complexity is less than half the complexity of the fastest two-channel RLS versions. Simulation results show the obtained performance.

A. Gilloire

Papers

Adaptive filtering algorithms for stereophonic acoustic echo cancellation

Using auditory properties to improve the behaviour of stereophonic acoustic echo cancellers

Comparison of three post-filtering algorithms for residual acoustic echo reduction

A new method to stabilize fast RLS algorithms based on a first-order of the propagation of numerical errors

A fast two-channel projection algorithm for stereophonic acoustic echo cancellation