47 research outputs found
Joint NN-Supported Multichannel Reduction of Acoustic Echo, Reverberation and Noise
We consider the problem of simultaneous reduction of acoustic echo,
reverberation and noise. In real scenarios, these distortion sources may occur
simultaneously and reducing them implies combining the corresponding
distortion-specific filters. As these filters interact with each other, they
must be jointly optimized. We propose to model the target and residual signals
after linear echo cancellation and dereverberation using a multichannel
Gaussian modeling framework and to jointly represent their spectra by means of
a neural network. We develop an iterative block-coordinate ascent algorithm to
update all the filters. We evaluate our system on real recordings of acoustic
echo, reverberation and noise acquired with a smart speaker in various
situations. The proposed approach outperforms in terms of overall distortion a
cascade of the individual approaches and a joint reduction approach which does
not rely on a spectral model of the target and residual signals
Single- and multi-microphone speech dereverberation using spectral enhancement
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, i.e., 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, i.e., 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, i.e., 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
Instantaneous PSD Estimation for Speech Enhancement based on Generalized Principal Components
Power spectral density (PSD) estimates of various microphone signal
components are essential to many speech enhancement procedures. As speech is
highly non-nonstationary, performance improvements may be gained by maintaining
time-variations in PSD estimates. In this paper, we propose an instantaneous
PSD estimation approach based on generalized principal components. Similarly to
other eigenspace-based PSD estimation approaches, we rely on recursive
averaging in order to obtain a microphone signal correlation matrix estimate to
be decomposed. However, instead of estimating the PSDs directly from the
temporally smooth generalized eigenvalues of this matrix, yielding temporally
smooth PSD estimates, we propose to estimate the PSDs from newly defined
instantaneous generalized eigenvalues, yielding instantaneous PSD estimates.
The instantaneous generalized eigenvalues are defined from the generalized
principal components, i.e. a generalized eigenvector-based transform of the
microphone signals. We further show that the smooth generalized eigenvalues can
be understood as a recursive average of the instantaneous generalized
eigenvalues. Simulation results comparing the multi-channel Wiener filter (MWF)
with smooth and instantaneous PSD estimates indicate better speech enhancement
performance for the latter. A MATLAB implementation is available online
Two-Channel Speech Enhancement and Implementation Considerations: Noise Reduction and Speech Quality
A Study into Speech Enhancement Techniques in Adverse Environment
This dissertation developed speech enhancement techniques that improve the speech quality in applications such as mobile communications, teleconferencing and smart loudspeakers. For these applications it is necessary to suppress noise and reverberation. Thus the contribution in this dissertation is twofold: single channel speech enhancement system which exploits the temporal and spectral diversity of the received microphone signal for noise suppression and multi-channel speech enhancement method with the ability to employ spatial diversity to reduce reverberation