Search results for: Jacob Benesty
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Applied Acoustics > 2017 > 126 > C > 26-35
The widely studied subspace and linear filtering methods for noise reduction require the noise correlation matrix to be invertible. In certain application scenarios, however, this matrix is either rank deficient or very ill conditioned, so this requirement cannot be fulfilled. In this paper, we investigate possible solutions to this important problem based on subspace techniques for single-channel...
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Circular differential microphone arrays (CDMAs) facilitate compact superdirective beamformers whose beampatterns are nearly frequency invariant, and allow perfect steering for all azimuthal directions. Herein, we eliminate the inherent limitation of symmetric beampatterns associated with a linear geometry, and introduce an analytical asymmetric model for Nth-order CDMAs. We derive the theoretical...
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In a recent work, we addressed the identification poblem of bilinear forms with the Wiener filter. In this context, a different approach was introduced, by defining the bilinear term with respect to the impulse responses of a spatiotemporal model, which resembles a multiple-input/single-output (MISO) system. However, in practice, the Wiener filter may not be always very efficient or convenient to...
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In this paper, we present a least-mean-square (LMS) adaptive filter tailored for bilinear forms. The bilinear term is defined with respect to the impulse responses of a spatiotemporal model, which resembles a multiple-input/single-output (MISO) system. A convergence analysis is also provided, outlining the main features of this algorithm. Simulation results support the theoretical findings.
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In this paper, we study the binaural noise-reduction problem using an array of microphones. The widely linear (WL) framework in the short-time-Fourier-transform (STFT) domain is adopted. In such a framework, the microphone array signals and binaural outputs are first merged into complex signals. These complex signals are subsequently transformed into the STFT domain. The WL estimation theory is then...
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This paper proposes a new and very simple method to perform identification of nonlinearities in systems that are supposed to be linear. The improvement with respect to the previous approaches lies in the low computational complexity of the solution and the convenient possibility of online evaluation. Simulation results show that the proposed method provides good performance in terms of coefficients...
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IEEE/ACM Transactions on Audio, Speech, and Language Processing > 2017 > 25 > 5 > 1140 - 1153
This paper deals with two critical issues about uniform circular arrays (UCAs): frequency-invariant response and steering flexibility. It focuses on some optimal design of frequency-invariant beampatterns in any desired direction along the sensor plane. The major contributions are as follows. 1) We explain how to include the steering information in the desired directivity pattern. 2) We show that...
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IEEE Signal Processing Letters > 2017 > 24 > 5 > 653 - 657
In this letter, the identification problem of bilinear forms with the Wiener filter is addressed. The contribution is twofold. First, a different approach is introduced, by defining the bilinear term with respect to the impulse responses of a spatiotemporal model, in the context of multiple-input/single-output systems. Second, two versions of the Wiener filter (namely direct and iterative) are developed...
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Time delay estimation (TDE) plays an important role in localizing and tracking radiating acoustic sources. Although many efforts have been devoted to this problem in the literature, the robustness of TDE with respect to noise and reverberation remains a great challenge for practical systems. In this paper, we investigate the TDE problem in acoustic single-input/multiple-output (SIMO) systems in reverberant...
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This paper presents an approach to the multichannel noise reduction problem. It first transforms the multichannel noisy speech signals into the frequency domain. A Householder transformation is then constructed, which converts the multichannel coefficients in each frequency bin into two components: one dominated by speech and the other dominated by noise. A Wiener filter is subsequently formed to...
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This paper deals with the problem of single-channel noise reduction. Thanks to the eigenvalue decomposition, we arrange the eigenvalues of the speech correlation matrix in such a way that all the spectral mode signal-to-noise ratios (SNRs) of the noisy speech are ordered in a descending manner. By maintaining no speech distortion in the spectral modes with high input SNRs while allowing some degree...
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In recent years, signal processing with ad hoc microphone arrays has attracted a lot of attention. Speech enhancement in noisy, interfered, and reverberant environments is one of the problems targeted by ad hoc microphone arrays. Most of the proposed solutions require knowledge of fingerprints, such as acoustic transfer functions, which may not be known as accurately as required in practical situations...
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The regularization parameter is required in most (if not all) adaptive algorithms, while its role becomes very critical in the presence of additive noise. In this paper, we focus on the regularized recursive least-squares (RLS) algorithm and present a method to find its regularization parameter, which is related to the signal-to-noise ratio (SNR). Also, using a proper estimation of the SNR, we further...
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IEEE/ACM Transactions on Audio, Speech, and Language Processing > 2016 > 24 > 12 > 2531 - 2543
Superdirective beamforming has attracted a significant amount of research interest in speech and audio applications, since it can maximize the directivity factor (DF) given an array geometry and, therefore, is efficient in dealing with signal acquisition in diffuse-like noise environments. However, this beamformer is very sensitive to sensor self-noise and mismatch among sensors, which considerably...
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Noise reduction has long been an active research topic in signal processing and many algorithms have been developed over the last four decades. These algorithms were proved to be successful in some degree to improve the signal-to-noise ratio (SNR) and speech quality. However, there is one problem common to all these algorithms: the volume of the enhanced signal after noise reduction is often perceived...
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Audio Signal Processing for Next-Generation Multimedia Communication Systems > Acoustic Echo Cancellation > 149-169
Double-talk detectors (DTDs) are vital to the operation and performance of acoustic echo cancelers. In this chapter, we highlight important aspects needed to be considered when choosing and designing a DTD. The generic double-talk detector scheme along with fundamental means of performance evaluation are discussed. A number of double-talk detectors suitable for acoustic echo cancelers are presented...
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In this chapter, we present some very useful measures that are necessary to properly design filters in the different domains. These definitions will also help us better understand how noise reduction works in real-world applications and what price we need to pay for this. As a matter of fact, all known approaches in the single-channel (one microphone) case add distortion to the desired signal. Therefore,...
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The recently proposed joint-optimized normalized least-mean-square (JO-NLMS) adaptive algorithm employs a variable step-size parameter to achieve both high convergence and low misadjustment. In this paper, an efficient hardware implementation design is proposed, which exploits the key numerical properties of the JO-NLMS algorithm to reduce the hardware workload (corresponding to the adaptive filter's...
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In the problem of acoustic source localization, time difference of arrival (TDOA) among multiple sensors is needed, which is often obtained through time delay estimation (TDE) techniques. Among the multiple TDE methods developed in the literature, the normalized multichannel frequency-domain least-mean-square (NMCFLMS) algorithm is shown robust to reverberation. The performance of this algorithm,...
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Conventional superdirective beamforming is a well-known multi-microphone enhancement method with superior directivity factor (DF). However, it suffers from an inferior white noise gain (WNG), which is expressed by high sensitivity to uncorrelated noise and array inaccuracies. In this work, a beamformer with tunable superdirective gain is introduced. The proposed approach plays a role of a regularized...
