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Time filter

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Talmon R.,Yale University | Habets E.A.P.,International Audio Laboratories Erlangen
ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings | Year: 2013

The reverberation time (RT) is a very important measure that quantifies the acoustic properties of a room and provides information about the quality and intelligibility of speech recorded in that room. Moreover, information about the RT can be used to improve the performance of automatic speech recognition systems and speech dereverberation algorithms. In a recent study, it has been shown that existing methods for blind estimation of the RT are highly sensitive to additive noise. In this paper, a novel method is proposed to blindly estimate the RT based on the decay rate distribution. Firstly, a data-driven representation of the underlying decay rates of several training rooms is obtained via the eigenvalue decomposition of a specially-tailored kernel. Secondly, the representation is extended to a room under test and used to estimate its decay rate (and hence its RT). The presented results show that the proposed method outperforms a competing method and is significantly more robust to noise. © 2013 IEEE. Source


Levin D.,Bar - Ilan University | Habets E.A.P.,International Audio Laboratories Erlangen | Gannot S.,Bar - Ilan University
Journal of the Acoustical Society of America | Year: 2012

A vector-sensor consisting of a monopole sensor collocated with orthogonally oriented dipole sensors is used for direction of arrival (DOA) estimation in the presence of an isotropic noise-field or internal device noise. A maximum likelihood (ML) DOA estimator is derived and subsequently shown to be a special case of DOA estimation by means of a search for the direction of maximum steered response power (SRP). The problem of SRP maximization with respect to a vector-sensor can be solved with a computationally inexpensive algorithm. The ML estimator achieves asymptotic efficiency and thus outperforms existing estimators with respect to the mean square angular error (MSAE) measure. The beampattern associated with the ML estimator is shown to be identical to that used by the minimum power distortionless response beamformer for the purpose of signal enhancement. © 2012 Acoustical Society of America. Source


Habets E.A.P.,International Audio Laboratories Erlangen | Benesty J.,University of Quebec
2011 Joint Workshop on Hands-free Speech Communication and Microphone Arrays, HSCMA'11 | Year: 2011

In this paper a two-stage beamforming approach is presented for dereverberation and noise reduction. The first stage comprises a delay-and-sum (DS) beamformer that generates a reference signal that contains a spatially filtered version of the desired speech and interference. In general, the desired speech component at the output of the DS beamformer contains less reverberation compared to reverberant speech signal received at the microphones. The second stage uses the filtered microphone signals and the noisy reference signal to estimate the desired speech component at the output of the DS beamformer. A major advantage over classical approaches is that the proposed approach is able to dereverberate the received desired signal with very low speech distortion. The dereverberation and noise reduction performance is evaluated for a circular microphone array. © 2011 IEEE. Source


Jarrett D.P.,Imperial College London | Habets E.A.P.,International Audio Laboratories Erlangen | Thomas M.R.P.,Imperial College London | Naylor P.A.,Imperial College London
Journal of the Acoustical Society of America | Year: 2012

Simulated room impulse responses have been proven to be both useful and indispensable for comprehensive testing of acoustic signal processing algorithms while controlling parameters such as the reverberation time, room dimensions, and source-array distance. In this work, a method is proposed for simulating the room impulse responses between a sound source and the microphones positioned on a spherical array. The method takes into account specular reflections of the source by employing the well-known image method, and scattering from the rigid sphere by employing spherical harmonic decomposition. Pseudocode for the proposed method is provided, taking into account various optimizations to reduce the computational complexity. The magnitude and phase errors that result from the finite order spherical harmonic decomposition are analyzed and general guidelines for the order selection are provided. Three examples are presented: an analysis of a diffuse reverberant sound field, a study of binaural cues in the presence of reverberation, and an illustration of the algorithm's use as a mouth simulator. © 2012 Acoustical Society of America. Source


Habets E.A.P.,International Audio Laboratories Erlangen | Benesty J.,University of Quebec at Montreal
IEEE Transactions on Audio, Speech and Language Processing | Year: 2012

Signals captured by a set of microphones in a speech communication system are mixtures of desired signals and noise. In this paper, a different perspective on frequency-domain beamformers in room acoustics is provided. Specifically, the observed noise signals are divided into coherent and incoherent signal components while no assumptions are being made regarding the number of coherent noise sources and the noise sound field. From this perspective, performance measures are defined and existing beamformers are deduced. In addition, a new and general tradeoff beamformer is proposed that enables a compromise between noise reduction and speech distortion on the one hand, and coherent noise versus incoherent noise reductions on the other hand. The presented performance evaluation shows how existing beamformers and the tradeoff beamformer perform in different scenarios. © 2006 IEEE. Source

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