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ABSTRACT: A method for dereverberation and noise reduction is presented. The method is designed for a spherical microphone array, and formulated in the spherical harmonics domain, based on an acoustic model that is also formulated in the spherical harmonics domain. The novelty in the proposed method is the dereverberation process, which exploits the useful formulation in the spherical harmonics domain, which facilitates dereverberation by employing DOA estimation, rather that room impulse response identification. Noise reduction is further performed by a linearly constrained minimum variance filter, where the array output power is minimized with constraint of distortionless response to the direct sound. The paper concludes with a simulation investigation and comparison to theoretical results.
Acoustics Speech and Signal Processing (ICASSP), 2010 IEEE International Conference on; 04/2010 · 4.63 Impact Factor
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ABSTRACT: The room impulse response (RIR) can be used to calculate many room acoustical parameters, such as the reverberation time (RT). However, estimating the room volume, another important room parameter, from the RIR is typically a more difficult task requiring extraction of other features from the RIR. Most of the existing fully-blind methods for estimating the room volume from the RIR do not combine features from different feature sets. This can be one reason to the fact that these methods are sensitive to differences in source-to-receiver distance and wall reflection coefficients. We propose a new approach in which hypothetical-volume room models are trained with room volume features from different feature sets. Estimation is performed by identifying the hypothesis with maximum-likelihood (ML) using background model normalization. The different feature sets are compared using equal error rate (EER) of hypothesis verification. A combination of features from the different feature sets is selected so that minimum EER is achieved. Using the selected features, we achieve average detection rate of 98.8% with a standard deviation (STD) of 1.5% for eight rooms with different volumes, source-to-receiver distances, and wall reflection coefficients.
Applications of Signal Processing to Audio and Acoustics, 2009. WASPAA '09. IEEE Workshop on; 11/2009
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ABSTRACT: In order to understand the acoustic behavior of the sound field in a room, it is important to know the volume of the room, as well as other room parameters, like reverberation time (RT). However, estimating the room volume from the room impulse response (RIR) is usually considered a more difficult task than estimating the RT from the RIR. Most of the existing fullyblind methods for estimating the room volume from the RIR do not have satisfactory performances. The reason is that they are sensitive to differences in source-to-receiver distance and wall reflection coefficients. We propose a new approach in which hypothetical-volume room models are trained, and estimation is performed via hypothesis verification using log-likelihood ratio test (LLRT). We achieve low equal error rate (EER) of 3.6% in hypothesis verification for eight rooms with different values of source-to-receiver distance and wall reflection coefficients.
Statistical Signal Processing, 2009. SSP '09. IEEE/SP 15th Workshop on; 10/2009
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ABSTRACT: The performance of speaker verification (SVR) systems degrades with the presence of room reverberation. Reverberation results in mismatched conditions between target models and test segments. Reverberation time (RT) is commonly used as a room parameter that represents reverberation. We investigate the effect of other room parameters such as room dimensions and reflection coefficients of the walls on SVR. Equal error rate (EER) is calculated by using room dimensions and reflection coefficients as parameters. Results of SVR with reverberant speech of the same RT are shown to be essentially different, when other room parameters are different. Feature normalization techniques are tested with reverberant speech of the same RT, and shown to be either improving or degrading the performance of SVR when other room parameters are different. This stands in contradiction to the approach in the literature towards RT as a dominant room parameter.
Electrical and Electronics Engineers in Israel, 2008. IEEEI 2008. IEEE 25th Convention of; 01/2009
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ABSTRACT: Reverberation and noise have a significant effect on the intelligibility of speech in rooms. Detection of clear speech in highly reverberant and noisy enclosures is an extremely difficult problem. In this work we take the approach of spatial filtering by spherical microphone array to overcome noise and reverberation. In order to investigate speech intelligibility, an acoustic model was developed. The model is based on U<sub>50</sub>, which was shown to have good correlation with speech intelligibility, while combining the effect of both reverberation and noise. This paper presents the array beamforming design for optimal speech intelligibility performance. The beamformer is designed to enhance the speech signal while minimizing noise power and maintaining robustness at a wide frequency range. Having such a model allows the prediction of the required array order for detecting speech clearly.
Electrical and Electronics Engineers in Israel, 2008. IEEEI 2008. IEEE 25th Convention of; 01/2009
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ABSTRACT: The performance of speaker recognition systems may degrade significantly when speech is recorded in reverberant environments by a microphone positioned far from the speaker. Most of the literature on speaker recognition uses the reverberation time to classify the reverberation effects. However, as described in this work, the reverberation time is mainly a room feature and is less affected by the distance between the source and the microphone. This paper presents a comprehensive study of room acoustics parameters and their relationship with speaker recognition performance. The definition and centra-time, acoustic parameters which are affected by both room properties and distance, were found to be more correlated with the degradation in the speaker recognition performance.
Hands-Free Speech Communication and Microphone Arrays, 2008. HSCMA 2008; 06/2008
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ABSTRACT: Speaker recognition is used today in a wide range of applications. The presence of reverberation, in hands-free systems for example, results in performance degradation. The effect of reverberation on the feature vectors and its relation to optimal GMM order are investigated. Optimal model order is calculated in terms of minimum BIC and KIC, and tested for EER of a GMM-based speaker recognition system. Experimental results show that for high reverberation time, reducing model order reduces EER values of speaker recognition. The effect of CMS on state of the art GMM and AGMM-based speaker recognition systems is investigated for reverberant speech. Results show that high reverberation time reduces the effectiveness of CMS.
Hands-Free Speech Communication and Microphone Arrays, 2008. HSCMA 2008; 06/2008
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ABSTRACT: Detection of clear speech in highly reverberant and noisy enclosures is an extremely difficult problem. Recently, spherical microphone arrays have been studied that are suitable for noise reduction and de-reverberation in three dimensions. This paper presents the development of a model for investigating speech intelligibility in noisy enclosures when recorded and processed by spherical microphones arrays. The model uses the image method, diffuse sound fields, spherical array beamforming and speech intelligibility measures, to predict the array order required to overcome noise and reverberation when detecting speech in noisy enclosures. Having such a model, one can design a spherical array that overcomes given acoustic conditions, or assess whether a given problem can be solved by a practical array configuration.
Hands-Free Speech Communication and Microphone Arrays, 2008. HSCMA 2008; 06/2008
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B. Rafaely
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ABSTRACT: Spherical microphone arrays have been recently studied for spatial sound recording, speech communication, and sound field analysis for room acoustics and noise control. Complementary theoretical studies presented progress in spatial sampling and beamforming methods. This paper reviews recent results in spatial sampling that facilitate a wide range of spherical array configurations, from a single rigid sphere to free positioning of microphones. The paper then presents an overview of beamforming methods recently presented for spherical arrays, from the widely used delay-and-sum and Dolph-Chebyshev, to the more advanced optimal methods, typically performed in the spherical harmonics domain.
Hands-Free Speech Communication and Microphone Arrays, 2008. HSCMA 2008; 06/2008
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B. Rafaely
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ABSTRACT: Spherical microphone arrays have been recently studied for a wide range of applications. In particular, microphones arranged around an open or virtual sphere are useful in scanning microphone arrays for sound field analysis. However, open-sphere spherical arrays have been shown to have poor robustness at frequencies related to the zeros of the spherical Bessel functions. This paper presents a framework for the analysis of array robustness using the condition number of a given matrix, and then proposes several robust array configurations. In particular, a dual-sphere configuration previously presented which uses twice as many microphones compared to a single-sphere configuration is analyzed. This paper then shows that high robustness can be achieved without increasing the number of microphones by arranging the microphones in the volume of a spherical shell. Another simpler configuration employs a single sphere and an additional microphone at the sphere center, showing improved robustness at the low-frequency range. Finally, the white-noise gain of the arrays is investigated verifying that improved white-noise gain is associated with lower matrix condition number.
IEEE Transactions on Audio Speech and Language Processing 06/2008; · 1.50 Impact Factor
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ABSTRACT: A nearfield spherical microphone array is presented. The nearfield criterion of the spherical array is defined in terms of array order, frequency and location. It is shown that given a source in the nearfield, significant attenuation of farfield interference is achieved. Also, nearfield sources may be attenuated relative to farfield sources. Dereverberation of a nearfield source in a reverberant enclosure is demonstrated using the nearfield microphone array.
Acoustics, Speech and Signal Processing, 2008. ICASSP 2008. IEEE International Conference on; 05/2008 · 4.63 Impact Factor
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ABSTRACT: Speech recorded by a distant microphone in a room may be subject to reverberation. Performance of a speaker verification system may degrade significantly for reverberant speech, with severe consequences in a wide range of real applications. This paper presents a comprehensive study of the effect of reverberation on speaker verification, and investigates approaches to reduce the effect of reverberation: training target models with reverberant speech signals and using acoustically matched models for the reverberant speech under test, score normalization methods to improve the reverberation robustness, and also reverberation classification via the background model scores. Experimental investigation is performed, using simulated and measured room impulse responses, NIST-based speech database, and AGMM based speaker verification system, showing significant improvement in performance.
Acoustics, Speech and Signal Processing, 2008. ICASSP 2008. IEEE International Conference on; 05/2008 · 4.63 Impact Factor
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ABSTRACT: Spherical microphone arrays have been studied for a wide range of applications, one of which is acoustic measurement and analysis. Since a minimal interaction between the array and the measured sound field is an advantage in this case, open-sphere arrays are preferable compared to rigid-sphere arrays. However, it has been shown that open-sphere arrays suffer from numerical ill-conditioning at frequencies which correspond to the nodal values of the spatial spherical modes with the result of excessive noise at these frequencies. A method for overcoming this problem using an open dual-sphere array is proposed in this correspondence and then investigated and compared to an array configured around a rigid sphere and an array composed of cardioid microphones. An optimal value for the ratio of the two spheres is derived, and simulation examples illustrating the advantage of the dual-sphere array are finally presented
IEEE Transactions on Audio Speech and Language Processing 03/2007; · 1.50 Impact Factor
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B. Rafaely
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ABSTRACT: Phase-mode spherical microphone array processing, also known as spherical harmonic array processing, has been recently studied for various applications. The spherical array configuration provides desired three-dimensional symmetry, while the phase modes provide frequency-independent spatial processing. This letter employs the spherical harmonic framework to compare the well-known delay-and-sum to the phase-mode processing for spherical arrays. The two approaches show similar performance at frequencies where the upper spherical harmonic order equals the product of the wave number and sphere radius. However, at lower frequencies, phase-mode processing maintains the same directivity, limited by signal-to-noise ratio, while for delay-and-sum, spatial resolution deteriorates.
IEEE Signal Processing Letters 11/2005; · 1.39 Impact Factor
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B. Rafaely
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ABSTRACT: Spherical microphone arrays have been recently studied for sound-field recordings, beamforming, and sound-field analysis which use spherical harmonics in the design. Although the microphone arrays and the associated algorithms were presented, no comprehensive theoretical analysis of performance was provided. This work presents a spherical-harmonics-based design and analysis framework for spherical microphone arrays. In particular, alternative spatial sampling schemes for the positioning of microphones on a sphere are presented, and the errors introduced by finite number of microphones, spatial aliasing, inaccuracies in microphone positioning, and measurement noise are investigated both theoretically and by using simulations. The analysis framework can also provide a useful guide for the design and analysis of more general spherical microphone arrays which do not use spherical harmonics explicitly.
IEEE Transactions on Speech and Audio Processing 02/2005; · 2.29 Impact Factor
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ABSTRACT: Spherical microphone arrays have been recently proposed for sound recordings, speech communications and room acoustics measurements, as they provide true three-dimensional detection of the sound field. Improved performance in all these applications can be achieved with arrays which have high spatial resolution and which in addition provide broad operating frequency and good noise rejection. The design of such arrays is a challenging task, as increase in spatial resolution typically comes at the expense of the other factors. In this paper a theoretical analysis of the factors affecting the performance of such arrays is discussed, and initial simulation and experimental results presented.
Electrical and Electronics Engineers in Israel, 2004. Proceedings. 2004 23rd IEEE Convention of; 10/2004
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ABSTRACT: Room acoustics measurements play an important role in the design and analysis of enclosures for music and speech. Spatial measures such as the directivity index use spatial sound field information and attempt to predict the performance of enclosures for speech and music. Although recently developed microphone arrays were suggested for the measurement of these measures, they do not have sufficient spatial resolution to perform a complete, detailed analysis of sound fields. In this paper, the decomposition of acoustic fields into plane waves is performed by spherical microphone arrays, which use a large number of samples over a sphere to achieve high spatial resolution. The array processing then employs the spherical Fourier transform for the decomposition into plane waves. The paper presents simulation results of spherical array processing in various sound fields and examines the requirements for successful application of the method in realistic sound fields.
Electrical and Electronics Engineers in Israel, 2004. Proceedings. 2004 23rd IEEE Convention of; 10/2004
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The Journal of the Acoustical Society of America 05/2001; 109(4):1747-9. · 1.55 Impact Factor
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B. Rafaely
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ABSTRACT: The derivation of Wiener filters, which are spatially optimal for
reverberant sound fields, is presented, with an application to
two-microphone noise cancellation in reverberant rooms. The optimal
filters, which are derived from the spatial correlation of diffuse sound
fields, capture the local behaviour of the sound field and are therefore
of lower order and potentially more spatially robust than conventional
adaptive noise cancellation filters in reverberant rooms. Formulations
for unconstrained and causally constrained optimal filters are
presented, and an example application to a two-microphone speech
enhancement is demonstrated using a computer simulation
Applications of Signal Processing to Audio and Acoustics, 2001 IEEE Workshop on the; 02/2001
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ABSTRACT: A bound on the variability of the feedback path is employed in the
design of fixed FIR hearing aid filters that are robust to the specified
variability, thus avoiding instability and howling in everyday use. A
design example is presented for a linear gain hearing aid filter with a
given maximal mismatch of the feedback cancellation filter
IEEE Transactions on Speech and Audio Processing 12/2000; · 2.29 Impact Factor
