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IEEE International Conference on Acoustics, Speech and Signal processing

作者： S. Chakrabarty O. Thiergart E. A. P. Habets Int. Audio Labs. Erlangen Erlangen Germany

ISBN: (纸本)9781467369985

A spatial filter, with L linear constraints that are based on instantaneous narrowband direction-of-arrival (DOA) estimates, was recently proposed to obtain a desired spatial response for at most L sound sources. In noisy and reverberant environments, it becomes difficult to get reliable instantaneous DOA estimates and hence obtain the desired spatial response. In this work, we develop a Bayesian approach to spatial filtering that is more robust to DOA estimation errors. The resulting filter is a weighted sum of spatial filters pointed at a discrete set of DOAs, with the relative contribution of each filter determined by the posterior distribution of the discrete DOAs given the microphone signals. In addition, the proposed spatial filter is able to reduce both reverberation and noise. In this work, the required diffuse sound power is estimated using the posterior distribution of the discrete set of DOAs. Simulation results demonstrate the ability of the proposed filter to achieve strong suppression of the undesired signal components with small amount of signal distortion, in noisy and reverberant conditions.

关键词： bayesian beamforming dereverberation microphone array processing

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3D LOCALIZATION OF MULTIPLE SOUND SOURCES WITH INTENSITY VECTOR ESTIMATES IN SINGLE SOURCE ZONES

3D LOCALIZATION OF MULTIPLE SOUND SOURCES WITH INTENSITY VEC...

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European Signal processing Conference

作者： Despoina Pavlidi Symeon Delikaris-Manias Ville Pulkki Athanasios Mouchtaris FORTH-ICS Aalto University Department of Signal Processing and Acoustics

ISBN: (纸本)9781479988518

This work proposes a novel method for 3D direction of arrival (DOA) estimation based on the sound intensity vector estimation, via the encoding of the signals of a spherical micro-phone array from the space domain to the spherical harmonic domain. The sound intensity vector is estimated on detected single source zones (SSZs), where one source is dominant. A smoothed 2D histogram of these estimates reveals the DOA of the present sources and through an iterative process, accurate 3D DOA information can be obtained. The performance of the proposed method is demonstrated through simulations in various signal-to-noise ratio and reverberation conditions.

关键词： Direction of arrival 3D Multiple sources microphone array processing Sound intensity sound intensity Direction of arrival estimation array processing Angle of arrival Three Dimensions United States Department of Agriculture signal to noise ratios SOUND Reverberation

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An Informed Parametric Spatial Filter Based on Instantaneous Direction-of-Arrival Estimates

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE processing 2014年第12期22卷 2182-2196页

作者： Thiergart, Oliver Taseska, Maja Habets, Emanuel A. P. Univ Erlangen Nurnberg Int Audio Labs Erlangen Joint Inst Friedrich Alexander Univ Erlangen Nur D-91058 Erlangen Germany

Extracting desired source signals in noisy and reverberant environments is required in many hands-free communication systems. In practical situations, where the position and number of active sources may be unknown and time-varying, conventional implementations of spatial filters do not provide sufficiently good performance. Recently, informed spatial filters have been introduced that incorporate almost instantaneous parametric information on the sound field, thereby enabling adaptation to new acoustic conditions and moving sources. In this contribution, we propose a spatial filter which generalizes the recently proposed informed linearly constrained minimum variance filter and informed minimum mean square error filter. The proposed filter uses multiple direction-of-arrival estimates and second-order statistics of the noise and diffuse sound. To determine those statistics, an optimal diffuse power estimator is proposed that outperforms state-of-the-art estimators. Extensive performance evaluation demonstrates the effectiveness of the proposed filter in dynamic acoustic conditions. For this purpose, we have considered a challenging scenario which consists of quickly moving sound sources during double-talk. The performance of the proposed spatial filter was evaluated in terms of objective measures including segmental signal-to-reverberation ratio and log spectral distance, and by means of a listening test confirming the objective results.

关键词： Dereverberation interference reduction microphone array processing optimal beamforming

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Location Feature Integration for Clustering-Based Speech Separation in Distributed microphone arrays

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE processing 2014年第2期22卷 354-367页

作者： Souden, Mehrez Kinoshita, Keisuke Delcroix, Marc Nakatani, Tomohiro Georgia Inst Technol Sch Elect & Comp Engn Atlanta GA 30308 USA NTT Commun Sci Labs Kyoto 6190237 Japan

In distributed microphone arrays (DMAs) the source location information can be defined at the intra and inter-node levels. Indeed, while the first type of information results from the diversity of acoustic channels recorded by microphones embedded in the same node, the second is attributed to the differences between the acoustic channels observed by spatially distributed nodes. Both cues are very useful in DMA processing, and the aim of this paper is to utilize both of them to cluster and separate multiple competing speech signals. To capture the intra-node information, we employ the normalized recording vector, while at the inter-node level, we consider different features including the energy level differences with and without the phase differences between nodes. We model the intra-node information using the Watson mixture model (WMM), and propose using the Gamma mixture model (GaMM), Dirichlet mixture model (DMM), and WMM to model different inter-node location features. Furthermore, we propose several integrations of the intra-node and inter-node feature contributions to cluster speech recordings using the expectation maximization algorithm. Finally, simulation results are provided to demonstrate the performance of all ensuing methods.

关键词： Blind source separation decision fusion distributed microphone array processing location-based speech clustering speech enhancement

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Speaker Localization by Humanoid Robots in Reverberant Environments 28

Speaker Localization by Humanoid Robots in Reverberant Envir...

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IEEE 28th Convention of Electrical and Electronics Engineers in Israel (IEEEI)

作者： Tourbabin, Vladimir Rafaely, Boaz Ben Gurion Univ Negev Dept Elect & Comp Engn IL-84105 Beer Sheva Israel

ISBN: (纸本)9781479959877

One of the important tasks of a humanoid-robot auditory system is speaker localization. It is used for the construction of the surrounding acoustic scene and as an input for additional processing methods. Localization is usually required to operate indoors under high reverberation levels. Recently, an algorithm for speaker localization under these conditions was proposed. The algorithm uses a spherical microphone array and the processing is performed in the spherical harmonics domain, requiring a relatively large number of microphones to efficiently cover the entire frequency range of speech. However, the number of microphones in the auditory system of a humanoid robot is usually limited. The current paper proposes an improvement of the previously published algorithm. The improvement aims to overcome the frequency limitations imposed by the insufficient number of microphones. The improvement is achieved by using a novel space-domain distance algorithm that does not requires the transformation to the spherical harmonics domain, thereby avoiding the frequency range limitations. A numerical study shows two important results. The first is that, using the improved algorithm, the operation frequency range can be significantly extended. The second important result is related to the fact that higher frequencies contain more detailed information about the surrounding sound field. Hence, the additional higher frequencies lead to improved localization accuracy.

关键词： array signal processing harmonics humanoid robots microphone arrays speaker recognition frequency range limitation humanoid robot auditory system microphone array processing reverberant environment speaker localization spherical harmonics domain arrays Direction-of-arrival estimation Estimation Humanoid robots microphones Multiple signal classification Speech

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SPATIAL PERCEPTION OF VIRTUAL X-Y RECORDINGS 14

SPATIAL PERCEPTION OF VIRTUAL X-Y RECORDINGS

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14th International Workshop on Acoustic Signal Enhancement (IWAENC)

作者： Kowalczyk, Konrad Craciun, Alexandra Dachmann, Christian Habets, Emanuel A. P. Int Audio Labs Erlangen Erlangen Germany Fraunhofer Inst Integrated Circuits IIS Erlangen Germany

ISBN: (纸本)9781479968084

Spatial sound acquisition aims at capturing the sound field such that spatial cues present in the sound scene at the recording side are preserved at the reproduction side. The virtual microphone technique is a spatial recording method that allows for an arbitrary selection of the recording position and orientation. It exploits the parametric information about the sound field estimated using distributed microphone arrays in order to synthesize the virtual microphone signals that sound perceptually similar to the signals that would be recorded using physical microphones located at the same positions. In this paper, the quality of the spatial image in virtual stereo recordings is evaluated through a set of subjective experiments, including dedicated tests for distance and angle perception. The listening test results show that the spatial image captured using virtual microphones is perceptually close to the image recorded with physical microphones positioned analogously.

关键词： spatial sound recording spatial audio perception microphone array processing spatial filtering

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User Determined Superdirective Beamforming 28

User Determined Superdirective Beamforming

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IEEE 28th Convention of Electrical and Electronics Engineers in Israel (IEEEI)

作者： Berkun, Reuven Cohen, Israel Benesty, Jacob Technion Israel Inst Technol IL-32000 Haifa Israel Univ Quebec INRS EMT Montreal PQ H5A 1K6 Canada

ISBN: (纸本)9781479959877

Superdirective beamforming is a well-known method for enhancement of reverberated speech signals. Nevertheless, it is very susceptible to errors in the sensor array characteristics, thermal noise, and white noise input, resulting in a low level of white noise gain, particularly at low frequencies. It is of great interest to develop a beamformer with superior enhancement of reverberated signals, having a high directivity factor, together with a relatively high white noise gain level. In this paper, a solution which controls both the directivity factor and the white noise gain is examined. We propose a linear weighted combination of two conventional beamformers, the regularized superdirective beamformer and the delay-and-sum beamformer. We analyze the beamformer gain responses, and consequently derive two user-determined frequency-dependent white noise gain and directivity factor beamformers, respectively. Simulation results approve our findings, and show of a robust user-controlled solution, with an effective tradeoff between the performance measures of the beamformer.

关键词： array signal processing delays speech synthesis white noise beamformer gain responses beamformers delay-and-sum beamformer directivity factor directivity factor beamformers microphone array processing reverberated signal enhancement reverberated speech signals sensor array characteristics speech signals superdirective beamforming thermal noise user-controlled solution user-determined frequency-dependent white noise gain array signal processing arrays microphones Robustness Vectors White noise

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A PLENACOUSTIC APPROACH TO ACOUSTIC SIGNAL EXTRACTION

A PLENACOUSTIC APPROACH TO ACOUSTIC SIGNAL EXTRACTION

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IEEE Workshop on Applications of Signal processing to Audio and Acoustics

作者： Lucio Bianchi Fabrizio D'Amelio Fabio Antonacci Augusto Sarti Stefano Tubaro Dipartimento di Elettronica Informazione e Bioingegneria Politecnico di Milano

ISBN: (纸本)9781479974511

This paper considers the problem of separation of acoustic sources from convolutive mixtures captured by a microphone array. The problem is approached through Plane Wave Decomposition (PWD) of the sound field measured at multiple points along the extension of the array. The directional components of the sound field are analyzed by means of the plenacoustic framework to accurately estimate the direction of arrival of the desired and undesired sources at every point at which the PWD is measured. Multiple spatial filters are designed, one for each PWD measurement point, to leave undistorted the desired source and attenuate the interferer. A successive stage of delay and sum of the outputs of the individual spatial filters enables the reconstruction of the desired source. The use of the plenacoustic framework allows us to gather intuitive and immediate interpretation of the acoustic scene. We prove the effectiveness of the proposed solution through simulations on speech data.

关键词： microphone array processing Informed spatial filtering Plenacoustic function

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Design of microphone arrays for hearing aids optimized to unknown subjects

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SIGNAL processing 2013年第11期93卷 3239-3250页

作者： Ayllon, David Gil-Pita, Roberto Rosa-Zurera, Manuel Univ Alcala Dept Signal Theory & Commun Madrid 28508 Spain

The improvement of speech intelligibility in hearing aids is still an unsolved problem. Modern devices, either monaural or binaural, may include microphone arrays to provide directivity by means of spatial filtering, but the signals that arrive at the array are distorted by the so-called head shadow effect, which must be considered in the design. The fact that these effects are highly dependent on a person causes the design of an array customized for a subject to need a correct measurement of these effects, which is not practical in real scenarios. The lack of information about the head causes directivity reduction and distortions. In this work we propose three different methods to compute optimized filter coefficients with the purpose of maximizing the array gain and minimizing distortions, and with the objective that these filter coefficients are valid for all users. Furthermore, measurements of the amplitude and phase distortions caused by a lack of head information in the design are introduced. In addition, the methods proposed here have been evaluated in 13 different array configurations. Results show that these approaches reduce significantly the gain reduction and distortions, obtaining a similar intelligibility in comparison to the case where head information is available. (C) 2013 Elsevier B.V. All rights reserved.

关键词： Hearing aids microphone array processing Speech enhancement

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SOUND ACQUISITION IN NOISY AND REVERBERANT ENVIRONMENTS USING VIRTUAL microphoneS

SOUND ACQUISITION IN NOISY AND REVERBERANT ENVIRONMENTS USIN...

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14th IEEE Workshop on Applications of Signal processing to AudNew Paltzio and Acoustics (WASPAA)

作者： Kowalczyk, Konrad Thiergart, Oliver Craciun, Alexandra Habets, Emanuel A. P. Fraunhofer Inst Integrated Circuits IIS Erlangen Germany Int Audio Labs Erlangen Erlangen Germany

ISBN: (纸本)9781479909728

In hands-free communication applications, the main goal is to capture desired sounds, while reducing noise and interfering sounds. However, for natural-sounding telepresence systems, the spatial sound image should also be preserved. Using a recently proposed method for generating the signal of a virtual microphone (VM), one can recreate the sound image from an arbitrary point of view in the sound scene (e.g., close to a desired speaker), while being able to place the physical microphones outside the sound scene. In this paper, we present a method for synthesizing a VM signal in noisy and reverberant environments, where the estimation of the required direct and diffuse sound components is performed using two multichannel linear filters. The direct sound component is estimated using a multichannel Wiener filter, while the diffuse sound component is estimated using a linearly constrained minimum variance filter followed by a single-channel Wiener filter. Simulations in a noisy and reverberant environment show the applicability of the proposed method for sound acquisition in a scenario in which two microphone arrays are installed in a large TV.

关键词： microphone array processing spatial filtering Wiener filtering spatial sound recording

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