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IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)

作者： Nakamura, Keisuke Gomez, Randy Honda Res Inst Japan Co Ltd 8-1 Honcho Wako Saitama 3510188 Japan

ISBN: (纸本)9781509047888

In this paper, we propose a novel approach of sound source separation for meeting conversations even when using an uncalibrated microphone array. Our method can blindly estimate three parameters for separation, namely Steering Vectors (SVs), speaker indices, and activity periods of each speaker. First, we estimate the number of speakers and SVs by clustering Time Delay Of Arrival (TDOA) of the observed signal and selecting major clusters to compute TDOA-based SVs. Then, speaker indices and activity periods are estimated by thresholding spatial spectrum using estimated SVs, whose threshold is blindly obtained. Finally, we separate overlapped speeches/noise based on dynamic design of noise correlation matrices of the minimum variance distortionless response (MVDR) beamformer using blindly estimated parameters. The proposed algorithm was evaluated in both separation objective measure and recognition correct rate and showed improvements in both single and simultaneous speech scenarios in a reverberant meeting room. Moreover, the blindly estimated parameters improved separation and recognition compared to geometrically obtained parameters.

关键词： microphone array processing Sound Source Separation Speaker Diarization Speaker Indexing

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A CONFORMAL, HELMET-MOUNTED microphone array FOR AUDITORY SITUATIONAL AWARENESS AND HEARING PROTECTION

A CONFORMAL, HELMET-MOUNTED MICROPHONE ARRAY FOR AUDITORY SI...

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IEEE Workshop on Applications of Signal processing to Audio and Acoustics (WASPAA)

作者： Calamia, Paul Davis, Shakti Smalt, Christopher Weston, Christine MIT Lincoln Lab 244 Wood St Lexington MA 02420 USA

ISBN: (纸本)9781538616321

Hearing protectors often are considered detrimental to auditory situational awareness due in part to the distortions they impose on localization cues. The addition of a helmet or other head-worn equipment can exacerbate this problem by covering the ears and effectively changing the shape of the head. In an effort to design a hearing-protection system that maintains natural auditory localization performance, we have built a prototype conformal microphone array integrated into a helmet. The system involves beamforming for spatial selectivity, filtering with head-related transfer functions to reintroduce spatial auditory cues to the beamformed data, dynamic range compression to attenuate potentially damaging sound levels, and insert earphones to deliver binaural audio and provide additional hearing protection. All directions are auralized simultaneously to provide full spatial coverage around the wearer. In this paper we describe the design, implementation, and testing of a prototype 32-channel system, including 3D modeling and simulations, signal-processing approaches, real-time processing, and localization performance.

关键词： Spatial Hearing microphone array processing Hearing Protection

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Real-World Evaluation of Multichannel Audio Enhancement Using Acoustic Pilot Signals 51

Real-World Evaluation of Multichannel Audio Enhancement Usin...

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51st IEEE Asilomar Conference on Signals Systems and Computers

作者： Corey, Ryan M. Singer, Andrew C. Univ Illinois Champaign IL 61820 USA

ISBN: (纸本)9781538618233

In reverberant environments, the performance of multichannel audio enhancement systems depends on the accuracy with which the acoustic impulse responses between sources and microphones can be estimated. Since these responses are difficult to estimate in real-world environments, many multichannel audio enhancement experiments are conducted using simulated data. We propose an experimental method that sidesteps the channel estimation problem by measuring the acoustic impulse responses using pilot signals emitted from a beacon placed near the source of interest. We demonstrate the utility of the method on a multichannel speech mixture recorded in a noisy and reverberant room.

关键词： Audio enhancement microphone array processing beamforming source separation system identification

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SoundCraft: Enabling Spatial Interactions on Smartwatches using Hand Generated Acoustics 17

SoundCraft: Enabling Spatial Interactions on Smartwatches us...

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30th Annual ACM Symposium on User Interface Software and Technology (UIST)

作者： Han, Teng Hasan, Khalad Nakamura, Keisuke Gomez, Randy Irani, Pourang Univ Manitoba Winnipeg MB Canada Honda Res Inst 8-1 Honcho Wako Saitama Japan

ISBN: (纸本)9781450349819

We present SoundCraft, a smartwatch prototype embedded with a microphone array, that localizes angularly, in azimuth and elevation, acoustic signatures: non-vocal acoustics that are produced using our hands Acoustic signatures are common in our daily lives, such as when snapping or rubbing our fingers, tapping on objects or even when using an auxiliary object to generate the sound. We demonstrate that we can capture and leverage the spatial location of such naturally occurring acoustics using our prototype. We describe our algorithm, which we adopt from the MUltiple Signal Classification (MUSIC) technique [31], that enables robust localization and classification of the acoustics when the microphones are required to be placed at close proximity. SoundCraft enables a rich set of spatial interaction techniques, including quick access to smartwatch content, rapid command invocation, in-situ sketching, and also multi-user around device interaction. Via a series of user studies, we validate SoundCraft's localization and classification capabilities in non-noisy and noisy environments.

关键词： microphone array processing smartwatch spatial input acoustic signatures classification and localization

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Real-World Evaluation of Multichannel Audio Enhancement Using Acoustic Pilot Signals

Real-World Evaluation of Multichannel Audio Enhancement Usin...

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Asilomar Conference on Signals, Systems, and Computers

作者： Ryan M. Corey Andrew C. Singer University of Illinois at Urbana-Champaign

关键词： Audio enhancement microphone array processing Beamforming Source separation System identification

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AN INFORMED SEPARATION ALGORITHM BASED ON SOUND FIELD MAPPING FOR SPEECH RECOGNITION SYSTEMS 15

AN INFORMED SEPARATION ALGORITHM BASED ON SOUND FIELD MAPPIN...

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15th International Workshop on Acoustic Signal Enhancement (IWAENC)

作者： Markovic, Dejan Popat, Jigyasa Antonacci, Fabio Sarti, Augusto Kumar, T. Kishore Politecn Milan Dipartmento Elettron Informaz & Bioingn Piazza Leonardo da Vinci 32 I-20133 Milan Italy Natl Inst Technol Dept Elect & Commun Engn Warangal 506004 Telangana India Politecn Milan Milan Italy

ISBN: (纸本)9781509020072

This paper considers the problem of separation of speech sources from signals captured by a microphone array, and its impact on speech recognition systems. The proposed method improves the novel separation procedure based on processing of sound field maps in the ray space by incorporating the estimates of signal spectral envelopes into the design of the separation filter. In particular, we resort to a two-stage algorithm in which the assumptions regarding the signal spectral densities made in the first stage are replaced by their parametric estimates in the second stage. The performance gain is evaluated in real experiments and the impact on speech recognition accuracy is examined using several commercial cloud-based speech recognition APIs.

关键词： microphone array processing source separation speech recognition soundfield imaging

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An Efficient Parameterization of the Room Transfer Function

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE processing 2015年第12期23卷 2217-2227页

作者： Samarasinghe, Prasanga Abhayapala, Thushara Poletti, Mark Betlehem, Terence Australian Natl Univ Res Sch Engn Canberra ACT 2601 Australia Australian Natl Univ Res Sch Engn Canberra ACT 0200 Australia Callaghan Innovat Lower Hutt 5040 New Zealand

This paper proposes an efficient parameterization of the room transfer function (RTF). Typically, the RTF rapidly varies with varying source and receiver positions, hence requires an impractical number of point to point measurements to characterize a given room. Therefore, we derive a novel RTF parameterization that is robust to both receiver and source variations with the following salient features: 1) The parameterization is given in terms of a modal expansion of 3D basis functions. 2) The aforementioned modal expansion can be truncated at a finite number of modes given that the source and receiver locations are from two sizeable spatial regions, which are arbitrarily distributed. 3) The parameter weights/coefficients are independent of the source/receiver positions. Therefore, a finite set of coefficients is shown to be capable of accurately calculating the RTF between any two arbitrary points from a pre-defined spatial region where the source(s) lie and a pre-defined spatial region where the receiver(s) lie. A practical method to measure the RTF coefficients is also provided, which only requires a single microphone unit and a single loudspeaker unit, given that the room characteristics remain stationary over time. The accuracy of the above parameterization is verified using appropriate simulation examples.

关键词： Room transfer function (RTF) room reverberation microphone array processing

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3D LOCALIZATION OF MULTIPLE SOUND SOURCES WITH INTENSITY VECTOR ESTIMATES IN SINGLE SOURCE ZONES 23

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

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23rd European Signal processing Conference (EUSIPCO)

作者： Pavlidi, Despoina Delikaris-Manias, Symeon Pulkki, Ville Mouchtaris, Athanasios FORTH ICS GR-70013 Iraklion Crete Greece Univ Crete Dept Comp Sci GR-70013 Iraklion Crete Greece Aalto Univ Dept Signal Proc & Acoust FI-00076 Espoo Finland

ISBN: (纸本)9780992862633

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 microphone 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

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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 (WASPAA)

作者： Bianchi, Lucio D'Amelio, Fabrizio Antonacci, Fabio Sarti, Augusto Tubaro, Stefano Politecn Milan Dipartimento Elettron Informaz & Bioingn Piazza Leonardo da Vinci 32 I-20133 Milan Italy

ISBN: (纸本)9781479974504

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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A Bayesian approach to spatial filtering and diffuse power estimation for joint dereverberation and noise reduction 40

A Bayesian approach to spatial filtering and diffuse power e...

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40th IEEE International Conference on Acoustics, Speech, and Signal processing, ICASSP 2015

作者： Chakrabarty, Soumitro Thiergart, Oliver Habets, Emanuel A.P. International Audio Laboratories Erlangen Am Wolfsmantel 33 Erlangen Germany

ISBN: (纸本)9781467369978

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. © 2015 IEEE.

关键词： Bayesian beamforming dereverberation microphone array processing

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