Sound System Engineering

Davis, Don; Patronis, Eugene T.

doi:10.4324/9780080959603

Cited by 22 publications

(27 citation statements)

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“…The pressure gradient sensors in the AVS produce a direct representation of the particle velocity [14] the frequency responses of these microphones are different to that of the omni-directional microphone which is a direct representation of the pressure at the array [15]. The frequency responses of the two gradient sensors have a high-pass effect [15].…”

Section: Multichannel Lp Of Reverberant Avs Recordingsmentioning

confidence: 96%

Speech dereverberation based on Linear Prediction: An Acoustic Vector Sensor approach

Shujau

Ritz

Burnett

2013

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

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This paper introduces a dereverberation algorithm based on Linear Prediction (LP) applied to the outputs of an Acoustic Vector Sensor (AVS). The approach applies adaptive beamforming to take advantage of the directional outputs of the AVS array to obtain a more accurate LP spectrum than can be obtained with a single channel or Uniform Linear Array (ULA) with a comparable number of channels. This is then used within a modified version of the Spatiotemporal Averaging Method for Enhancement of Reverberant Speech (SMERSH) algorithm derived for the AVS to enhance the LP residual signal. In a highly reverberant environment, the approach demonstrates a significant improvement compared to a ULA as measured by both the Signal to Reverberant Ratio (SRR) and Speech to Reverberation Modulation Energy Ratio (SRMR) for sources ranging from at 1m to 5m from the array.The effect of reverberation on speech and audio plays an important role in making the sound more natural, but when the level of reverberation increases (beyond = 1s) it significantly degrades the quality of the speech signals in terms of intelligibility due to box effect and distant talker effect [1,2]. Reverberation also leads to altering of the parameters derived for source-filter models of speech typically used in applications such as distant speech recognition and hence degrades their performance [2]. There are several methods that can be used for enhancing speech through dereverberation including beamforming methods; traditional speech enhancement methods; and blind system identification and equalization methods, where the acoustic impulse response of a room is identified blindly and then used to design an equalization filter [2]. These methods can also be categorized into single or multichannel approaches. This paper focuses on the latter methods, which generally result in significant performance gains compared to single channel approaches [2].Estimation of a room impulse response or acoustic transfer function (ATF) is difficult to obtain accurately [2] as it depends on the geometry, the furniture in the room and materials used in the construction of the room. In contrast multichannel algorithms that do not rely on the ATF but instead are based on Linear Predictive Coding have been shown to be highly successful [3][4][5], including the Spatiotemporal Averaging Method for Enhancement of Reverberant Speech (SMERSH) [2] [6] that was designed for a Uniform Linear Array (ULA) of microphones. In this paper, a new multichannel dereverberation method based on adapting the SMERSH algorithm for an Acoustic Vector Sensor (AVS) is proposed. The AVS has three co-located velocity gradient microphones and one omni-direction microphone arranged orthogonally in an area occupying no more than 1cm 3 [7]. The use of gradient sensors allows for precise recording of directional sound component and has been shown to be affective for speech enhancement in additive noise and mildly reverberant environments [8-10] and here it is shown that adapting SMERSH to the AVS provides a superio...

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Section: Multichannel Lp Of Reverberant Avs Recordingsmentioning

confidence: 96%

Speech dereverberation based on Linear Prediction: An Acoustic Vector Sensor approach

Shujau

Ritz

Burnett

2013

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

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“…Other names have subsequently been used to describe it, including the "time response" 20 and the "envelope time curve". 21 Regardless, the Polar ETC does allow the localization of points of reflection and has been widely used for that purpose.…”

Section: Theoretical Predictionsmentioning

confidence: 99%

The effects of non-cardioid directivity on incidence angle estimation using the polar energy time curve

Esplin

Anderson

Leishman

et al. 2011

The Journal of the Acoustical Society of America

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Assessment of desirable reflections and control of undesirable reflections in rooms are best accomplished if the reflecting surfaces are properly localized. Several measurement techniques exist to identify the incident direction of reflected sound, including the useful polar energy time curve (Polar ETC), which requires six cardioid impulse response measurements along the Cartesian axes. The purpose of this investigation is to quantify the incidence angle estimation error introduced into the Polar ETC by non-cardioid microphone directivities. The results demonstrate that errors may be minimized with a cardioid-family microphone possessing a certain range of directivities and by maximizing the measurement signal-to-noise ratio.

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“…The level of the direct signal is dependent on the distance from the speaker to the listener or microphone. This is characterised by the so called "6 dB rule" where the average level of direct speech falls by 6 dB for every doubling of distance from the lips [10].…”

Section: Distance Effect and Normalisationmentioning

confidence: 99%

Audio head pose estimation using the direct to reverberant speech ratio

Barnard

Wang

Kittler

2013

2013 IEEE International Conference on Acoustics, Speech and Signal Processing

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Head pose is an important cue in many applications such as, speech recognition and face recognition. Most approaches to head pose estimation to date have used visual information to model and recognise a subject's head in different configurations. These approaches have a number of limitations such as, inability to cope with occlusions, changes in the appearance of the head, and low resolution images. We present here a novel method for determining coarse head pose orientation purely from audio information, exploiting the direct to reverberant speech energy ratio (DRR) within a highly reverberant meeting room environment. Our hypothesis is that a speaker facing towards a microphone will have a higher DRR and a speaker facing away from the microphone will have a lower DRR. This hypothesis is confirmed by experiments conducted on the publicly available AV16.3 database. © 2013 IEEE

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Sound System Engineering

Cited by 22 publications

References 0 publications

Speech dereverberation based on Linear Prediction: An Acoustic Vector Sensor approach

Speech dereverberation based on Linear Prediction: An Acoustic Vector Sensor approach

The effects of non-cardioid directivity on incidence angle estimation using the polar energy time curve

Audio head pose estimation using the direct to reverberant speech ratio

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