Adaptive array noise suppression of handsfree speaker input in cars

Nordholm, Sven; Claesson, Ingvar; Bengtsson, Bengt

doi:10.1109/25.260760

Cited by 61 publications

(31 citation statements)

References 9 publications

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“…using a more robust fixed blocking matrix design [4][5][6][7]; using an adaptive blocking matrix [8][9][10]; or by constructing a blocking matrix based on estimating the ratios of the acoustic transfer functions from the speech source to the microphone array [11]; • limiting the distorting effect of the remaining speech leakage components by · updating the multichannel ANC only during periods (and for frequencies)…”

Section: Introductionmentioning

confidence: 99%

Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction

Doclo

Spriet²,

Wouters³

et al. 2007

Speech Communication

132

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Section: Introductionmentioning

confidence: 99%

Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction

Doclo

Spriet²,

Wouters³

et al. 2007

Speech Communication

132

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“…Techniques in the first class reduce the leakage components in the noise references i) by using a more robust fixed blocking matrix design [7]- [10], ii) by using an adaptive blocking matrix [11]- [13], or iii) by constructing a blocking matrix based on estimating the ratios of the acoustic transfer functions from the speech source to the microphone array [6], [14], [15]. Techniques in the second class limit the distorting effect of the leakage components i) by controlling the step-size of the multichannel adaptive algorithm, such that the multichannel ANC is only updated during periods 1558-7916/$31.00 © 2013 IEEE (and for frequencies) where the signal-to-noise ratio is low [6], [7], [11]- [13], [16]- [18], ii) by constraining the update formula for the multichannel adaptive filter [9], [12], [19], [20], or iii) by taking speech distortion due to speech leakage into account using a speech distortion weighted multichannel Wiener filter [21], [22]. In practice, several of these techniques are usually combined.…”

Section: E a P Habets Is With The International Audio Laboratoriesmentioning

confidence: 99%

A Two-Stage Beamforming Approach for Noise Reduction and Dereverberation

Habets

Benesty

2013

IEEE Trans. Audio Speech Lang. Process.

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Abstract-In general, the signal-to-noise ratio as well as the signal-to-reverberation ratio of speech received by a microphone decrease when the distance between the talker and microphone increases. Dereverberation and noise reduction algorithm are essential for many applications such as videoconferencing, hearing aids, and automatic speech recognition to improve the quality and intelligibility of the received desired speech that is corrupted by reverberation and noise. In the last decade, researchers have aimed at estimating the reverberant desired speech signal as received by one of the microphones. Although this approach has let to practical noise reduction algorithms, the spatial diversity of the received desired signal is not exploited to dereverberate the speech signal. In this paper, a two-stage beamforming approach is presented for dereverberation and noise reduction. In the first stage, a signal-independent beamformer is used to generate a reference signal which contains a dereverberated version of the desired speech signal as received at the microphones and residual noise. In the second stage, the filtered microphone signals and the noisy reference signal are used to obtain an estimate of the dereverberated desired speech signal. In this stage, different signal-dependent beamformers can be used depending on the desired operating point in terms of noise reduction and speech distortion. The presented performance evaluation demonstrates the effectiveness of the proposed two-stage approach.

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“…In order to make the array work satisfactorily in the car interior context where the speaker is a source in the near-"eld subject to motions, preventive measures must be taken in order to avoid cancellation of the speaker's voice. A simple and straightforward method to control the optimisation of the "lter is to constraint spatial "ltering in the blocking matrix of the general sidelobe canceller, in order to select angular intervals over which the array is not allowed to cancel the target signal [54]. For "xed beamformers combined with post-processing treatments, an interesting choice of microphone positions can be found in [40].…”

Section: Multi-microphones Techniquesmentioning

confidence: 99%

Innovative speech processing for mobile terminals: an annotated bibliography

et al. 2000

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This paper gives an overview of recent bibliographic references dealing with speech processing in mobile terminals. Its purpose is to point out state of the art issues in the area; thus a fairly large list of references taken from many conferences proceedings and journals is given and commented. General considerations about speech processing in mobile communications are "rstly introduced; then we deal with audio processing for speech enhancement in mobile terminals and with low bit-rate speech coding. Speech recognition is addressed with some accent put on mobile applications. A short overview of implementation aspects of speech processing algorithms in mobile terminals is also given. Finally, open issues and problems are listed.2000 Published by Elsevier Science B.V. All rights reserved. ZusammenfassungDiese Arbeit gibt einen U G berblick auf neue bibliogra"sche Referenzen, die sich mit der Sprachverarbeitung in mobilen Einheiuten beschaK ftigen. Unser Ziel ist es, den Stand der technischen VeroK !entlichungen auf diesem Gebiet zu ermitteln; somit ist eine recht lange Liste von Referenzen aus vielen Konferenzen und Zeitschiften und die zugehoK rigen Kommentare angegeben. Allgemeine U G berlegungen zur Sprachverarbeitung in der Mobilkommunikation werden als erstes angefuK hrt; anschlie{end behandeln wir Verfahren zur Spracheverbesserung in mobilen Systemen basierend auf niedrigen Bitraten. Bei der anschlieend behandelten Spracherkennung wird ein gewisser Schwerpunkt auf mobile Systeme gelegt. Ein kurzer U G berblick zum Implementierungsaspekt der Sparchverarbeitungsverfahren in mobilen EndgeraK ten ist ebenfalls angegeben. Schlielich werden o!ene Themen und Probleme aufgezeigt.2000 Published by Elsevier Science B.V. All rights reserved. Re2 sume2Cet article passe en revue la bibliographie reH cente concernant le traitement de la parole dans les terminaux mobiles. Son objectif est de mettre l'accent sur l'eH tat de l'art dans le domaine; en conseH quence une importante liste de reH feH rences recueillies dans des actes de congre`s et des revues est donneH e et commenteH e. En premier lieu, des consideH rations geH neH rales sur le traitement de la parole dans les communications mobiles sont preH senteH es; puis nous discutons de traitements de rehaussement de la parole dans les terminaux mobiles ainsi que du codage de la parole a`bas deH bits. La reconnaissance vocale est traiteH e avec un accent particulier mis sur les applications mobiles. L'implantation des algorithmes de traitement de la parole dans les terminaux mobiles est aussi brie`vement traiteH e. En"n, certaines questions et proble`mes ouverts sont citeH s.

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Adaptive array noise suppression of handsfree speaker input in cars

Cited by 61 publications

References 9 publications

Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction

Frequency-domain criterion for the speech distortion weighted multichannel Wiener filter for robust noise reduction

A Two-Stage Beamforming Approach for Noise Reduction and Dereverberation

Innovative speech processing for mobile terminals: an annotated bibliography

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