| 1. |
- Cornelius, Per, et al.
(författare)
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A Spatially Constrained Subband Beamforming Algorithm for speech enhancement
- 2004
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Konferensbidrag (refereegranskat)abstract
- This paper discusses speech enhancement in an enclosed environment such as communication in a motorcycle helmet. A new constrained subband adaptive beamformer is proposed, which uses the concept of an earlier proposed calibrated beamformer mainly developed for a hands-free in-car environment. The highly non-stationary nature of the disturbing sound field encountered in an motorcycle helmet and the fact that the source is situated in the extreme nearfield of the array, causes the beamformer to produce an unwanted fluctuation in the output level. The spatially constrained beamformer proposed in this paper makes sure that the output maintains a constant gain, as long as the corresponding source originates from the desired location.
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| 2. |
- Yermeche, Zohra, et al.
(författare)
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A Calibrated Subband Beamforming Algorithm for Speech Enhancement
- 2002
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Konferensbidrag (refereegranskat)abstract
- This paper proposes a new calibrated adaptive frequency domain beamformer for speech enhancement. The beamformer is based on the principle of a soft constraint formed from calibration data, rather than precalculated from free-field assumptions. The benefit is that the real room acoustical properties will be taken into account. The proposed algorithm continuously estimates the spatial information for each frequency band, based on weighting of the received data. The update of the beamforming weights is done recursively where the initial precalculated correlation estimates of the speech constitute a soft constraint. The soft constraint secures the spatial-temporal passage of the desired source signal, without the need of any speech detection. The performance is evaluated in real world scenarios, both in a car-and a restaurant- environment. Interference and noise sup-pression of more than 15 dB is accomplished, while very small distortion is measured for the source signal.
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| 3. |
- Yermeche, Zohra, et al.
(författare)
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Spatial Filter Bank Design for Speech Enhancement Beamforming Applications
- 2004
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Ingår i: Sensor Array and Multichannel Signal Processing Workshop Proceedings, 2004. - Barcelona : IEEE. - 0780385454 ; , s. 557-560
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Konferensbidrag (refereegranskat)abstract
- In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constraint of signal passage at one position (and closing in other positions corresponding to known disturbing sources). By performing the directional opening towards the desired location in the fixed filter bank structure, the beamformer is left with the task of tracking and suppressing the continuously emerging noise sources. This algorithm has been implemented in MATLAB and tested on real speech recordings conducted in a car hands-free communication situation. Results show that a reduction of the total complexity can be achieved while maintaining the noise suppression performance and reducing the speech distortion.
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| 4. |
- Yermeche, Zohra
(författare)
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Subband Beamforming for Speech Enhancement in Hands-Free Communication
- 2004
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Licentiatavhandling (övrigt vetenskapligt/konstnärligt)abstract
- Speech enhancement by means of microphone array signal processing has a major role in voice communication applications such as audio-conferencing, hands-free telephony, voice recognition and hearing aids. In these communication scenarios, the speaker is positioned at a remote distance from the microphones, which causes problems of environment noise and interfering sound corrupting the received speech. Additionally, reverberations of the voice from walls or ceilings, also impairs the received speech signal. In the case of a duplex communication, the acoustic feedback constitutes another disturbance for the talker who hears his or her voice echoed. Successful speech enhancement solutions should achieve speech dereverberation, efficient noise and interference reduction, and for mobile environments, they should also provide an adaptation capacity to speaker motion. Microphone arrays spatially sample the sound pressure field. When combined with spatio-temporal filtering techniques known as {\em beamforming}, they can extract the sound source information from signals, of which only a mixture is observed. This is based on the inherent ability of sensor arrays to exploit the spatial correlation of multiple received signals. A subband beamforming structure can be used in order to improve the performance of the time-domain filters and reduce their computational complexity. Each of the received signals is decomposed into a set of narrow-band signals and the filtering operations of the beamformer are performed for each frequency band separately. The output of the subband beamformers are then used to reconstruct a full-band output signal. In this thesis an adaptive subband RLS beamforming approach is investigated and evaluated in real hands-free acoustical environments. The proposed methodology is defined such to perform background noise and acoustic coupling reduction, while producing an undistorted filtered version of the signal originating from a desired location. The beamformer recursively minimizes a Least Squares error based on the continuously received data. This adaptive structure allows for a tracking of the noise characteristics, such to accomplish its attenuation in an efficient manner. A soft constraint built from calibration data in low noise conditions guarantee the integrity of the desired signal without the need of any speech detection. Additionally, a new spatial filter bank design method for beamforming applications, which includes the constraint of signal passage at one position and closing in other undesired positions, is suggested. Furthermore, to allow for source mobility tracking, a soft constrained beamforming approach with built-in speaker localization, is proposed. The source of interest is modelled as a cluster of point sources and source motion is accommodated by revising the point source cluster. Real speech signals are used in the simulations and results show accurate speaker movement tractability with maintained noise and interference suppression of about 10-15 dB, when using a four-microphone array.
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