60 research outputs found

    Optimal Binaural LCMV Beamforming in Complex Acoustic Scenarios: Theoretical and Practical Insights

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    Binaural beamforming algorithms for head-mounted assistive listening devices are crucial to improve speech quality and speech intelligibility in noisy environments, while maintaining the spatial impression of the acoustic scene. While the well-known BMVDR beamformer is able to preserve the binaural cues of one desired source, the BLCMV beamformer uses additional constraints to also preserve the binaural cues of interfering sources. In this paper, we provide theoretical and practical insights on how to optimally set the interference scaling parameters in the BLCMV beamformer for an arbitrary number of interfering sources. In addition, since in practice only a limited temporal observation interval is available to estimate all required beamformer quantities, we provide an experimental evaluation in a complex acoustic scenario using measured impulse responses from hearing aids in a cafeteria for different observation intervals. The results show that even rather short observation intervals are sufficient to achieve a decent noise reduction performance and that a proposed threshold on the optimal interference scaling parameters leads to smaller binaural cue errors in practice.Comment: To appear in Proc. IWAENC 201

    A spatial enhancement approach for binaural rendering of head-worn microphone arrays

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    This paper builds upon a recently proposed spatial enhancement approach, which has demonstrated im- provements in the perceived spatial accuracy of binaurally rendered signals using head-worn microphone arrays. The foundation of the approach is a parametric sound-field model, which assumes the existence of a single source and an isotropic diffuse component for each time-frequency index. The enhancement approach involves the post-processing of an initial estimate of the binaural signals, in order to obtain a refined esti- mate of binaural signals which more closely represent the inter-aural cues corresponding to the sound-field model. In this contribution, the enhancement approach has been implemented as an open-source framework, written in both the MATLAB and C programming languages, and as a real-time audio plug-in. The frame- work was also extended to offer direction-dependent gain control of sound sources relative to the listener, and a frequency-dependent control of the direct-to-diffuse balance, which are modifications that may find application within future augmented reality headsets and assistive hearing devices.publishedVersionNon peer reviewe

    Speech enhancement in binaural hearing protection devices

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    The capability of people to operate safely and effective under extreme noise conditions is dependent on their accesses to adequate voice communication while using hearing protection. This thesis develops speech enhancement algorithms that can be implemented in binaural hearing protection devices to improve communication and situation awareness in the workplace. The developed algorithms which emphasize low computational complexity, come with the capability to suppress noise while enhancing speech

    Audio source separation into the wild

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    International audienceThis review chapter is dedicated to multichannel audio source separation in real-life environment. We explore some of the major achievements in the field and discuss some of the remaining challenges. We will explore several important practical scenarios, e.g. moving sources and/or microphones, varying number of sources and sensors, high reverberation levels, spatially diffuse sources, and synchronization problems. Several applications such as smart assistants, cellular phones, hearing aids and robots, will be discussed. Our perspectives on the future of the field will be given as concluding remarks of this chapter

    Model-based speech enhancement for hearing aids

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