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

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

The personal hearing system- A software hearing aid for a personal communication system

A concept and architecture of a personal communication system (PCS) is introduced that integrates audio communication and hearing support for the elderly and hearing-impaired through a personal hearing system (PHS). The concept envisions a central processor connected to audio headsets via a wireless body area network (WBAN). To demonstrate the concept, a prototype PCS is presented that is implemented on a netbook computer with a dedicated audio interface in combination with a mobile phone. The prototype can be used for field-testing possible applications and to reveal possibilities and limitations of the concept of integrating hearing support in consumer audio communication devices. It is shown that the prototype PCS can integrate hearing aid functionality, telephony, public announcement systems, and home entertainment. An exemplary binaural speech enhancement scheme that represents a large class of possible PHS processing schemes is shown to be compatible with the general concept. However, an analysis of hardware and software architectures shows that the implementation of a PCS on future advanced cell phone-like devices is challenging. Because of limitations in processing power, recoding of prototype implementations into fixed point arithmetic will be required and WBAN performance is still a limiting factor in terms of data rate and delay

Spatial hearing rendering in wireless microphone systems for binaural hearing aids

In 2015, 360 million people, including 32 million children, were suffering from hearing impairment all over the world. This makes hearing disability a major worldwide issue. In the US, the prevalence of hearing loss increased by 160% over the past generations. However, 72% of the 34 million impaired American persons (11% of the population) still have an untreated hearing loss. Among the various current solutions alleviating hearing disability, hearing aid is the only non-invasive and the most widespread medical apparatus. Combined with hearing aids, assisting listening devices are a powerful answer to address the degraded speech understanding observed in hearing-impaired subjects, especially in noisy and reverberant environments. Unfortunately, the conventional devices do not accurately render the spatial hearing property of the human auditory system, weakening their benefits. Spatial hearing is an attribute of the auditory system relying on binaural hearing. With 2 ears, human beings are able to localize sounds in space, to get information about the acoustic surroundings, to feel immersed in environments... Furthermore, it strongly contributes to speech intelligibility. It is hypothesized that recreating an artificial spatial perception through the hearing aids of impaired people might allow for recovering a part of these subjects' hearing performance. This thesis investigates and supports the aforementioned hypothesis with both technological and clinical approaches. It reveals how certain well-established signal processing methods can be integrated in some assisting listening devices. These techniques are related to sound localization and spatialization. Taking into consideration the technical constraints of current hearing aids, as well as the characteristics of the impaired auditory system, the thesis proposes a novel solution to restore a spatial perception for users of certain types of assisting listening devices. The achieved results demonstrate the feasibility and the possible implementation of such a functionality on conventional systems. Additionally, this thesis examines the relevance and the efficiency of the proposed spatialization feature towards the enhancement of speech perception. Via a clinical trial involving a large number of patients, the artificial spatial hearing shows to be well appreciated by disabled persons, while improving or preserving their current hearing abilities. This can be considered as a prominent contribution to the current scientific and technological knowledge in the domain of hearing impairment

Effect of dynamic range compression on attending to sounds based on spatial location

Thesis (Ph. D. in Speech and Hearing Bioscience and Technology)--Harvard-MIT Program in Health Sciences and Technology, 2013.Cataloged from PDF version of thesis.Includes bibliographical references (pages 118-125).Many hearing aids introduce nonlinear compressive gain to accommodate the reduced dynamic range that often accompanies hearing loss. Unfortunately, when applied independently at either ear, this gain can introduce fluctuations in interaural level difference (ILD), which is an important cue for spatial perception and attending to sounds in an acoustic mixture. Moreover, natural sounds produce complicated interactions between different sounds in a mixture, as a compressor's gain is driven by whichever source dominates the mixture within a specified temporal window. While independent compression can interfere with spatial perception of sound, it does not always interfere with localization accuracy or speech identification. This thesis investigates the role of dynamic range compression on the ability to attend to target speech in the presence of interfering speech. First, the fundamental concepts behind dynamic range compression and its use are introduced, and used to develop a framework to understand some of the possible effects on ILD and spatial perception. This framework is applied toward the interpretation of the existing literature regarding dynamic range compression and spatial perception, bringing together a seemingly contradictory range of results. In particular, the framework presented here predicts that dynamic range compression will only affect performance in tasks for which relatively small spatial separations are tested, whereas many existing studies compare only large spatial separations to no spatial separation. We describe and analyze the results of an experiment designed to test this prediction by systematically varying the spatial separation between different speech sources that normal-hearing listeners attended to. We found a robust but modest detrimental effect of dynamic range compression on listeners' performance. Linking the left and right compressors so that ILD was unaltered restored performance. Lastly, we develop a model to describe the utility of ILD for such tasks. The results of this model provide insight into the reported behavioral results, and generate predictions for how hearing impairment may alter the observed pattern of results.by Andrew H. Schwartz.Ph.D.in Speech and Hearing Bioscience and Technolog

Control of feedback for assistive listening devices

Acoustic feedback refers to the undesired acoustic coupling between the loudspeaker and microphone in hearing aids. This feedback channel poses limitations to the normal operation of hearing aids under varying acoustic scenarios. This work makes contributions to improve the performance of adaptive feedback cancellation techniques and speech quality in hearing aids. For this purpose a two microphone approach is proposed and analysed; and probe signal injection methods are also investigated and improved upon