Acoustic Impulse Responses for Wearable Audio Devices

We present an open-access dataset of over 8000 acoustic impulse from 160 microphones spread across the body and affixed to wearable accessories. The data can be used to evaluate audio capture and array processing systems using wearable devices such as hearing aids, headphones, eyeglasses, jewelry, and clothing. We analyze the acoustic transfer functions of different parts of the body, measure the effects of clothing worn over microphones, compare measurements from a live human subject to those from a mannequin, and simulate the noise-reduction performance of several beamformers. The results suggest that arrays of microphones spread across the body are more effective than those confined to a single device.Comment: To appear at ICASSP 201

Influence of microphone housing on the directional response of piezoelectric mems microphones inspired by Ormia ochracea

The influence of custom microphone housings on the acoustic directionality and frequency response of a multiband bio-inspired MEMS microphone is presented. The 3.2 mm by 1.7 mm piezoelectric MEMS microphone, fabricated by a cost-effective multi-user process, has four frequency bands of operation below 10 kHz, with a desired first-order directionality for all four bands. 7×7×2.5 mm3 3-D-printed bespoke housings with varying acoustic access to the backside of the microphone membrane are investigated through simulation and experiment with respect to their influence on the directionality and frequency response to sound stimulus. Results show a clear link between directionality and acoustic access to the back cavity of the microphone. Furthermore, there was a change in direction of the first-order directionality with reduced height in this back cavity acoustic access. The required configuration for creating an identical directionality for all four frequency bands is investigated along with the influence of reducing the symmetry of the acoustic back cavity access. This paper highlights the overall requirement of considering housing geometries and their influence on acoustic behavior for bio-inspired directional microphones

CABE : a cloud-based acoustic beamforming emulator for FPGA-based sound source localization

Microphone arrays are gaining in popularity thanks to the availability of low-cost microphones. Applications including sonar, binaural hearing aid devices, acoustic indoor localization techniques and speech recognition are proposed by several research groups and companies. In most of the available implementations, the microphones utilized are assumed to offer an ideal response in a given frequency domain. Several toolboxes and software can be used to obtain a theoretical response of a microphone array with a given beamforming algorithm. However, a tool facilitating the design of a microphone array taking into account the non-ideal characteristics could not be found. Moreover, generating packages facilitating the implementation on Field Programmable Gate Arrays has, to our knowledge, not been carried out yet. Visualizing the responses in 2D and 3D also poses an engineering challenge. To alleviate these shortcomings, a scalable Cloud-based Acoustic Beamforming Emulator (CABE) is proposed. The non-ideal characteristics of microphones are considered during the computations and results are validated with acoustic data captured from microphones. It is also possible to generate hardware description language packages containing delay tables facilitating the implementation of Delay-and-Sum beamformers in embedded hardware. Truncation error analysis can also be carried out for fixed-point signal processing. The effects of disabling a given group of microphones within the microphone array can also be calculated. Results and packages can be visualized with a dedicated client application. Users can create and configure several parameters of an emulation, including sound source placement, the shape of the microphone array and the required signal processing flow. Depending on the user configuration, 2D and 3D graphs showing the beamforming results, waterfall diagrams and performance metrics can be generated by the client application. The emulations are also validated with captured data from existing microphone arrays.</jats:p

FPGA-based architectures for acoustic beamforming with microphone arrays : trends, challenges and research opportunities

Over the past decades, many systems composed of arrays of microphones have been developed to satisfy the quality demanded by acoustic applications. Such microphone arrays are sound acquisition systems composed of multiple microphones used to sample the sound field with spatial diversity. The relatively recent adoption of Field-Programmable Gate Arrays (FPGAs) to manage the audio data samples and to perform the signal processing operations such as filtering or beamforming has lead to customizable architectures able to satisfy the most demanding computational, power or performance acoustic applications. The presented work provides an overview of the current FPGA-based architectures and how FPGAs are exploited for different acoustic applications. Current trends on the use of this technology, pending challenges and open research opportunities on the use of FPGAs for acoustic applications using microphone arrays are presented and discussed

The Effect of a Voice Activity Detector on the Speech Enhancement

A multimicrophone speech enhancement algorithm for binaural hearing aids that preserves interaural time delays was proposed recently. The algorithm is based on multichannel Wiener filtering and relies on a voice activity detector (VAD) for estimation of second-order statistics. Here, the effect of a VAD on the speech enhancement of this algorithm was evaluated using an envelope-based VAD, and the performance was compared to that achieved using an ideal error-free VAD. The performance was considered for stationary directional noise and nonstationary diffuse noise interferers at input SNRs from 10 to +5dB. Intelligibility-weighted SNR improvements of about 20dB and 6dB were found for the directional and diffuse noise, respectively. No large degradations (<1dB) due to the use of envelope-based VAD were found down to an input SNR of 0dB for the directional noise and 5dB for the diffuse noise. At lower input SNRs, the improvement decreased gradually to 15dB for the directional noise and 3dB for the diffuse noise.12 page(s