Frequency-invariant (FI) beamforming aims at recovering signals without any distortion and maintaining spatial selectivity over the entire bandwidth. However, most existing FI beamforming ...(FIB)methods consider the weighted ℓ1-norm or modified ℓ1-norm of the filters as the objective function for sparse FIB design, which cannot assure the optimal sparse solution. To deal with this drawback, an optimization problem for the FIB design of sparse microphone array in terms of ℓp-norm (0<p<1) minimization is formulated, where distortion-less response, mainlobe and sidelobe constraints are considered. Due to the existence of the ℓp-norm objective function, the resultant problem is nonconvex, and therefore, an alternating direction method of multipliers (ADMM) algorithm is devised. Specifically, the corresponding problem is decomposed into multi-block subproblems to determine all unknown variables separately. Then, the most dominant sensor positions for each frequency are determined by using the principal component analysis (PCA) and K-mean clustering algorithm. Numerical examples show that the proposed design achieves good directivity factor, frequency invariance and better sparsity.
Robot audition systems with multiple microphone arrays have many applications in practice. However, accurate calibration of multiple microphone arrays remains challenging because there are many ...unknown parameters to be identified, including the relative transforms (i.e., orientation, translation) and asynchronous factors (i.e., initial time offset and sampling clock difference) between microphone arrays. To tackle these challenges, in this paper, we adopt batch simultaneous localization and mapping (SLAM) for joint calibration of multiple asynchronous microphone arrays and sound source localization. Using the Fisher information matrix (FIM) approach, we first conduct the observability analysis (i.e., parameter identifiability) of the above-mentioned calibration problem and establish necessary/sufficient conditions under which the FIM and the Jacobian matrix have full column rank, which implies the identifiability of the unknown parameters. We also discover several scenarios where the unknown parameters are not uniquely identifiable. Subsequently, we propose an effective framework to initialize the unknown parameters, which is used as the initial guess in batch SLAM for multiple microphone arrays calibration, aiming to further enhance optimization accuracy and convergence. Extensive numerical simulations and real experiments have been conducted to verify the performance of the proposed method. The experiment results show that the proposed pipeline achieves higher accuracy with fast convergence in comparison to methods that use the noise-corrupted ground truth of the unknown parameters as the initial guess in the optimization and other existing frameworks.
Purpose
One of the major reasons that totally implantable cochlear microphones are not readily available is the lack of good implantable microphones. An implantable microphone has the potential to ...provide a range of benefits over external microphones for cochlear implant users including the filtering ability of the outer ear, cosmetics, and usability in all situations. This paper presents results from experiments in human cadaveric ears of a piezofilm microphone concept under development as a possible component of a future implantable microphone system for use with cochlear implants. This microphone is referred to here as a drum microphone (DrumMic) that senses the robust and predictable motion of the umbo, the tip of the malleus.
Methods
The performance was measured by five DrumMics inserted in four different human cadaveric temporal bones. Sensitivity, linearity, bandwidth, and equivalent input noise were measured during these experiments using a sound stimulus and measurement setup.
Results
The sensitivity of the DrumMics was found to be tightly clustered across different microphones and ears despite differences in umbo and middle ear anatomy. The DrumMics were shown to behave linearly across a large dynamic range (46 dB SPL to 100 dB SPL) across a wide bandwidth (100 Hz to 8 kHz). The equivalent input noise (over a bandwidth of 0.1–10 kHz) of the DrumMic and amplifier referenced to the ear canal was measured to be about 54 dB SPL in the temporal bone experiment and estimated to be 46 dB SPL after accounting for the pressure gain of the outer ear.
Conclusion
The results demonstrate that the DrumMic behaves robustly across ears and fabrication. The equivalent input noise performance (related to the lowest level of sound measurable) was shown to approach that of commercial hearing aid microphones. To advance this demonstration of the DrumMic concept to a future prototype implantable in humans, work on encapsulation, biocompatibility, and connectorization will be required.
Time-delay estimation algorithms for speaker localization usually suffer from adverse effects of background noise and reverberation. The multichannel cross-correlation coefficient (MCCC) algorithm ...exploits spatial redundancy among multiple microphone signals to boost the robustness of the time-delay estimator. The MCCC algorithm, however, does not completely exploit the useful redundancy among the signals received at an array of microphones for the time-delay estimation. This issue is investigated in this paper from an information theory perspective. An equivalent time-delay estimation algorithm is derived to disclose the mechanism and limitation of the MCCC algorithm. Two new time-delay estimation algorithms are proposed on the basis of the mutual information among an array of microphones. The first algorithm fully utilizes the mutual information between all the different microphone signals to enhance its robustness to reverberation. The second algorithm emphasizes the monotone factors in these mutual information functions to promote its robustness to noise and reverberation. The effectiveness of the new time-delay estimators is demonstrated in noisy and reverberant environments.
Drones pose a hidden threat to public safety, and effective and accurate detection technology for the drone that exist in the environment is imminent, in which how to weaken the influence of target ...sound source localization deviation and strong background interference noise on the detection task is the key to improve the accuracy of drone sound event detection. In this paper, a method has been proposed to improve the accuracy of drone acoustic event detection, named as the linear shrinkage-subspace projection-power spectral density filter method (LSP). This method mainly including covariance matrix reconstruction, steering vector recalibration, and filter coefficient redesign method. Firstly, based on Minimum Variance Distortionless Response, the linear shrinkage method is used to suppress the interference and noise components in the signal plus interference covariance matrix, and the sample covariance matrix is reconstructed to eliminate background interference noise. Then, the correlation between the steering vector and the eigenvector is used to eliminate the angle correlation term, and the subspace projection method is combined to recalibrate the steering vector, so as to improve the ability of the beamforming method to resist the angle deviation and realize the correction of the target source positioning deviation. Next, a redesign method for wiener filter coefficients based on the estimated power spectral density is used to further weaken background interference noise. In order to verify the accuracy of the proposed method, a complete drone sound event detection system is constructed by combining the deep learning drone sound event detection classifier, and the evaluation is carried out according to different angular deviations and interference sound distances. In addition, a new evaluation criterion is proposed, named as the Machine-Human Extreme Hearing Distance Rate (MHDR), which analogizes the system's detection ability with the ear's auditory detection ability. The research results of this article indicate that the detection accuracy of the detection system shows satisfactory accuracy when the proposed method is applied to circular microphone array, that improved by 15.12 % compared to existing methods. The proposed method improves the detection accuracy of the drone acoustic event detection task under the influence of sound source position deviation and strong background speech interference, and provides a reference for the development of anti-drone technology.
Display omitted
•The linear shrinkage method overcomes the influence of strong background noise on the drone detection task.•The subspace projection method eliminates the impact of the angle deviation of the target source.•The improved wiener filter redesigned by signal power spectral densities further suppress strong interference.•The proposed LSP can achieve high detection accuracy with larger mismatches or strong interferences.•The proposed MHDR provides a more quantitative description for the drone sound event detection.
A new National Institute of Standards and Technology (NIST) measurement service has been developed for determining the pressure sensitivities of American National Standards Institute and ...International Electrotechnical Commission type LS2aP laboratory standard microphones over the frequency range 31.5 Hz to 20 000 Hz. At most frequencies common to the new service and the old service, the values of the expanded uncertainties of the new service are one-half the corresponding values of the old service, or better. The new service uses an improved version of the system employed by NIST in the Consultative Committee for Acoustics, Ultrasound, and Vibration (CCAUV) key comparison CCAUV.A-K3. Measurements are performed using a long and a short air-filled plane-wave coupler. For each frequency in the range 31.5 Hz to 2000 Hz, the reported sensitivity level is the average of data from both couplers. For each frequency above 2000 Hz, the reported sensitivity level is determined with data from the short coupler only. For proof test data in the frequency range 31.5 Hz to 2000 Hz, the average absolute differences between data from the long and the short couplers are much smaller than the expanded uncertainties.
We present a fiber-optic microphone (FOM) based on graphene oxide (GO) membrane in this study. A Fabry-Perot cavity consisting of a single-mode fiber and a piece of GO membrane works as the acoustic ...sensing structure. Using the GO as the core acoustic sensing component, the fabricating process of the FOM is demonstrated to be simple and efficient. Acoustic tests show that this FOM achieves an average minimum detectable pressure of 10.2 μPa/Hz 1/2 , while maintaining a linear acoustic pressure response and a flat frequency response in the range of 100 Hz to 20 kHz. These results indicate the excellent suitability of this FOM for acoustic detection in the audible range with high sensitivity and high fidelity.
Phased microphone array beamforming has become a standard technique to localize sound sources. For localization of rotating sound sources at free-space conditions in the frequency domain, a number of ...algorithms have been investigated. Two most famous algorithms were proposed by Pannert and Maier (J. Sound Vib., 333, 2014) and Herold and Sarradj (Noise Control Eng. J., 63, 2015). The relationship between the cross-spectral matrixes used in these two algorithms is still unclear. This paper investigated their relationship by proposing a new approach to calculate the cross-spectral matrix. Their relationship can then be interpreted from different interpolations used to calculate the sound pressures at virtual rotating array microphones from pressures at real stationary microphones. The former uses Fourier interpolation, while the latter uses linear interpolation. Compared with the latter algorithm, the cross-spectral matrix in the former algorithm has lower computational efficiency, nearly equal sound source location precision, and better sound source strength precision at high frequencies due to its better spectrum reconstruction capability. Additionally, the steering vector based on numerically solving the transcendental equation is proposed as an alternative with higher computational efficiency to the steering vector in the former algorithm.
This review collates around 100 papers that developed micro-electro-mechanical system (MEMS) capacitive microphones. As far as we know, this is the first comprehensive archive from academia on this ...versatile device from 1989 to 2019. These works are tabulated in term of intended application, fabrication method, material, dimension, and performances. This is followed by discussions on diaphragm, backplate and chamber, and performance parameters. This review is beneficial for those who are interested with the evolutions of this acoustic sensor.
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