Microphone Characteristics Research Articles

Performance evaluation of automotive Hands-free microphones using subjective and objective metrics and their correlation

Hands-free, or voice, microphone is a standard device used in cars to enable hands-free telephony. The speech intelligibility and quality (SI&SQ) associated with such a microphone represents the ultimate performance considered by users. However, the SI&SQ cannot be derived from microphone datasheets. Furthermore, the complex and constantly changing automotive acoustical environment under different driving conditions makes it very challenging to choose the proper hands-free microphone design(s) for certain vehicle model(s). To establish the relationship between microphone characteristics commonly described by directivities and frequency-responses with the SI&SQ perceived by drivers and passengers in automotive applications, a study is conducted using three common types of automotive hands-free microphones. Their SI&SQ are evaluated using both subjective and objective metrics described in ANSI S3.2-2009, S3.5-1997 and ITU-T Recommendation P.862.2. The correlation of results obtained from different evaluation metrics is analyzed. It is shown that the speech intelligibility index (SII) calculated objectively using ANSI S3.5-1997 correlates approximately linearly with the SI data obtained from the subjective S3.2-2009 method. With some mathematical mapping, a modified SII can be derived that also correlates well with the mean-opinion-score predicted by P.862.2. The newly proposed modified SII method can be an effective tool to guide automotive hands-free microphone designs.

Influence of microphone characteristics on demodulated sound measurement in near field of parametric loudspeaker

This study evaluates the accuracy of demodulated sound measurements using a condenser microphone in the near field of a parametric loudspeaker system. Microphones with different sensitivities placed at incidence angles of 0° and 90° were used to measure demodulation frequency components without special acoustic filters. The measured components were compared with theoretical predictions. The results show that the measured sound pressure using microphones placed at 0° was up to several tens of decibels larger than the theoretical predictions and significantly inaccurate in the near field. This was due to the nonlinear response of the microphone, which had high sensitivity at primary sound frequencies, inducing spurious signals. This result suggests that using a microphone with low sensitivity at primary sound frequencies placed at an appropriate angle that reduces sensitivity improves parametric sound measurement accuracy.

Fingerprinting Smartphones Based on Microphone Characteristics From Environment Affected Recordings

Fingerprinting devices based on unique characteristics of their sensors is an important research direction nowadays due to its immediate impact on non-interactive authentications and no less due to privacy implications. In this work, we investigate smartphone fingerprints obtained from microphone data based on recordings containing human speech, environmental sounds and several live recordings performed outdoors. To comply with real-world circumstances, we also consider the presence of several types of noise that is specific to the scenarios which we address, e.g., traffic and market noise at distinct volumes, which may reduce the reliability of the data. We analyze several classification techniques based on traditional machine learning algorithms and more advanced deep learning architectures that are put to test in recognizing devices from the recordings they made. The results indicate that the classical Linear Discriminant classifier and a deep-learning Convolutional Neural Network have comparable success rates while outperforming all the rest of the algorithms.

Electroacoustic analysis of a directional moving-coil microphone using combined lumped-parameter models and BEM

This article presents a lumped-parameter model combined with external sound pressure field simulation, based on the boundary element method (BEM) for utilization in a directional microphone. A study about microphone capsule of a typical handheld moving-coil microphone with hyper-cardioid pattern has been conducted. In the BEM simulation, a simplified 3D model of the capsule that met sound waves which radiated from a point sound source has been taken into account. The blocked average pressure at the openings of the capsule is used to drive the lumped element equivalent circuit model. Through the model, the sensitivity, directivity, and impedance of the microphone were analyzed. Hence, the proximity effect is observed and discussed. To verify our simulation, the microphone characteristics were experimentally measured by B&K devices with SoundCheck software in an anechoic chamber. The simulation data are in good agreement with the measurement data at low and medium frequencies below 13 kHz, and there were larger errors at higher frequencies, which may indicate that the lumped parameters is less applicable at higher frequencies. The proposed method gains more precise and comprehensive prediction for the microphone characteristics, which therefore could be utilized in optimizing the choice of materials and size of the microphone capsule design.

1Pb4-5 Influence of Microphone Characteristics on Measurement of Near-field of Parametric Acoustic Array

1Pb4-5 Influence of Microphone Characteristics on Measurement of Near-field of Parametric Acoustic Array

Numerical Study on the Phase Sensitivity Variation in Low Frequency Primary Microphone Calibrations

The low frequency phase characteristics of microphones in a monitoring system are crucial for characterizing large-scale natural and artificial activities—e.g., earthquakes, nuclear explosions, or rocket launchings. At present, microphones are simultaneously calibrated using in-situ or calibrator methods to get their phase consistency. However, the essential primary calibration, which traces their phase sensitivity to basic physical quantities, is grossly overlooked. Recently, we speculated that the microphone phase sensitivity is acoustically controlled by the pressure leakage and heat conduction effects in its back chamber, which will vary at low frequencies. Therefore, by means of the FEA (Finite Element Analysis) technique, simulations of laser pistonphone-based primary microphone calibrations are conducted both in the frequency and time domains. The frequency domain simulation quantifies the phase variation, while the time domain analysis helps us to understand the variation mechanism. It is found that the low frequency phase sensitivity is greatly influenced by its geometries and the venting state and should be pre-calibrated before serving.

Design and modeling of a highly sensitive microelectromechanical system capacitive microphone

A single-chip microelectromechanical system (MEMS) capacitive microphone is designed and modeled. The mechanical model of the structure is extracted and the mathematical equations for a description of the microphone behavior are obtained. Then the proposed microphone characteristics are considered. In this structure, by adding Z-shape arms around the diaphragm, diaphragm hardness is decreased and diaphragm displacement becomes uniform. The sensitivity and the pull-in voltage are improved despite the decreasing size. The perforated diaphragm of this microphone is supported by Z-shape arms at its four corners. These arms around the diaphragm decrease the stiffness and air damping of the microphone. The behavior of this microphone is also analyzed by the finite element method. The structure has a diaphragm thickness of 2 μm, a diaphragm size of 0.32 × 0.32 mm2, an air gap of 2 μm, and a highly doped monocrystalline silicon wafer as a backplate. The proposed microphone is simulated with IntelliSuite software. According to the results, the new microphone has a sensitivity of 14.245 mV / Pa and a pull-in voltage of 5.83 V. The results show that the proposed MEMS capacitive microphone is one of the best structures in performance. The obtained mathematical equations for description of the microphone’s behavior have good agreement with the simulation results.

The application of artificial neural networks in solid-state photoacoustics for the recognition of microphone response effects in the frequency domain

An analysis of the application of neural networks as a reliable, precise, and fast tool in open-cell photoacoustics setups for the recognition of microphone effects in the frequency domain from 10 Hz to 100 × 104 Hz is presented. The network is trained to achieve simultaneous recognition of microphone characteristics, which are the most important parameters leading to the distortion of photoacoustic signals in both amplitude and phase. The training is carried out using a theoretically obtained database of amplitudes and phases as the input and five microphone characteristics as the output, based on transmission measurements obtained using an open photoacoustic cell setup. The results show that the network can precisely and reliably interpolate the output to recognize microphone characteristics including electronic effects in the low and acoustic effects in the high frequency domain. The simulations reveal that the network is not capable of interpolating an input including modulation frequencies. Consequently, in real applications, the network training must be adapted to the experimental frequencies, or vice versa. The total number of frequencies used in the experiment must also be in accordance with the total number of frequencies used in the network training.

Condenser MEMS Microphone

The market for MEMS microphones is growing every year. The aggregate average annual growth rate is 11.7%. This is due to the increase in devices using voice control, the increase in the number of microphones in smartphones, and the increasing role of MEMS microphones in the Internet of Things. A MEMS microphone consists of a MEMS element for converting acoustic pressure, a preamplification chip, a board with an acoustic inlet, and a cover. The MEMS conversion element is a variable capacitor with polysilicon plates. The main parameters of MEMS microphones are the signal-to-noise (S/N) ratio, non-linear distortion coefficient, and non-uniformity of the amplitude-frequency characteristics. When developing a MEMS microphone, computer simulation is used in the ANSYS software package. Using the ANSYS Mechanical program, the design is calculated and optimized to achieve the necessary sensitivity parameters of the MEMS conversion element. Using electrostatic analysis, the CV characteristics are calculated to determine the active capacitance and collapse voltage of MEMS-CE membranes. The influence of the housing dimensions, position, size and number of acoustic holes, the sizes above the membrane and under membrane volumes of the microphone frequency response is calculated in order to determine the effect of the housing parameters on the MEMS microphone’s characteristics. An experimental prototype of the first domestic MEMS microphone is developed and manufactured at the NPK Technological Center. New designs are being actively designed to improve the technical characteristics of MEMS microphones.

Improving Cochlear Implant Performance in the Wind Through Spectral Masking Release: A Multi-microphone and Multichannel Strategy.

Adopting the omnidirectional microphone (OMNI) mode and reducing low-frequency gain are the two most commonly used wind noise reduction strategies in hearing devices. The objective of this study was to compare the effectiveness of these two strategies on cochlear implant users' speech-understanding abilities and perceived sound quality in wind noise. We also examined the effectiveness of a new strategy that adopts the microphone mode with lower wind noise level in each frequency channel. A behind-the-ear digital hearing aid with multiple microphone modes was used to record testing materials for cochlear implant participants. It was adjusted to have linear amplification, flat frequency response when worn on a Knowles Electronic Manikin for Acoustic Research to remove the head-related transfer function of the manikin and to mimic typical microphone characteristics of hearing devices. Recordings of wind noise samples and hearing-in-noise test sentences were made when the hearing aid was programmed to four microphone modes, namely (1) OMNI; (2) adaptive directional microphone (ADM); (3) ADM with low-frequency roll-off; and (4) a combination of omnidirectional and directional microphone (COMBO). Wind noise samples were recorded in an acoustically treated wind tunnel from 0° to 360° in 10° increment at a wind velocity of 4.5, 9.0, and 13.5 m/s when the hearing aid was worn on the manikin. Two wind noise samples recorded at 90° and 300° head angles at the wind velocity of 9.0 m/s were chosen to take advantage of the spectral masking release effects of COMBO. The samples were then mixed with the sentences recorded using identical settings. Cochlear implant participants listened to the speech-in-wind testing materials and they repeated the sentences and compared overall sound quality preferences of different microphone modes using a paired-comparison categorical rating paradigm. The participants also rated their preferences of wind-only samples. COMBO yielded the highest speech recognition scores among the four microphone modes, and it was also preferred the most often, likely due to the reduction of spectral masking. The speech recognition scores generated using ADM with low-frequency roll-off were either equal to or lower than those obtained using ADM because gain reduction decreased not only the level of wind noise but also the low-frequency energy of speech. OMNI consistently yielded speech recognition scores lower than COMBO, and it was often rated as less preferable than other microphone modes, suggesting the conventional strategy to switch to the omnidirectional mode in the wind was undesirable. Neither adopting an OMNI nor reducing low-frequency gain generated higher speech recognition scores or higher sound quality ratings than COMBO. Adopting the microphone with lower wind noise level in different frequency channels can provide spectral masking release, and it is a more effective wind noise reduction strategy. The natural 6 dB/octave low-frequency roll-off of first-order directional microphones should be compensated when speech is present. Signal detection and decision rules for wind noise reduction applications are discussed in hearing devices with and without binaural transmission capability.

Integration of a Priori and Estimated Constraints Into an MVDR Beamformer for Speech Enhancement

Conventionally, the single constraint of the minimum variance distortionless response (MVDR) beamformer for speech enhancement has been defined using one of two approaches. Either it is based on a priori assumptions such as microphone characteristics, position, speech source location, and room acoustics, or on a relative transfer function (RTF) vector estimate using a data dependent method. Each approach has its respective merits and drawbacks and a decision usually has to be made between one of the approaches. In this paper, an alternative approach of using an integrated MVDR beamformer is investigated, where both the hard constraints from the two conventional approaches are softened to yield two tuning parameters. It will be shown that this integrated MVDR beamformer can be expressed as a convex combination of the conventional MVDR beamformers, a linearly constrained minimum variance (LCMV) beamformer, and an all-zero vector, with real, positive-valued coefficients. By analysing how the tuning parameters affect these coefficients, two tuning rules for a practical implementation of the integrated MVDR are subsequently proposed. An evaluation with simulated and recorded data demonstrates that the integrated MVDR beamformer can be beneficial as opposed to relying on either of the conventional MVDR beamformers.

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.

Robust broadband beamformer design for noise reduction and dereverberation

In this paper, we investigate robust design methods for broadband beamformers in reverberant environments. In the design formulation room reverberation as well as robustness to amplitude and phase mismatches in the microphones have been included. Particularly, the direct path and the reflections are separated in the design such that there is a penalty on the reflective part. This approach is different from the commonly studied problem of dereverberation of a single point source as the investigated design is made over a region in space. A single point derverberation is not a very practical approach due to a high sensitivity to position changes. Thus in order to obtain more practical microphone array designs, we study methods that optimize performance over areas in space. The design problem has been formulated in four different ways; (i) using direct path only representing a traditional beamformer design method, (ii) using a robust design method which considers robustness against the microphone characteristics (gain and phase) by optimizing the mean performance, (iii) by including room impulse response in the design and finally, (iv) using both robustness and room impulse response in the design. Simulation results show that robust direct path based beamformer can achieve approximately the same performance as including room response in the design in many reverberation environments. The proposed method provides robustness over larger variations in the reverberation environment. This means that the robust direct path based method which is based on mean variations in gain and phase can be used in low- to medium ($$T_{60}$$ = 100–300 ms) reverberant environments with good result.

Influence of primary elements parameters on result of voice information protection estimation

The purpose of the study is to investigate the main characteristics of measurement microphones. An experiment with use of different parameters microphones was conducted. Influence of the parameters on word intelligibility was estimated. All measurements were made in the same fix point by different microphones. Essential influence of inherent noise level of microphones was discovered by protection estimation in a fix point without active protection means (measurements are made on inherent noise level).

Field-effect transistor-based transduction and acoustic receiving transducers

Microphones and hydrophones are representative acoustic receiving transducers. To properly receive sound waves, a receiver must be smaller than the wavelength of the target sound. The target wave characteristics do not impose any lower limits on the size of microphones. When the performance of a smaller microphone or hydrophone will be satisfactory, users generally choose a smaller device since smaller receivers are easier to install and use. However, miniaturized microphones are less sensitive at low frequencies and conventional infrasound detectors are considerably larger than those for higher frequency sounds. These trends in receiver size can be explained by considering the transduction characteristics of microphones and hydrophones. We describe two transduction mechanisms based on field-effect transistors (FET) and use them to develop new microphones and hydrophones. We used theoretical analysis and experiments to show that the sensitivity and frequency response functions of FET-based microphones and hydrophones are size-independent. These results suggest that more sensitive micro-machined microphones and hydrophones, with better frequency response functions, may be available for use in the near future. [Work supported by the Institute of Civil Military Technology Cooperation Center (16-CM-SS-18).]

Regularization Approaches for Synthesizing HRTF Directivity Patterns

As an alternative to traditional artificial heads, it is possible to synthesize individual head-related transfer functions (HRTFs) using a so-called virtual artificial head (VAH), consisting of a microphone array with an appropriate topology and filter coefficients optimized using a narrowband least squares cost function. The resulting spatial directivity pattern of such a VAH is known to be sensitive to small deviations of the assumed microphone characteristics, e.g., gain, phase and/or the positions of the microphones. In many beamformer design procedures, this sensitivity is reduced by imposing a white noise gain (WNG) constraint on the filter coefficients for a single desired look direction. In this paper, this constraint is shown to be inappropriate for regularizing the HRTF synthesis with multiple desired directions and three alternative different regularization approaches are proposed and evaluated. In the first approach, the measured deviations of the microphone characteristics are taken into account in the filter design. In the second approach, the filter coefficients are regularized using the mean WNG for all directions. The third approach additionally takes into account several frequency bins into both the optimization and the regularization. The different proposed regularization approaches are compared using analytic and measured transfer functions, including random deviations. Experimental results show that the approach using multiple frequency bands mimicking the spectral resolution of the human auditory system yields the best robustness among the considered regularization approaches.

Temperature characteristics of single-crystal silicon electret microphones

Silicon Electret Condenser Microphones (Si-ECMs) are being studied to achieve small size, high performance, and reliability using Micro Electro Mechanical System (MEMS) technology. Increased robustness and stability are important to ensure wider use. In particular, temperature characteristics for microphones must be evaluated. Here, we focus on the diaphragm and impedance converter, which are the main factors determining the temperature characteristics of microphones, and present a method for measuring temperature characteristics and evaluating results for each parameter. The experimental Si-ECMs were found to have good temperature characteristics compared to a conventional ECM. These results indicate that the Si-ECM has the potential to be applied for measurement.

Photoacoustic signal and noise analysis for Si thin plate: signal correction in frequency domain.

Methods for photoacoustic signal measurement, rectification, and analysis for 85 μm thin Si samples in the 20-20 000 Hz modulation frequency range are presented. Methods for frequency-dependent amplitude and phase signal rectification in the presence of coherent and incoherent noise as well as distortion due to microphone characteristics are presented. Signal correction is accomplished using inverse system response functions deduced by comparing real to ideal signals for a sample with well-known bulk parameters and dimensions. The system response is a piece-wise construction, each component being due to a particular effect of the measurement system. Heat transfer and elastic effects are modeled using standard Rosencweig-Gersho and elastic-bending theories. Thermal diffusion, thermoelastic, and plasmaelastic signal components are calculated and compared to measurements. The differences between theory and experiment are used to detect and correct signal distortion and to determine detector and sound-card characteristics. Corrected signal analysis is found to faithfully reflect known sample parameters.

A Study of the Effect of the Microphone Characteristics for Transaural Systems

트랜스오럴 시스템은 3차원 음향 신호를 스피커를 이용하여 청취자의 두 귀 입구에서 재현시키는 방식의 3차원 음향 시스템이다. 본 논문에서는 트랜스오럴 시스템의 임펄스응답 측정에 이용되는 두 마이크로폰 사이에 주파수응답 특성이 다를 경우, 청취자의 두 귀 입구에서 재현되는 신호에서 두 귀간 레벨 차이의 오차가 발생하고 있음을 입증한다. 더욱이 임펄스응답 측정용으로 널리 이용되고 있는 소형 콘덴서 마이크로폰의 주파수응답 특성을 실측하여 두 귀간 레벨 차이의 오차가 실제로 발생하고 있음을 증명한다. 실험 결과, 약 10KHz 이하의 주파수대역에서 최대 3dB 정도의 두 귀간 레벨 차이의 오차가 발생하였고, 3dB의 오차는 청취자의 정면에 위치하는 3KHz의 음원일 경우, 약 20도의 방향 제시 오차를 야기할 수 있음을 증명했다.

System-level Modeling of Silicon Microphones Including Distributed Effects

We present a hierarchical modeling methodology, which combines short simulation times with the attention to detail as provided by finite element-based techniques. It enables predictive simulation of micro-devices even if exhibiting complex geometries like the MEMS microphone considered in this work. The challenge to properly account for mechanical, electrical and fluidic effects including their respective couplings has been solved by applying a mixed-level approach realized in a commercial system-level simulation environment. This enables to systematically include not only lumped but also distributed effects, like viscous damping and inhomogeneously distributed electrostatic forces. Applying the calibrated and validated model we obtain important microphone characteristics and, moreover, insights into distributed phenomena affecting the device operation. Hence, this constitutes a powerful tool for re-design and optimization as well as for the development of new prototypes.