Microphone Placement Research Articles

Computational study of head geometry effects on sound pressure gradient with applications to head-related transfer function

Effects of object geometry on sound scattering of far-field incident sound waves for two different head models (spherical and ellipsoidal) have been studied using an optimized Finite Element Method (FEM) based on frequency-dependent adaptive dimensions. This optimized FEM technique is proven both efficient and accurate when compared with analytical results for the spherical model, with maximum deviation of 0.6 dB. Comparisons between models have been made on the equivalent Head-Related Transfer Functions (HRTFs) for acoustic pressure, and for the first and second order pressure gradients on the surface. It is shown that while directionality cannot be achieved at lower frequencies using only pressure, pressure gradients provide sound cancellation for certain source orientations. Hence, it is possible to cancel incoming sound from the front (or behind) or sides depending on the direction (radial, azimuthal) and order of pressure gradients. While the pattern of pressure gradient directionality remains similar between the spherical and ellipsoidal models, the difference in dimensions affects the amplitude of the equivalent HRTFs for these parameters. This study provides insight into the placement of directional microphones on the head for hearing aids.

A distributed network of compact microphone arrays for drone detection and tracking

This work focuses on the development of a distributed network of compact microphone arrays for unmanned aerial vehicle (UAV) detection and tracking. Each compact microphone array extends in a 10 cm length aperture and consists in an arrangement of digital MEMS microphones. Several arrays are connected to a computing substation using the I2S, ADAT and MADI protocols using optical fiber. These protocols used together allow to collect the signals from hundreds of microphones spread over a distance of up to 10 km. Sound source localization is performed on each array using measured pressure and particle velocities. The pressure is estimated by averaging the signals of multiple microphones, and the particle velocity is estimated with high order finite differences of microphone signals. Multiple calibration procedures are compared experimentally. Results in sound source localization, noise reduction by spatial filtering and UAV recognition using machine learning are presented.

Some effects of microphone format and location on assistive listening system performance

Approximately 10—14% of the general population (USA & Northern Europe) suffer from a noticeable degree of hearing loss and would benefit from some form of hearing assistance or deaf-aid. However, many assistive listening systems do not provide the benefit that they should, as they are often let down by their poor acoustic performance. The paper investigates the acoustic and speech intelligibility requirements for assistive listening systems and examines a number of microphone pick-up scenarios and configurations in terms of their potential intelligibility and sound quality performance. The results of testing carried out in a number of rooms and venues are presented, mainly in terms of the resultant Speech Transmission Index (STI) measurements. The paper concludes by providing a number of recommendations and “rules of thumb” for optimal microphone formats and placement. Although the research has primarily been directed towards Audio Frequency Induction Loop Systems, the acoustic aspects are equally applicable to other technologies such as Infrared and Wireless Systems.

Rapid impedance tube determination of fabric transparency

Acousticians are continually being asked to verify fabric transparency for applications with absorptive and diffusive surfaces, as well as in sound reinforcement. Standard reverberation chamber methods can be used, but require large fabric and fiberglass samples. A quick and simple impedance tube method has been developed requiring only a 160 mm x 160 mm sample. Two measurements are made. One with an anechoic wedge termination and another with a rigid termination in which the fabric is applied to a 50 mm 6pcf fiberglass substrate. Four microphones are placed at a quarter and three quarters of the square tube’s width and height and the signal is summed. This unique microphone placement minimizes the 1st, 2nd, and 3rd harmonics resulting in a fourfold increase in the upper frequency limit. Impulse response measurements are made at three distances from the sample to calculate the reflection factor, impedance and normal incidence absorption coefficient from 63 Hz to 4000 Hz in one measurement. The small fabric sample is large enough to minimize non-homogeneous effects. A study of fabrics with a wide range of transparencies reveals how both acoustically transparent and backed fabrics can be used as sonic equalizers depending on the application.

High-resolution directivities of played musical instruments

Recent experimental developments have enabled directivity measurements of played musical instruments with high angular resolution. The measurement system assesses directional radiation characteristics while including diffraction and absorption effects of seated musicians. The results are advantageous to better understand and visualize sound produced by the instruments, provide benchmarks for physical modeling, predict and auralize sound in rehearsal and performance venues, and improve microphone placement techniques. This paper explores steady-state directivities, contrasting key differences between brass, tone-hole, and string instruments. As expected, brass instruments generate relatively predictable results because of their single radiation elements. Tone-hole instruments produce notable interference patterns due to radiation from multiple instrument openings. String instruments produce even more complex directivities because of radiation from distributed vibrating structures and instrument openings. To illustrate the effects, the paper focuses on an instrument from each category, within the same pitch range.

Attenuating the ear canal feedback pressure of a laser-driven hearing aid

Microphone placement behind the pinna, which minimizes feedback but also reduces perception of the high-frequency pinna cues needed for sound localization, is one reason why hearing-aid users often complain of poor sound quality and difficulty understanding speech in noisy situations. In this paper, two strategies are investigated for minimizing the feedback pressure (thereby increasing the maximum stable gain, MSG) of a wide-bandwidth light-activated contact hearing aid (CHA) to facilitate microphone placement in the ear canal (EC): (1) changing the location of the drive force and its direction at the umbo, and (2) placing an acoustic damper within the EC to reduce the feedback pressure at the microphone location. The MSG and equivalent pressure output (EPO) are calculated in a 3D finite element model of a human middle ear based on micro computed tomography (micro-CT) images. The model calculations indicate that changing the umbo-force direction can decrease feedback pressure, but at the expense of decreased EPO. However the model shows improvements in MSG without sacrificing EPO when an acoustic damper is placed in the EC. This was verified through benchtop experimentation and in human cadaver temporal bones. The results pave the path towards a wide-bandwidth hearing aid that incorporates an EC-microphone design.

Extension of sonic anemometry to high subsonic Mach number flows

In the literature, the application of sonic anemometry has been limited to low subsonic Mach number, near-incompressible flow conditions. To the best of the authors’ knowledge, this paper represents the first time a sonic anemometry approach has been used to characterize flow velocity beyond Mach 0.3. Using a high speed jet, flow velocity was measured using a modified sonic anemometry technique in flow conditions up to Mach 0.83. A numerical study was conducted to identify the effects of microphone placement on the accuracy of the measured velocity. Based on estimated error strictly due to uncertainty in time-of-acoustic flight, a random error of 4 m s−1 was identified for the configuration used in this experiment. Comparison with measurements from a Pitot probe indicated a velocity RMS error of 9 m s−1. The discrepancy in error is attributed to a systematic error which may be calibrated out in future work. Overall, the experimental results from this preliminary study support the use of acoustics for high subsonic flow characterization.

A Retrospective Multicentre Cohort Review of Patient Characteristics and Surgical Aspects versus the Long-Term Outcomes for Recipients of a Fully Implantable Active Middle Ear Implant

Objective: To summarise treatment outcomes compared to surgical and patient variables for a multicentre recipient cohort using a fully implantable active middle ear implant for hearing impairment. To describe the authors' preferred surgical technique to determine microphone placement. Study Design: Multicentre retrospective, observational survey. Setting: Five tertiary referral centres. Patients: Carina recipients (66 ears, 62 subjects) using the current Cochlear® Carina® System or the legacy device, the Otologics® Fully Implantable Middle Ear, with a T2 transducer. Methods: Patient file review and routine clinical review. Patient outcomes assessed were satisfaction, daily use and feedback reports at the first fitting and ≥12 months after implantation. Descriptive and statistical analysis of correlations of variables and their influence on outcomes was performed. Independently reported preferred methods for microphone placement are collectively summarised. Results: The average implant experience was 3.5 years. Satisfaction increased significantly over time (p < 0.05). No correlation with covariates examined was observed. Feedback significantly decreased over time, showing a significant correlation with microphone location, primary motivation, gender, age at implantation, and contralateral hearing aid use (p < 0.05). Patient satisfaction was inversely correlated with reports of system feedback (p < 0.05). The implantable microphone was most commonly on the posterior inferior mastoid line, in 42/66 (65%) cases, correlating with less likelihood for feedback and consistent with author surgical preference. Conclusion: Carina recipients in this study present as satisfied consistent daily users with very few reports of persistent feedback. As microphone location is an influencing factor, a careful surgical consideration of microphone placement is required. The authors prefer a posterior inferior mastoid line position whenever possible.

On determining the acoustic properties of main helicopter rotor models on an open test bench

The paper presents the results of experimental studies on developing a technique to determine the acoustic properties of models of main helicopter rotors on an open test bench. The method of maximum length sequences is used to choose the optimum arrangement of microphones for an open test bench that would minimize the influence of parasitic echo. The results of processing the data of an acoustic experiment with a model rotor are detailed.

The Classical wind player

This batch of recordings, all of wind-centric pieces, divides into two groups: one of chamber works and one of concertos or other works featuring a solo instrument. Including some well-known and often-recorded works and some that have never before been available on CD, this set demonstrates the virtuosity of today’s historically informed performers and hints at the range and quantity of Classical wind music, which is still easy to overlook as a genre to listen to and to perform. Ignaz Joseph Pleyel: Partitas for winds (Accent acc24276, rec 2012, 68′) includes four works written by Pleyel for string ensembles and arranged for winds by multiple other composers and one partita which may be original or arranged. I find this genre fascinating and the arrangements interesting to contemplate, but I wish that the CD case indicated the arranged nature of the pieces. The Amphion Wind Octet, a long-standing and award-winning group, exhibits good ensemble and very slick and flashy fingers, paired with occasional wild or rough notes. Clarinettists Christian Leitherer and Daniel Beyer’s impressions of rapid string arpeggios are particularly impressive. However, despite the comfortable and sensitive ensemble playing, each instrument sounds distractingly separate, perhaps due to microphone placement, and the bassoon is occasionally drowned out. I found this recording primarily interesting as a way to compare pieces arranged from string works and the different styles that can be attributed to arrangers or to the translation from strings to winds. Compositionally, I am particularly fond of the surprising jazziness of the Sextetto in C minor. The liner notes discuss Pleyel’s career in the context of the once substantial and now neglected Harmoniemusik repertory for wind ensembles. While the notes are very informative and the information about the history of the composition and arrangement of each work is well presented, I found myself wishing that the track listings included the opus numbers used in the notes. This would have ensured no confusion over which track is being discussed at each point in the notes.

The Effect of Microphone Placement on Interaural Level Differences and Sound Localization Across the Horizontal Plane in Bilateral Cochlear Implant Users.

This study examined the effect of microphone placement on the interaural level differences (ILDs) available to bilateral cochlear implant (BiCI) users, and the subsequent effects on horizontal-plane sound localization. Virtual acoustic stimuli for sound localization testing were created individually for eight BiCI users by making acoustic transfer function measurements for microphones placed in the ear (ITE), behind the ear (BTE), and on the shoulders (SHD). The ILDs across source locations were calculated for each placement to analyze their effect on sound localization performance. Sound localization was tested using a repeated-measures, within-participant design for the three microphone placements. The ITE microphone placement provided significantly larger ILDs compared to BTE and SHD placements, which correlated with overall localization errors. However, differences in localization errors across the microphone conditions were small. The BTE microphones worn by many BiCI users in everyday life do not capture the full range of acoustic ILDs available, and also reduce the change in cue magnitudes for sound sources across the horizontal plane. Acute testing with an ITE placement reduced sound localization errors along the horizontal plane compared to the other placements in some patients. Larger improvements may be observed if patients had more experience with the new ILD cues provided by an ITE placement.

Effect of smartphone orientation on sound level measurement

The proliferation of smartphones provides an opportunity for acoustic measurements to be made by citizen scientists. However, there are concerns about the accuracy of data collected by the public. This study reports on the variability in smartphone sound level measurements due to how the device is being carried. A sample of smartphones running either Android or iOS and with various microphone placements were selected and calibrated for dBA measurements. These devices were tested in anechoic and semi-reverberant rooms while being situated to reflect common carrying positions. We found that the measured sound level in the anechoic chamber could vary by >10 dB depending on the orientation of the device, the location of the microphone, and the relationship of the device to the sound source. Smaller but still substantial variation was observed in the semi-reverberant space. These data highlight the variability probable in participatory sensing efforts, and underscore the potential usefulness of providing guide...

Large-Area Microphone Array for Audio Source Separation Based on a Hybrid Architecture Exploiting Thin-Film Electronics and CMOS

We present a system for reconstructing-independent voice commands from two simultaneous speakers, based on an array of spatially distributed microphones. It adopts a hybrid architecture, combining large-area electronics (LAE), which enables a physically expansive array ( ${>} 1\text{ m}$ width), and a CMOS IC, which provides superior transistors for readout and signal processing. The array enables us to: 1) select microphones closest to the speakers to receive the highest SNR signal; 2) use multiple spatially diverse microphones to enhance robustness to variations due to microphones and sound propagation in a practical room. Each channel consists of a thin-film transducer formed from polyvinylidene fluoride (PVDF), a piezopolymer, and a localized amplifier composed of amorphous silicon (a-Si) thin-film transistors (TFTs). Each channel is sequentially sampled by a TFT scanning circuit, to reduce the number of interfaces between the large-area electronics (LAE) and CMOS IC. A reconstruction algorithm is proposed, which exploits the measured transfer function between each speaker and microphone, to separate two simultaneous speakers. The algorithm overcomes 1) sampling-rate limitations of the scanning circuits and 2) sensitivities to microphone placement and directionality. An entire system with eight channels is demonstrated, acquiring and reconstructing two simultaneous audio signals at 2 m distance from the array achieving a signal-to-interferer (SIR) ratio improvement of ${\sim} 12\text{ dB}$ .

Signal processing techniques for seat belt microphone arrays

Microphones integrated on a seat belt are an interesting alternative to conventional sensor positions used for hands-free telephony or speech dialog systems in automobile environments. In the setup presented in this contribution, the seat belt consists of three microphones which usually lay around the shoulder and chest of a sitting passenger. The main benefit of belt microphones is the small distance from the talker’s mouth to the sensor. As a consequence, an improved signal quality in terms of a better signal-to-noise ratio (SNR) compared to other sensor positions, e.g., at the rear view mirror, the steering wheel, or the center console, can be achieved. However, the belt microphone arrangement varies considerably due to movements of the passenger and depends on the size of the passenger. Furthermore, additional noise sources arise for seat belt microphones: they can easily be touched, e.g., by clothes, or might be in the path of an air-stream from the automotive ventilation system. This contribution presents several robust signal enhancement algorithms designed for belt microphones in multi-seat scenarios. The belt microphone with the highest SNR (usually closest to the speaker’s mouth) is selected for speech signal enhancement. Further improvements can be achieved if all belt microphone signals are combined to a single output signal. The proposed signal enhancement system for belt microphones includes a robust echo cancelation scheme, three different microphone combining approaches, a sophisticated noise estimation scheme to track stationary as well as non-stationary noise, and a speech mixer to combine the signals from each seat belt to a single channel output in a multi-seat scenario.

Genetic optimisation of a plane array geometry for beamforming. Application to source localisation in a high speed train

Thanks to its easy implementation and robust performance, beamforming is applied for source localisation in several fields. Its effectiveness depends greatly on the array sensor configuration. This paper introduces a criterion to improve the array beampattern and increase the accuracy of sound source localisation. The beamwidth and the maximum sidelobe level are used to quantify the spatial variation of the beampattern through a new criterion. This criterion is shown to be useful, especially for the localisation of moving sources. A genetic algorithm is proposed for the optimisation of microphone placement. Statistical analysis of the optimised arrays provides original results on the algorithm performance and on the optimal microphone placement. An optimised array is tested to localise the sound sources of a high speed train. The results show an accurate separation.

Optimized vector sound intensity measurements with a tetrahedral arrangement of microphones in a spherical shell.

Recent times have seen the introduction of small spherical arrays whose usefulness as sound intensity probes is the focus of this paper. The presented probe consists of a spherical shell, 30 mm in diameter, housing four 14 in. microphones arranged in a regular tetrahedral configuration. Classical formulae may be used to estimate the sound intensity vector, as may methods based on spherical harmonics decomposition. Results are shown to be comparable to those obtained from classical sound intensity probes. The existence of an analytical model for a plane wave's diffraction about a sphere provides a means for adopting common optimization techniques for potentially improving the intensity vector estimate, however. This paper examines the validity of non-linear least squares optimization in conjunction with the proposed spherical sound intensity probe when placed in the following sound fields: (1) a simple plane wave; (2) a plane wave corrupted by noise; and (3) multiple incident plane waves. Under certain conditions, the probe is shown to greatly extend the operational frequency range of classical sound intensity probes. The optimization algorithm is found to lack robustness against deviations from plane wave conditions, however.

Electric guitar—From measurement arrays to recording studio microphones

A joint research effort by the audio recording programs at the University of Massachusetts Lowell and New York University has made use of a 32-microphone measurement array in the quantification and visualization of the spectral radiation of musical instruments. Work to date has focused on electric guitar and piano. The measured directivities of the guitar amplifiers offer rich insight for the recording engineer. Traditional microphone selection and placement strategies formed over decades, before such data existed, are found to have merit. The data also shed light on those potentially unattractive microphone locations to be avoided. The measurements, taken with high spatial resolution, reveal a process for microphone placement as much as providing a window into showing exactly where to place them. Measurements of the acoustic radiation from electric guitar amplifiers reveal a spatial complexity that many recording engineers anticipate, and add valuable further insight.

Beamformer configuration design in reverberant environments

Many speech-related products rely on the deployment of microphone arrays and standard regular configurations are often used. In enhancing speech quality, the placement of microphones is indeed an important factor. Moreover, for indoor applications, the room acoustics further increases the difficulty. In this paper, these problems are addressed. First, we define the LCMV beamformer design problem using estimated room impulse responses with reverberation. Then, we study the performance limit on the filter length and formulate the configuration design problem. Finally, we employ a hybrid descent method with the genetic algorithm for solving the design problem. Numerical examples demonstrate the effectiveness of the proposed method.

Improvement in airborne position measurements based on an ultrasonic linear-period-modulated wave by 1-bit signal processing

In this paper, we describe an expansion of the airborne ultrasonic systems for object localization in the three-dimensional spaces of navigation. A system, which revises the microphone arrangement and algorithm, can expand the object-position measurement from +90° in a previous method up to +180° for both the elevation and azimuth angles. The proposed system consists of a sound source and four acoustical receivers. Moreover, the system is designed to utilize low-cost devices, and low-cost computation relying on 1-bit signal processing is used to support the real-time application on a field-programmable gate array (FPGA). An object location is identified using spherical coordinates. A spherical object, which has a curved surface, is considered a target for this system. The transmit pulse to the target is a linear-period-modulated ultrasonic wave with a chirp rate of 50–20 kHz. Statistical evaluation of this work is the experimental investigation under repeatability.

Static flow resistivity measurements based on the ISO 10534.2 standard impedance tube

Static flow resistivity is a fundamental parameter governing the sound absorption behavior of different materials, and several methods based on the use of impedance tubes have been developed for measuring the flow resistivity acoustically. However, these methods cannot be directly implemented in the standard impedance tubes that are widely used for measurements of sound absorption coefficients and impedance as defined in ISO 10534.2, because the termination conditions of impedance tubes and the placement of microphones in these existing methods are different. In this paper, a new acoustic method based on the impedance transfer function is proposed to determine the static flow resistivity with a standard impedance tube that complies with ISO 10534.2. The accuracy and the robustness of the proposed method are investigated through the finite element analysis and the feasibility of determining the static flow resistivity is validated through experiments. It is found that the accuracy of the proposed method is sensitive to the acoustic center position of the measurement microphones and the accuracy can be enhanced by increasing the back cavity depth or decreasing the measurement frequency.