Center Microphone Research Articles

Comparison of computational and experimental microphone array results for an 18% scale aircraft model

An 18%-scale, semispan model is used as a platform for examining the efficacy of microphone array processing using synthetic data from numerical simulations. Two hybrid RANS/LES codes coupled with Ffowcs Williams-Hawkings solvers are used to calculate 97 microphone signals at the locations of an array employed in the NASA LaRC 14×22 tunnel. Conventional, DAMAS, and CLEAN-SC array processing is applied in an identical fashion to the experimental and computational results for three different configurations involving deploying and retracting the main landing gear and a part-span flap. Despite the short time records of the numerical signals, the beamform maps are able to isolate the noise sources, and the appearance of the DAMAS synthetic array maps is generally better than those from the experimental data. The experimental CLEAN-SC maps are similar in quality to those from the simulations indicating that CLEAN-SC may have less sensitivity to background noise. The spectrum obtained from DAMAS processing of synthetic array data is nearly identical to the spectrum of the center microphone of the array, indicating that for this problem array processing of synthetic data does not improve spectral comparisons with experiment. However, the beamform maps do provide an additional means of comparison that can reveal differences that cannot be ascertained from spectra alone.

Infrasonic emissions from aircraft wake vortices: Field installation

An infrasonic field installation was set up at Newport News-Williamsburg International Airport in early 2013. The system is made up of three PCB 377M06 microphones installed into non-porous subsurface windscreens [POMA 1pNS9, 18, 040005 (2013)], which limit the bandwidth to 100 Hz. The microphones are placed 250 ft (76.2 m) orthogonal to the runway and 200 ft (60.96 m) apart. The data acquisition system is the B&K Pulse, from which time histories, spectra, and coherence between microphone channels are derived. The system is placed inside an instrumentation vehicle just behind the center microphone. Perforated drainage hoses are installed from the subsurface windscreens to adjacent drainage ditches and weight is added to the windscreens for additional stability. The drainage system has proved successful even on occasions of heavy downpour, revealing a truly all-weather system. A pistonphone calibration at 14 Hz in the field reveals that the three channels are matched to within 2 dB. This capability permits long-term monitoring of the health of the system. A sample time history of signals received from an aircraft takeoff will be presented.

Design and performance evaluation of a broadband three dimensional acoustic intensity measuring system

A seven-microphone three dimensional (3D) intensity measuring system has been developed and evaluated for performance for a broad frequency band (200 Hz-6.5 kHz). Six microphones are arranged in a concentric array with one microphone at the center of the probe. The screw adjustable center microphone is the probe reference microphone, and is used for calibrations of the other microphones in the probe. This probe addresses limitations of the traditional two-microphone system in measuring acoustical properties in a 3D space from the one dimensional measurements. This probe also eliminates the need of spacers used in the existing 3D probes for broadband measurements. Diffraction and reflection effects on calibrations due to presence of the microphones and the probe supporting structure are negligible. This seven-microphone probe provided better results in the intensity measurements for the wide frequency band than that of a similar four-microphone array probe.

Acoustic source localization with eye array

A novel signal-processing algorithm and array for sound source localization (SSL) in three-dimensional space is presented. This method, which has similarity to the eye in localization of light rays, consists of a novel hemispherical microphone array with 26 microphones on the shell and one microphone in the sphere center. The microphones on the shell map a geodesic hemisphere called two-frequency icosahedron; hence, each microphone has at least four other orthogonal microphones. A signal-processing scheme utilizes parallel creation of a special closeness function for each microphone direction on the shell in the time domain. Each closeness function cell (lens cell) consists of center microphone, shell microphone, and one of its orthogonals. The closeness function output values are linearly proportional to spatial angular difference between the sound source direction and each of the shell microphone directions. By choosing microphone directions corresponding to the highest closeness function values and implementing a linear weighted spatial averaging on them, the sound source direction is estimated. Contrary to traditional SSL techniques, this method is based on simple parallel mathematical calculations in the time domain with low computational costs. The laboratory implementation of the array and algorithm shows reasonable accuracy in a reverberant room.

Development of a multi-beam array microphone for multi-channel pickup of sound fields and its applications to room acoustics

A new multi-beam array microphone (MAM for short) has been developed. The MAM is composed of four concentric rings, each of which has 12 microphones elements, and a common center microphone. The MAM synthesizes the output phase of each microphone by compensating for the path differences. It has 12 main beams that point in 12 different directions at intervals of 30 deg in the horizontal plane. These main beams are sharp over a wide frequency range. By utilizing these characteristics, the MAM can be applied to many fields such as precise acoustic measurement of sound fields, automatic talker locations, and beam steering. When reproducing the original sound field by this system, 12 loudspeakers arranged in a circle around a central point are connected to the corresponding 12 outputs of the MAM. A new acoustical parameter based on spatial correlation of the sound field (correlation between the 12 MAM outputs) is proposed by utilizing the MAM.

Study on a “spatially selective” microphone system

Spatial selectivity of a microphone system is defined as a function that rejects all sound signals coming from the region specified as external by the system. It can be effectively realized by signal processing of the multiport output signals of a receiving system that comprises several microphone pairs located on a virtual spherical boundary surface and one center microphone. Because the signal processing part of the system consists of adders, constant multipliers, differentiators, and a single delay, it is essentially linear and can be realized in either analog or digital form according to requirements of precision. The practical advantage of the spatially selective microphone system is that it has the capability of suppressing howling in public address systems and conference telephony, of improving the signal-to-noise ratio of microphone output to automatic speech recognition machines, and of improving the separability of the signal picked up from a particular musical instrument from those of other instruments played simultaneously in the same studio. The design method is presented after the simulation results for the system.

A multimicrophone system with spatial selectivity

The principle of spatial selectivity of a receiving system, which comprises several microphones distributed on a spherical boundary surface and one microphone at the center of the space enclosed by the boundary, is introduced in accordance with the theorem expressed by Kirchhoff's integral equation. The output from each of the boundary microphones are processed in the following way. Each signal is first delayed and then differentiated in both time and space in the direction normal to the spherical boundary. All of the delayed and differentiated signals are added together with appropriate weighting. The system output is finally obtained by subtracting the weighted sum from the output of the center microphone. The system output contains no components of external noise from the outside of the spherical boundary. Simulation studies were carried out for various numbers of microphones and for different ratios of signal wavelength to the radius of the spherical boundary and for both plane and spherical incident waves.

Pivoted plural microphone assembly

A novel microphone assembly including three microphones which are rotatably or pivotably supported relative to each other and relative to a supporting handle housing structure wherein the left and right microphones rotate the same amount relative to a center axis of the microphone structure and wherein the center microphone can be extended different lengths and directed in different directions so as to vary the mixing ratio relative to the other microphones and including a ganged variable resistor to which the center microphone is connected so as to automatically control the mixing ratio between the signal from the center microphone with the two outer left and right microphones. The microphone assembly can be folded into a compact package and the microphone has superior localization characteristics for reproduction of sound.

Three-channel stereo playback of two tracks derived from three microphones

Playback of two-track stereo source material with a derived center channel offers accurate reproduction of the original stereo geometry, and requires very simple implementation. Essentially this two-track three-channel stereo depends on the principle that if two microphones are properly placed relative to each other and to the plane of the sound source, their combined output is that of a single microphone in the center, which output may be recovered by recombination of the two tracks. When a physical third microphone in the center is employed to feed the two tracks, its recovery from two tracks depends on relative polarity and amplitudes and in one recombination method the center microphone could be cancelled instead of being reproduced. An all-pass network may be used to shift the phase of one track so that on recombination the physical third microphone is always recovered, regardless of the manner in which it was mixed into the two tracks. The all-pass network produces 90° phase shift at only one frequency, but by choice of this frequency and the expectancy of additive polarity of original mixing, the center channel is recovered with excellent acuity, based on tests similar to those of Steinberg and Snow. Experimental recovery of a center output channel from a single center microphone feeding the two tracks, with flanking input signals zero, resulted in a center track output which was substantially indistinguishable from a normal monophonic reproduction.

A Compatible Photographic Stereophonic Sound System

This paper describes a 2-channel photographic stereophonic system, variable-density and variable-area, the two soundtracks occupying the same position as the normal single track on release film. Methods of recording using either a straight 2-microphone or an alternate system with a bridged center microphone are described. Methods of reproducing over a 2-loudspeaker or over a 3-loudspeaker system with the center bridged to the outer two are described. Tracks recorded in this manner may be played over standard theater systems equipped for single soundtrack only. Standard single tracks may be played over this system by means of a simple operating change in the booth.