Microphone Locations Research Articles

Hybrid Digital–Analog transmission for wireless acoustic sensor networks

A crucial task of a Wireless Acoustic Sensor Network (WASN) is the transmission of the acoustic signal to a central fusion node. Due to different microphone locations, the microphones may experience different or even varying radio channel qualities. Usually, the design of the digital transmission scheme is governed by the worst-case radio channel. For given radio resources, the end-to-end audio quality is specified by the compromise between the fidelity of the audio encoder and the required error protection capabilities of the channel encoder. The quantization noise of the audio encoder limits the end-to-end SNR even for increasing radio channel qualities. In the fusion node, the decoded signals are combined (fusion algorithm) and the overall output audio quality is limited due to the quantization noise even for error free transmission. In this paper it is shown that this unfavorable behavior can be avoided by using a Hybrid Digital-Analog (HDA) transmission concept. The key feature of HDA is that the audio output quality improves with increasing channel qualities. For typical fusion algorithms the HDA system supersedes purely digital transmission for all radio channel qualities. HighlightsNew transmission mode of signal from microphones to central fusion node.Avoiding saturation of quality due to quantization noise of purely digital system.Increased quality of output signal of fusion node for any type of signal combining.Simulation results for different types of fusion audio signal processing algorithms.Hybrid digital analog transmission always supersedes purely digital transmission.

An Experimental Setup for the Calibration of Acoustic Antenna

One of the most common applications of the acoustic antennas is the detection and the localization of sources. The geometric distribution of the sensors is not the only parameter that influences the measure; also the electromechanical features of the microphones play an important role, strongly affecting the delay of the signals. For this reason, a dedicated calibration process, considering both the geometric location of the sensors and their intrinsic features, can increase the accuracy level of the measure. In this paper at hand, a dedicated calibration method is proposed, providing a global measure of the locations, comprehensive of all the above mentioned causes of delay. The acoustic signals detected by the sensors are used to determine their distances from a reference microphone; on this basis, through a triangulation method, their locations are computed. First, the calibration method was tuned in a simulation environment: the effect of the location of sound sources and reference microphone was investigated using a dedicated numerical model to theoretically predict signal produced by each microphone. A dedicated optimization process was adopted to identify layout configuration guaranteeing the right calibration also for large size antenna compatibly with anechoic room available space constraint. The proposed solution was experimentally tested on a linear shape acoustic antenna.

Acoustic transfer functions and speech intelligibility in permanent wearing headsets

A detailed analysis of the acoustic transfer functions has been investigated for different locations of the headset microphones in the area of human head. Amplitude-frequency and phase-frequency characteristics of acoustic signals are discussed in the report. The signals have been recorded with the micro-mechanical-electrical-system (MEMS) microphones located on a mannequin, simulating the acoustic properties of the human torso. The the source of the signals is an emitter built in the mouth of the manikin's head. Analysis of the characteristics has shown that the acoustic signals undergo significant phase and amplitude changes associated with the interference and diffraction processes. These changes result in significant nonlinearity phase characteristics, and this nonlinearity depends on the installation location of the MEMS microphone. Also quality of speech intelligibility has been estimated by the method of expert evaluation and the speech has been recorded for different positions of the microphones and speaker head. A comparative analysis of the results has shown the optimal location of the microphones in the ear and on the breast of man.

The Effect of Changes in Position of Sound Source and Microphone on Acoustic Measurements in Small Rooms. A Case Study

Results of computer simulations and measurements in real interior for varying location of sound source and microphone are shown in the paper. A small room with a volume of 47 m was used for this purpose. The objective of measurements and calculations was to determine the sound pressure level and other parameters derived from the room impulse response (T30, EDT, C80, STI) followed by the sensitivity analysis of those parameters to changing the location and orientation of the sound source and the receiver. In order to determine these parameters, the room impulse response was measured using MLS method. Experimental studies have been used to verify the acoustic room model built with use of enhanced radial method algorithms and its sensitivity. That allowed complementary and extended simulation studies on the room acoustic characteristics and nally determination of sensitivity of output parameters to changes of location and orientation of the measurement channel elements.

Comparison of computational predictions of broadband interior sound fields with experimental measurements

Predictions of broadband interior noise using an energy-intensity boundary element method (EIBEM) are compared to experimental measurements. Steady-state sound fields with reflection boundaries are modeled with the EIBEM method. The specular reflection field is represented using uncorrelated broadband directional sources, expressed as spherical harmonics. For each boundary element, the amplitudes of the harmonics are determined from the incident field from all other elements and sources, and are subject to energy conservation using a Lagrange multiplier integral constraint. The computational solution utilizes a relaxation method starting with 3-D diffuse reflection. Earlier EIBEM results were compared to exact analytical solutions obtained from modal analysis, and to a broadband image method. For the experimental study, 1/3-octave band measurements were made within a full-scale steel-walled enclosure with different absorption levels, distributions, and source locations. Measurements were made along three 10-microphone linear arrays spanning the enclosure, and by microphones at other selected locations. Source power was measured independently, and random incidence absorption was inferred from both reverberation times and from steady state processing using multiple microphone locations. Predicted and measured spatial variation of the steady-state fields were in good agreement. The sensitivities of the predictions to various assumptions, geometric, anomalies, and experimental uncertainties are discussed.

디지털영상연기방법 고찰 - 스타니슬랍스키의 주의집중의 범위를 중심으로

본 논문에서는 디지털시대에 중요한 영상 매체로 대두되는 영화의 연기 방법론에 대해 연구하였다. 사실주의 연극에서 많이 연습되어지는 스타니슬랍스키(Konstantin Sergeevich Stanislavsky, 1863~1938)의 주의집중의 범위를 중심으로 영화에 부합되는 방법들을 고찰하였다. 정신적인 측면만을 고려한 연기 접근 방법이 아닌 영화의 스토리보드를 참고한 신체적인 접근 방법과 마이크 위치를 고려한 사운드의 적응 방법 등을 서술하였다. 스타니슬랍스키의 주의집중의 범위를 통한 인물창조는 사실주의 연극에서 뿐만 아니라 영화에도 큰 영향을 미칠 수 있다는 것을 확인할 수 있다. The purpose of this study analyzes the basic elements of film acting based on Stanislavsky Circles of Attention. This research also demonstrates the basic elements on how to become a good film actor. I attempt to interpret Stanislavsky Circles of Attention, in order to adapt the size of camera shots and the location of microphones. This Circles of Attention is an effective method utilized for actors to help them understand and be absorbed in the work of art both in film and stage. It is useful for building a character in realism drama as well as in film. I believe Stanislavsky Circles of Attention is certainly the most advanced acting style in film because it proposes the importance of physical action for actors.

The Average Direct Current Offset Values for Small Digital Audio Recorders in an Acoustically Consistent Environment

In this research project, nine small digital audio recorders were tested using five sets of 30-min recordings at all available recording modes, with consistent audio material, identical source and microphone locations, and identical acoustic environments. The averaged direct current (DC) offset values and standard deviations were measured for 30-sec and 1-, 2-, 3-, 6-, 10-, 15-, and 30-min segments. The research found an inverse association between segment lengths and the standard deviation values and that lengths beyond 30 min may not meaningfully reduce the standard deviation values. This research supports previous studies indicating that measured averaged DC offsets should only be used for exclusionary purposes in authenticity analyses and exhibit consistent values when the general acoustic environment and microphone/recorder configurations were held constant. Measured average DC offset values from exemplar recorders may not be directly comparable to those of submitted digital audio recordings without exactly duplicating the acoustic environment and microphone/recorder configurations.

실험실 실험을 통한 벽체 차음성능 측정의 신뢰성 향상을 위한 실험적 연구

Sound pressure levels in the receiving room while testing airborne sound insulation performance are varied by the measuring points. This may increase the measurement error, then decrease the measurement reliability. With this reason the research has carried out on the method to reduce deviations of sound pressure level in the ISO type rectangular laboratory focusing on the measurement of airborne sound insulation performance. Tests were made to see the effect of sound absorption in the receiving room, loudspeaker locations, microphones locations and flanking transmission path. Consequently, it was resulted that sound absorption in the receiving room and the loudspeaker location have influence on the sound level deviations especially in the low frequency. The microphone location was very important to get measurement reliability. The effective measuring point, which the sound level difference with average sound pressure level is within 2 dB, could yield most reliable average sound pressure level. Therefore it is necessary to find the effective measuring points in the receiving room. Flanking transmission path should be sealed using sound absorber or magnet etc. to prevent from lowering the sound insulation performance.

Characterization of the available feedback gain margin at two device microphone locations, in the fossa triangularis and Behind the Ear, for the light-based contact hearing device

Assistive devices compensate for hearing impairment by amplifying sounds with gain, which is limited by acoustic feedback. The light-based Contact Hearing Device (CHD) provides amplification to 10 kHz by mechanically vibrating the umbo with a wireless Tympanic-Contact Actuator (TCA). Driving the eardrum mechanically generates a pressure wave, which travels laterally down the ear canal and produces feedback. Placing the microphone in the fossa triangularis (FT) may preserve more natural acoustic cues than the BTE location, although it may reduce available feedback gain margin (FGM). Thirteen subjects with bilateral mild-to-severe hearing impairment were fit with CHDs (26 ears). The TCA was driven with light-pulses and the feedback pressure was measured at the FT and above the pinna at the BTE microphone locations. The mean FGM varied from 32 to 48 dB and 38 to 60 dB for the FT and BTE microphone locations, respectively. FGM was lowest in the 3–5 kHz range and highest at about 7 kHz (below 1 kHz FGM is not measurable). The STD of FGM varied from 5.3 to 13 dB due individual anatomies. A microphone at the BTE has 6 to 12 dB additional FGM over the FT location, and allows a broader inclusion range to fit patients with amplification to 10 kHz.

Acoustic determination of impedance tube microphone locations

Impedance tube methods allow for convenient, rapid characterization of the normal-incidence acoustic properties of materials and constructions. Many such methods rely on the determination of precise phase relationships between microphones sensing the sound field inside the impedance tube; knowledge of the location of these microphones is crucial to the accuracy of the measurement. Due to the large physical size of typical microphone diaphragms, physical measurements of the microphone positions (such as with a ruler or caliper) are inadequate and have a large uncertainty. This paper presents a method to determine the impedance tube microphone acoustic center locations from a broadband acoustic reflection coefficient measurement of a rigid termination. Utilizing the Bayesian inference framework, the estimation procedure provides information about microphone locations and uncertainties in the position estimates.

Modified FxLMS Algorithm for Active Noise Control and Its Real-Time Implementation

This paper presents a modified filtered-x least mean square (FxLMS) algorithm to improve the stability of active noise control (ANC) system in realistic environment. A real-time ANC system employing modified FxLMS is designed and implemented on digital signal processor (DSP) board. The ANC system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz and the performance is measured in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone with maximum 20 dB SPL attenuation can be generated around the location of error microphone.

Light‐based Contact Hearing Device: Characterization of Available Feedback Gain Margin at Two Device Microphone Locations

Objectives: Assistive devices compensate for hearing impairment by amplifying sounds with gain, which is limited by acoustic feedback. The light-based Contact Hearing Device (CHD) provides amplification to 10 kHz by mechanically vibrating the umbo with a wireless Tympanic Contact Actuator (TCA). Driving the eardrum mechanically generates a pressure wave, which travels laterally down the ear canal and produces feedback. Placing the microphone closer to the canal entrance may preserve more natural acoustic cues than the BTE location, although it may reduce available FGM. Our objective was to characterize the feedback gain margin (FGM) at two microphone locations using the CHD to determine if both locations are compatible with treating a broad range of hearing-impaired subjects. Methods: Thirteen subjects with bilateral mild-to-severe hearing impairment were fitted with CHDs (26 ears). The TCA was driven with light-pulses, and the feedback pressure was measured at two microphone locations: at the fossa triangularis (FT) and above the pinna at the BTE microphone location. The FGM from 1-10 kHz was computed (FGM below 1 kHz is not measureable). Results: The mean FGM varied from 32-48 dB and 38-60 dB for the FT and BTE microphone locations respectively. FGM was lowest in the 3-5 kHz range and highest at about 7 kHz. The STD of FGM varied from 5.3-13 dB due individual anatomies. Conclusions: A microphone at the BTE has 6-12 dB additional FGM over a microphone at a FT location and allows a broader inclusion range to fit impaired patients with amplification to 10 kHz.

링크분석에 의한 우리나라 근해대형트롤선의 선교 레이아웃 개선에 관한 기초연구

The purpose of this study is to obtain a basic data on layout of the trawlers` bridge equipment. The task activities of bridge workers involved in the navigation and fishing operation were analyzed by link analysis methods. The results are as follows. It was found that the movement pattern and frequency of bridge workers are different accordance with the bridge work (navigation, casting net, towing net and hauling net). The central workstation of movement of the bridge workers was a radar workstation, a steering workstation and a trawl winch workstation in the bridge work. But the radar did not show up as the center of movement during the hauling net. Workstations related deeply to the central workstations of the movement on the bridge were as below. Radar workstation was related to a GPS plotter, a microphone location for external communication with VHF and MF/HF equipment and a steering in the case of the navigation, the steering, the GPS plotter and the net monitor in the case of the fishing operation. Steering workstation was related deeply to the GPS plotter, the radar in the case of the navigation, a speed controller, the GPS plotter, a fish finder, the net monitor and the microphone location in the case of the fishing operation. Trawl winch workstation showed deep relation with the GPS plotter and the speed control during the fishing operation. Through this study, it was found that Workstations related deeply to the central workstation of the movement of the bridge workers in accordance with the bridge work. The results of this study might be utilized as the basic data on the bridge layout to minimize the fatigue degree due to a physical movement of the bridge workers.

The effect of bone conduction microphone placement on intensity and spectrum of transmitted speech items

Speech signals can be converted into electrical audio signals using either conventional air conduction (AC) microphone or a contact bone conduction (BC) microphone. The goal of this study was to investigate the effects of the location of a BC microphone on the intensity and frequency spectrum of the recorded speech. Twelve locations, 11 on the talker's head and 1 on the collar bone, were investigated. The speech sounds were three vowels (/u/, /a/, /i/) and two consonants (/m/, /∫/). The sounds were produced by 12 talkers. Each sound was recorded simultaneously with two BC microphones and an AC microphone. Analyzed spectral data showed that the BC recordings made at the forehead of the talker were the most similar to the AC recordings, whereas the collar bone recordings were most different. Comparison of the spectral data with speech intelligibility data collected in another study revealed a strong negative relationship between BC speech intelligibility and the degree of deviation of the BC speech spectrum from the AC spectrum. In addition, the head locations that resulted in the highest speech intelligibility were associated with the lowest output signals among all tested locations. Implications of these findings for BC communication are discussed.

Finite element computation of elliptical vocal tract impedances using the two-microphone transfer function method

A two-microphone transfer function (TMTF) method is adapted to a numerical framework to compute the radiation and input impedances of three-dimensional vocal tracts of elliptical cross-section. In its simplest version, the TMTF method only requires measuring the acoustic pressure at two points in an impedance duct and the postprocessing of the corresponding transfer function. However, some considerations are to be taken into account when using the TMTF method in the numerical context, which constitute the main objective of this paper. In particular, the importance of including absorption at the impedance duct walls to avoid lengthy numerical simulations is discussed and analytical complex axial wave numbers for elliptical ducts are derived for this purpose. It is also shown how the direct impedance of plane wave propagation can be computed beyond the TMTF maximum threshold frequency by appropriate location of the virtual microphones. Virtual microphone spacing is also discussed on the basis of the so-called singularity factor. Numerical examples include the computation of the radiation impedance of vowels /a/, /i/, and /u/ and the input impedance of vowel /a/, for simplified vocal tracts of circular and elliptical cross-sections.

Issues in binaural hearing in bilateral cochlear implant users

Despite the success of bilateral cochlear implants (CIs) in restoring sound localization abilities in profoundly deaf individuals, their localization accuracy is still poorer than that of normal hearing listeners. One factor could be the behind-the-ear location of the microphones. However, when CI users were tested with stimuli filtered through individualized head-related transfer functions (HRTFs), they showed very little difference in sound localization performance with different microphone locations (behind-the-ear versus in-the-ear). Another factor is the different implantation depths of the electrode arrays at the two ears. Since CIs are typically fitted independently in each ear at the clinic, it is likely that binaural information at a particular frequency can be presented mismatched across the ears. By simulating different amounts of interaural frequency mismatch at single electrode pairs, CI users showed poorer fusion and lower binaural sensitivity with increasing interaural mismatch. Good lateralization and fusion was achieved on or near a pitch-matched pair of electrode. Additionally, results from a separate study showed lateralization performance was typically maintained with simultaneous stimulation of multiple, pitch-matched pairs of electrodes. These results demonstrate methods beyond just changing the microphone position are needed to improve sound localization performance in CI users. [Work supported by NIH-NIDCD (R01-DC003083) and NICHD (P30-HD03352).]

Characterization of microphone placement and noise sensitivity in a global active noise control system for a compact noise source

This paper presents a sensitivity analysis to error microphone placement and extraneous noise sources for two control source configurations used for the active control of noise radiated from a compact source. The two control configurations analyzed are a linear array of four control sources with the primary source in the middle of the array and a symmetric array of four control sources surrounding the primary source. For modeling purposes, the primary source and the control sources are modeled as simple sources with a baffled free space Green's function to predict the noise measured by the four error microphones. The control source strengths which minimize the squared pressure at the error microphones are solved for based on error microphone locations. If the microphones are placed properly with no error in spatial location, this minimization process has been previously shown to result in radiated power minimization. Simulation results show the linear array is significantly more sensitive to microphone placement errors than the symmetric array. The linear array configuration is also shown to be more sensitive to added random noise in the error signals. Experimental results suggest that for the configurations used, the sensitivity to microphone placement errors was the dominant effect. The reduction in performance for the linear array configuration is significant with microphone placement errors and/or added random noise, while the reduction in performance for the symmetric array is minimal.

Semi-anechoic music recording using multi-microphone technique to simulate orchestra directivity in room acoustic auralizations

Over the last two decades, computer generated auralizations have become increasingly relevant to the design process of new performance spaces. Auralizations are based on anechoic material, and simulation realism is often limited by a small repertoire of recordings and the challenges of simulating an orchestra’s complex sound radiation characteristics. To address some of these issues, this paper describes a method to record—in semi-anechoic conditions—large musical ensembles for use in computer simulations. The method uses a multi-microphone hemispherical array distributed over an orchestra playing on a hard floor. A ray-tracing program models the orchestra as sectors of a hemisphere corresponding to each microphone location. Microphone signals are then mapped to their respective source sector. The method permits to record and simulate an entire orchestra with its directivity, including the movement of the musicians while playing. The paper describes the influence of the radius and angular resolution of the microphone array; it also investigates different source configurations in the computer model. The method has been implemented with renowned performers and orchestras and has been used for auralizations of several new concert hall design projects.

A model for predicting localization performance in cochlear implant users

Mathematical models can be very useful for understanding complicated systems, and for testing algorithms through simulation that would be difficult or expensive to implement. This paper describes the proposal for a model that would simulate the sound localization performance of profoundly hearing-impaired persons using bilateral cochlear implants (CIs). The expectation is that this model could be used as a tool that could prove useful in developing new signal processing algorithms for neural encoding strategies. The head related transfer function (HRTF) is a critical component of this model, and provides the base characteristics of head shadow, torso and pinna effects. This defines the temporal, intensity and spectral cues that are important to sound localization. This model was first developed to simulate normal hearing persons and validated against published literature on HRTFs and localization. The model was then further developed to account for the differences in the signal pathway of the CI user due to sound processing effects, and the microphone location versus ear canal acoustics. Finally, the localization error calculated from the model for CI users was compared to published localization data obtained from this population.

SLAM-based Online Calibration for Asynchronous Microphone Array

This paper addresses online calibration of an asynchronous microphone array. Although microphone array techniques are effective for sound localization and separation, these techniques have two issues; geometry information on a microphone array or time-consuming measurements of transfer functions between a microphone array and a sound source is necessary, and a fully synchronous multichannel analog-to-digital converter should be used. To solve these issues, we proposed an online framework for microphone array calibration by combining simultaneous localization and mapping (SLAM), and beamforming. SLAM simultaneously calibrates locations of microphones and a sound source, and clocks differences between microphones every time a microphone array observes a sound event. Beamforming works as a cost function to decide the convergence of calibration by localizing the sound using the transfer functions calculated from the estimated microphone locations and clock differences. We implemented a prototype system based on the proposed framework using extended Kalman filter-based SLAM and delay-and-sum beamforming. The experimental results showed that the proposed framework successfully calibrated an eight-channel asynchronous microphone array both in a simulated and a real environment even when system parameters such as variances are set to be 10 times larger than the optimal values. Furthermore, the error of sound localization with the calibrated microphone array was as small as the desired one, that is, the grid size for beamforming.