We design a digital pre-amplifier which can be directly connected to an electret microphone. The amplifier can convert analog signals into digital signals, has a wide voltage swing and low power consumption, as is req...We design a digital pre-amplifier which can be directly connected to an electret microphone. The amplifier can convert analog signals into digital signals, has a wide voltage swing and low power consumption, as is required in portable applications. Measurement results show that the dynamic range of the digital pre-amplifier reaches 88 dB, the equivalent input referred noise is 5μVrms, the typical power consumption is 540μW, and in standby mode the current does not exceed 10μA. Compared with an analog microphone, an electret microphone with digital pre-amplifier offers a better SNR, higher integration, lower power consumption, and higher immunity to system noise.展开更多
PurposeThe purpose of the study was to investigate the effect of bimodal beamforming on speech recognition and comfort for cochlear implant (CI) users with the bimodal hearing solution made up by linking a hearing aid...PurposeThe purpose of the study was to investigate the effect of bimodal beamforming on speech recognition and comfort for cochlear implant (CI) users with the bimodal hearing solution made up by linking a hearing aid to the CI sound processor.Methods19 subjects participated in this study. Speech tests were conducted in quiet and in noisy environments, with the target speech presented from 0° and the noise signal from 45°. Speech recognition thresholds (SRTs) were compared among the previously used bimodal hearing configuration (baseline, any CI sound processor plus any hearing aid), the Naída Bimodal Hearing Solution with omnidirectional microphone, and with directional microphone (so called StereoZoom) switched on. In addition, the study participants provided subjective feedback on their hearing impressions.ResultsThe SRT results showed no significant difference among the three hearing conditions in the quiet environment. No significant improvement was reported when using Naída bimodal system with omnidirectional microphone in noise compared to the baseline (p=0.27). When applying StereoZoom, SRT in noise showed significant improvements compared to omnidirectional settings (p<0.05). Subjective feedback showed that 13 participants were satisfied with Naída Bimodal Hearing Solution, and wanted to continue using it after the trial.ConclusionThe Naída Bimodal Hearing Solution with the same pre-processing algorithm can provide satisfying hearing performance. Beamforming technology can further improve speech perception in noisy environments.展开更多
At an age when most teens are figuring out high school,Siddharth is already shaping the future of medical tech.The 14⁃year⁃old boy from Dallas has created an AI⁃powered app,Circadian AI,capable of detecting heart dise...At an age when most teens are figuring out high school,Siddharth is already shaping the future of medical tech.The 14⁃year⁃old boy from Dallas has created an AI⁃powered app,Circadian AI,capable of detecting heart disease in just 7 seconds using only a smartphone's microphone.展开更多
Aiming at the problem that the traditional SRP-PHAT sound source localization method performs intensive search in a 360-degree space,resulting in high computational complexity and difficulty in meeting real-time requi...Aiming at the problem that the traditional SRP-PHAT sound source localization method performs intensive search in a 360-degree space,resulting in high computational complexity and difficulty in meeting real-time requirements,an innovative high-precision sound source localization method is proposed.This method combines the selective SRP-PHAT algorithm with real-time visual analysis.Its core innovations include using face detection to dynamically determine the scanning angle range to achieve visually guided selective scanning,distinguishing face sound sources from background noise through a sound source classification mechanism,and implementing intelligent background orientation selection to ensure comprehensive monitoring of environmental noise.Experimental results show that the method achieves a positioning accuracy of±5 degrees and a processing speed of more than 10FPS in complex real environments,and its performance is significantly better than the traditional full-angle scanning method.展开更多
The emerging millimeter-wave microphones have garnered considerable attention in recent years due to their potential for sound detection in various applications,particularly in situations where traditional microphones...The emerging millimeter-wave microphones have garnered considerable attention in recent years due to their potential for sound detection in various applications,particularly in situations where traditional microphones may be impractical.However,despite their promise,there is a notable lack of evidence demonstrating high-quality sound recovery of moving sources,which remains a significant challenge in thefield.This paper addresses this critical gap by proposing a novel method for displacement alignment that improves the detection and recovery of sound signals from moving sources.The proposed method works byfirst aligning the displacement of the sound source over time,which ensures that the signals are synchronized and avoids interference from movement of sources.Subsequently,precise surface vibrations are extracted from the aligned signals,providing data for sound recovery.Afinite impulse response(FIR)filter is applied to remove low-frequency motion,which often interferes with the clarity of the detected sound.Experimental results demonstrate the method’s effectiveness in recovering high-quality sound from moving sources,offering a promising solution for advancing the emerging millimeter-wave microphone technology in real-world applications.This work could pave the way for more accurate and reliable sound detection systems,particularly in dynamic environments.展开更多
This paper presents a literature review exploring the potential of piezoelectric field-effect transistors(piezo-FETs)as bionic microelectromechanical systems(MEMS).First,piezo-FETs are introduced as bionic counterpart...This paper presents a literature review exploring the potential of piezoelectric field-effect transistors(piezo-FETs)as bionic microelectromechanical systems(MEMS).First,piezo-FETs are introduced as bionic counterparts to natural mechanoreceptors,highlighting their classic configuration and working principles.Then,this paper summarizes the existing research on piezo-FETs as sensors for pressure,inertial,and acoustic sensors.Material selections,design characteristics,and key performance metrics are reviewed to demonstrate the advantage of piezo-FETs over traditional piezoelectric sensors.After identifying the limitations in these existing studies,this paper proposes using bionic piezoelectric coupling structures in piezo-FETs to further enhance the sensing capabilities of these artificial mechanoreceptors.Experimentally validated manufacturing methods for the newly proposed piezo-FET structures are also reviewed,pointing out a novel,feasible,and impactful research direction on these bionic piezoelectric MEMS sensors.展开更多
The vortex shedding noise has been revealed as an important wing noise source on some modern commercial aircraft based on the fly-over measurements with a planar microphone array by Michel (1998). In this paper, an an...The vortex shedding noise has been revealed as an important wing noise source on some modern commercial aircraft based on the fly-over measurements with a planar microphone array by Michel (1998). In this paper, an analytical model is presented for predicting this vortex shedding noise. The downstream wake of a 2-dimensional airfoil is assumed to be dominated by the von Karman vortex street, and the strength and the shedding frequency of the wake vortex are determined from the wake structure model. An aero-acoustic model is developed based on the Howe's unified theory of trailing edge noise and is incorporated with the wake model to predict the sound pressure level and directivity of vortex shedding noise. The predicted vortex shedding frequencies, sound pressure levels and directivities compare favorably with the measured results for 6 modern commercial aircraft.展开更多
A new technique to fabricate silicon condenser microphone is presented.The technique is based on the use of oxidized porous silicon as sacrificial layer for the air gap and the heavy p+-doping silicon of approximately...A new technique to fabricate silicon condenser microphone is presented.The technique is based on the use of oxidized porous silicon as sacrificial layer for the air gap and the heavy p+-doping silicon of approximately 15μm thickness for the stiff backplate.The measured sensitivity of the microphone fabricated with this technique is in the range from -45dB(5.6mV/Pa) to -55dB(1.78mV/Pa) under the frequency from 500Hz to 10kHz,and shows a gradual increase at higher frequency.The cut-off frequency is above 20kHz.展开更多
Automatically Updated Soundmaps are maps that convey the sound rather than the visual information content of an area of interest, at a certain time instant or period. Sound features encapsulate information that can be...Automatically Updated Soundmaps are maps that convey the sound rather than the visual information content of an area of interest, at a certain time instant or period. Sound features encapsulate information that can be combined with the visual features of the landscape, thus leading to useful environmental conclusions. This work aims to construct an Automatically Updated Soundmap of an area of environmental interest. A hierarchical pattern recognition approach method is proposed here, that can exploit sound recordings collected by a network of microphones. Hence, after appropriate signal processing, the large amounts of information, originally in the raw form of sound recordings, can be presented in the concise yet meaningful form of a periodically updated soundmap.展开更多
When using a miniature single sensor boundary layer probe, the time sequences of the stream-wise velocity in the turbulent boundary layer （TBL） are measured by using a hot wire anemometer. Beneath the fully develope...When using a miniature single sensor boundary layer probe, the time sequences of the stream-wise velocity in the turbulent boundary layer （TBL） are measured by using a hot wire anemometer. Beneath the fully developed TBL, the wall pressure fluctuations are attained by a microphone mechanism with high spatial resolution. Analysis on the statistic and spectrum properties of velocity and wall pressure reveals the relationship between the wall pressure fluctuation and the energy-containing structure in the buffer layer of the TBL. Wavelet transform shows the multi-scale natures of coherent structures contained in both signals of velocity and pressure. The most intermittent wall pressure scale is associated with the coherent structure in the buffer layer. Meanwhile the most energetic scale of velocity fluctuation at y＋ = 14 provides a specific frequency f9 ≈ 147 Hz for wall actuating control with Ret = 996.展开更多
A large planar microphone array, which consists of 111 microphones, was successfully applied to measure a two dimensional mapping of the sound sources on landing aircraft. The focus was on the flap side edge noise s...A large planar microphone array, which consists of 111 microphones, was successfully applied to measure a two dimensional mapping of the sound sources on landing aircraft. The focus was on the flap side edge noise source in this paper. The spectra, directivity and sound pressure level of flap side edge noise of 10 aircraft were presented in this paper. It is found that the spectrum of flap side edge noise is a broadband noise with some tones in some cases. Two different types of tone sources are found. It is proposed that one type of these tone sources is trailing edge semi baffled dipole source, and another is produced from the shedding of vortex from the wing cusp. The total sound pressure level of flap side edge broadband noise has no obvious directionality. However, the directivity of the tone noise in the flap side edge noise spectrum is obvious. It is demonstrated that the local flow field is the key to controlling the flap side edge noise.展开更多
Ammonia （NH3） volatilization is one of the important pathways of nitrogen loss in alkaline soil, and the NH3 concentration in soil headspace is directly linked with the NH3 volatilization. Ammonia was characterized ...Ammonia （NH3） volatilization is one of the important pathways of nitrogen loss in alkaline soil, and the NH3 concentration in soil headspace is directly linked with the NH3 volatilization. Ammonia was characterized by Fourier transform mid-infrared photoacoustic spectroscopy （FTIR-PAS） and two typical absorption bands in the region of 850-1 200 cm-1 were observed, which could be used for the prediction of NH3 concentration in the soil headspaze. An alkaline soil from North China was involved in the soil incubation, pot and field experiments under three fertilization treatments （control without N input （CK）, urea and coated urea）. Ammonia concentrations in the soil headspace were determined in each experiment. In the soil incubation experiment, the NH3 emissions were initiated by the N input, reached the highest concentration on day 2, and decreased to the level as measured in CK after 8 d, with significantly higher NH3 emissions in the urea treatment compared to coated urea treatment, especially during the first 4 d. The NH3 concentration in soil headspace of the pot experiment showed the similar dynamics as that in the incubation experiment; however, the NH3 concentration in the soil headspace in the field experiment demonstrated a significantly different emission pattern with those of the incubation and pot experiments, and there was a 4-d delay for the NH3 concentration. Therefore, the NH3 concentration in the incubation and pot experiments could not be directly used to model the real NH3 emission in the field due to the differences in fertilization method and application rate as well as soil temperature and soil disturbance. It was recommended that light irrigation in the second week after fertilization and involvement of controlled release coated urea could be used to significantly decrease N loss from the perspective of NH3 volatilization. Key Words： ammonia volatilization, cantilevel-type microphone, nitrogen, principal component regression, soil incubation.展开更多
This paper proposes a novel microphone array speech denoising scheme based on tensor filtering methods including truncated HOSVD(High-Order Singular Value Decomposition), low rank tensor approximation and multi-mode W...This paper proposes a novel microphone array speech denoising scheme based on tensor filtering methods including truncated HOSVD(High-Order Singular Value Decomposition), low rank tensor approximation and multi-mode Wiener filtering. Microphone array speech signal is represented in three-order tensor space with channel, time, and spectrum modes and then tensor filtering model can be designed to process the multiway array data. As to the first method, noise can be reduced through the truncated HOSVD which is a simple scheme in tensor processing. It is more accurate to find the lower-rank approximation of the three-order tensor with Tucker model. Then MDL(Minimum Description Length) criterion is used to estimate the optimal tensor rank in the second method. Further, multimode Wiener filtering approach upon tensor analysis can be considered as the spanning of one-mode wiener filtering. How to take advantages of tensor model to obtain a set of filters is the heart of the novel scheme. The performances of the proposed three approaches are evaluated with objective indexes and listening quality test. The experimental results indicate that the proposed tenor filtering methods have potential ability of retrieving the target signal from noisy microphone array signal and the multi-mode Wiener filtering method provides the best denoising results among the three ones.展开更多
Wideband acoustic imaging,which combines compressed sensing(CS)and microphone arrays,is widely used for locating acoustic sources.However,the location results of this method are unstable,and the computational efficien...Wideband acoustic imaging,which combines compressed sensing(CS)and microphone arrays,is widely used for locating acoustic sources.However,the location results of this method are unstable,and the computational efficiency is low.In this work,in order to improve the robustness and reduce the computational cost,a DCS-SOMP-SVD compressed sensing method,which combines the distributed compressed sensing using simultaneously orthogonal matching pursuit(DCS-SOMP)and singular value decomposition(SVD)is proposed.The performance of the DCS-SOMP-SVD is studied through both simulation and experiment.In the simulation,the locating results of the DCS-SOMP-SVD method are compared with the wideband BP method and the DCS-SOMP method.In terms of computational efficiency,the proposed method is as efficient as the DCS-SOMP method and more efficient than the wideband BP method.In terms of locating accuracy,the proposed method can still locate all sources when the signal to noise ratio(SNR)is−20 dB,while the wideband BP method and the DCS-SOMP method can only locate all sources when the SNR is higher than 0 dB.The performance of the proposed method can be improved by expanding the frequency range.Moreover,there is no extra source in the maps of the proposed method,even though the target sparsity is overestimated.Finally,a gas leak experiment is conducted to verify the feasibility of the DCS-SOMP-SVD method in the practical engineering environment.The experimental results show that the proposed method can locate both two leak sources in different frequency ranges.This research proposes a DCS-SOMP-SVD method which has sufficient robustness and low computational cost for wideband acoustic imaging.展开更多
Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transf...Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transform(SRP-PHAT)spatial spectrum as input feature is presented in this paper.Since the SRP-PHAT spatial power spectrum contains spatial location information,it is adopted as the input feature for sound source localization.DNN is exploited to extract the efficient location information from SRP-PHAT spatial power spectrum due to its advantage on extracting high-level features.SRP-PHAT at each steering position within a frame is arranged into a vector,which is treated as DNN input.A DNN model which can map the SRP-PHAT spatial spectrum to the azimuth of sound source is learned from the training signals.The azimuth of sound source is estimated through trained DNN model from the testing signals.Experiment results demonstrate that the proposed algorithm significantly improves localization performance whether the training and testing condition setup are the same or not,and is more robust to noise and reverberation.展开更多
In this paper, the frequency-domain Frost algorithm is enhanced by using conjugate gradient techniques for speech enhancement. Unlike the non-adaptive approach of computing the optimum minimum variance distortionless ...In this paper, the frequency-domain Frost algorithm is enhanced by using conjugate gradient techniques for speech enhancement. Unlike the non-adaptive approach of computing the optimum minimum variance distortionless response （MVDR） solution with the correlation matrix inversion, the Frost algorithm implementing the stochastic constrained least mean square （LMS） algorithm can adaptively converge to the MVDR solution in mean-square sense, but with a very slow convergence rate. In this paper, we propose a frequency-domain constrained conjugate gradient （FDCCG） algorithm to speed up the convergence. The devised FDCCG algorithm avoids the matrix inversion and exhibits fast convergence. The speech enhancement experiments for the target speech signal corrupted by two and five interfering speech signals are demonstrated by using a four-channel acoustic-vector-sensor （AVS） micro-phone array and show the superior performance.展开更多
To improve localization accuracy, the spherical microphone arrays are used to capture high-order wavefield in- formation. For the far field sound sources, the array signal model is constructed based on plane wave deco...To improve localization accuracy, the spherical microphone arrays are used to capture high-order wavefield in- formation. For the far field sound sources, the array signal model is constructed based on plane wave decomposition. The spatial spectrum function is calculated by minimum variance distortionless response （MVDR） to scan the three-dimensional space. The peak values of the spectrum function correspond to the directions of multiple sound sources. A diagonal loading method is adopted to solve the ill-conditioned cross spectrum matrix of the received signals. The loading level depends on the alleviation of the ill-condition of the matrix and the accuracy of the inverse calculation. Compared with plane wave decomposition method, our proposed localization algorithm can acquire high spatial resolution and better estimation for multiple sound source directions, especially in low signal to noise ratio （SNR）.展开更多
Audio applications such as mobile communication and hearing aid devices demand a small size but high performance, stable and low cost microphone to reproduce a high quality sound. Capacitive microphone can be designed...Audio applications such as mobile communication and hearing aid devices demand a small size but high performance, stable and low cost microphone to reproduce a high quality sound. Capacitive microphone can be designed to fulfill such requirements with some trade-offs between sensitivity, operating frequency range, and noise level mainly due to the effect of device structure dimensions and viscous damping. Smaller microphone size and air gap will gradually decrease its sensitivity and increase the viscous damping. The aim of this research was to develop a mathematical model of a spring-supported diaphragm capacitive MEMS microphone as well as an approach to optimize a microphone’s performance. Because of the complex shapes in this latest type of diaphragm design trend, analytical modelling has not been previously attempted. A novel diaphragm design is proposed that offers increased mechanical sensitivity of a capacitive microphone by reducing its diaphragm stiffness. A lumped element model of the spring-supported diaphragm microphone is developed to analyze the complex relations between the microphone performance factors and to find the optimum dimensions based on the design requirements. It is shown analytically that the spring dimensions of the spring-supported diaphragm do not have large effects on the microphone performance com pared to the diaphragm and backplate size, diaphragm thickness, and air-gap distance. A 1 mm2 spring-supported diaphragm microphone is designed using several optimized performance parameters to give a –3 dB operating bandwidth of 10.2 kHz, a sensitivity of 4.67 mV/Pa (–46.5 dB ref. 1 V/Pa at 1 kHz using a bias voltage of 3 V), a pull-in voltage of 13 V, and a thermal noise of –22 dBA SPL.展开更多
The Steered Response Power(SRP)method works well for sound source localization in noisy and reverberant environment.However,the large computation complexity limits its practical application.In this paper,a fast SRP se...The Steered Response Power(SRP)method works well for sound source localization in noisy and reverberant environment.However,the large computation complexity limits its practical application.In this paper,a fast SRP search method is proposed to reduce the computational complexity using small-aperture microphone array.The proposed method inspired by the SRP spatial spectrum includes two steps:first,the proposed method estimates the azimuth of the sound source roughly and determines whether the sound source is in far field or near field;then,different fine searching operations are performed according to the sound source being in far field or near field.Experiments both in simulation environments and real environments have been performed to compare the localization accuracy and computation complexity of the proposed method with those of the conventional SRP-PHAT algorithm.The results show that,the proposed method has a comparative accuracy with the conventional SRP algorithm,and achieves a reduction of 93.62%in computation complexity compared to the conventional SRP algorithm.展开更多
文摘We design a digital pre-amplifier which can be directly connected to an electret microphone. The amplifier can convert analog signals into digital signals, has a wide voltage swing and low power consumption, as is required in portable applications. Measurement results show that the dynamic range of the digital pre-amplifier reaches 88 dB, the equivalent input referred noise is 5μVrms, the typical power consumption is 540μW, and in standby mode the current does not exceed 10μA. Compared with an analog microphone, an electret microphone with digital pre-amplifier offers a better SNR, higher integration, lower power consumption, and higher immunity to system noise.
基金supported by grants from Capital’s Funds for Health Improvement and Research(No.2022-1-2023)the National Natural Science Foundation of China(No.82371148)Open project National Clinical Research Center for Otolaryngologic Diseases(202200010).
文摘PurposeThe purpose of the study was to investigate the effect of bimodal beamforming on speech recognition and comfort for cochlear implant (CI) users with the bimodal hearing solution made up by linking a hearing aid to the CI sound processor.Methods19 subjects participated in this study. Speech tests were conducted in quiet and in noisy environments, with the target speech presented from 0° and the noise signal from 45°. Speech recognition thresholds (SRTs) were compared among the previously used bimodal hearing configuration (baseline, any CI sound processor plus any hearing aid), the Naída Bimodal Hearing Solution with omnidirectional microphone, and with directional microphone (so called StereoZoom) switched on. In addition, the study participants provided subjective feedback on their hearing impressions.ResultsThe SRT results showed no significant difference among the three hearing conditions in the quiet environment. No significant improvement was reported when using Naída bimodal system with omnidirectional microphone in noise compared to the baseline (p=0.27). When applying StereoZoom, SRT in noise showed significant improvements compared to omnidirectional settings (p<0.05). Subjective feedback showed that 13 participants were satisfied with Naída Bimodal Hearing Solution, and wanted to continue using it after the trial.ConclusionThe Naída Bimodal Hearing Solution with the same pre-processing algorithm can provide satisfying hearing performance. Beamforming technology can further improve speech perception in noisy environments.
文摘At an age when most teens are figuring out high school,Siddharth is already shaping the future of medical tech.The 14⁃year⁃old boy from Dallas has created an AI⁃powered app,Circadian AI,capable of detecting heart disease in just 7 seconds using only a smartphone's microphone.
基金the research result of the 2024 Guangxi Higher Education Undergraduate Teaching Reform Project“OBE-Guided,Digitally Empowered‘Hadoop Big Data Development Technology’Course Ideological and Political Construction Innovation Exploration and Practice”(Project No.:2024JGA396).
文摘Aiming at the problem that the traditional SRP-PHAT sound source localization method performs intensive search in a 360-degree space,resulting in high computational complexity and difficulty in meeting real-time requirements,an innovative high-precision sound source localization method is proposed.This method combines the selective SRP-PHAT algorithm with real-time visual analysis.Its core innovations include using face detection to dynamically determine the scanning angle range to achieve visually guided selective scanning,distinguishing face sound sources from background noise through a sound source classification mechanism,and implementing intelligent background orientation selection to ensure comprehensive monitoring of environmental noise.Experimental results show that the method achieves a positioning accuracy of±5 degrees and a processing speed of more than 10FPS in complex real environments,and its performance is significantly better than the traditional full-angle scanning method.
基金supported by the National Natural Science Foundation of China under Grant No.51905341the Natural Science Foundation of Shanghai under Grant 22ZR1433900.
文摘The emerging millimeter-wave microphones have garnered considerable attention in recent years due to their potential for sound detection in various applications,particularly in situations where traditional microphones may be impractical.However,despite their promise,there is a notable lack of evidence demonstrating high-quality sound recovery of moving sources,which remains a significant challenge in thefield.This paper addresses this critical gap by proposing a novel method for displacement alignment that improves the detection and recovery of sound signals from moving sources.The proposed method works byfirst aligning the displacement of the sound source over time,which ensures that the signals are synchronized and avoids interference from movement of sources.Subsequently,precise surface vibrations are extracted from the aligned signals,providing data for sound recovery.Afinite impulse response(FIR)filter is applied to remove low-frequency motion,which often interferes with the clarity of the detected sound.Experimental results demonstrate the method’s effectiveness in recovering high-quality sound from moving sources,offering a promising solution for advancing the emerging millimeter-wave microphone technology in real-world applications.This work could pave the way for more accurate and reliable sound detection systems,particularly in dynamic environments.
文摘This paper presents a literature review exploring the potential of piezoelectric field-effect transistors(piezo-FETs)as bionic microelectromechanical systems(MEMS).First,piezo-FETs are introduced as bionic counterparts to natural mechanoreceptors,highlighting their classic configuration and working principles.Then,this paper summarizes the existing research on piezo-FETs as sensors for pressure,inertial,and acoustic sensors.Material selections,design characteristics,and key performance metrics are reviewed to demonstrate the advantage of piezo-FETs over traditional piezoelectric sensors.After identifying the limitations in these existing studies,this paper proposes using bionic piezoelectric coupling structures in piezo-FETs to further enhance the sensing capabilities of these artificial mechanoreceptors.Experimentally validated manufacturing methods for the newly proposed piezo-FET structures are also reviewed,pointing out a novel,feasible,and impactful research direction on these bionic piezoelectric MEMS sensors.
基金the Bundersministerium for Building und Forschung(BMBF) of Germany
文摘The vortex shedding noise has been revealed as an important wing noise source on some modern commercial aircraft based on the fly-over measurements with a planar microphone array by Michel (1998). In this paper, an analytical model is presented for predicting this vortex shedding noise. The downstream wake of a 2-dimensional airfoil is assumed to be dominated by the von Karman vortex street, and the strength and the shedding frequency of the wake vortex are determined from the wake structure model. An aero-acoustic model is developed based on the Howe's unified theory of trailing edge noise and is incorporated with the wake model to predict the sound pressure level and directivity of vortex shedding noise. The predicted vortex shedding frequencies, sound pressure levels and directivities compare favorably with the measured results for 6 modern commercial aircraft.
文摘A new technique to fabricate silicon condenser microphone is presented.The technique is based on the use of oxidized porous silicon as sacrificial layer for the air gap and the heavy p+-doping silicon of approximately 15μm thickness for the stiff backplate.The measured sensitivity of the microphone fabricated with this technique is in the range from -45dB(5.6mV/Pa) to -55dB(1.78mV/Pa) under the frequency from 500Hz to 10kHz,and shows a gradual increase at higher frequency.The cut-off frequency is above 20kHz.
文摘Automatically Updated Soundmaps are maps that convey the sound rather than the visual information content of an area of interest, at a certain time instant or period. Sound features encapsulate information that can be combined with the visual features of the landscape, thus leading to useful environmental conclusions. This work aims to construct an Automatically Updated Soundmap of an area of environmental interest. A hierarchical pattern recognition approach method is proposed here, that can exploit sound recordings collected by a network of microphones. Hence, after appropriate signal processing, the large amounts of information, originally in the raw form of sound recordings, can be presented in the concise yet meaningful form of a periodically updated soundmap.
基金Project supported by the National Basic Research Program of China(Grant Nos.2012CB720101 and 2012CB720103)the National Natural Science Foundation of China(Grant Nos.11272233,11332006,and 11411130150)
文摘When using a miniature single sensor boundary layer probe, the time sequences of the stream-wise velocity in the turbulent boundary layer （TBL） are measured by using a hot wire anemometer. Beneath the fully developed TBL, the wall pressure fluctuations are attained by a microphone mechanism with high spatial resolution. Analysis on the statistic and spectrum properties of velocity and wall pressure reveals the relationship between the wall pressure fluctuation and the energy-containing structure in the buffer layer of the TBL. Wavelet transform shows the multi-scale natures of coherent structures contained in both signals of velocity and pressure. The most intermittent wall pressure scale is associated with the coherent structure in the buffer layer. Meanwhile the most energetic scale of velocity fluctuation at y＋ = 14 provides a specific frequency f9 ≈ 147 Hz for wall actuating control with Ret = 996.
基金F inancially supported by the Bundersministerium fur Bildung und Forschung ( BMBF) of Germ any
文摘A large planar microphone array, which consists of 111 microphones, was successfully applied to measure a two dimensional mapping of the sound sources on landing aircraft. The focus was on the flap side edge noise source in this paper. The spectra, directivity and sound pressure level of flap side edge noise of 10 aircraft were presented in this paper. It is found that the spectrum of flap side edge noise is a broadband noise with some tones in some cases. Two different types of tone sources are found. It is proposed that one type of these tone sources is trailing edge semi baffled dipole source, and another is produced from the shedding of vortex from the wing cusp. The total sound pressure level of flap side edge broadband noise has no obvious directionality. However, the directivity of the tone noise in the flap side edge noise spectrum is obvious. It is demonstrated that the local flow field is the key to controlling the flap side edge noise.
基金supported by the National Natural Science Foundation of China (No. 41130749)the National Basic Research Program (973 Program) of China (No. 2015CM15043)
文摘Ammonia （NH3） volatilization is one of the important pathways of nitrogen loss in alkaline soil, and the NH3 concentration in soil headspace is directly linked with the NH3 volatilization. Ammonia was characterized by Fourier transform mid-infrared photoacoustic spectroscopy （FTIR-PAS） and two typical absorption bands in the region of 850-1 200 cm-1 were observed, which could be used for the prediction of NH3 concentration in the soil headspaze. An alkaline soil from North China was involved in the soil incubation, pot and field experiments under three fertilization treatments （control without N input （CK）, urea and coated urea）. Ammonia concentrations in the soil headspace were determined in each experiment. In the soil incubation experiment, the NH3 emissions were initiated by the N input, reached the highest concentration on day 2, and decreased to the level as measured in CK after 8 d, with significantly higher NH3 emissions in the urea treatment compared to coated urea treatment, especially during the first 4 d. The NH3 concentration in soil headspace of the pot experiment showed the similar dynamics as that in the incubation experiment; however, the NH3 concentration in the soil headspace in the field experiment demonstrated a significantly different emission pattern with those of the incubation and pot experiments, and there was a 4-d delay for the NH3 concentration. Therefore, the NH3 concentration in the incubation and pot experiments could not be directly used to model the real NH3 emission in the field due to the differences in fertilization method and application rate as well as soil temperature and soil disturbance. It was recommended that light irrigation in the second week after fertilization and involvement of controlled release coated urea could be used to significantly decrease N loss from the perspective of NH3 volatilization. Key Words： ammonia volatilization, cantilevel-type microphone, nitrogen, principal component regression, soil incubation.
基金supported by the National Natural Science Foundation of China(No.61571044,No.11590772,No.61473041 and No.61620106002)
文摘This paper proposes a novel microphone array speech denoising scheme based on tensor filtering methods including truncated HOSVD(High-Order Singular Value Decomposition), low rank tensor approximation and multi-mode Wiener filtering. Microphone array speech signal is represented in three-order tensor space with channel, time, and spectrum modes and then tensor filtering model can be designed to process the multiway array data. As to the first method, noise can be reduced through the truncated HOSVD which is a simple scheme in tensor processing. It is more accurate to find the lower-rank approximation of the three-order tensor with Tucker model. Then MDL(Minimum Description Length) criterion is used to estimate the optimal tensor rank in the second method. Further, multimode Wiener filtering approach upon tensor analysis can be considered as the spanning of one-mode wiener filtering. How to take advantages of tensor model to obtain a set of filters is the heart of the novel scheme. The performances of the proposed three approaches are evaluated with objective indexes and listening quality test. The experimental results indicate that the proposed tenor filtering methods have potential ability of retrieving the target signal from noisy microphone array signal and the multi-mode Wiener filtering method provides the best denoising results among the three ones.
基金Supported by National Natural Science Foundation of China(Grant Nos.51675425,52075441)Shaanxi Provincial Key Research Program Project of China(Grant No.2020ZDLGY06-09)+1 种基金Dongguan Municipal Social Science and Technology Development(key)Project of China(Grant No.20185071021600)Science and Technology on Micro-system Laboratory Foundation of China(Grant No.6142804200405).
文摘Wideband acoustic imaging,which combines compressed sensing(CS)and microphone arrays,is widely used for locating acoustic sources.However,the location results of this method are unstable,and the computational efficiency is low.In this work,in order to improve the robustness and reduce the computational cost,a DCS-SOMP-SVD compressed sensing method,which combines the distributed compressed sensing using simultaneously orthogonal matching pursuit(DCS-SOMP)and singular value decomposition(SVD)is proposed.The performance of the DCS-SOMP-SVD is studied through both simulation and experiment.In the simulation,the locating results of the DCS-SOMP-SVD method are compared with the wideband BP method and the DCS-SOMP method.In terms of computational efficiency,the proposed method is as efficient as the DCS-SOMP method and more efficient than the wideband BP method.In terms of locating accuracy,the proposed method can still locate all sources when the signal to noise ratio(SNR)is−20 dB,while the wideband BP method and the DCS-SOMP method can only locate all sources when the SNR is higher than 0 dB.The performance of the proposed method can be improved by expanding the frequency range.Moreover,there is no extra source in the maps of the proposed method,even though the target sparsity is overestimated.Finally,a gas leak experiment is conducted to verify the feasibility of the DCS-SOMP-SVD method in the practical engineering environment.The experimental results show that the proposed method can locate both two leak sources in different frequency ranges.This research proposes a DCS-SOMP-SVD method which has sufficient robustness and low computational cost for wideband acoustic imaging.
基金This work is supported by the National Nature Science Foundation of China(NSFC)under Grant No.61571106Jiangsu Natural Science Foundation under Grant No.BK20170757the Natural Science Foundation of the Jiangsu Higher Education Institutions of China under grant No.17KJD510002.
文摘Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transform(SRP-PHAT)spatial spectrum as input feature is presented in this paper.Since the SRP-PHAT spatial power spectrum contains spatial location information,it is adopted as the input feature for sound source localization.DNN is exploited to extract the efficient location information from SRP-PHAT spatial power spectrum due to its advantage on extracting high-level features.SRP-PHAT at each steering position within a frame is arranged into a vector,which is treated as DNN input.A DNN model which can map the SRP-PHAT spatial spectrum to the azimuth of sound source is learned from the training signals.The azimuth of sound source is estimated through trained DNN model from the testing signals.Experiment results demonstrate that the proposed algorithm significantly improves localization performance whether the training and testing condition setup are the same or not,and is more robust to noise and reverberation.
基金supported by the Human Sixth Sense Programme at the Advanced Digital Sciences Center from Singapore’s Agency for Science,Technology and Research
文摘In this paper, the frequency-domain Frost algorithm is enhanced by using conjugate gradient techniques for speech enhancement. Unlike the non-adaptive approach of computing the optimum minimum variance distortionless response （MVDR） solution with the correlation matrix inversion, the Frost algorithm implementing the stochastic constrained least mean square （LMS） algorithm can adaptively converge to the MVDR solution in mean-square sense, but with a very slow convergence rate. In this paper, we propose a frequency-domain constrained conjugate gradient （FDCCG） algorithm to speed up the convergence. The devised FDCCG algorithm avoids the matrix inversion and exhibits fast convergence. The speech enhancement experiments for the target speech signal corrupted by two and five interfering speech signals are demonstrated by using a four-channel acoustic-vector-sensor （AVS） micro-phone array and show the superior performance.
基金Project supported by the National Natural Science Foundation of China (Grant No.61001160)the Doctoral Foundation of Ministry of Education (Grant No.20093108120018)the Shanghai Leading Academic Discipline Project (Grant No.S30108)
文摘To improve localization accuracy, the spherical microphone arrays are used to capture high-order wavefield in- formation. For the far field sound sources, the array signal model is constructed based on plane wave decomposition. The spatial spectrum function is calculated by minimum variance distortionless response （MVDR） to scan the three-dimensional space. The peak values of the spectrum function correspond to the directions of multiple sound sources. A diagonal loading method is adopted to solve the ill-conditioned cross spectrum matrix of the received signals. The loading level depends on the alleviation of the ill-condition of the matrix and the accuracy of the inverse calculation. Compared with plane wave decomposition method, our proposed localization algorithm can acquire high spatial resolution and better estimation for multiple sound source directions, especially in low signal to noise ratio （SNR）.
文摘Audio applications such as mobile communication and hearing aid devices demand a small size but high performance, stable and low cost microphone to reproduce a high quality sound. Capacitive microphone can be designed to fulfill such requirements with some trade-offs between sensitivity, operating frequency range, and noise level mainly due to the effect of device structure dimensions and viscous damping. Smaller microphone size and air gap will gradually decrease its sensitivity and increase the viscous damping. The aim of this research was to develop a mathematical model of a spring-supported diaphragm capacitive MEMS microphone as well as an approach to optimize a microphone’s performance. Because of the complex shapes in this latest type of diaphragm design trend, analytical modelling has not been previously attempted. A novel diaphragm design is proposed that offers increased mechanical sensitivity of a capacitive microphone by reducing its diaphragm stiffness. A lumped element model of the spring-supported diaphragm microphone is developed to analyze the complex relations between the microphone performance factors and to find the optimum dimensions based on the design requirements. It is shown analytically that the spring dimensions of the spring-supported diaphragm do not have large effects on the microphone performance com pared to the diaphragm and backplate size, diaphragm thickness, and air-gap distance. A 1 mm2 spring-supported diaphragm microphone is designed using several optimized performance parameters to give a –3 dB operating bandwidth of 10.2 kHz, a sensitivity of 4.67 mV/Pa (–46.5 dB ref. 1 V/Pa at 1 kHz using a bias voltage of 3 V), a pull-in voltage of 13 V, and a thermal noise of –22 dBA SPL.
基金Supported by the National Natural Science Foundation of China(No.61201345)the Beijing Key Laboratory of Advanced Information Science and Network Technology(No.XDXX1308)
文摘The Steered Response Power(SRP)method works well for sound source localization in noisy and reverberant environment.However,the large computation complexity limits its practical application.In this paper,a fast SRP search method is proposed to reduce the computational complexity using small-aperture microphone array.The proposed method inspired by the SRP spatial spectrum includes two steps:first,the proposed method estimates the azimuth of the sound source roughly and determines whether the sound source is in far field or near field;then,different fine searching operations are performed according to the sound source being in far field or near field.Experiments both in simulation environments and real environments have been performed to compare the localization accuracy and computation complexity of the proposed method with those of the conventional SRP-PHAT algorithm.The results show that,the proposed method has a comparative accuracy with the conventional SRP algorithm,and achieves a reduction of 93.62%in computation complexity compared to the conventional SRP algorithm.
