Sound Pressure Information Research Articles

Robot assisted bone milling state classification network with attention mechanism

In the process of medical robot assisted bone milling surgery, the accuracy of recognition of milling state is crucial for surgical safety. However, previous studies have rarely used neural networks for signal analysis and processing, and not included attention mechanisms in neural networks to distinguish the weights of different signal features. In this paper a tactile-auditory attention model for milling state recognition is proposed. The model combines attention mechanism with fully connected neural networks. First, the milling state is divided into four types: idling, cortical bone, cancellous bone, and muscle. The acceleration and sound pressure information are extracted in 13 dimensions each, including 3-dimensional time-domain features and 10-dimensional frequency-domain features. Second, a milling state classification network with attention mechanism was established, including pre-connected attention mechanism (Pre-AT) and embedded attention mechanism (Emb-AT). The experimental results showed greater performance than other traditional methods, with test set accuracy of 94.57% and 95.29%, respectively. Afterwards, the impact of single signal and fused signal on recognition results was explored. From the experimental results, fused tactile-auditory signals had higher accuracy than single signal recognition. The accuracy rates of the test set using fused signals and acceleration and sound pressure signals were 95.29%, 92.09% and 90.07%. In addition, attention vectors are visualized to identify the degree of emphasis on different signals during the recognition process.

Non-iterative reconstruction of time-domain sound pressure and rapid prediction of large-scale sound field based on IG-DRBEM and POD-RBF

In this paper, a novel non-iterative reconstruction method of time-domain sound pressure and rapid prediction scheme of large-scale sound field are proposed respectively based on the isogeometric dual reciprocity boundary element method (IG-DRBEM) and proper orthogonal decomposition - radial basis functions (POD-RBF). By establishing the non-iterative inversion theory in the meaning of least squares, infinite domain acoustic holography is achieved with a small number of measured sound pressure information. Furthermore, the POD-RBF network is adopted to predict large-scale sound pressure. The presented work further expands the scope of application of IG-DRBEM, and gives full play to its advantage of not having to calculate the coefficient matrix repeatedly when analyzing the sound field at different moments. Moreover, in order to satisfy the integral regularization conditions, a suitable convergence condition about variation coefficients is investigated. Numerical results show that the proposed method has high accuracy and good stability even when the sound pressure of complex airship model is predicted with large-scale freedom.

Spatially extended sound source reconstruction based on the non-synchronous measurements of microphone arrays and Bayesian compressive sensing

The Non-synchronous Measurements (NSM) was proposed to enhance the accuracy of reconstruction and reduce the measurement cost. However, the investigated object of conventional NSM are only sparsely distributed sound sources, and little emphasis placed on the reconstructed accuracy of the spatially extended sound source. In this article, the traditional NSM is further investigated in the spatially extended sound source, and the sound pressure information is reconstructed by Bayesian compressive sensing algorithm (BCS). Firstly, the spatially extended sound source is scanned by moving the rectangular microphone array, and the measurement effects of the array with large aperture and high density are approximately obtained by using NSM technique. Then, BCS is applied to reconstruct sound field by the NSM of sound pressure. The results of numerical simulations provide evidence for the feasibility of applying the proposed investigation to the spatially extended sound source. NSM has advantages in reconstruction accuracy compared with the single measurement. At the same time, the reconstructed sound pressure under NSM can replace the simultaneous measurement (SM) results of multi-array, which avoids the limitation of using reference microphone in increasing the number of channels and spatial layout. The experimental results of the driven steel plate provide additional evidence to support the effectiveness and feasibility of the investigation.

Sound field visualization system by MVDR beamforming with augmented reality and point clouds

In previous research, we proposed an augmented reality (AR) and mixed reality (MR) system visualizing the sound field data measured by the beamforming method. Our system visualized the sound pressure as a colormap on the AR/MR device in real-time. Free viewpoint and visualization of a colormap along the spatial geometry observed by the AR/MR sensor enabled us to understand the location of the sound source. On the other hand, the spacial geometry information of the measurement space was used only for display purposes and not in the processing of sound source location estimation. And then the previous system displaying a flat colormap projected on the spatial geometry has not been able to make measurements that precisely in space. In this paper, we propose a sound field measurement and AR/MR visualization system using point cloud data acquired by light detection and ranging (LiDAR) sensor. The MVDR (minimum variance distortionless response) beamforming calculates the beamforming output by using point cloud data. The sound sources are assumed to be located on point clouds, and the system allows fast localization of a large number of sources. In addition, it is possible to interpret the sound field in various ways based on the scattered sound pressure information. We show some basic experiments conducted in an anechoic chamber.

A Low Frequency Noise Source Localization and Identification Method Based on a Virtual Open Spherical Vector Microphone Array

Aiming at the problem of poor spatial resolution of low-frequency noise sources in a small-aperture spherical microphone array (SMA), this paper proposes a method for localizing and identifying low-frequency noise sources based on virtual-vector open SMA (‘p+v’ joint processing method of pressure and velocity). Firstly, a virtual open SMA with a larger aperture is obtained using a virtual array extrapolation method. In this method, the virtual SMA and the actual SMA are regarded as a dual-radius SMA, and velocity information is obtained using finite difference elements of the same direction (azimuth and elevation) array of the virtual and actual SMA. At the same time, the sound pressure at the velocity position is obtained using the virtual SMA extrapolation method and the virtual vector array element SMA, whereby both velocity and sound pressure information is obtained. Finally, the vector signal processing technology is introduced into the generalized inverse beamforming algorithm (GIB). After determining the vector transfer function of the ‘p+v’ joint processing mode, a low-frequency-noise-source localization and identification method based on the vector signal processing GIB is proposed. The simulation and experiment results show that a virtual SMA with a large aperture can be obtained using a virtual array extrapolation method, and the GIB with sound pressure and velocity joint processing has a better spatial resolution.

Design and Implementation of a Composite Hydrophone of Sound Pressure and Sound Pressure Gradient.

The bionic cilium MEMS vector hydrophone has the characteristics of low power consumption, small volume, and good low-frequency response. Nevertheless, there exists the problem of left–right ambiguity in the azimuth estimation of a single hydrophone. In order to solve the engineering application problem, a sound-pressure sound-pressure-gradient hydrophone is designed in this paper. The new composite hydrophone consists of two channels. The bionic cilium microstructure is optimized and used as the vector channel, to collect the sound pressure gradient information, and a scalar channel, based on a piezoelectric ceramic tube, is added, to receive the sound pressure information. The theoretical analysis, simulation analysis, and test analysis of the composite hydrophone are carried out, respectively. The test results show that the sensitivities of the hydrophone can reach up to −188 dB (vector channel) and −204 dB (scalar channel). The problem of left–right ambiguity is solved by combining the sound pressure and sound pressure gradient in different ways. This is of great significance in the engineering application of single cilium MEMS hydrophone orientation.

Optimization of Shell packaging for cilium MEMS bionic vector hydrophone

Packaging of vector hydrophone influences its engineering application scope. However, the sensitivity of vector hydrophone is greatly reduced after packaging in previous works. Moreover, the vector hydrophone with dipole directivity has the problem of left-right ambiguity. To solve these problems, the shell packaging of cilium MEMS bionic vector hydrophone is optimized in terms of sound-transparent cap and the capability of underwater localization. A blue nitrile butadiene rubber is used as the material for the sound-transparent cap to solve the problem of poor sensitivity after packaging. And in order to solve the problem of left-right ambiguity of a single vector hydrophone, a piezoelectric ceramic tube is integrated into the shell to obtain sound pressure information. In this paper, the packaging process of optimized shell is introduced in detail. Then, the sensitivity test, directivity test and pressure-resisting test of optimized shell are finished. The testing results show that sensitivity of vector information part is increased average 5.6 dB after packaging. And the sensitivity curve depicts that the sensitivity increases 6 dB per octave in the working bandwidth in the range of 63−1000 Hz. Therefore, the optimized vector hydrophone shows a good performance in engineering application.

Optimization for rotor blade-tip planform with low high-speed impulsive noise characteristics in forward flight

Rotor noise is one of the most important issues for helicopter designer, and high-speed impulsive noise is particularly intense among the various rotor noise sources due to compressibility. Based on Computational Fluid Dynamics/Ffowcs Williams and Hawkings equations with Penetrable Data Surface (CFD/FW-Hpds) methods and hybrid optimization technique, a new optimization design procedure for rotor blade planform with low high-speed impulsive noise characteristics is established. First, in order to accurately capture the unsteady aerodynamic characteristics of rotor, based on the moving-embedded grid methodology, a CFD simulation is developed by solving the compressible Reynolds-average Navier–Stokes equations with Baldwin–Lomax turbulence model. The low dissipation Roe-Monotone Upwind-centered Scheme for Conservation Laws (MUSCL) scheme and highly efficient implicit lower–upper symmetric Gauss–Seidel scheme are used for spatial and temporal discretization, respectively. Second, taking the CFD results as sound pressure information input, the high-speed impulsive noise characteristics generated by transonic rotor are analyzed through a robust numerical method based on FW-Hpds equations. Third, the genetic algorithm and surrogated model based on radial basis function are combined as a hybrid optimization technique; during the optimization process, the blade grids are generated by a highly efficient parameterized method. Aiming at the minimization of the sound pressure level of rotor in forward flight, the parametric effect analyses of blade-tip shapes on transonic noise have been conducted first. Then, optimization analyses based on the rotor blade with double-swept and tapered tip have been accomplished with the aerodynamic performance as constraints. Compared with the baseline blade, it shows that the sound pressure level of rotor with optimized blade-tip shape can be decreased obviously at the present calculating condition due to its weaker transonic delocalization phenomenon in the region of blade tip. In the rotor plane, absolute peak value of sound pressure produced by the optimized blade planform is reduced about 59.1% of that by the baseline one, and the reduced value in sound pressure level is up to 5.6 dB.

A new technique to measure directional impulse responses using a 64-channel spherical microphone

A spherical microphone is an array consisting of numerous microphones arranged almost uniformly on the surface of a rigid sphere. Recently, spherical microphone arrays have received attention in architectural acoustics for their ability to characterize the sound field over a full sphere, which can yield additional information about the spatial characteristics of the field, including field diffusivity and directionality of individual reflections. Typically, these arrays are used as real-time analyzers to capture sound pressure information. However, in architectural acoustics, an impulse response measurement is required to adequately characterize the space. An effective approach for utilizing spherical microphones, particularly those with a large number of channels, to capture directional impulse responses has yet to be realized. The current project has utilized a 64-channel spherical microphone to carry out the integrated impulse response measurement technique using a sine-sweep signal. MATLAB has been used to evaluate sound fields in a reverberant chamber using a variety of directionality patterns (e.g., omnidirectional, cardioid, and figure-of-eight). Additionally, work is in progress to allow for arbitrary beamforming and other directionality patterns. Results and analysis will be presented.

A straightforward method for wall impedance eduction in a flow duct

The development of the advanced liner technology for aeroengine noise control necessitates the impedance measurement method under realistic flow conditions. Currently, the methods for this need are mainly based on the inverse impedance eduction principle, confronting with the problems of initial guess, high computation cost, and low convergence. In view of this, a new strategy is developed that straightforwardly educes the impedance from the sound pressure information measured on the duct wall opposing to the test acoustic liner embedded in a flow duct. Here, the key insight is that the modal nature of the duct acoustic field renders a summed-exponential representation of the measured sound pressure; thus, the characterizing axial wave number can be readily extracted by means of Prony's method, and further the unknown impedance is calculated from the eigenvalue and dispersion relations based on the classical mode-decomposition analysis. This straightforward method is simple in its basic principle but remarkably has the advantages of ultimately overcoming the drawbacks inherent to the inverse methods, incorporating the realistic multimode nonprogressive wave effects, high computational efficiency, possibly reducing the measurement points, and even avoiding the necessity of the duct exit impedance that bothers perhaps all the existing waveguide methods.

Measurement of Sound Velocity in Water Using Optical Probe and Acoustical Holography

We propose a method for a determination of a three‐dimensional sound velocity using optical computerized tomography (O‐CT) and near‐field acoustical holography (NAH) for measuring a phase object. Ultrasonic waves affect a phase of the light passing through radiated sound fields. The zeroth order diffraction light including sound pressure information is acquired by an avalanche photodiode. Eighteen projections are acquired by rotational scanning, and each projection along the optical axis is obtained by single linear scanning and electronically quadrature‐detected as the complex amplitude. The complex sound fields are reconstructed by O‐CT in a region of 40 × 40 mm2. Then the sound field on another plane is propagated using NAH from the acquired sound fields, and the sound field on the same plane is reconstructed by O‐CT. Comparing the phase of propagated and reconstructed sound fields in wavenumber domain, we can obtain the sound velocity in three dimensional space in a region of 30 × 30 × 10 mm3. The experimental results are in agreement with the reference value measured by another mean.

Determination of Sound Velocity in Three-Dimensional Space by Optical Probe

We propose a method of determining a three-dimensional sound velocity using a Michelson interferometer, optical computerized tomography (O-CT) and near-field acoustical holography (NAH). Ultrasonic waves affect the phase of the test light passing through radiated sound fields. The zeroth-order diffraction light including sound pressure information is electrically acquired by an avalanche photodiode (APD). Projection data along the optical axis is obtained by single linear scanning in the range of ±30 mm and electronically quadrature-detected as a complex amplitude. Eighteen projections are acquired in the range of 0≤θ<π rad, and the complex sound fields are reconstructed in a region of 40×40 mm2 by O-CT. Then another plane separated by 1 mm is propagated using NAH from the acquired sound fields, and the same plane is reconstructed. Comparing the phase of the reconstructed and propagated sound fields in wave number domain, we determine the three-dimensional sound velocity in a region of 40×40×1 mm3. The experimental results are in agreement with the reference value.

Optical Computerized Tomography for Visualization of Ultrasonic Fields Using Michelson Interferometer

We propose a method for the noncontact visualization of ultrasonic fields using the Michelson interferometer and optical computerized tomography (O-CT). The light source of the interferometer is a He–Ne laser. The test light passing through radiated sound fields is interfered by the reference light. Therefore, the index of refraction gradient including sound pressure information is transformed into the intensity of the interference light that can be electrically acquired by an avalanche photo-diode. Projection data along the optical axis is obtained by single linear scanning in the range of ±32 mm and electronically quadrature-detected as complex amplitude. Thirty-six projections are acquired in the range of 0≤θ<π (rad), and complex sound fields are reconstructed in a region of 64×64 mm2 by CT. The experimental results are in agreement with the numerical results.

Visualization of Ultrasonic Beam Using Michelson Interferometer

Ultrasonic beams radiated from a circular transducer are visualized using a Michelson interferometer. The light source of the interferometer is a 632.8 nm-He–Ne laser. The test light that passes through the radiated sound fields is interfered with by the reference light. Therefore, the index of refraction gradient including sound pressure information is transformed into the intensity of the interference light that can be electrically acquired by a photodetector. The transducer is 8 mm in diameter and its center frequency is 1.416 MHz. The visualization region is 100 mm in the sound axial direction, and ±30 mm in the lateral direction, where the origin is the center of the vibrating surface. Information acquired as a result of the experiment is the two-dimensional image that is a projection of the three-dimensional sound pressure distribution along the optical axis.

Speechreading supplemented with auditorily presented speech parameters.

Results are reported from two experiments in which the benefit of supplementing speechreading with auditorily presented information about the speech signal was investigated. In experiment I, speechreading was supplemented with information about the prosody of the speech signal. For ten normal-hearing subjects with no experience in speechreading, the intelligibility score for sentences increased significantly when speechreading was supplemented with information about the overall amplitude of the speech signal, information about the fundamental frequency, or both. Binary information about voicing appeared not to be a significant supplement. In experiment II, the best-scoring supplements of experiment I were compared with two supplementary signals from our previous studies, i.e., information about the sound-pressure levels in two 1-oct filter bands centered at 500 and 3160 Hz, or information about the frequencies of the first and second formants from voiced speech segments. Sentence-intelligibility scores were measured for 24 normal-hearing subjects with no experience in speechreading, and for 12 normal-hearing experienced speechreaders. For the inexperienced speechreaders, the sound-pressure levels appeared to be the best supplement (87.1% correct syllables). For the experienced speechreaders, the formant-frequency information (88.6% correct), and the fundamental-frequency plus amplitude information (86.0% correct), were equally efficient supplements as the sound-pressure information (86.1% correct). Discrimination of phonemes (both consonants and vowels) was measured for the group of 24 inexperienced speechreaders. Percentage correct responses, confusion among phonemes, and the percentage of transmitted information about different types of manner and place of articulation and about the feature voicing are presented.

Speechreading supplemented with frequency-selective sound-pressure information and formant-frequency information

The benefit of supplementing speechreading with frequency-selective sound-pressure information or formant-frequency information was studied by auditorily presenting this information to normal-hearing listeners without experience in speechreading. In a first experiment the sound-pressure levels in one or two, octave or 1/3-octave bands were used to amplitude-modulate pure-tone carriers with frequencies equal to the center frequencies of the filterbands (500, 1600, or 3160 Hz). Eighteen listeners scored 65.7% correct syllables (from totally unknown sentences) for speechreading with sound-pressure information in a single one-octave band at 500 Hz, and 86.7% correct syllables for speechreading with two one-octave bands at 500 and 3160 Hz, respectively. Speechreading-only scored 22.8%. In a second experiment the frequencies of the first and second formant from voiced speech were presented either as pure tones, or as a complex speechlike signal. In this experiment 18 (other) listeners scored 82.0% correct syllables on speechreading with formant-frequency information. No significant difference was found between presenting this information as pure tones or as a complex speechlike signal. Speechreading-only scored again 22.8% correct syllables.

Speechreading supplemented with frequency-selective sound-pressure information.

The benefit of supplementing speechreading with frequency-selective sound-pressure information was studied by auditorily presenting this information to normal-hearing listeners. The sound-pressure levels in one or two frequency bands of the speech signal with center frequencies of 500, 1600, and 3160 Hz, respectively, and with 1- or 1/3-oct bandwidth were used to amplitude-modulate pure-tone carriers with frequencies equal to the center frequencies of the filter bands. Short sentences were presented to 18 normal-hearing listeners under the conditions of speechreading-only and speechreading combined with the sound-pressure information. The mean number of correctly perceived syllables increased from 22.8% for speechreading-only to 65.7% when sound-pressure information was supplied in a single 1-oct band at 500 Hz and to 86.7% with two 1-oct bands at 500 and 3160 Hz, respectively. The latter signal scored only 26.7% correct syllables without accompanying visual information.