239 results about "Acoustic property" patented technology

The invention relates to the technical field of defect detection, and discloses a flange forging defect detection method and system. The method comprises the following steps: performing three-dimensional scanning and material acoustic characteristic measurement on a to-be-detected wind power flange to obtain layer partition sound isolation path data; generating an array element excitation control file of the double-array phased array ultrasonic detection system; applying the array element excitation control file to a double-array phased array ultrasonic detection system, and performing micro defect feature enhancement on a received echo signal to obtain a feature-enhanced signal data set; performing ultrasonic emission and data acquisition on the wind power flange to obtain a global detection data set; and inputting the global detection data set into the two-stage defect detection model for defect feature extraction and classification evaluation, and generating a defect detection evaluation report. According to the method, the detection rate and the classification accuracy of the micro forging defects are improved.

The air-pulse generating device comprises a film structure, operating at an ultrasonic operating frequency, and a resonant chamber, formed on a side of the film structure. A resonance is formed within the resonance chamber. Due to the resonant chamber, the APG device has a peak on a frequency response of an acoustic property of the APG device at the ultrasonic operating frequency.

The invention relates to the technical field of voiceprint recognition, in particular to a photovoltaic equipment fault detection method and system based on voiceprint recognition, and the method comprises the following steps: collecting the sound of the surrounding environment of a current photovoltaic power station, and building scene associated equipment spectrum peak features; according to the method, classification of environment sound is introduced in the sound acquisition stage, and scene association is performed on the operation sound of the photovoltaic equipment and the surrounding background sound in combination with Mel-frequency cepstrum coefficient extraction and acoustic scene recognition, so that a more targeted reference framework is provided for subsequent analysis. Modeling of multi-source sound features is achieved by synchronously collecting operation sound of an inverter and a cooling fan in equipment and extracting spectrum peak frequency points and amplitudes. In the signal processing process, multiple sub-bands are divided on the basis of scene correlation spectrum peak frequency points, so that each sub-band is aligned with the acoustic characteristics of equipment, the structural precision of short-time energy envelope extraction is improved, and it is ensured that the cross-correlation analysis result is more reliable.

The invention relates to the technical field of material analysis, and discloses an intelligent raw material composition screening method and system based on a sound insulation material, and the method comprises the steps: collecting acoustic characteristic test data, microstructure data and chemical composition data of a candidate sound insulation material; performing frequency response curve decomposition on the acoustic characteristic test data to obtain a characteristic frequency band sound absorption coefficient, and constructing an acoustic impedance tensor of the sound insulation material to analyze the sound wave dissipation performance of the candidate sound insulation material; calculating the pore distribution entropy and connectivity index of the candidate sound insulation material to analyze the sound wave propagation retardation efficiency of the candidate sound insulation material; analyzing the molecular chain flexibility and interfacial compatibility of the candidate sound insulation material to evaluate the acoustic performance stability of the candidate sound insulation material for multi-component compounding; and screening a target material from the candidate sound insulation materials by using sound wave dissipation performance, sound wave propagation retardation efficiency and acoustic performance stability. The performance matching degree of the raw materials screened out of the sound insulation material in actual composite use can be improved.

Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for generating an audio signal from input text. In one aspect, a method includes receiving a request to convert an input text to an audio signal, where the input text includes a plurality of lexical meta text inputs; generating, using a first generative neural network, a semantic representation of the lexical text input, the semantic representation comprising semantic terms representing semantic content of the lexical text input, each semantic term selected from a word list of semantic terms; generating an acoustic representation of the semantic representation using a second generative neural network and on at least the semantic representation, the acoustic representation comprising one or more respective acoustic lems representing acoustic properties of the audio signal; and processing the acoustic representation using a decoder neural network to generate an audio signal.

The invention provides a cell acoustic characteristic parameter estimation method and system based on a full-connection neural network and a computer medium, and relates to the technical field of biomedical engineering and micro-nano operation, and the method comprises the steps: constructing a three-layer spherical shell model of a target cell, defining the radius, density and sound velocity of each layer and a surrounding medium, and constructing a three-layer spherical shell model of the target cell; the acoustic impedance and the wave number of each layer are calculated; obtaining a transducer driving parameter, a displacement speed of the cell or equivalent particle solid sphere particle, and physical parameters of the cell and a medium; inputting the input feature data into a pre-trained full-connection neural network model to output acoustic characteristic parameters such as a scattering coefficient and an internal refraction coefficient; and calculating acoustic radiation force based on the acoustic characteristic parameters to realize dynamic control. Based on the full-connection neural network and the three-layer spherical shell model, the cell acoustic parameters are efficiently and accurately estimated, the problems that a traditional method is complex in calculation and insufficient in precision are solved, and accurate operation of the acoustic tweezers on the cells is achieved.

The invention provides an ultrasonic detection method for internal defects of an insulating part of a power system based on a piezoelectric ultrasonic transducer, and belongs to the technical field of insulating part detection. After wavelet denoising and deep attenuation compensation are carried out on echo signals, sound path distance time delay and sound pressure amplitude parameters are extracted, defect identification and position estimation are carried out by utilizing a layered variational auto-encoder, a high-frequency transducer is switched to aiming at tiny defects, a harmonic component is extracted by adopting a nonlinear ultrasonic method, and a characteristic size is calculated; a defect distribution space feature matrix is generated based on a synthetic aperture focusing technology, a unified or personalized processing strategy is selected according to an acoustic characteristic homogenization degree evaluation value, a complete defect detection result is output, and the technical problem that deep weak defects in the electric insulation part are difficult to accurately detect and recognize is solved.

The invention discloses a spatial sound effect test method and system for a sound system. The method comprises the following steps: S1, synchronously acquiring multi-modal sensing data of a test environment in a preset stage; s2, generating a three-dimensional sound pressure distribution thermodynamic diagram, and constructing an initial sound field model based on the three-dimensional sound pressure distribution thermodynamic diagram; s3, generating a corrected sound field model, and synchronously recording vibration data acquired by the triaxial accelerometer; s4, generating a sound image positioning trajectory diagram, and calculating a deviation angle between an actual sound image position and a theoretical position; and S5, determining the sound effect quality score of each test stage. According to the method, a closed-loop test framework of physical environment perception, sound field real-time modeling, man-machine interaction analysis and fault linkage traceability is constructed, a full-process test solution covering a hardware state, acoustic characteristics and human factor perception is provided for an immersive audio system, and the spatial sound effect reliability of digital playing equipment is remarkably improved.

The present disclosure relates to the technical field of audio control. Provided are a computer-implemented method for audio control, and an electronic device and a storage medium. The computer-implemented method for audio control comprises: on the basis of a specified measurement signal, performing acoustic effect testing on a target space, in order to obtain environmental response information, wherein the environmental response information comprises impulse response information and post-separation impulse response information; analyzing the environmental response information, in order to obtain acoustic characteristic information of the target space; on the basis of the acoustic characteristic information, performing compensation processing on audio to be processed, in order to obtain regulated audio; regulating gains of multiple channels corresponding to the regulated audio, in order to obtain re-regulated audio; and performing multi-channel collaborative optimization processing on the re-regulated audio, in order to obtain audio having a target effect. The method achieves precise compensation for spatial acoustic defects, resolves inter-channel crosstalk, improves the acoustic image localization and spatial perception, significantly improves the perceptual audio quality, and optimizes an output.

The invention discloses a method and a system for generating a sound field capable of perceiving rhythm characteristics. The method comprises the following steps of: determining a group of acoustic guidance parameters containing target rhythm information, and driving an excitation mechanism to excite one or more physical resonators according to the acoustic guidance parameters so as to generate a basic acoustic environment of which the acoustic characteristics are completely dominated by the physical attributes of the resonators. The physical acoustic environment forms a sound field which covers a range of 1-100 Hz and has a perceivable rhythm characteristic through an endogenous acoustic beat effect or isochronous sound effect response of the physical acoustic environment. By returning the source of physical vibration, the technical defects of lack of physical reality and unnatural auditory experience of the existing electronic synthesis acoustic technology are solved, a sound field generation mode which is real and natural in acoustic performance and can be efficiently synchronized with a human body perception rule is created, and the sound field generation mode can be collaboratively presented with conventional audio to enhance immersion.

The invention belongs to the technical field of digital design of Roots blower equipment, and particularly relates to a digital twinning-based dynamic performance digital design method of a blower, which comprises the following steps of: constructing a multi-component parameterized structure model of the blower, establishing a multi-physics field coupling model of pneumatic, structural, thermal, electromagnetic and sound fields, and constructing a digital twinning model of the blower. Dynamic boundary conditions such as flow disturbance, inertia force and base disturbance are loaded, a digital twin model synchronous with a physical fan state is constructed, master-slave type strong coupling joint solution is executed, multi-field response indexes are extracted and normalized analysis is carried out, and finally optimization design parameters of the fan are output through a multi-target optimization feedback mechanism. According to the method, multi-domain balance design among fan structure performance, electromagnetic response, thermal stability and acoustic characteristics can be achieved, modeling precision and parameter optimization capacity are improved, and the method is suitable for digital development of high-reliability roots blower equipment under complex operation conditions.

Some implementations relate to methods, systems, and computer-readable media to reproduce a voice stream with removed personally identifiable information (PII). An original voice stream is received from a user by a hardware processor. The original voice stream is converted into text using a machine learning (ML) voice transcription model. Then, the text is used for generating an anonymized voice stream corresponding to the text having similar auditory properties to the original voice stream other than PII using a machine learning (ML) voice synthesis model. The similar properties are determined using metadata corresponding to the original voice stream. The metadata may be obtained by analyzing acoustic characteristics of the original voice stream. The metadata may include volume changes and pitch frequency changes stored as metadata annotations associated with tokens in the text as key-value pairs. The generation may use the metadata annotations to determine the similar audio properties.

The invention discloses a sound wave detection method and system for an underground pipeline, and the method comprises the following steps: 1, dynamically adjusting the detection height in the radial direction of a pipeline through a lifting adjustment device of a first sound wave detection mechanism, and enabling a sound wave transmitting end to adapt to the inner walls of pipelines with different pipe diameters; 2, a rotation driving device of a second sound wave detection mechanism controls a sound wave transmitting end to conduct multi-angle rotation scanning around the axis of the pipeline, and the circumferential area of the cross section of the pipeline is covered; 3, based on the pipeline inner diameter D, the material acoustic characteristics and the preset coverage rate eta, motion parameters are generated through a dynamic optimization algorithm; 4, collecting a reflected sound wave signal and carrying out multi-stage processing; through the implementation of the method, a dynamic optimization algorithm of full-dimensional scanning coverage enables the screw pitch error of a spiral track to be small, 100% non-blind area coverage of the cross section of the pipeline is achieved, the self-adaptive focusing lens controls the sound beam focusing angle error to be within + / -2 degrees, and the detection sensitivity of the thin-walled pipe is improved.

An acoustic wave device is disclosed. The acoustic waved device can be a shear horizontal mode surface acoustic wave device. The acoustic wave device can include a piezoelectric layer, an interdigital transducer electrode over the piezoelectric layer, and a temperature compensation layer over the interdigital transducer electrode. The piezoelectric layer can be a lithium niobate layer with a cut angle in a range of −20° YX to 25° YX. The interdigital transducer electrode including a first layer and a second layer. The first layer affects acoustic properties of the acoustic wave device and the second layer affects electrical properties of the acoustic wave device. The second layer is positioned between the piezoelectric layer and the first layer such that a frequency response of the acoustic wave device includes a Rayleigh mode response at a frequency higher than a shear horizontal mode response.

A method can include determining carbon dioxide acoustic properties for at least supercritical carbon dioxide using thermodynamics that relate isothermal compressibility and adiabatic compressibility: determining fluid-saturated rock acoustic properties using the carbon dioxide acoustic properties; and performing a seismic workflow using the fluid-saturated rock acoustic properties.

Methods, systems, and devices for determining an acoustic parameter of a downhole fluid using an acoustic assembly. Methods include transmitting a plurality of pulses; measuring values for at least one wave property measured for reflections of the plurality of pulses received at at least one acoustic receiver, including: a first value for a first reflection traveling a first known distance from a first acoustically reflective surface having a first known acoustic impedance, a second value for a second reflection traveling a second known distance substantially the same as the first known distance from a second acoustically reflective surface having a second known acoustic impedance, and a third value for a third reflection traveling a third known distance from a third acoustically reflective surface having a third known acoustic impedance substantially the same as the second acoustic impedance; and estimating the acoustic parameter using the values.

The invention relates to the technical field of ocean monitoring, and particularly discloses a transducer array-based seabed sound velocity gradient in-situ measurement device, which comprises a transducer array unit, a depth adjusting unit, a transmitting and receiving control unit and a data processing unit, the transducer array unit comprises a plurality of transmitting transducers and a plurality of receiving transducers, the plurality of transmitting transducers form a transmitting array, the plurality of receiving transducers form a receiving array, and the opening angles of the transducers in the transmitting array and the receiving array are set to ensure that the plurality of receiving transducers can receive signals from the transmitting array. A protective shell is arranged outside the transmitting array and the receiving array, and the strength of the protective shell is enough to ensure that the protective shell can penetrate into seabed sediments with various hardness degrees; the depth adjusting unit is respectively connected with the protective shells of the transmitting array and the receiving array; by adopting the technical scheme of the invention, fine measurement of the acoustic characteristics of the seabed shallow sedimentary layer can be realized.

A scalable conferencing system can include a set of conferencing devices configured to be placed on a surface (such as, a table), mounted on a ceiling, or attached to a wall. The conferencing devices can be configured to be positioned in a room and pick up and / or reproduce audio in the room. Depending on the configuration and / or characteristics of the room (such as, dimensions, geometry, acoustical properties, furnishings, or the like), a plurality of conferencing devices of a particular type can be deployed in the room and / or conferencing devices of another particular type may not be deployed. The conferencing system can be used for audio and / or video conferencing.

The invention provides an underwater environment area dynamic search method, which comprises the following steps of: transmitting and receiving a sound wave signal by using a multi-beam sonar system, obtaining a high-resolution underwater environment image, and extracting particle size distribution data of submarine sediments through image analysis; establishing a mapping model by combining the sound wave frequency and the sediment acoustic characteristics to calculate a scattering intensity distribution map; adopting a pattern recognition algorithm to recognize the sediment particle size and the sound wave incident angle, and correcting inertial navigation data errors; a spatial filtering parameter optimization model is established based on sediment particle size distribution, filtering parameters are adjusted to remove noise, and scattering signal features are extracted; the configuration of the receiving array and the search area are dynamically adjusted according to the particle size change of the sediment, accurate and real-time analysis and environment adaptability search of the particle size distribution of the underwater sediment are realized, and the efficiency and accuracy of underwater detection are greatly improved.

Techniques for providing neuroacoustic modeling for an audio environment are disclosed herein. Examples may include receiving audio data and image data associated with an audio environment; generating an image set comprising a plurality of images, each image associated with an audio sample representing an acoustic property of the audio environment; and generating an audio rendering model for the audio environment based at least in part on the image set and the audio samples. The audio rendering model may include a neural rendered volumetric representation of the audio environment enhanced with audio coding.

Disclosed herein are systems and methods for generating and presenting virtual audio for mixed reality systems. A method may include determining a collision between a first object and a second object, wherein the first object comprises a first virtual object. A memory storing one or more audio models can be accessed. It can be determined if the one or more audio models stored in the memory comprises an audio model corresponding to the first object. In accordance with a determination that the one or more audio models comprises an audio model corresponding to the first object, an audio signal can be synthesized, wherein the audio signal is based on the collision and the audio model corresponding to the first object, and the audio signal can be presented to a user via a speaker of a head-wearable device. In accordance with a determination that the one or more audio models does not comprise an audio model corresponding to the first object, an acoustic property of the first object can be determined, a custom audio model based on the acoustic property of the first object can be generated, an audio signal can be synthesized, wherein the audio signal is based on the collision and the custom audio model, and the audio signal can be presented, via a speaker of a head-wearable device, to a user.

The present invention discloses a method, apparatus, device, and storage medium for measuring the acoustic properties of contrast agent microbubbles. The method includes: a calibration phase, obtaining a pulse signal emitted by a transmitting probe and an echo signal received by a receiving probe, and calculating the receiving transfer function of the receiving probe using the pulse signal and the echo signal; a measurement phase, controlling the transmitting probe to sweep within a preset frequency range to emit an ultrasonic excitation signal, and controlling the receiving probe to receive scattered signals from microbubbles in a phantom blood vessel; extracting the subharmonic signal voltage amplitude from the scattered signal; using the receiving transfer function of the receiving probe to convert the subharmonic signal voltage amplitude into actual subharmonic scattering pressure at different frequencies; and determining the optimal subharmonic driving frequency and subharmonic scattering pressure based on the actual subharmonic scattering pressure. The present invention improves the measurement accuracy of the nonlinear scattering properties of microbubbles by obtaining the receiving transfer function of the receiving probe and using it to obtain the actual scattered acoustic pressure signal.

The invention relates to the technical field of audio processing, and provides an audio signal processing method, electronic equipment and a computer readable storage medium. According to the method, in the process that the electronic equipment plays stereo audio through two loudspeakers with different acoustic characteristics and / or asymmetrical layout, sound channel signals of the two loudspeakers can be calibrated according to the frequency response difference of the two loudspeakers, so that the difference of the sound channel signals of the two loudspeakers is reduced, and the sound channel signals of the two loudspeakers can be corrected. And the sound images of the sound emitted by the two loudspeakers are prevented from shifting. And then, according to the position relationship between the two loudspeakers and the head of the user, the two calibrated sound channel signals are calibrated again, so that crosstalk generated between the two sound channel signals is eliminated, and the hearing feeling of the user is improved.

This application is entitled "Apparatus and Method for Acoustic Modeling of Defects in Composite Materials with Calibration Panels Formed by Additive Manufacturing." Disclosed herein is a calibration panel that models acoustic properties of defects in a vehicle component made of a composite material. The calibration panel includes a body formed by a first digital material through additive manufacturing, the acoustic properties of which are selected to approximate acoustic properties of the composite material, and an insert embedded within the body, the insert formed at least of a second digital material, the acoustic properties of which are selected to approximate acoustic properties of a defect within the composite material. Also disclosed herein is a method for manufacturing a vehicle component that complies with non-destructive evaluation (NDE). The method includes receiving a proposed design of a component, manufacturing a calibration panel, and testing the calibration panel to determine whether the insert is detectable using acoustic emission testing. The method further includes rejecting the proposed design of the component in response to a non-detectable state of the insert.

A computer having a system for mitigating mechanical rattle arising due to a mechanical coupling of an audio output transducer to a part of the computer device, the system comprising: a processing subsystem configured to receive an input audio signal and to output a drive signal for driving the audio output transducer to produce a transducer output, wherein the processing subsystem is configured to selectively apply an attenuation function to the input audio signal to attenuate a signal component of the input audio signal at a frequency that causes mechanical rattle in the computer; and an analyser subsystem configured to receive the input audio signal and to output a control signal to the processing subsystem to control application of the attenuation function to the input audio signal by the processing subsystem based on a spectral content of the received input audio signal; wherein the attenuation function is based on characteristic acoustic behaviour of the computer.

This disclosure provides a wedge and design method for a multipath acoustic attenuation phased array ultrasonic testing probe. The wedge includes a wedge body, multiple silencing grooves, and a damping block. The extension direction, spacing, and depth of each silencing groove are configured such that at least a portion of the reflected transverse wave generated at the detection surface can be guided to an attenuation interface formed by the groove wall of the silencing groove and the damping block. Furthermore, the reflected longitudinal wave generated at the detection surface undergoes at least six interface reflections within the wedge body before returning to the probe direction. The acoustic characteristics of the damping block are configured to absorb the reflected transverse wave. The reflected transverse wave is attenuated at the attenuation interface through a combination of scattering by the silencing groove and absorption by the damping block. After at least six interface reflections, the acoustic component of the reflected longitudinal wave returning to the probe direction forms a 180° phase difference with the incident longitudinal wave emitted by the probe, thus being attenuated through phase interference.

In order to adapt the frequency response of headphones, various filters are used, including shelving or frequency-shelving filters. However, these are usually implemented electronically. In the case of wired headphones without their own voltage supply, however, no electronic signal processing is possible. In order to realise an acoustic shelving filter, an acoustic damping element (DD) having a flat, acoustically effective damping material, which has a defined specific acoustic resistance, is placed in front of the radiating surface of the sound transducer at a small distance (d) in relation to the diameter of the acoustic damping element. It thus forms a flat air gap (AG), which is open at the edge, with respect to the radiating surface of the transducer. Acoustic properties of the shelving filter can be controlled by the size and material of the damping element (DD) and its distance from the sound panel (SW) or from the transducer (W).

Disclosed herein is a method for determining a characteristic of a hearing instrument, the hearing instrument including at least one input transducer operable to provide an input audio signal responsive to sensing sound in the environment of the hearing instrument, a signal processing unit and at least one output transducer, the method comprising: emitting an acoustic probe signal by the output transducer, receiving an input audio signal from the microphone, analyzing the received input audio signal to determine the characteristic of the hearing instrument from an input transducer response to the emitted acoustic probe signal, wherein the method further comprises filtering the received input audio signal to selectively attenuate one or more signal components corresponding to the acoustic probe signal and wherein emitting the acoustic probe signal comprises emitting a combined acoustic output signal comprising the acoustic probe signal and an acoustic hearing instrument signal obtained from the filtered input audio signal.

The present invention relates to a sound modulator for modifying the amplitude and / or phase of an ultrasonic wave comprising: a thermally-responsive material, wherein at least one acoustic property of the thermally-responsive material is variable in response to a change in temperature; a source of ultrasonic waves, which is suitable to emit an ultrasonic pulse or a continuous ultrasonic wave toward the thermally-responsive material; a tempering device, which is in thermal communication with the thermally-responsive material, to trigger a local change of at least one property of the thermally-responsive material; a control signal, which is suitable to control heating and / or cooling of the tempering device. Furthermore, the invention relates to a method for spatially and temporally modulate the amplitude and / or phase of an ultrasonic wave, comprising the steps of: transmitting an ultrasonic wave toward a thermally-responsive material of a sound modulator, wherein at least one acoustic property of the thermally-responsive material is variable in response to a change in temperature; transmitting a control signal to the sound modulator; and tempering one or a plurality of areas of the thermally-responsive material, in response to the control signal, causing a local change of an acoustic property of the thermally-responsive material.

The embodiments of the present disclosure provide an electronic apparatus and a method performed by the same, a storage medium, and a computer program product which relates to the field of artificial intelligence. The method includes: acquiring a plurality of first audio signals recorded at a plurality of different locations in a space where a sound source is located; rendering, for each first audio signal, using a transmission response corresponding to at least one other first audio signal to obtain a rendered audio signal; estimating parameters for reflecting acoustic characteristics of the space based on the plurality of rendered audio signals; and performing, based on the estimated parameters, audio rendering of a virtual audio to obtain a reverberation audio. Alternatively, the above method executed by electronic apparatus may be executed by using artificial intelligence models.