139 results about "Sound source location" patented technology

The invention discloses a conference sound reinforcement system based on an AI intelligent algorithm and 360-degree omni-directional noise reduction, and relates to the technical field of audio signal processing, the conference sound reinforcement system comprises a conference management center, the conference management center is in communication connection with the following modules: a multi-sound-source sensing module used for capturing 360-degree omni-directional sound field information in a conference environment and constructing a sound source space distribution model; according to the invention, the omnidirectional microphone array unit covers all directions of a conference space, synchronously collects audio data streams, eliminates the limitation of a conventional unidirectional microphone, combines the sound source positioning space mapping unit, constructs a sound source space distribution model based on the time difference of arrival and the phase difference through a deep learning model, and improves the sound source positioning accuracy. The azimuth angle, pitch angle and distance parameters of the sound source are accurately analyzed, the position of the sound source is mapped to a virtual space coordinate system, a dynamically updated 3D sound source distribution diagram is generated, the position change and intensity distribution of the sound source are reflected in real time, and the positioning precision in a complex acoustic environment is remarkably improved.

The invention relates to a digital conference voice processing method, system and device and a storage medium, and the method comprises the following steps: carrying out the pickup of a conference voice, obtaining a mixed voice signal, carrying out the framing sampling, and forming a voice sampling sequence; carrying out sound source direction estimation based on the sequence to obtain multi-sound-source position information, and carrying out beam forming and spatial filtering on the voice sampling sequence according to the multi-sound-source position information to obtain a sound source separation signal; performing voice segment segmentation on the signal to obtain a voice segment sequence, extracting voiceprint features of each segment, and generating a speaker feature mark; and performing time sequence recombination on the voice fragment sequence by using the mark, constructing a speaking time sequence table, selectively outputting the voice fragment sequence according to the table, and finally generating clear and ordered conference voice. The technical problems that due to the fact that a traditional voice processing method lacks effective space-voiceprint joint constraint, voice separation is not thorough, identities of speakers are confused, and the speaking time sequence is disordered are solved.

The invention relates to the field of microphone sound reception, in particular to a precise sound reception system and method based on double microphone arrays and 3D visual identification, and the system comprises a sound source positioning module, a visual identification module, an audio processing module, a sound reception control module and a precise tracking module. The visual identification module is used for calculating the listening angle of the microphone array, the audio processing module is used for obtaining the sound arrival time difference of each microphone, the sound reception control module is used for outputting the position of a target sound source, and the precise tracking module is used for enabling the camera to track the target sound source. The method is advantaged in that requirements of real-time interaction scenes are satisfied, sound reception capability of the microphone array in complex scenes is improved, good robustness and accuracy are realized, sound source positioning precision and dynamic tracking capability are improved, and environment adaptability and real-time performance of microphone sound reception are improved.

The invention discloses a multi-modal speech enhancement method and device based on a deep learning model, and relates to the technical field of artificial intelligence, and the method comprises the steps: obtaining the head posture data, binaural audio signals and visual context information of a user in a virtual reality environment, coding the binaural audio signals into three-dimensional space acoustic features, and carrying out the coding of the three-dimensional space acoustic features; and extracting virtual sound source position features and lip motion features from the visual context information, inputting the features into an immersive fusion enhancement network, selectively enhancing or inhibiting acoustic features from different spatial directions, generating an enhanced audio stream, and outputting the enhanced audio stream through a binaural rendering engine. According to the method, the technical problems that in the prior art, due to the fact that multi-modal prior information cannot be effectively fused, voice enhancement lacks spatial selectivity, and an interference sound source irrelevant to vision is difficult to restrain are solved, and head posture dynamic attention and lip motion cross-modal constraint are fused; the technical effects of natural binding of auditory attention and a visual focus and effective suppression of an interference sound source are achieved.

A sound source positioning device based on multi-microphone and time difference algorithm, the device comprises a sound signal acquisition module, a signal processing module, a positioning and communication module, a display module and a power module; the sound signal acquisition module adopts multiple output analog signal microphones to realize omnidirectional acquisition of sound signals; the signal processing module takes a single-chip microcomputer as the core and integrates analog signal amplification, filtering, peak detection, analog-to-digital conversion and time difference algorithm processing functions; the positioning and communication module combines a Beidou positioning module and a mobile communication module to realize sound source position determination and remote data transmission; the power module stably supplies power to each module in multiple gears. The present application combines multi-microphone cooperative acquisition with the time difference algorithm to improve the accuracy and stability of local single-point sound source positioning and can be applied to security monitoring, environmental monitoring, industrial inspection and other scenes.

The invention provides a sound wave propagation spatial distribution and sound source position control method, and relates to the technical field of loudspeaker control, and the method comprises the steps: building an analysis control model based on a sound source and a propagation auxiliary object; constructing a first area of sound wave propagation of the sound source in the analysis control model, and marking a second area of spatial distribution of sound wave propagation of the current sound source; mapping the target position to an analysis control model to generate a mapping point location; when the mapping point is in the second area or the mapping point is not in any first area, the sound source position is not controlled; and when the mapping point location is not in the second area but is in any first area, determining a target sound source in the first area, and determining a sound source position control parameter based on the control library correspondingly associated with the target sound source and the mapping point location. According to the sound wave propagation spatial distribution and sound source position control method, sound delivery is carried out according to the position of a film viewer, and the viewing experience of the film viewer is guaranteed.

The application discloses a near-field strong interference suppression method based on spherical wave deconvolution beam forming positioning, which comprises the following steps: performing fine grid division on a near-field scanning area of a linear array; performing spherical wave focusing beam forming on the near-field scanning area and two-dimensional deconvolution beam forming about distance-angle under the condition of unknown sound source position; calculating the distribution entropy characteristics of the deconvolution beam forming result in the angle dimension, and using a preset threshold to correlate the interference positioning of a target; constructing an array flow pattern matrix A of each near-field target l and generating a covariance matrix A; calculating the orthogonal complement space projection matrix of the covariance matrix A, and applying the orthogonal complement space projection matrix to the array element domain covariance matrix C to perform near-field interference suppression; using a far-field plane wave model to perform deconvolution beam forming calculation on the covariance matrix for suppressing near-field interference, so as to obtain passive detection results of the near-field strong interference suppression. The application can realize near-field interference positioning and interference suppression under the condition that the interference position is unknown.

To provide an inspection device capable of locating a position of an abnormal sound source even if an abnormal sound of a rotating machine has periodicity.SOLUTION: An inspection device includes: multiple microphones detecting sound generated from each of multiple rotating machines; a sound waveform processing unit obtaining a feature amount of a sound waveform detected by the multiple microphones; an abnormality detection unit detecting an abnormal sound on the basis of the feature amount; a waveform time difference calculation unit calculating the minimum time difference of sound waveforms among the multiple microphones; and an abnormal sound source position locating unit locating a rotating machine that generates an abnormal sound from among the multiple rotating machines on the basis of the minimum time difference.SELECTED DRAWING: Figure 1

This application relates to a conference screen processing method, device, storage medium, and system. It involves acquiring multiple audio data streams using a data acquisition device, determining the sound source location information based on the first audio data stream where the human voice energy exceeds a first threshold, acquiring the image corresponding to the sound source location information, identifying the speaker in the image whose human voice energy meets preset conditions, and obtaining the facial deflection angle between a preset position representing the speaker's face and the data acquisition device. The speaker's image and facial deflection angle are then sent to a data processing device. The data processing device can display the image of the speaker with the smallest facial deflection angle using a display device. This eliminates the need for the data processing device to expend significant computing resources for image recognition, reducing its computational resource consumption and improving the efficiency of displaying the speaker's face at the conference.

The present application relates to a kind of multi-sensor wideband sound source positioning method suitable for deep-sea reliable sound path environment, utilize the same sound source D-SR time delay measurement value observed by deep-sea large depth multi-sensor to carry out three-dimensional positioning to sound source.In the first stage, based on the position of each sensor, depth, observed D-SR time delay and water body sound velocity profile information, a matrix and a vector are constructed, there is a linear relationship between the two and the position of sound source, the least square method is used to solve this linear relationship, obtain the estimation result of the two-dimensional position of sound source, based on two-dimensional position estimation result, the depth estimation result of sound source is obtained using depth solution formula;In the second stage, based on the three-dimensional position of sound source estimated in the first stage, the D-SR time delay value of the position of each sensor is predicted, then an auxiliary variable matrix is constructed, and new sound source two-dimensional horizontal position estimation result is obtained according to the rule of auxiliary variable estimator (IVE), and finally the final sound source depth estimation result is obtained according to depth solution formula.

The invention discloses a steel plate slitting cutting parameter correction method and system, a medium and a program product, and relates to the field of intelligent monitoring devices.The method comprises the steps that initial cutting parameters and a quality judgment benchmark are obtained, and a slitting unit is started; acquiring real-time sound data of a slitting unit and real-time image data of the edge of the steel strip; identifying an abnormal sound event, and determining a sound source position as an abnormal cutter position; extracting a target edge image corresponding to the abnormal tool position from the real-time image data, determining a quality defect of the abnormal tool position, and generating a defect confidence coefficient corresponding to the quality defect; when the confidence coefficient of the quality defect event exceeds a preset confidence threshold value, generating a parameter adjustment instruction according to the sound feature of the abnormal sound event and the geometric defect type of the target edge image; and correcting the cutting process parameters of the slitting unit based on the parameter adjustment instruction. By implementing the method, the adjustment response time of the production line to the abnormal cutting state can be shortened.

Methods and vehicle with sound visualization are provided. A vehicle includes a graphic projection display; exterior microphones mounted to the vehicle; and a processing device programmed to receive audio signals from the exterior microphones; retrieve map data in a region around the vehicle; identify from the audio signals a sound of interest; identify a location of a source of the sound of interest; determine a stationary status or a moving status of the source; when the source has the moving status, determining a direction and speed of movement of the source; ascertain from the map data and from the location, stationary status, moving status, direction, and / or speed, a preferred driving maneuver; determine a graphic exemplifying the preferred driving maneuver; and display the graphic upon the graphic projection display.

The present disclosure relates to a method and device for acoustically imaging fan noise, an electric vehicle charging pile, and a storage medium. The method comprises: acquiring a noise time-domain signal generated by a fan when the fan is running; performing time-frequency conversion on the noise time-domain signal based on a short-time Fourier transform algorithm to obtain a noise frequency-domain signal; inputting the noise frequency-domain signal into a delay-and-sum beamformer to perform beamforming to obtain beam data; positioning principal component data of the beam data to obtain sound power data; and generating a sound image containing a sound source position according to the sound power data. In this embodiment, the resolution of the noise can be improved by positioning the principal component of the beam data, which achieves the effect of accurately positioning the sound source position and is conducive to obtaining a more accurate sound image.

According to the room acoustic parameter determination method and device, the electronic equipment, the wearable equipment and the storage medium provided by the invention, the reverberation time in the acoustic scene is determined, and the acoustic scene comprises the listener position, the sound source position and the room property; the reflectivity of each reflecting surface in a room with room properties is determined through random light tracing and reverberation time, the random light tracing is used for determining a light propagation path reaching a listener, and the light propagation path reaching the listener represents a propagation path of light emitted from a sound source position of the room and received by the listener at the listener position; the descent curve of the reverberation energy can be determined through the reverberation time, and the random light tracing result is corrected based on the descent curve of the reverberation energy, so that the reflectivity of each reflecting surface in a room is corrected, and a more real spatial audio signal can be generated based on the estimated reflectivity of each reflecting surface.

The present application relates to a kind of all-weather power grid equipment abnormal sound monitoring device and its sound source positioning method, belong to the technical field of power equipment online monitoring and fault diagnosis.The device includes: multiple microphone units, each microphone unit includes microphone body and the e-PTFE sound-permeable film covered on its surface, for efficiently permeating sound wave while blocking wind and rain erosion;Multiple three-legged stand poles with water-blocking umbrella covers, at least one microphone unit is arranged on each stand pole, and is distributed around the power grid equipment to be monitored;Data processing unit;And solar power supply module and protection cabin.The method includes: setting loudspeaker array at the suspected abnormal sound source position, using exponential sine sweep method to measure the frequency response function matrix of loudspeaker to microphone unit;After removing the loudspeaker, based on the measured frequency response function matrix and the sound pressure signal collected in real time, the abnormal sound source signal is extracted and the path contribution is calculated using the transfer path analysis method based on regularization.

The application discloses a wind direction control method and an air conditioner. The method comprises the following steps: obtaining at least two control commands from different sound source positions, wherein each control command comprises control information, and the control information is used for indicating whether the wind blows towards the sound source position; determining the swing angle value of the air deflector of the air conditioner at each sound source position according to the control information in each control command; determining the target swing angle value corresponding to the at least two control commands according to the swing angle value of the air deflector at each sound source position; and performing the wind direction control operation according to the target swing angle value. The technical scheme provided by the application aims to solve the problem of low wind direction control accuracy in the prior art.

A rendering method and a related device are disclosed, and may be applied to a scenario such as production of music or film and television works, or the like. The method may be performed by a rendering device, or may be performed by a component (for example, a processor, a chip, a chip system, or the like) of a rendering device. The method includes: obtaining a first single-object audio track based on a multimedia file, where the first single-object audio track corresponds to a first sound object; determining a first sound source position of the first sound object based on reference information; and performing spatial rendering on the first single-object audio track based on the first sound source position, to obtain a rendered first single-object audio track.

System and method for projecting audio signals to an operator of a teleoperational surgical system to convey spatial orientation associated with the audio signals to the operator. Characteristics of the audio signals such as direction and volume may be selected to give the impression that the remote operator is positioned locally next to the patient. Characteristics of the audio signals may also be modified to provide spatial translations between actual locations of audio sources within a physical audio environment and simulations locations of the audio sources with a virtual audio environment.

The invention discloses a multi-node cooperative sound source positioning system and method for pipeline state monitoring, relates to the field of pipeline state monitoring, and introduces a BDS (Beidou satellite navigation system) second pulse signal as a global synchronization reference to realize nanosecond synchronous acquisition and packaging of multi-node acoustic data. On the basis, abnormal events are screened and converged on the edge side based on a signal amplitude threshold value. And furthermore, the same abnormal event data captured by multiple nodes is utilized, and high-precision positioning of the sound source is realized through a TDOA (Time Difference Of Arrival) algorithm in combination with linear solution of a nonlinear hyperbolic positioning equation set. And finally, according to a positioning result, reversely extracting a source signal section and performing time-frequency characteristic analysis, so that not only is the position of a sound source locked, but also specific classification of pipeline states (such as leakage and damage) is completed, and thus all-weather and high-precision intelligent pipeline monitoring is achieved.

The application provides a voice matching method and device, equipment and storage medium, and the method comprises the following steps: obtaining target voice data and the sound source position coordinates of the target voice data; calculating the target distance between the sound source position and the voice receiving position according to the sound source position coordinates; determining the voice matching range in the preset voice database according to the target distance, wherein the voice data contained in the voice matching range forms a first voice data set, the voice data contained in the voice database forms a second voice data set, and the first voice data set is a subset of the second voice data set; and performing voice matching processing on the target voice data and the voice data in the first voice data set to obtain the voice matching result corresponding to the target voice data. The target distance between the sound source position and the voice receiving position is calculated, and the target distance is used as a reference basis for voice matching, which can greatly improve the accuracy of voice matching.

This invention proposes a multi-source localization method combining adaptive spatial grid refinement and sparse reconstruction. It is a parameterized localization method based on physical sound propagation models and array signal processing theory. The method uses a microphone array to collect sound signals, constructs an array guidance model, and combines sparse reconstruction and continuous parameter optimization to estimate the sound source location. The steps include: Step 1: Array signal acquisition and preprocessing; Step 2: Constructing a coarse-resolution spatial grid and guidance dictionary; Step 3: Selecting candidate sound source locations using a sparse algorithm, and performing coarse multi-source localization based on sparse reconstruction; Step 4: Adaptively refining the local spatial grid; Step 5: Performing multi-parameter joint optimization in continuous space through nonlinear least-squares optimization of position and amplitude for fine localization; Step 6: Outputting the multi-source localization results. This invention is applicable to sound source spatial location estimation under planar microphone array conditions.

We provide the system. [Solution] A means of filtering noise, A means of amplifying a specific sound, A means of analyzing audio data to report on the situation of disaster victims, A means of synchronizing multiple devices to determine the location of the sound source, A system that includes this.

To provide a sound source position estimate method capable of estimating the sound source position of a call of a prey bird for identifying the nesting point of the prey bird.SOLUTION: The sound source position estimation method is a method for estimating the sound source position of a call of a prey bird using a computer. The sound source position estimate method includes at least an estimate step (S2) of estimating the sound source position by using the sound pressure level attenuation corresponding to the distance between the sound acquisition position and the sound source position based on the sound pressure level of the call included in the audio data acquired at each of the multiple audio acquisition points.SELECTED DRAWING: Figure 3

To provide a technique for matching a display position of an object with a sound source position for a viewer.SOLUTION: A broadcast receiving device includes a receiver that receives a broadcast wave including a video signal, an audio signal, and metadata related to position information of a sound source of the audio signal, a display screen that displays video corresponding to the video signal, an audio output unit that outputs audio corresponding to the audio signal, a detection device that detects a viewing position of a viewer who views the display screen, and a correction circuit that positions the detected viewing position, a display position of an object corresponding to the sound source on the display screen, and a sound source position based on the position information on a straight line.SELECTED DRAWING: Figure 69

The present application relates to a kind of sound leakage detection method, system, equipment and storage medium of secret meeting room, its method includes obtaining image data and the visual angle range data corresponding to image data;According to image data, corresponding sound data is called, sound data includes sound spectrum data and sound imaging data, sound spectrum data is the sound spectrum diagram of sound, sound imaging data can reflect the image of sound source distribution position;Based on the matching rule of pre-established, according to visual angle range data and sound imaging data, the sound source position in visual angle range data is matched, and target sound source is marked in sound imaging data;According to the preset image analysis rule and sound imaging data, adjust sound spectrum data, obtain target sound spectrum data;Based on the preset sound spectrum transformation model, target sound spectrum data is inversely transformed, and the target audio data corresponding to image data is obtained.The present application realizes the effect that the sound condition in the limited range outside secret meeting room is monitored.

This disclosure provides a display device and its audio acquisition and transmission method and related equipment. The display device includes a plurality of first acquisition units disposed on a first side of the display device. The method includes: acquiring a plurality of first audio data acquired by the plurality of first acquisition units; determining whether the location of a sound source is within a preset location range based on the plurality of first audio data; and in response to determining that the location of the sound source is within the preset location range, determining target audio data based on the plurality of first audio data and sending it to a target device.

The present application belongs to the technical field of voice interaction, and particularly relates to a human-machine voice precise interaction method based on a large model. In view of the problems of voice source confusion, low instruction recognition rate and insufficient interaction safety caused by the complex acoustic environment in the vehicle cabin, the present application fuses reverberation features and harmonic attenuation features to construct acoustic fingerprints, and realizes high-precision sound source positioning in combination with seat occupancy state verification; performs priority filtering and semantic pre-alignment on non-main driver voice, and eliminates fragmented interference; introduces a large language model to standardize the mapping of structured text into instructions conforming to syntax, format and semantic specifications, and embeds a dynamic permission verification mechanism based on sound source position. The scheme significantly improves the voice instruction recognition accuracy and system robustness, effectively suppresses false triggering, enhances user privacy protection, provides an efficient and precise voice interaction experience for the intelligent cabin, and has outstanding technical practical value and industrial application prospects.

This invention discloses an outer frame for a cooling fan used for acoustic modal modulation noise reduction, comprising an outer frame body (1) and an air intake shroud (2); the outer frame body (1) includes a casing (4), an outer frame support (5), and a motor support (6); the casing (4), the outer frame support (5), and the motor support (6) are mounted on the outer frame body (1); the air intake shroud (2) is a parameterized shaped pipe section controlled by a sine or cosine function, fixedly connected to and adapted to the outer frame body (1); the axial length L, the average radius R of the lip profile, and the lobe amplitude A of the air intake shroud (2) are determined by the size of the cooling fan rotor blades and the location of the sound source; the combination of the number of lobes k of the lip profile of the air intake shroud (2) is determined by the number of main acoustic modes of the cooling fan; and the initial phase φ of the lip profile of the air intake shroud (2) is determined by the number of main acoustic modes of the cooling fan. i Determined by the phase of the primary acoustic modes. This invention can modulate different circumferential acoustic modes generated by the dynamic and static interference of any cooling fan and significantly suppress noise.

The invention relates to the technical field of electric power, and discloses a partial discharge detection method and device, a medium and a product. The method comprises the following steps: acquiring an initial sound source position, adjusting a detection direction of a microphone array according to the initial sound source position, acquiring pulse density according to a partial discharge signal, and acquiring current sampling frequency according to the pulse density; carrying out discharge pulse sampling according to the current sampling frequency, obtaining each target discharge pulse and a corresponding power frequency phase angle, and generating a phase analysis partial discharge map according to each target discharge pulse and the corresponding power frequency phase angle; and through a pre-trained feature extraction model and a partial discharge type identification model, analyzing the partial discharge map according to the phase, and obtaining the partial discharge type. Partial discharge can be efficiently and accurately detected and positioned, automatic identification of partial discharge types can be realized, and detection precision and reliability in a complex environment can be ensured.

A method for delivering audible feedback to a driver within a cabin of a road vehicle, the vehicle comprising a plurality of wheels and an electric motor, the method comprising determining the torque delivered to at least two wheels of the plurality of wheels; determining a wheel torque distribution for the vehicle based on the determined torques; determining a primary sound location in the sound space in accordance with the wheel torque distribution; and inputting the primary sound location to a spatial audio system, the spatial audio system providing the audible feedback to the vehicle cabin such that the audible feedback is perceived by the driver to be originating from a virtual audio source at the primary sound location.