149 results about "Directional microphone" patented technology

A real-time system for the multimodal generation of virtual reality scenes based on artificial intelligence for the creation of immersive three-dimensional environments from natural language narratives, consisting of: a speech capture module configured to continuously record a user's spoken narrative via one or more directional microphones, preprocesses the captured signal by noise reduction and temporal alignment, and outputs a digital speech stream; A speech-to-text processing unit that is operationally coupled to the speech capture module and configured for real-time speech recognition using a continuous neural transformer model. The unit is trained to transcribe natural language utterances into structured text data while maintaining contextual continuity throughout the evolving narrative. a semantic interpretation processing unit that is communicatively linked to the speech recognition unit and configured to perform natural language understanding techniques to extract contextual entities, spatial references, temporal relationships, and object attributes from the transcribed narrative; the engine includes a large language model that is fine-tuned for spatial reasoning tasks; a scene graph generation module configured to transform the interpreted semantic data into a structured, hierarchical representation that defines nodes for identified entities and edges for corresponding relationships, with each node associated with metadata describing geometry, position, orientation, texture, and linking attributes between objects; a multimodal image-language model processor coupled with the scene graph generation module, wherein the processor is configured to retrieve, adapt, or synthesize appropriate three-dimensional elements from a pre-trained visual-lexical embedding space and align these elements with their semantic and spatial definitions derived from the scene graph; a scene assembly and rendering controller configured to create a cohesive virtual scene from the aligned assets, perform real-time rendering using a GPU-accelerated ray tracing pipeline, and produce a stereoscopic visual output that corresponds to the evolving narrative; A head-mounted virtual reality visualization device connected to the rendering engine and configured to display the generated immersive environment to the user in real time. The device features motion sensors and inside-out tracking cameras to detect head and body movements, dynamically updating viewing angles and perspective within the rendered scene; and a bidirectional feedback module integrated into the head-mounted device and connected to the semantic interpretation processing unit; the module is configured to interpret corrective commands, gestures, or supplementary comments from the user to refine or modify specific scene elements without interrupting the real-time visualization; The system continuously updates the virtual scene as the narrative develops, ensuring temporal synchronization between speech input and rendered output below a defined latency threshold, thus enabling a natural, dialogic construction of complex three-dimensional virtual environments.

A system for real-time analysis of emotional feedback during motivational speeches, consisting of: a series of multimodal sensors, including at least one visual sensor configured to capture facial expressions of spectators, at least one directional microphone configured to capture the audio responses of the audience, and optionally one or more physiological sensors configured to capture biometric signals from spectators; an edge-based processing unit that is communicatively coupled to the arrangement of multimodal sensors, wherein the edge-based processing unit comprises the following: (a) a feature extraction module configured to extract visual features from captured facial images, acoustic features from voice responses, and physiological features from biometric signals; (b) an emotion inference machine configured to process the features using a deep learning-based emotion recognition model comprising a convolutional neural network (CNN) for classifying facial expressions, a recurrent neural network (RNN) for classifying voice emotions, and a multimodal late fusion layer configured to compute a composite emotion state vector representing the aggregated emotions of the audience; (c) a timestamp and speech alignment module configured to correlate the calculated composite emotion state vector with segmented portions of a live motivational speech based on real-time speech-to-text transcription and semantic analysis; and (d) a session-based storage unit configured to log time-indexed emotional state vectors and corresponding speech segments for post-event analysis; A speaker feedback interface comprising a portable display or a podium-mounted visualization panel, wherein the interface is configured to display visual indicators of emotional feedback in real time, the indicators being derived from the emotional state vector and including at least emotional trend graphs, threshold alerts, or engagement indices.

The invention relates to the technical field of data processing, in particular to an automatic conference recording and abstract generating method for an intelligent conference, which comprises the following steps of: capturing voice streams of multiple speakers in real time through a directional microphone array, generating an original text stream with a time sequence mark based on real-time voiceprint clustering, and synchronously extracting intention intensity parameters in the voice streams; performing intention-driven dynamic segmentation processing on the original text stream; performing agenda-perceived abstract block generation on the segmented text, wherein decision expressions meeting a semantic density threshold value are extracted as abstract core blocks; and assembling the abstract core blocks into a structured abstract document according to the hierarchical structure of the agenda template. Compared with a traditional linear transcription mode based on voice recognition, the method has the advantages that the mapping precision between the speech content and the identity of the participant is remarkably improved, and a solid foundation is provided for semantic segmentation and decision tracing.

The invention discloses a gas pipeline leakage detection method based on an unmanned aerial vehicle, and the method comprises the following steps: S1, controlling the unmanned aerial vehicle to fly along a preset pipeline inspection path, and synchronously collecting the following data: S2, carrying out the real-time fusion processing of the data collected in S1, and generating a leakage probability score; s3, when the leakage probability score exceeds a preset threshold value, controlling the unmanned aerial vehicle to hover and marking a leakage point coordinate; and S4, generating an evaluation report containing the leakage aperture, the risk level and the environmental parameters, and encrypting and transmitting the evaluation report to the emergency terminal through the hybrid network. The unmanned aerial vehicle is adopted to carry the laser methane sensor, the multispectral thermal imager and the directional microphone array to detect whether the gas pipeline leaks or not, so that the labor intensity of workers is reduced, and meanwhile, the large-scale pipe network detection time can be shortened by adopting multi-machine cooperative task allocation, so that the detection efficiency is improved.

A concentric circular microphone array (CCMA) may include a number of omnidirectional microphones and an equal number of directional microphones, wherein the omnidirectional microphones and the directional microphones form a plurality of concentric rings on a substantially planar platform. Each of the plurality of concentric rings includes a subset of the omnidirectional microphones and a subset of the directional microphones (e.g., arranged in mixed pairs of microphones). Responsive to a sound source, the omnidirectional microphones and the directional microphones may respectively generate first and second electronic signals. A target beampattern of Nth order may be specified for the CCMA. An Nth order beamformer for the CCMA, that is steerable in a three-dimensional space including the sound source, may be determined based on the specified target beampattern. The beamformer may be executed to calculate an estimate of the sound source based on the first electronic signals and the second electronic signals.

The embodiment of the invention relates to a method and a device for target identification based on audio and laser radar point cloud. The method comprises the following steps: carrying out audio acquisition on a current environment based on a first directional microphone array to obtain a corresponding first array audio set; performing radar scanning on the current environment based on the first laser radar in the audio acquisition process to obtain a corresponding first environment point cloud; performing sound source target recognition processing according to the first array audio set to obtain a corresponding first recognition target set; performing point cloud target recognition processing according to the first environment point cloud to obtain a corresponding second recognition target set; and performing target type correction on the second recognition target set based on the first recognition target set. According to the invention, the point cloud target recognition result can be effectively corrected based on the audio target recognition result in an environment with bad weather or poor visibility.

The invention relates to the technical field of sound reception, and discloses a system for multi-directional sound reception and noise reduction of a smart glasses microphone array, and the system comprises at least five microphones which are disposed at different positions of a smart glasses frame, and the first microphone is a unidirectional microphone, is located below the glasses frame, and is used for picking up the voice of a wearer; the second microphone and the third microphone are bi-directional microphones, are located right in front of the glasses frame and are used for picking up voice signals right in front of the wearer. The directivity weight of the microphone array is adjusted in real time through the dynamic optimization module, and the pickup effect of the target voice signal is enhanced preferentially. The voice enhancement module is matched to extract a target voice signal by using a Bayesian inference method, so that noise components in a mixed signal can be effectively separated, voice distortion is reduced, the definition and fidelity of the voice signal are remarkably improved, and the voice has better intelligibility in various complex noise environments.

The invention discloses an interactive seven-stringed plucked instrument music body and mind healing system, relates to the technical field of intelligent health management, and constructs an immersive body and mind conditioning environment by integrating multi-sensory stimulation of sound, light and smell and a seven-stringed plucked instrument music healing function. The system comprises a simulated seven-stringed plucked instrument sound box, an intelligent seven-stringed plucked instrument base, a millimeter wave radar, an infrared camera, a directional microphone, an intelligent aromatherapy machine and an intelligent light module, physiological and psychological data of a user are collected in real time, and a physical and psychological state report is generated through AI model analysis. And dynamically matching guqin music tracks (such as'ghost blue 'for relieving anxiety), fragrance types (such as sandalwood for soothing nerves) and intelligent light scenes (simulating morning dawn / moonlight). A non-contact sensing technology is innovatively adopted, and a seven-stringed plucked instrument tone color characteristic and a natural light environment algorithm are combined, so that cooperative adjustment of music frequency, light color temperature and fragrance concentration is realized, the problems of single sense organ and insufficient dynamic adaptation of a traditional healing system are solved, and a personalized and non-intrusive body and mind conditioning scheme is provided for sub-health people.

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 provides a factory boundary noise online monitoring traceability system and method, and belongs to the technical field of noise monitoring. The system comprises a data acquisition module; the data processing module is used for judging whether factory boundary noise processing is carried out or not; when the boundary noise processing is determined to be carried out, determining a direction corresponding to the directional microphone with the maximum sound level as a noise source direction; based on the noise source direction, a data acquisition instruction is generated and sent to the data acquisition module, and audio frequency information and video information, collected in real time and fed back by the data acquisition module, in the noise source direction are recycled; and taking the audio frequency information and the video information in the noise source direction as input parameters, and based on a pre-established abnormal sound source positioning model and a pre-established sound feature recognition model, obtaining a noise source visual sound field cloud picture and a noise classification recognition result. The purposes of noise comprehensive response, noise orientation definition and noise accurate traceability are achieved, evidence is provided for factory boundary noise source definition, and disputes are effectively avoided.

The invention discloses a noise measuring device and a noise filtering method thereof, and belongs to the technical field of acoustic signal processing. The noise measuring device comprises an annular microphone array, directional microphones and a processor; the annular microphone array and the directional microphone are electrically connected with the processor; the annular microphone array comprises a plurality of omnidirectional microphones arranged in an annular array, one part of the omnidirectional microphones correspond to the non-interested area, and the other part of the omnidirectional microphones correspond to the interested area; the directional microphone is arranged in the center of the annular microphone array, the directional microphone has a target sound receiving direction, and the target sound receiving direction corresponds to the region of interest; by adopting a plurality of omnidirectional microphones which are annularly arranged and a directional microphone, the device can comprehensively utilize the information of a plurality of microphones, and a beam forming algorithm is combined, so that a noise source in a factory boundary is accurately positioned. The method has flexibility and adaptability, and has a wide application prospect in the aspects of noise detection, voice recognition and audio monitoring.

The utility model relates to the technical field of microphone protection devices, in particular to an omnidirectional microphone protection device which comprises a body, a connecting plate, a base, a protection shell and a connecting mechanism, the body is arranged at the top of the base, the connecting plate is arranged on the body, the protection shell is connected to the top of the body in a sleeved mode, and the connecting mechanism is arranged on the base. The protective shell is connected with the connecting plate through the connecting mechanism, the connecting mechanism comprises a long plate, a sliding groove, a sliding block, an open groove, a groove, a spring, a square block and a limiting plate, one end of the long plate is connected with the protective shell, the other end of the long plate penetrates through the connecting plate, the sliding groove is formed in the connecting plate, and the sliding block is arranged in the open groove. The long plate is provided with a sliding groove and a groove, the groove is arranged on the long plate, one end of the sliding block is connected with the sliding groove in a sliding mode, the other end of the sliding block extends into the groove, the groove is arranged on the sliding block, and the protection device of the omni-directional microphone solves the problem that an existing device is inconvenient to disassemble.

The invention provides an AI glasses automatic shooting method and system based on voice control, and relates to the technical field of intelligent wearable equipment, accurate interaction is realized through a multi-channel directional microphone array and an end-cloud collaborative voice recognition engine, the microphone array optimizes the pickup angle based on the wearing position characteristics of a user, and the user experience is improved. In combination with real-time voice activity detection, environmental noise is filtered out, it is ensured that a clear voice instruction can still be captured in a noisy environment, an end-cloud cooperation mode operates a lightweight model locally to guarantee the off-line response speed, a cloud large model is called when a network is available to improve the complex instruction analysis capability, and a composite statement containing parameter adjustment can be recognized; and an operation intention and parameters are automatically bound through a natural language processing engine, so that one-step execution of the instruction is realized.

A concentric circular microphone array (CCMA) may include a number of omnidirectional microphones and an equal number of directional microphones, wherein the omnidirectional microphones and the directional microphones form a plurality of concentric rings on a substantially planar platform. Each of the plurality of concentric rings includes a subset of the omnidirectional microphones and a subset of the directional microphones (e.g., arranged in mixed pairs of microphones). Responsive to a sound source, the omnidirectional microphones and the directional microphones may respectively generate first and second electronic signals. A target beampattern of Nth order may be specified for the CCMA. An Nth order beamformer for the CCMA, that is steerable in a three-dimensional space including the sound source, may be determined based on the specified target beampattern. The beamformer may be executed to calculate an estimate of the sound source based on the first electronic signals and the second electronic signals.

The invention discloses a machine tool state real-time intelligent monitoring and abnormity early warning method. The method comprises the following steps that a directional microphone array is used for collecting cutting voiceprint signals in the machining process of a machine tool; focusing sound in a cutting area through a sound beam forming technology, and inhibiting noise in a non-target area; performing frequency separation, de-noising enhancement and signal segmentation preprocessing on the collected voiceprint signals; extracting a time domain feature, a frequency domain feature and a time frequency feature of the voiceprint; based on process steps in a machine tool CNC program, dynamically matching the real-time voiceprint features with the reference voiceprint model; recognizing the voiceprint feature deviation condition through an anomaly detection algorithm, and triggering a multi-stage early warning signal according to the deviation degree; through the unique signal acquisition and processing technology, a dynamic matching mechanism and intelligent early warning classification, extremely high creativity is shown, the limitation of the prior art is exceeded, and the accuracy, adaptability and intelligent level of the machine tool state monitoring system are remarkably improved.

The invention discloses a wind turbine blade internal damage diagnosis method based on a sound field graph neural network, and belongs to the technical field of wind turbine blade state detection, and the method comprises the steps: deploying a microphone array in a cabin, and collecting an acoustic signal when a blade rotates; constructing a space sound field graph structure by taking the microphone as a node and the sound wave propagation path as an edge; extracting nonlinear acoustic features by using a physical constraint graph neural network PC-GNN; generating a damage embedding vector based on self-supervised training contrast learning; and outputting a damage probability thermodynamic diagram and positioning information. According to the method, a directional microphone array is deployed in a cabin, and a sound wave propagation space diagram structure is constructed; designing a physical constraint graph neural network PC-GNN, and embedding an acoustic wave equation as a regularization item; the problem of scarcity of damaged samples is solved by adopting self-supervised contrast learning; and finally outputting a positioning thermodynamic diagram of the internal damage of the blade.

The invention discloses a signal generation method and device, a readable storage medium and a computer program product, and relates to the technical field of signal processing, the signal generation method and device are applied to audio equipment, the audio equipment comprises an omnidirectional microphone and a splayed directional microphone, and the method comprises the following steps: acquiring a picked original audio signal, the original audio signal comprises a first audio signal and a second audio signal, the first audio signal is an audio signal picked up by an omnidirectional microphone, and the second audio signal is an audio signal picked up by a splayed directional microphone; sound channel separation processing is carried out on the original audio signal to obtain a dual-sound-channel signal, and the dual-sound-channel signal comprises a left sound channel signal and a right sound channel signal; sound source parameters corresponding to the sound source are calculated based on the dual-track signal, the original audio signal is rendered according to the sound source parameters, a generated target audio signal is obtained, and the sound source parameters comprise the sound source orientation and / or the sound source size. According to the invention, the universality of stereo recording is improved.

To determine whether sound from a specific site is normal in the process of inspecting a finished automobile and specify the generation source if the sound is abnormal.SOLUTION: The inspection display device includes a sound acquisition unit 1, a vibration acquisition unit 10, an abnormal sound determination unit 20, an abnormal sound vibration source estimation unit 30, a data storage unit 40, and a display unit 50. A directional microphone of the sound acquisition unit 1 and a laser doppler vibration meter of the vibration acquisition unit 10 are arranged to point to the specific site. When the acquired sound is determined to be abnormal, vibration data of a corresponding predetermined time length is acquired and the possibility that the specific site is the generation source of the abnormal sound is determined from the correlation between the amplitudes and the consistency degree of the frequencies.SELECTED DRAWING: Figure 1

The invention relates to the technical field of multi-modal data processing, and discloses an elevator advertisement audience behavior analysis system and method based on multi-modal data acquisition, and the system comprises a data acquisition layer which is used for collecting multi-modal sensing information in an elevator space, the data acquisition layer comprises a millimeter wave radar module, a low-resolution thermal imaging module, an ambient light and distance sensing module, a directional microphone array module and an elevator state interface module; and the data processing and fusion layer is connected with the data acquisition layer and used for preprocessing and dynamically fusing the multi-modal sensing information, and the data processing and fusion layer comprises a space-time alignment unit, a dynamic weight adjustment unit and a conflict processing unit. According to the method, a complete audience behavior analysis scheme for the elevator closed space is formed through non-intrusive multi-modal data acquisition, a dynamic fusion strategy adaptive to the elevator scene and fine-grained behavior modeling.

The invention relates to a method and an apparatus for improving the intelligibility of a group of persons engaged in conversation, each person in the group being able at times to be a speaker and at times to be a listener. The persons are situated at two or more positions P. For this purpose, a system (1) is used comprising: two or more directional microphones M which are directed towards the positions P; two or more directional loudspeakers L which are likewise directed towards the positions P; and a digital signal processor DSP which is capable of processing and forwarding acoustic signals in real time. Each microphone M continuously receives acoustic signals Ai and forwards them to the DSP. The DSP identifies, in real time, a speech signal Si1 from each acoustic signal Ai and generates therefrom a processed speech signal aSi1. The DSP transmits the processed speech signal aSi1 to one or more other loudspeakers L. The DSP detects later-arriving speech signals Si2, Si3 having the same characteristics as reflections of the speech signal Si1, and generates therefrom, in real time, processed reflected signals aSi2, aSi3, which the DSP transmits to another loudspeaker L within a predetermined time window (4) having a duration of at most 40 ms.

The application relates to the technical field of data processing, in particular to an audio acquisition method, an electronic device and a storage medium. The audio acquisition method of the application uses an acoustic vector sensor (AVS) array with directivity as a pickup device, which is better than an omnidirectional microphone array, to acquire voice signals in a space, wherein each AVS in the AVS array comprises an omnidirectional microphone and a directional microphone, then the weights of the voice signals acquired by the omnidirectional microphone and the directional microphone in each AVS are adjusted according to a target direction, the voice signals of each AVS after enhancement in the target direction are obtained, then a designed super-directivity beamformer is applied to the enhanced signals acquired by each AVS for further enhancement processing, and the voice signals of the entire AVS array after enhancement in the target direction are obtained.

This invention relates to the field of acoustic signal analysis technology, specifically to a sound collector and diagnostic method for SCR system fault diagnosis. It includes multiple directional microphone arrays arranged in a circular pattern on the outer ring of an elastic clamp. The elastic clamp is fitted onto the tube at the end of the jet valve. The directional microphone arrays are inserted into a mounting base, which is slidably connected to the outer ring of the elastic clamp. The mounting base has slots on its surface, and clamping blocks are inserted into the side walls of the slots. The advantages are: the use of an elastic clamp fitted onto the tube at the end of the jet valve allows the clamp to adapt to tubes of different sizes, simplifying the installation process. Furthermore, the rubber gaskets clamping the clamp between the clamp and the tube prevent loosening of the clamp and effectively reduce vibrations transmitted through the tube, improving the stability and accuracy of sound collection.

This utility model discloses an adjustable pickup angle microphone, relating to the field of microphone technology. It includes a directional microphone body and an adjustment component. The bottom of the directional microphone body has a support frame, and the adjustment component is located on one side of the support frame. The adjustment component includes a side plate, a side cover, a first motor, a worm gear, a worm wheel, and an adjustment shaft. The side cover is mounted on one side of the side plate, and the first motor is fixed to the bottom of the side cover. Through the adjustment component, when the pickup angle of the directional microphone body needs to be adjusted, the first motor drives the worm gear to rotate, which in turn drives the adjustment shaft to rotate, adjusting the pitch angle of the directional microphone body. In conjunction with a second motor, a gear and a gear ring drive the base plate to rotate, changing the orientation of the directional microphone body. This allows for precise pickup at specific angles, further expanding the device's application range.

The present application relates to the technical field of sound recognition, and particularly relates to a sound recognition system for sheep feeding behavior. The technical scheme comprises a sound collection module, a voice enhancement module, a voiceprint feature extraction module, a multi-modal classification module, a data fusion unit, the sound collection module is arranged on a wearable device on the neck of a sheep, is provided with an anti-wind-noise directional microphone array, and is used for collecting environmental sound signals in real time; the voice enhancement module is connected with the sound collection module. The present application realizes accurate recognition and analysis of sheep feeding behavior, effectively solves the problems of sound signal processing in a complex environment, individual and group monitoring, privacy protection and energy supply, not only improves the intelligent level and efficiency of pasture management, reduces labor costs, but also can timely find health problems of sheep, protect the health of the sheep, protect the privacy of sheep data, and improve the energy utilization efficiency of equipment.

The invention provides an electronic device and a pickup method, and belongs to the technical field of terminals.The electronic device comprises a directional microphone array, and the directional microphone array at least comprises a first directional microphone and a second directional microphone. The pickup enhancement direction of the first directional microphone is perpendicular to the pickup enhancement direction of the second directional microphone. The direction with the maximum sound signal gain of the directional microphone array is a target direction, and the target direction is jointly determined by the first directional microphone and the second directional microphone. According to the scheme provided by the invention, the array structure of the first directional microphone and the second directional microphone is combined with the back-end algorithm, so that pickup enhancement in a specific direction can be realized.

The invention provides an audio acquisition device and method, electronic equipment and a computer program product. The device comprises a plurality of directional microphone units which are used for respectively collecting audio signals corresponding to the directional directions of the directional microphone units; the plurality of filter units are used for eliminating the audio signals in the non-pointing direction collected by each of the plurality of pointing microphone units; wherein the plurality of directional microphone units comprise a first directional microphone unit corresponding to a first directional direction and a second directional microphone unit corresponding to a second directional direction; one of the plurality of filter units is configured to cancel the audio signal in the first directional direction acquired by the second directional microphone unit. According to the invention, by calibrating the coefficients of the plurality of filter units, the extraction of the audio signal in the specific pointing direction is realized while the far-end audio signal is suppressed, so that the acquisition effect is enhanced, and the conference quality is improved.

The utility model discloses a single-directivity condenser microphone connection structure, which belongs to the technical field of single-directivity microphones and comprises a printed circuit board and a parallel plate capacitor which are arranged at intervals. The cylinder is located between the printed circuit board and the parallel plate capacitor, one end of the cylinder abuts against the printed circuit board, and the other end of the cylinder abuts against the parallel plate capacitor; the circular ring is arranged in the cylinder body, and the outer edge of the circular ring is connected with the inner wall surface of the cylinder body to form a damping plate; according to the utility model, the damping plate can be formed by utilizing the circular ring arranged in the cylinder body, so that the structure at the damping plate is simpler, the material cost can be reduced, the assembly is simpler and more convenient, the assembly efficiency is improved, and automatic mass production is convenient to realize; furthermore, when the electrical performance parameters of the microphone need to be adjusted, only the position of the circular ring in the barrel needs to be adjusted or the barrel and the circular ring are replaced as a whole, and the adjustment mode of the electrical performance parameters is simpler and more convenient.

The embodiment of the utility model discloses audio and video recording equipment, the audio and video recording equipment comprises a base assembly and a holder assembly, the base assembly comprises a base body, a panoramic camera and an omnidirectional microphone, and the panoramic camera and the omnidirectional microphone are arranged on the base body at an interval; the holder assembly comprises a shell, a zoom camera and a plurality of directional microphones; the zoom camera and the directional microphones are both connected with the shell, the shell is provided with a daylighting opening and a plurality of pickup holes, the pickup holes are annularly formed in the peripheral side of the daylighting opening, the directional microphones correspond to the pickup holes one to one, the zoom camera can shoot through the daylighting opening, the directional microphones correspond to the pickup holes, the shell is connected with the base body, and the base body is connected with the shell. The zoom camera and the directional microphone are arranged on the base body and can move relative to the base body, so that the zoom camera and the directional microphone are aligned with a spokesman to carry out audio and video recording, the pickup effect of the directional microphone can be further optimized, and the pickup definition of the spokesman is improved.

The application discloses a kind of sports dance training rhythm synchronism quantitative analysis and feedback method, it is related to sports dance training data analysis technical field, comprising the following steps: training data multimodal acquisition, 12 camera 16 key joint data are collected, 48kHz directional microphone audio is collected, wrist heart rate instrument is collected heart rate;Preprocessing, action data filter noise completion is removed exception, audio is extracted feature and is removed noise, heart rate is smoothed, unified time sequence;Rhythm characteristic bidirectional extraction, audio is extracted beat cycle and stress, action is extracted force and action cycle;Synchronism quantization, calculate time difference, sequence consistency, identify synchronism decline section;Analysis result, grade, mark weak link, calculate progress amplitude;Generation feedback, AR and voice prompt, adjust plan and push APP.This method solves the problem of traditional training synchronism subjective fuzzy;Real-time multi-sensory feedback helps immediate adjustment, individualized report and plan fit demand.

A robot includes: a light detection and ranging (LiDAR) sensor; a plurality of directional microphones; and at least one processor configured to identify an object in the vicinity of the robot based on sensing data obtained by the LiDAR sensor, identify a weight applied to an audio signal received through a directional microphone corresponding to a position of the object among the plurality of directional microphones based on a type of the object, obtaining contextual information of the robot based on the sensed data, identifying a preprocessing model corresponding to each of the plurality of directional microphones based on the contextual information, applying a weight to an audio signal received by a directional microphone corresponding to the position of the object among a plurality of audio signals received by the plurality of directional microphones, a plurality of preprocessed audio signals are obtained by inputting the audio signal to which the weight has been applied and the remaining audio signals into preprocessing models corresponding to the respective directional microphones, and speech recognition is performed based on the plurality of preprocessed audio signals.