75 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 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 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.

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 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 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 conference terminal, an echo cancellation method and apparatus, and a sound pickup device are provided. The conference terminal comprises a loudspeaker and at least one omni-directional microphone group. The omni-directional microphone group comprises at least two omni-directional microphones. According to the conference terminal, a weight vector of a beam former enabling the at least two omni-directional microphones to form a dipole beam mode is determined, so that an echo signal in the direction of the loudspeaker is suppressed, and a sound signal in a target direction is enhanced. The sound signal is collected by means of the omni-directional microphones. For the at least two omnidirectional microphones, the weighted sum of at least two sound signals is determined according to the weight vector as an echo cancellation signal.

The application discloses a sports action intelligent teaching system and method based on a multimodal large model, and relates to the technical field of intelligent teaching.The method is characterized in that a directional microphone and a high-speed camera are arranged in a badminton training ground, and the method combines band-pass filtering, frame processing and short-time energy calculation of acoustic signals, and target detection and optical flow analysis of visual images, so that the method can obtain a hitting window energy Ehit and a background noise energy Ebase, and further construct an acoustic energy ratio index Rste.The method guarantees high signal-to-noise ratio extraction of the hitting acoustic signal in a complex environment, and can accurately locate a visual contact time Tvis in a visual mode.Compared with the existing teaching method which only relies on visual detection, the method still has high robustness under the conditions of illumination change and field noise interference, and significantly improves the accuracy of hitting action data acquisition and analysis.

To provide a method and an apparatus for detecting the position of a wasp with high accuracy.SOLUTION: A method for detecting a wasp includes collecting an environmental sound in front of a directional microphone by using the directional microphone, performing a frequency analysis on the environmental sound to identify, from the environmental sound, a magnitude of a sound of a fundamental frequency derived from a movement of a wing of a wasp, magnitudes of harmonics of the fundamental frequency, and a high-frequency noise, and determining whether each of the sound of the fundamental frequency, the sounds of the harmonics, and the high-frequency noise is greater than a predetermined threshold. The hornet detection device includes a sound collection means, a determination means, and an output means.SELECTED DRAWING: Figure 11

A conference speech presentation system, a sound localization method and apparatus, a conference system and a pickup device. The method includes the following steps: collecting (S101) a multi-channel voice signal through a directional microphone array; determining (S103) a steering vector including phase information and amplitude information according to array shape information and microphone pointing direction information; determining (S105) sound direction information according to the steering vector and the voice signal. By adopting this processing mode, both the phase information and the amplitude information are considered when determining the steering vector, which can effectively improve the accuracy of sound localization.

The application provides a directional microphone, which comprises a device substrate, a MEMS microphone sensor assembly, an intermediate plate, a top substrate and a MEMS comb mechanism; the MEMS microphone sensor assembly is fixed in the middle area of the top surface of the device substrate, the intermediate plate is covered on the device substrate and has an inner hole surrounding the MEMS microphone sensor assembly; the top substrate is covered on the intermediate plate and has a horizontal channel on the top surface and a lower end opening on the bottom surface, the horizontal channel is communicated with the inner hole through the lower end opening; the MEMS comb mechanism comprises a movable comb on the top surface of the device substrate and capable of controlled horizontal movement, the movable comb has an opening structure and a shielding structure; during the movement of the movable comb, the opening structure is communicated with the horizontal channel at different positions respectively. The directional microphone has good versatility based on the adjustable directional design of the microphone sensor assembly.

This invention discloses a subject data acquisition system based on multi-mode interaction. The system includes an edge computing gateway, and a flexible piezoelectric thin film array, a millimeter-wave radar module, and a directional microphone array, all connected to the edge computing gateway. The edge computing gateway is configured to control the millimeter-wave radar module to operate at a low duty cycle and extract the compression profile in basic scanning mode. When the piezoelectric signal envelope variance exceeds the limit, it triggers entry into a directional diagnostic mode, calculating the geometric centroid coordinates and extracting the main peak of ventricular ejection. Based on these geometric centroid coordinates, the beamforming weight vector of the directional microphone array is updated to extract the audio signal sequence. The cross-correlation function between the main peak and the Doppler phase signal trough of the chest wall displacement is calculated to obtain the time delay parameter. After sliding window compensation alignment and feature vector extraction, the results are input into the model to output the state confidence assessment result. This application can achieve high signal-to-noise ratio and low false alarm rate for accurate spatiotemporal alignment of multi-source heterogeneous physiological signals and imperceptible monitoring of abnormal states.

A vehicle system uses OEM sensors and OEM speakers to spatially convey, inside the cabin, the location and distance of real-world entities relative to a driver. A processor determines a directional bearing and, when available, a distance from proximity sensors, integrated cameras, or a directional microphone. An audio generation module produces a warning that is routed only to the speakers on the side that corresponds with the bearing, with audio gain set as a function of distance so closer entities sound louder. For emergency vehicles, the system synthesizes a siren or isolates and re-broadcasts the original siren, then plays the siren only through the speakers that correspond with the detected bearing. An interior microphone gates synthesis when the original siren is already audible. The approach enables drivers to localize pedestrians and other hazards by sound.

The present application discloses a signal generation method, a device, a readable storage medium, and a computer program product. The present application relates to the technical field of signal processing, and is applied to an audio device. The audio device comprises an omnidirectional microphone and a figure-8 directional microphone. The method comprises: acquiring a picked-up original audio signal, wherein 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 the omnidirectional microphone, and the second audio signal is an audio signal picked up by the figure-8 directional microphone; performing channel separation processing on the original audio signal to obtain a two-channel signal, wherein the two-channel signal comprises a left channel signal and a right channel signal; and on the basis of the two-channel signal, calculating a sound source parameter corresponding to a sound source, and rendering the original audio signal on the basis of the sound source parameter, so as to obtain a generated target audio signal, wherein the sound source parameter comprises a sound source azimuth and / or a sound source size. The present application improves the universality of stereo recording.

The invention relates to the field of air-guided hearing aids, in particular to a directional sound transmission method and system of an air-guided hearing aid, comprising a hearing aid module, an angle adjustment module, a direction adjustment module, a telescopic module and a key module, the hearing aid module comprises a hearing aid assembly, a broadcast assembly and an external annular airbag; the angle adjusting module comprises an angle adjusting assembly and an angle fixing assembly; the hearing aid is started to make a sound to check whether the hearing aid operates normally or not, and then the mode switching key adjusts the system of the control key for switching, so that the angle and the direction of the broadcasting component are respectively switched, and the directions of the ear canals of the broadcasting component are fit; then the storage assembly is controlled to store the broadcasting assembly so as to fix the angle and the direction of the broadcasting assembly, and finally the external annular air bag is started to be expanded to be conical, so that the situation that the broadcasting assembly is pushed outwards due to the shape problem to affect use is avoided while the broadcasting assembly is fixed again.

The embodiment of the invention discloses a multi-party visiting device and system based on VR panoramic visiting, and relates to the technical field of medical visiting. The multi-party visiting device comprises a ward end mobile visiting vehicle, and the visiting vehicle is provided with a panoramic camera which is used for collecting a panoramic video of a ward in real time and supporting remote control of panoramic view angle rotation; the voice transmission module is integrated with a directional microphone and a noise reduction module and is used for realizing multi-party talkback of the patient, the visitor and the medical personnel; the video communication module is used for realizing multi-party remote video communication of patients, visitors and medical staff; the information storage module is used for automatically storing visiting information; the handheld device is used for providing visiting appointment service and remote video service for a visitor; and the visiting end VR equipment is used for providing VR video service for the visitor. According to the embodiment of the invention, remote visiting is realized by using the VR technology, and real visiting experience is provided.

This application discloses a multimodal guided emergency supplies location and medical order recording system, relating to the field of medical equipment technology. The system includes: a central processing unit, a mobile interactive terminal, and a visually guided medicine cabinet. The mobile interactive terminal has a built-in directional microphone array, used to collect on-site voice through the directional microphone array and transmit the voice to the central processing unit. The central processing unit issues LED control commands to the visually guided medicine cabinet based on the on-site voice and records the medical order information identified from the on-site voice. The visually guided medicine cabinet includes multiple partitions, each corresponding to an independently addressable LED indicator. The visually guided medicine cabinet has a built-in medicine cabinet controller, used to control the corresponding LED indicator to light up according to the LED control commands. This application improves the accuracy of emergency supplies guidance and the efficiency of medical order recording.

The invention discloses a directional fan-shaped range pickup method and system based on a microphone linear array, a terminal and a medium, and the method comprises the steps: building the microphone linear array which comprises at least one directional microphone and a plurality of omnidirectional microphones; determining a sound source angle and a distance between a sound source and the center of the omni-directional microphone based on the phase difference between the omni-directional microphones, and determining a fan-shaped pickup area based on the sound source angle and the distance to realize fan-shaped range pickup; and determining a position relationship between the sound source and the microphone linear array based on an amplitude difference between the directional microphone and any omnidirectional microphone, and obtaining a directional fan-shaped pickup area based on the position relationship and the fan-shaped pickup area, thereby realizing directional fan-shaped range pickup. The position relation means that the sound source is located in front of or behind the microphone linear array. According to the invention, the pickup distance and angle can be accurately controlled, and the directional fan-shaped range area pickup can be effectively solved.

The invention discloses a linear array differential beam forming method and system based on non-uniform directional microphone orientation optimization, a storage medium and electronic equipment. The method comprises the following steps: constructing a linear microphone array of a directional microphone combination of an omnidirectional directional microphone and a directional microphone combination with a non-uniform microphone orientation angle; calculating a steering vector of the microphone array based on the directivity and the position of each microphone in the microphone array; a linear equation set is obtained based on the steering vector, the undistorted constraint condition and the zero point constraint condition; optimizing a microphone orientation angle of the pointing microphone by adopting a grid search strategy; and constructing an optimization problem based on a linear equation set to obtain a weight vector in an actual wave beam. Compared with a traditional design, higher white noise gain is achieved, deflection of beams in any direction is supported, and excellent performance is kept at all deflection angles; compared with a zero point constraint method, the method has the advantage that the aspects of WNG-error equalization, beam approximation, frequency consistency and the like are remarkably improved.

The invention discloses an intelligent sports action teaching system and method based on a multi-modal large model, and relates to the technical field of intelligent teaching. According to the method, directional microphones and high-speed cameras are arranged in a badminton training field, and band-pass filtering, framing processing and short-time energy calculation of acoustic signals are combined; according to the method, the energy Ehit of the ball hitting window and the energy Ebase of the background noise can be obtained, and an acoustic energy ratio index Rst is further constructed. According to the mode, high signal-to-noise ratio extraction of the ball hitting acoustic signal in a complex environment is guaranteed, and the visual contact moment Tvis can be accurately positioned in a visual mode. Compared with an existing teaching method which only depends on visual detection, the method can still keep high robustness under the conditions of illumination variation, site noise interference and the like, and the accuracy of ball hitting action data collection and analysis is remarkably improved.

The invention provides an intelligent smoke heat training monitoring system and method based on multi-modal fusion, and the system comprises a multispectral detection module which is used for outputting infrared, visible light and fusion images in real time, and can penetrate smoke to recognize a fire source and a personnel contour; the position sensing module is used for outputting personnel position coordinates in real time; the high and low temperature monitoring module is used for monitoring the temperature of a key area and superposing data to a video picture; the voice acquisition module comprises a directional microphone array supporting noise reduction and sound source positioning, and is used for acquiring the voice of the trainee and the environment sound; the communication module is used for transmitting audio, video and temperature data to a command center server; and the main control module is used for coordinating and managing the work of each module. According to the system, the safety, scientificity and actual combat level of fire-fighting training are greatly improved, key technical support is provided for danger early warning and tactical decision, and meanwhile, the deployment and operation and maintenance cost of the system is remarkably reduced.

Headphones comprising: a headband having first and second opposing ends; a first earpiece coupled to the first end of the headband and comprising a first audio driver; a second earpiece coupled to the second end of the headband and comprising a second audio driver, wherein at least one of the first and second earpieces comprises a first directional microphone configured to and aligned to capture a user's voice; a boom microphone configured to be removably attached to one of the first or second earpieces, the boom microphone comprising a second directional microphone disposed at a distal end of a flexible boom; and a processor configured to, in response to detecting that the boom microphone is operably attached to the first or second earpiece, deactivate the first directional microphone.

This invention discloses an interactive TR robot and its interaction method, relating to the field of robot interaction technology. It includes a workbench and a voice acquisition device. A lower microphone is located at the top of the outer wall of the workbench, and the lower microphone has a wider bottom and narrower top structure. A voice acquisition device is located at the top of the outer wall of the lower microphone, and eight upward-tilted acquisition holes are located on the side of the outer wall of the voice acquisition device. An upper microphone is located at the top of the outer wall of the voice acquisition device, and the upper microphone has a wider top and narrower bottom structure. This invention, by incorporating the upper microphone, lower microphone, voice acquisition device, eight sets of tilted acquisition holes, and a directional microphone, achieves converged sound wave acquisition and directional noise reduction from eight groups, suppressing noise from non-target directions, significantly improving the strength and accuracy of voice signal acquisition, and clearly capturing operator voice even in noisy environments.