6478 results about "Microphone" patented technology

An online social wager-based gaming system is disclosed, featuring dynamic cross-provider game filtering and persistent cross-provider voice-interactive group play. The system allows for automated multi-seat group game reservations and incorporates a distributed ledger for bet verification. Features include a Group Connect module for dynamic group formation and live audio across various game providers, enabling real-time play. A Professional Companion Connect module facilitates live webcam or microphone gambling sessions with professional companions, under contractual agreements. The User Games module enhances personalization by integrating user-uploaded images into the gaming environment. A cryptocurrency / blockchain-based bet tracking system ensures transparency and security in managing bet transactions through a controlled digital wallet system. This integrated platform aims to provide a cohesive, interactive, and personalized online casino experience, addressing limitations of traditional solitary online gaming by fostering social connections and enhancing user trust and engagement.

The invention discloses an SF6 gas leakage detection method and system based on a photoacoustic spectrum analyzer, and the method comprises the steps: arranging a sampling end to collect an SF6 gas sample, and obtaining stable gas input through constant-current sampling and steady-state pretreatment; steady-state gas is guided into the photoacoustic spectrum analyzer, and resonance frequency stabilization and signal amplification output are achieved through the self-tuning unit; executing double-microphone differential detection and digital filtering processing, and outputting a stable SF6 detection signal with a high signal-to-noise ratio; performing time calibration, abnormity elimination and consistency processing on the detection signal to generate standardized detection data; inputting the standardized data to a Transform model, and performing inversion to generate an SF6 leakage source position and a diffusion path; and displaying an inversion result on a monitoring interface, triggering a sound-light alarm when the inversion result exceeds a limit, and uploading the inversion result to a cloud monitoring platform. According to the invention, the intelligent photoacoustic spectrum system combining photoacoustic-fluid steady-state control and self-tuning detection is constructed, so that high-sensitivity detection and accurate positioning of SF6 gas leakage are realized.

A system may include a storage device, configured to store computer-executable instructions. A system may include an ear-bud configured to be positioned within an ear canal of a user, the ear-bud comprising: a speaker, a microphone; and one or more processors in communication with the storage device, wherein the computer-executable instructions, when executed by the one or more processors, cause the one or more processors to: obtain a user hearing profile, obtain an equal-loudness hearing profile, receive audio data from the microphone, and generate a second audio data based on a first sound-pressure level, a second sound-pressure level, a first frequency; and cause the speaker to emit the second audio data within the ear canal of the user, such that the user perceives the audio data as if the user has normal hearing.

The invention relates to the technical field of multi-modal sensors, in particular to a multi-modal sensor embedded self-calibration system and a real-time compensation method, and the method comprises the following steps: S1, data acquisition: a multi-modal data acquisition mode is adopted, a multi-modal data acquisition module is responsible for acquiring original data of a camera, an inertial measurement unit and a microphone, and the original data of the camera, the inertial measurement unit and the microphone are acquired; the camera collects image data at the frequency of 30 Hz. The method has the advantages of comprehensive data acquisition, accurate data processing, comprehensive and deep feature extraction, scientific and flexible calibration parameters and efficient and continuous real-time compensation, and a multi-modal data fusion technology is adopted in the actual use process; a composite sensing system covering space positioning, dynamic sensing and acoustic analysis is constructed, and the multi-source heterogeneous data fusion scheme can comprehensively capture multi-dimensional features in a complex scene.

Provided herein are techniques to facilitate synchronizing audio streams for a conference call involving multiple microphones utilized at a same location or proximity to one another. In one example, a method may include obtaining, by an aggregating node, each of an audio data stream from each of a plurality of participant devices that are proximate to each other within a conference space for a conference session in which each audio data stream obtained from each participant device comprises audio data and synchronization information, wherein the synchronization information is based on a synchronization sound broadcast during the conference session and received by each of the plurality of participant devices; and synchronizing, by the aggregating node, the audio data of each audio data stream based, at least in part, on the synchronization information included in each audio stream obtained from each of the plurality of participant devices.

The invention relates to the field of biological medicine, and discloses a method and system for analyzing cough sound by using disease characteristics to diagnose respiratory diseases, and the method comprises the steps: deploying a six-microphone annular array to achieve the precise positioning and triggering of a sound source; self-adaptive spectral subtraction and Wiener filtering cascade are adopted to enhance the audio; segmenting a cough segment based on energy envelope; fusing the Mel-cepstrum, the linear prediction residual error, the harmonic energy ratio and the transient zero-crossing rate to construct a pathological feature matrix; extracting local, medium-range and global time sequence features through a three-branch parallel convolutional network; inputting a disease specific classifier to discriminate asthma, pneumonia and laryngitis respectively, and applying a feature decoupling regular term to improve interpretability. The system correspondingly realizes the modularized processing flow. According to the method, the cough sound collection quality and the disease subtype recognition accuracy in a complex environment are improved, meanwhile, the thermodynamic diagram is output to assist clinical decision making, and the diagnosis credibility and practicability are enhanced.

The present disclosure discloses a loudspeaker apparatus. The loudspeaker apparatus may include a support connection member configured to contact with a user's head. The loudspeaker apparatus may include at least one loudspeaker assembly. The loudspeaker assembly may include an earphone core and a core housing configured to accommodate the earphone core. The core housing may be fixedly connected to the support connection member. At least one button module may be arranged on the core housing. The interior of the core housing may further include at least two microphones, and the at least two microphones may be arranged at a position different from a user's mouth. The loudspeaker apparatus may further include a control circuit or a battery accommodated in the support connection member. The control circuit or the battery may drive the earphone core to vibrate to generate sound.

Disclosed are a method, system, and apparatus of a body-worn camera system with integrated artificial intelligence for real-time field assistance and automated incident reporting. In one embodiment, a body-worn safety device includes a body-worn camera configured to capture a video of an incident from a perspective of a wearer of the body-worn camera. In this embodiment, a microphone is configured to capture audio concurrently with the video. A processing unit includes an artificial intelligence module in this embodiment. The artificial intelligence module is configured to respond to a voice command from the wearer by analyzing the captured audio, the captured video, and / or an external data of the artificial intelligence module. In another embodiment, a method of a wearable safety system provides an audible answer and a guidance in real-time using the natural language queries; and outputting an audio response and an alert to the wearer.

Smart dental treatment chair, and a system utilizing Artificial Intelligence (AI) and computerized vision analysis to dynamically monitor real-time progress of a dental treatment, to dynamically report the progress to the patient, and to provide additional benefits to dental patients. A dental treatment chair includes video cameras that capture real time video, and a microphone that captures sound and speech. Computerized vision unit perform analysis of the video, and a Large Language Model (LLM) performs analysis of text extracted from uttered speech, to determine the current step in a multiple-step dental procedure. A display unit is oriented towards the patient, and displays a dynamically-updated progress bar and percentage value, indicating the actual progress of the ongoing dental treatment. Optionally, the smart dental treatment chair also integrally plays music that the patient selects and controls, sprays an aromatic agent, and provides other benefits to the dental patient.

The invention discloses a vending machine multi-mode emotion perception and personalized dialogue generation method and system, and the method comprises the steps: collecting a face image and a voice signal of a user through a camera and a microphone, and outputting a dual-mode emotion tag through the analysis of a face expression and a voice emotion; fusing the label and the dialogue text to generate a structured Prompt containing emotion, intention and context; based on an Agent framework multi-modal large model, combining context history and a personalized strategy to generate a reply text matched with the emotional scene; according to the text and the comprehensive emotion label, TTS parameters are dynamically adjusted, and anthropomorphic voice is output; in multiple rounds of dialogues, styles are kept consistent through an emotion smoothing formula, and emotion-verbal skill-conversion data are precipitated in combination with a user feedback optimization strategy. The interaction limitation of a traditional vending machine is broken through, multi-mode emotion perception, personalized personification interaction and continuous evolution are achieved, the vending machine is promoted to be upgraded to an emotional shopping guide platform, user experience and sales transformation are improved, and a technical model is provided for intelligent retail.

In aspects of presenting relevant audio data, a mobile device implements an audio playback manager that monitors audio in an environment for a trigger word. The audio playback manager detects the trigger word via a microphone associated with a headset in communication with a mobile device. The audio playback manager determines whether a portion of the audio preceding the trigger word is relevant to a user of the mobile device and presents the portion of the audio that is relevant to the user.

A refuse vehicle includes a body coupled at least one of the frame and the chassis, a battery configured to provide electrical power to a first motor, a vehicle body supported by the chassis and defining a receptacle for storing refuse therein, an electric power take-off system coupled to the vehicle body, the electric power take-off system including a second motor configured to convert electrical power received from the battery into hydraulic power, one or more microphones coupled to the body and configured to detect noise, one or more speakers coupled to the body configured to emit noise reducing sounds, and a controller configured to receive data related to the detected noise from the one or more microphones and cause the one or more speakers to emit the noise reducing sounds in response to the data received from the one or more microphones.

In an embodiment, the disclosure relates to a device for assisting a respondent in a conversation. The device includes a microphone configured to detect a voice input, and a transmitter communicatively coupled to a server and configured to transmit the voice input to the server. The server is to generate vectors associated with the voice input, feed the vectors associated with the voice input to an Artificial Intelligence utilizing a trained Machine Learning (ML) model, and obtain, from the trained ML model, an output corresponding to the vectors. The device further includes a receiver communicatively coupled to the server, and configured to receive from the server, the output generated by the ML model. A speaker is communicatively coupled with the receiver and is configured to generate a voice-based response based on the output, for assisting the respondent in responding to the conversation.

The invention provides a box body and a signal transmission method. The box body comprises an upper shell, a lower shell, a first microphone, keys, a rotating shaft and an antenna assembly, the lower shell and the upper shell are installed in a matched mode to form a containing cavity, the containing cavity comprises a first surface, opposite to the upper shell, in the lower shell, the lower shell comprises an annular side face connected with the first surface, and the annular side face comprises a first side face; the second side face and the third side face are connected with the first side face, the second side face and the third side face are oppositely arranged, a first microphone pickup hole is formed in the first side face, the first microphone is arranged in the lower shell, communicated with the first microphone pickup hole and used for collecting a first audio signal outside the box body, and the key is arranged on the second side face and used for collecting a second audio signal outside the box body. The rotating shaft is arranged corresponding to the third side surface and is connected between the lower shell and the upper shell, the upper shell and the lower shell are rotatably connected through the rotating shaft, and the antenna assembly is arranged on the first surface and is used for at least transmitting at least part of the first audio signal to the mobile terminal.

The invention discloses a wind turbine generator impeller anomaly detection method and system based on sound vibration signal identification. According to the method, firstly, impeller response is collected through a microphone / vibration sensor, and a Mel spectrogram is generated; then, a Teager-Kaiser energy operator and self-correlation analysis are utilized, and under the condition of not depending on a rotating speed signal, the rotating period of the impeller is recognized, and frequency spectrum segmentation is carried out; calculating a cross-period Mel frequency spectrum dynamic deviation, and normalizing the cross-period Mel frequency spectrum dynamic deviation through an amplitude correction coefficient related to the rotating speed; generating a periodic coherent energy diagram by adopting an improved structural similarity algorithm; performing filtering enhancement by using a harmonic resonance template, performing morphological deconstruction and parameterization on an abnormal region in the graph, and extracting geometric features; and finally, calculating a comprehensive abnormal score based on the multi-dimensional features and realizing automatic early warning. According to the method, the problems of variable working condition interference and rotating speed dependence are effectively solved, and accurate and stable detection of early abnormality of the impeller can be realized.

The invention relates to a speaker authentication and deep forgery detection cascade voice privacy protection device which comprises a microphone, an ASV module, an ADD module, a control unit and an audio output module, the microphone, the ASV module, the ADD module, the control unit and the audio output module are connected in sequence, the input end of the control unit is connected with the output end of the ASV module, and the output end of the ADD module is connected with the output end of the audio output module. A voice input signal is firstly collected by a microphone and then is input into an ASV module for identity verification, and meanwhile voice data is input into an ADD module for forgery detection; the outputs of the two are transmitted to the control unit, and the control unit determines whether to allow the audio signal to be output according to preset logic; according to the invention, the ASV and ADD modules are cascaded, and dual judgment of identity and authenticity is realized on a physical equipment level for the first time; the control unit performs joint decision through a signal fusion strategy, so that the safety is improved; and the structure is clear, the function partition is reasonable, and modular deployment and replacement can be realized.

The invention relates to the technical field of artificial intelligence monitoring, in particular to a bid evaluation site multi-modal behavior compliance monitoring method and system, and the method comprises the following steps: deploying an array microphone and a high-definition camera in a bid evaluation room, and synchronously collecting the sound and image data of bid evaluation experts and workers; meanwhile, personnel position information is collected in real time through a Bluetooth low-power-consumption or ultra-wideband tag; the bid evaluation method has the beneficial effects that sensitive languages, private communication, file or storage medium transmission, illegal use of electronic equipment, unauthorized leaving and other behaviors of judges and workers are recognized in real time in the bid evaluation process, and risk early warning or forced intervention is generated in real time, so that hidden dangers of camera obscura operation and serial fraud are eliminated. Meanwhile, the invention also aims to automatically store traceable audio and video evidence and behavior logs and provide reliable support for subsequent auditing and law enforcement evidence collection, so that the openness, justice and compliance levels of bid evaluation activities are comprehensively improved.

The invention discloses a sign language synthesis service method based on multi-modality, and the method comprises the following steps: S10, carrying out end-cloud collaborative rendering setting: deploying an edge end as a lightweight model to generate a basic action, and deploying a cloud end as operating a MoMask and 3D rendering engine; s20, performing multi-modal data acquisition: acquiring a voice signal through a microphone, acquiring a 48 * 48 pixel face grayscale image through a camera, and acquiring action posture data through an IMU sensor; s30, performing emotion feature extraction on the collected multi-modal data: performing voice emotion recognition to output six types of emotion probability distributions, and performing facial expression recognition to output seven types of expression probability distributions; s40, performing cross-modal feature fusion: dynamically fusing the voice and facial features based on a confidence weighting strategy, and generating a three-dimensional emotion intensity vector; and S50, sign language action generation is carried out, and a 3D skeleton sequence matched with emotion is generated through RVQ layering quantification and MoMask Transformer.

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source.

This invention relates to the field of speech recognition technology, specifically disclosing a cross-language AI voiceprint cloning method, system, and storage medium. The method includes: a speech collection end gating the raw microphone speech, with qualified samples entering preprocessing; gating, restricted spectrum, and unified conditional signals are integrated across the upstream and downstream processes to significantly improve robustness under noise / echo conditions; a speech processing end using AI adaptive filtering to denoise and obtaining representational data according to restricted parameters; a feature extraction and recognition end extracting voiceprints from the spectrum and embedding them into parallel language recognition, storing the voiceprint-language-quality association; a voiceprint cloning end performing cosine search in a template library to obtain a similarity queue, which is then weighted and aggregated after being rearranged based on quality and language consistency to obtain a target template; small-sample adaptation improves cross-language generalization and scalability; and finally, obtaining the target language and generating cloned speech by combining it with the target template.

An earphone includes a housing having a space therein and having a path capable of ventilation from one end side on an external auditory canal side of a wearer to the other end side on an ambient environment side, a valve accommodated in the housing and configured to switch the path between an open state and a close state, a microphone disposed on one end side of the housing and configured to collect uttered voice of the wearer, and a control unit configured to control the open state and the close state. The control unit switches the path to the open state during a first operation in a call including an operation in which the uttered voice of the wearer is collected by the microphone, and switches the path either the open state or the close state during a second operation different from the first operation.

This disclosure relates to methods, systems, and techniques for visualizing external audio captured by a microphone of the vehicle. Using the techniques described herein, external audio signals may be interpreted and converted into clear visual indicium that can be understood by individuals associated with the vehicle, such as passengers inside the vehicle or individuals awaiting pickup.

The invention discloses a fully-implanted artificial cochlea system and a noise reduction method thereof.The fully-implanted artificial cochlea system comprises an implant and a fully-implanted microphone, the fully-implanted microphone comprises a shell, a PCBA board, a first sound sensor and a bone conduction sensor are arranged in the shell, the first sound sensor is connected with the PCBA board and used for receiving a first external sound signal and transmitting the first external sound signal to the PCBA board, and the bone conduction sensor is connected with the implant. The bone conduction sensor is also connected with the PCBA board and is used for collecting an in-vivo noise signal and transmitting the in-vivo noise signal to the PCBA board; the PCBA board is used for receiving the first external sound signal and the in-vivo noise signal, and at least performing noise reduction on the first external sound signal and then outputting the first external sound signal; the implant or the fully-implanted microphone is connected with a second sound sensor, the second sound sensor is connected with the PCBA, the second sound sensor receives a second external sound signal and transmits the second external sound signal to the PCBA board, and the PCBA board receives the second external sound signal, at least carries out noise reduction on the second external sound signal and then outputs the second external sound signal; according to the system, the wearing comfort of a user is effectively improved, the efficient noise reduction performance is met, and the hearing requirement of a user patient can be met.

The present disclosure is of a bone conduction microphone. The bone conduction microphone comprises of a laminated structure and a base structure. The laminated structure is formed by a vibration unit and an acoustic transducer unit. The base structure is configured to load the laminated structure. At least one side of the laminated structure is physically connected to the base structure. The base structure vibrates based on an external vibration signal, the vibration unit deforms in response to the vibration of the base structure, and the acoustic transducer unit generates an electrical signal based on the deformation of the vibration unit. A resonant frequency of the bone conduction microphone is within a range of 2.5 kHz-4.5 KHz.

The invention relates to an array microphone noise reduction recording method based on cascade noise reduction and blind source separation, which belongs to the technical field of voice signal processing and recording, and comprises the following steps: configuring a multi-channel array microphone, ensuring that the amplitude and phase of a channel signal are consistent, and collecting an original multi-channel voice signal; a weighted kernel function blind source separation algorithm is adopted, signal-to-noise ratio distribution characteristics of signals are extracted, kernel function weights are given, and target voice and interference signal components are obtained through decoupling of an independent component analysis model; executing target-oriented adaptive cascade noise reduction, locking the voice of a keynote speaker through directional pickup, reducing noise, filtering out reverberation, and enhancing the voice of a far-field target by combining a voice mask neural network with a far-field pickup algorithm in sequence; and processing the target voice through voice feature perception lossless coding and storing the target voice. According to the invention, stable acquisition of multi-channel signals, accurate separation of mixed signals and layered suppression of noise reverberation are realized, the signal-to-noise ratio and definition of far-field voice are significantly improved, and the method is suitable for single-person speaking or multi-person dialogue scenes.

The invention provides an active noise reduction method capable of restraining divergence and a microphone vibration reduction device, belongs to the technical field of automobile active noise reduction, and detects the sealing condition of a cab through automobile door sensors, automobile window sensors and skylight sensors. If the microphone is closed, noise signals are collected by using errors on the microphone vibration damping device and a reference microphone, and secondary noise signals are generated by an active noise reduction filter and are played by a loudspeaker; if yes, the engine rotating speed in the vehicle CAN signal is read, and a reference noise signal is generated and played. A cab sealing state and an active noise reduction system operation state are displayed on an instrument panel, and the operation state is determined according to noise signal acquisition and noise reduction signal playing conditions. The noise reduction mode can be automatically switched according to the cab sealing state, the noise in the vehicle is effectively reduced when the cab is sealed, the reference noise is reasonably played when the cab is opened, and the influence on the robustness of a system filter is reduced. Related information is displayed on the instrument panel, so that a user can visually know the cab state and the system operation condition.

The invention discloses a directional pickup method for a microphone. The microphone comprises a left microphone core, a middle microphone core and a right microphone core. Wherein the middle microphone core is located on the central axis of the microphone, and the left microphone core and the right microphone core are symmetrically distributed on the left side and the right side of the middle microphone core; sound of a sound source is synchronously collected through a left microphone core, a middle microphone core and a right microphone core of the microphone, and three paths of sound signals are obtained; respectively filtering the three paths of sound signals to obtain left, middle and right microphone core signals; calculating an included angle between a connecting line of a sound source and the middle microphone core and the central axis of the microphone based on the time difference and the distance difference of the sound reaching the left, middle and right microphone cores to obtain a sound source direction; based on the left microphone core signal, the middle microphone core signal, the right microphone core signal and the sound source direction, through time delay compensation and gain adjustment, the sound signal in the target direction is enhanced, the sound signal in the non-target direction is suppressed, the final output signal is obtained, and the directionality and the sound quality of pickup are improved.

A communications headset includes one or more directional speakers mounted in left and right temple housings positioned near a wearer's ears, with acoustic channels within the temple housings configured to direct sound waves toward the ear canals. The temple housings form an open-ear configuration, allowing external noises to reach the ears unobstructed. Each directional speaker is acoustically coupled to a contoured channel that delivers focused audio to the wearer's ear with minimal sound wave leakage. An in-ear speaker, magnetically secured and tethered to the temple housing, provides optional noise-isolated listening. A user-controlled selection switch allows a wearer to toggle between open-ear and in-ear modes. The headset also includes a boom microphone, volume controls, and USB charging and communication ports.

A voice assistance system for holding a spoken conversation with a person. The system can include at least one microphone configured for detecting a voice utterance of the person, at least one speaker configured for outputting a sound to the person, at least one processor configured for executing computer instructions, and at least one memory. The at least one memory stores computer instructions configured for operating the system to perform steps including: providing at least one machine learning (ML) model configured for generating contextually relevant and varied responses in natural language conversations, detecting a voice utterance using the microphone, providing the voice utterance as an input to the ML model, prompting the ML model to generate an output based on the input, and providing the output to the speaker to be output to the person.

The invention relates to the technical field of voice recognition, and discloses a voice instruction recognition and cleaning control method for a sweeping robot, which comprises the following steps: synchronously reading a dust collection motor driving pulse width modulation value and a rolling brush motor load current value of the sweeping robot while acquiring an audio signal acquired by a microphone; retrieving and interpolating in a preset working condition noise spectrum mapping table according to the non-acoustic operation state data, determining a predicted mechanical noise power spectrum at the current moment, taking the predicted mechanical noise power spectrum as a priori reference, calculating an instantaneous signal-to-noise ratio of the predicted mechanical noise power spectrum and the frequency domain feature sequence, and constructing a frequency domain confidence mask; according to the method, a feedforward prediction mechanism is constructed by introducing non-acoustic prior information, and response lag of traditional acoustic posteriori estimation during sudden change of working conditions is avoided.