8576 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 a smart park safety protection system and method based on smart video analysis and multi-mode integration, and relates to the technical field of smart park management. The system comprises a camera, an environment microphone and an access control terminal which are deployed in a park, an edge computing node is used for extracting structured video, audio and access control features, a central server realizes multi-modal space-time alignment through a self-calibration module, a multi-modal fusion module constructs a joint representation vector based on a low-rank tensor algorithm, and the joint representation vector is used for realizing multi-modal space-time alignment. And the security decision module identifies an abnormal event and triggers a response strategy in combination with a classification and anomaly detection model. According to the invention, multi-mode cooperative identification, high-robustness behavior analysis and intelligent response linkage are realized, and the real-time performance, accuracy and intelligent level of a park security system are improved.

The invention discloses a multi-modal data fusion method and system based on a large-model intelligent agent, belongs to the technical field of artificial intelligence, big data processing and intelligent agents, and aims to solve the technical problem of how to effectively integrate data of different modalities such as texts, images and voices by using the large-model intelligent agent. According to the technical scheme, the method comprises the steps that data collection and preprocessing are conducted, specifically, various modal data of texts, images and voice are collected through a web crawler, an API interface, a camera and microphone equipment, the collected data are preprocessed, the preprocessed multi-modal data are obtained, and the data quality is ensured; feature extraction and mapping: extracting corresponding modal features from the preprocessed multi-modal data through CNN and Transform models, mapping the different modal features to the same space, and combining the aligned features to form comprehensive feature representation; carrying out multi-modal fusion processing; and performing intelligent decision and feedback.

The invention discloses a speech recognition and authentication method and system based on multi-modal features and dynamic evaluation in the technical field of speech recognition and authentication, and the method comprises the steps: collecting an original speech signal of a user through a microphone, and carrying out the preprocessing of the original speech signal, and obtaining the preprocessing speech data; and extracting feature data of the preprocessed voice data by adopting a multi-dimensional feature hierarchical extraction technology, and injecting a multi-source noise sample into an acoustic feature space of a voiceprint feature model based on an initial training stage established by the voiceprint feature model to construct an anti-noise mixed voiceprint map. Through integrating voiceprint, semantics, behavior characteristics and an environment adaptation mechanism, an authentication threshold is adjusted in real time according to dynamic risk assessment, meanwhile, a risk scoring model is utilized to calculate a comprehensive risk value, authentication modes of different levels are started according to risk scenes of different degrees, and two-factor authentication is forcibly implemented for high-risk scenes. And the authentication security and reliability can be obviously enhanced.

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 audio signal processing, and discloses a room sound correction method, device and equipment for audio equipment and a storage medium, which are used for improving the accuracy and correction effect of correction parameters and effectively improving the acoustic effect of the audio equipment in a room. The room sound correction method of the audio equipment comprises the following steps: acquiring response data of a room to a multi-band preset test signal, and extracting room acoustic data by adopting a time-frequency conjoint analysis method, the room acoustic data comprising frequency attenuation, reflection paths, phase shifts and reverberation time information of different bands; based on the room acoustic data, performing compensation processing by using preset microphone calibration data to obtain a calibrated frequency response curve; performing fitting calculation on a preset ideal frequency curve and the calibrated frequency response curve according to an optimization algorithm to obtain specific correction parameters of the room; and correspondingly adjusting the output of the audio equipment according to the correction parameter.

A logarithmic amplifier includes programmable gain amplifiers each having a different gain, wherein an input of each of the programmable gain amplifiers is coupled to an input of the logarithmic amplifier; and a summing circuit having inputs coupled to a corresponding output of each of the programmable gain amplifiers and an output coupled to an output of the logarithmic amplifier, wherein the summing circuit generates a logarithmic transfer function having piecewise linear segments.

Various embodiments of an ear-wearable electronic device are disclosed. The device includes a housing having a top shell and a bottom shell, a microphone disposed within the housing, and an acoustic path disposed at least partially within the housing and extending between an acoustic port disposed at an outer surface of the housing and an outlet disposed within the housing that acoustically couples the acoustic path to an inlet of the microphone. The device further includes an acoustic filter disposed at least partially within an inner surface of the top shell and includes a neck and a resonance cavity acoustically coupled to the neck. The acoustic filter is acoustically coupled to the acoustic path via the neck. Further, the acoustic filter is configured to reduce an intensity of acoustic waves sensed by the microphone in a first frequency range.

A communications system for a restaurant includes multiple field communications units, a portable communications device, and processing circuitry. The field communications units include a speaker and a microphone. The portable communications device is transportable by an order taker of the restaurant. The processing circuitry obtains a user input to transition the portable communications device between multiple different channels. The multiple different channels correspond to one of the multiple field communications units. The processing circuitry transitions the portable communications device to one of the multiple different channels according to the user input. The processing circuitry facilitates end-to-end bi-directional audio communication between the order taker and a customer by operating the portable communications device and the one of the multiple field communications units to exchange audio data over the one of the multiple different channels.

The invention relates to the technical field of safety and environmental protection, in particular to a negative-pressure circulating explosion-control dust removal method suitable for combustible dust places, which comprises the following steps: firstly, sealing a bin body and injecting carbon dioxide-nitrogen inert gas to generate a low-oxygen environment; then, a flow guide angle-adjustable cyclone separator, a dielectric barrier discharge plasma reactor and a waste heat thermo-acoustic transducer are driven to cooperatively capture dust, and sound node amplitude synthesis capture data are obtained through compressed sensing of a microphone; inputting the environment data and the capture data into a reinforcement learning process containing a micro-thermochemistry model, and outputting instructions for adjusting negative pressure pulse, inert flow, sound field intensity and cyclone geometry; in combination with distributed radio frequency imaging data, an adversarial digital twinborn model is generated to predict explosion free energy, and automatic rollback or shutdown is carried out when the explosion free energy exceeds a threshold, so that a safe, efficient and low-energy-consumption closed loop is formed.

The invention discloses an intelligent diagnosis and maintenance system for power equipment, and relates to the technical field of power equipment detection, and the system comprises the following components: an array microphone module which is composed of a plurality of microphones, is used for accurately calculating the spacing, is arranged in a three-dimensional or plane matrix manner, and is used for collecting sound signals generated in the operation process of the power equipment; according to the invention, the array microphone module is combined with a beam forming technology, the sound signals are collected efficiently, a voiceprint topological map is constructed, sound source distribution and energy conduction paths of all parts of the equipment are represented in a visual form, and the sound source distribution and energy conduction paths of all the parts of the equipment are displayed in a visual manner. Meanwhile, voiceprint features and vibration modal features are extracted through a feature extraction module, and a joint inversion module is used for performing joint analysis on the two features based on a mapping relation model and a machine learning algorithm and in combination with a modal database, so that the damage degree of the mechanical part of the power equipment is accurately inverted.

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.

Techniques for utilizing invisible and / or inaudible watermarking techniques in media streams (e.g., video and / or audio streams) as a form of in-band steganography to authenticate the source capture camera and / or microphone to prevent deepfakes in network-based communication systems. User device(s) may establish cloud-based connections to a network-based communication system hosting media stream(s). An encoder of a watermarking component of a user device may receive video / audio data directly from a camera / microphone thereof and encode the video / audio data with watermark(s) representing a signed signature. A decoder of a watermarking component of a recipient user device and / or another device associated with the communication system may extract the watermarks and authenticate the signed signature using public certificate(s) associated with the sending user device to validate authenticity of the capture source(s) of the sending user device and that the video / audio data has not been modified since the time of capture.

The invention relates to the technical field of game development and artificial intelligence, and discloses a Qt interactive game role action response control method fused with AI emotion recognition, which comprises the following steps: S1, acquiring multi-modal data such as facial expression, voice intonation and limb action of a player by using sensors such as a camera and a microphone on game equipment; s2, the collected multi-modal data are transmitted to an AI emotion recognition module, a deep learning algorithm is adopted to analyze and process the data, and the current emotion state of the player is recognized; and S3, obtaining current game scene information and other operation instructions input by the player through the Qt framework. According to the Qt interactive game role action response control method and system fused with AI emotion recognition, through the AI emotion recognition technology, a game role can sense the emotion state of a player and make the emotion interaction action matched with the game role, the emotion resonance between the game role and the player is enhanced, and the interaction experience and immersion of a game are greatly improved.

The invention relates to the technical field of content transmission methods, in particular to a content transmission method based on a multi-modal streaming media fusion technology system, which comprises the following steps: S1, multi-source data acquisition and time-space alignment: adopting an IEEE 1588PTP protocol, deploying a hardware clock synchronization module at a camera, a microphone and IMU equipment, improving the timestamp precision to a mu s level, and performing time-space alignment on the camera, the microphone and the IMU equipment; carrying out sampling rate normalization on the sensor data; s2, feature level cross-modal fusion: extracting each modal feature by using a lightweight Transform model, executing cross-modal attention calculation in an edge server, and generating a fusion feature tensor Ffusion belonging to RN * D; the system content transmission method based on the multi-modal streaming media fusion technology solves the problems that a traditional multi-modal transmission system usually adopts a separated transmission architecture, video streams are transmitted through an RTMP protocol, audio streams are encoded through Opus and are transmitted through WebRTC, sensor data are sent through an MQTT protocol in a JSON format, ABR only adjusts the video code rate, and the transmission efficiency is low. And joint optimization of multi-modal data is not coordinated.

The present invention discloses a multi-modal system for real-time plant hydration monitoring and irrigation management. The system comprises at least one acoustic sensor configured to emit controlled sound waves through plant tissues and detect corresponding vibrations and resonance frequencies using sensitive microphones or piezoelectric sensors to assess plant hydration. The system includes at least one bioelectrical sensor for monitoring bioelectrical signals. At least one nanotechnology-based sensor is configured to detect molecular changes in water content within plant cells through embedded or externally applied nanosensors. An aerial imaging device mounted on an unmanned aerial vehicle captures data related to canopy temperature, leaf color, and hydration indicators using infrared, multi-spectral, and thermal cameras. A central processing unit receives and analyzes data using predictive models, integrates the data via data fusion algorithm to generate a plant hydration profile, and controls water delivery through an irrigation management unit based on the hydration profile.

The invention belongs to the field of intelligent information system management, and particularly discloses an intelligent voice telephone robot system and method, voice and image multi-mode data are collected through a microphone and a camera, and after preprocessing, voice, emotion and semantic features are fused through an improved Transform architecture to achieve accurate recognition of user intentions; a double-layer decision network based on reinforcement learning is combined with a dynamic reward function to generate an optimal response strategy; and realizing rapid task migration and parameter optimization of the model by adopting a meta-learning mechanism. The system also has the functions of adaptive noise robustness, multi-language interaction, user portrait dynamic updating, man-machine collaboration and the like. Compared with a traditional scheme, the intention recognition accuracy, the task completion rate and the scene adaptability are remarkably improved, the interaction experience is effectively improved, and the method can be widely applied to the fields of customer service, intelligent marketing and the like.

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.

A work machine includes a lower traveling body; an upper slewing body slewably mounted on the lower traveling body; an external microphone configured to collect sound around a work machine; an internal speaker configured to output sound to an operator of the work machine; an information transmission device configured to enable a worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine; and a processor, and a memory storing instructions that cause the processor to execute a process. The process includes collecting, by the external microphone, sound around the work machine; outputting, by the internal speaker, the sound to the operator of the work machine; and enabling, by the information transmission device, the worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine.

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.

A mobile phone case includes a cradle for receiving the smartphone in a friction fit configuration, an upper frame portion coupled to an upper section of the cradle and a lower frame portion coupled to a lower section of the cradle via upper and lower rail assemblies, respectively. The upper and lower frame portions are slidably movable between a closed configuration covering the upper and lower microphones and the front and rear cameras of the smartphone, respectively, and an open configuration not covering them. The upper and lower frame portions include dampening members configured to block audio and visual signals when closed. The mobile phone case includes a spring-biased selector assembly coupled to the cradle portion and operatively coupled to the upper and lower frame portions for simultaneously urging the upper and lower frame portions toward the open configuration when actuated.

Systems and methods for annotating, also known as tagging, visually identifiable objects using natural language (i.e. spoken voice) are provided. More specifically, but not exclusively, this disclosure relates to systems and methods for tagging identified objects related to underground utilities and assets, and communication systems. Once tagged, data representing identified objects can be stored, transmitted, and / or mapped. In an exemplary embodiment, a utility service worker or other personnel may walk or drive around an area of interest using utility locating equipment or systems to collect data. As data is being collected, the user may visually identify utility related or other items using a laser pointer, and then use their voice to tag items of interest. A headset including a microphone may be provided for capturing the user's voice. Hardware and / or software may be provided for processing tagged items, and relating them to a specific location, and / or utility asset.

To effectively resolve the issue of inappropriate handovers during communication sessions, methods, systems, and machine-readable mediums which utilize speech features captured by microphones in the original wireless peripheral device and / or the wireless peripheral device to which the communication session is to be handed over to determine if a handover should proceed or should be reversed. Speech features refer to the composite attributes of spoken language that encompass both acoustic and linguistic features. Acoustic features characterize the sound properties of speech and include, but are not limited to, timbre, pitch, intonation, speaking rate, articulation, prosody, melody, spectral features, formant frequencies, and the like. Linguistic features pertain to the actual content conveyed, comprising words, phrases, syntax, and semantics. This includes the analysis of words, phrases, syntax, and semantics to understand the context and continuity of the conversation.

The utility model discloses an ear canal embedded hearing aid, which relates to the technical field of medical instruments, and comprises a bearing assembly, the bearing assembly comprises a first packaging shell, a second packaging shell and a filter screen, a signal processing piece and a lithium battery are respectively arranged in the first packaging shell, and the second packaging shell is provided with a second packaging shell. The signal processing piece comprises a signal amplifier, a filter and a central processing unit, and a micro air pump, a control switch, a potentiometer and a microphone are installed on the right side of the first packaging shell. The beneficial effects of the utility model are that through the arrangement of the miniature air pump, the air delivery pipe and the expansion anti-skid assembly, when the ear canals of different users have large differences or need to be worn stably, the miniature air pump can be controlled to work, so that air enters the hollow rubber ring through the air delivery pipe and the delivery pipe, and the hollow rubber ring is gradually expanded to be in contact with the ear wall; therefore, the wearing stability of the bearing assembly is ensured, and the sound definition after conversion of the receiver is ensured.

Various embodiments of an ear-wearable electronic device are disclosed. The device includes a faceplate having an outer surface and an inner surface, where the faceplate defines a cavity disposed in the inner surface of the faceplate; a microphone disposed at least partially within the cavity and including an inlet; and an acoustic path disposed within the faceplate and extending between a microphone port disposed at the outer surface of the faceplate and an outlet disposed in the cavity. The acoustic path is acoustically coupled to the inlet of the microphone via the outlet. The inlet of the microphone is disposed in the cavity or in the outlet of the acoustic path. The acoustic path is configured such that all acoustic waves propagating within the acoustic path pass directly from the acoustic path to the inlet of the microphone.

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.

The disclosed technology includes a donation kiosk comprising a microphone; a speaker; an input device configured to receive donation information; a user interface; one or more processors; and a memory storing instructions. The instructions, when executed by the one or more processors, are configured to cause the donation kiosk to determine whether to operate in an emergency mode. In response to determining that the donation kiosk should operate in the emergency mode, the instructions can cause the donation kiosk to display an emergency call option on the user interface. In response to the emergency call option being selected, the instructions can further cause the donation kiosk to facilitate communication with an emergency service via the microphone and speaker.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.