3743results about "Mouthpiece/microphone attachments" patented technology

The present disclosure provides an earphone comprising a housing assembly, a speaker, and an ear hook assembly. The housing assembly includes a first accommodating cavity and a connecting portion that is provided with a socket extending along a first direction. The first accommodating cavity and a first channel and a second channel included in the socket are sequentially connected. A first abutment surface is disposed at transition between the first and second channels. The first abutment surface is inclined to the first direction. The ear hook assembly includes a first connecting end that includes a plug portion. The plug portion is inserted into the socket and includes a second abutment surface and an annular surface. The second abutment surface is inclined to the first direction and faces the first abutment surface. The annular surface extends from an inner edge of the second abutment surface toward the first abutment surface.

Provided are a conversational artificial intelligence (AI) system and method based on real-time multimodal emotion recognition. The system includes a model server configured to provide a machine learning-based conversational model, a terminal configured to perform a conversation with the machine learning-based conversational model through the model server, display a virtual human responding to a user during a conversation with the user, and capture a facial image of the user during the conversation, and a multimodal empathetic conversation-generation system configured to access the model server and receive a response to a question of the user from the terminal, and assess an emotion of the user from the facial image of the user and control, based on the assessed emotion, an expression of the virtual human displayed on the terminal.

The invention relates to an intelligent microphone pickup and speech enhancement method, system and device, and the method comprises the steps: obtaining an audio signal collected by a dual-silicon microphone array, and carrying out the cross-correlation analysis of the audio signal, and obtaining a corresponding human voice signal correlation feature; performing sound source positioning analysis on the audio signal based on the human voice signal correlation feature to obtain a corresponding target human voice signal; performing frequency characteristic analysis on the target human voice signal, and performing segmented dynamic gain processing on the signal according to a preset frequency band range to obtain a corresponding voice enhancement signal; performing noise component adaptive filtering processing on the target human voice signal to obtain a corresponding noise suppression parameter; and inputting the speech enhancement signal and the noise suppression parameter into a preset echo cancellation model for joint optimization to obtain a corresponding output human voice signal. According to the invention, the coupling problem of multiple acoustic interferences can be effectively solved.

An acoustic output device is provided. The acoustic output device includes a housing; a bone conduction vibrator configured to generate bone conduction sound waves and transmit the bone conduction sound waves to a cochlea through the housing to produce sound; an air conduction vibrator configured to generate air conduction sound waves, the air conduction sound waves being transmitted to an ear through sound guiding holes in the housing; and a processing module configured to provide the bone conduction vibrator and the air conduction vibrator with a first audio signal and a second audio signal, respectively, the first audio signal and the second audio signal having a frequency division point, the first audio signal and the second audio signal including components with frequencies above and below the frequency division point, respectively. The frequency division point is not less than 600 Hz.

The invention relates to the technical field of environment monitoring, solves the problem that a monitoring system and method cannot meet the requirements for accuracy and real-time performance of animal behavior research and climate environment research in a plateau area at the same time, and particularly discloses a monitoring system and method integrating environment monitoring and animal monitoring recognition. Comprising a sensing system, a data transmission system and a data processing system, the sensing system is used for collecting environment and animal data of a monitoring site, and the data processing system is used for receiving and processing the data and analyzing the change trend and the correlation influence degree of the environment data and the animal data; the monitoring method comprises the following steps: acquiring environment and animal data through the environment monitoring network and the animal behavior sensing network; data is transmitted to the data processing system in real time through the data transmission system; the data processing system analyzes the dynamic change trend of the environment data and the animal data; and generating early warning information and triggering a protection measure execution instruction. The method can be used for monitoring the plateau environment and animals, and is high in accuracy and high in real-time performance.

Various implementations include a method of training a road noise cancelation (RNC) system for a vehicle, including: providing inputs to RNC system, the inputs obtained from: a set of ear-mounted microphones on a user, at least one transducer, an accelerometer, a set of cabin microphones in the vehicle, and a controller area network (CAN) bus, the inputs from the set of ear-mounted microphones on the user approximating a signal detected by the ears of the user; adapting a set of parameters in the RNC system defining an estimated signal detected at respective ears of the user based on the inputs; and generating at least one of the following for input during an operating mode of the RNC system: estimated ear microphone signals based on the adapted set of parameters, or a set of projection filters for use in determining an estimated ear signal at the respective ears of the user.

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.

Provided is a sound reproduction device which reduces air noise. The sound reproduction device includes a casing in which an opening section communicating with an inside and an outside is formed, and a microphone that is disposed in an internal space communicating with the opening section, in which a value obtained by dividing a volume of the internal space from the opening section to the microphone by an area of the opening section is 2.9 or more.

A method that includes determining a set of one or more reverberation parameters of an extended reality (XR) environment in which a user is to participate, determining whether an audio source device is wirelessly communicatively coupled to send audio to an audio output device, in response to determining that the audio source device is not wirelessly communicatively coupled to the audio output device obtaining a microphone signal produced by a microphone of the audio source device, producing a reverberant audio signal from the microphone signal according to the set of one or more reverberation parameters, and sending the reverberant audio signal to drive a speaker driver of the audio source device.

To provide headphones with an improved noise canceling function and the like.Headphones include a headband, a housing, an ear pad attached to the housing, a hanger that supports the housing, and a slider that connects the hanger to an end portion of the headband. The headphones further include a microphone arranged at a place of the housing outside a movable range of the hanger. The housing includes a case that covers a movable range of the hanger. The hanger and the microphone are arranged in the same space formed in the housing on an outer peripheral side of the hanger.

The invention relates to the technical field of remote volume adjustment, in particular to a control system and method for stage intelligent sound control sound equipment, and aims to accurately capture time distribution and amplitude characteristics of volume fluctuation by collecting environmental noise data in real time and combining a dynamic data analysis method, predict a future volume trend based on historical data, and improve the sound control efficiency. According to the logic based on dynamic prediction, the response capacity to sound changes is enhanced in volume control, the Kriging interpolation method is adopted to accurately analyze sound field requirements, the requirements of different areas are matched by adjusting sound field parameters, the sound field uniformity and the coverage accuracy are improved, and the volume control accuracy is improved. Meanwhile, the intelligence and automation level of the sound system is improved, the output power and frequency distribution of the sound equipment are dynamically adjusted through the particle swarm optimization algorithm, the output volume can be continuously optimized according to the real-time environment and sound field requirements, and the individual requirements of different stage performances are met.

The invention provides a single-microphone multi-array pickup method and system based on sound attenuation simulation, and the method comprises the steps: obtaining an original audio signal recorded by a single microphone and sound source direction information, and carrying out the sound attenuation simulation based on the sound source direction information and a preset virtual microphone array orientation parameter, calculating the included angle between the sound source direction and each virtual microphone; generating an audio intensity attenuation coefficient of each virtual microphone according to the included angle; and acting the audio intensity attenuation coefficient on the original audio signal to generate multi-channel audio data simulating the multi-microphone array. By adopting the method, the response difference of different microphone positions to the sound source can be simulated, the pickup result of the multi-microphone array is obtained, and high hardware cost and complex installation flow caused by actual deployment of a complex microphone array are avoided; and low-cost and high-diversity training data sources are provided for sound source localization, noise suppression and other models based on deep learning.

An audio processing method implemented by an electronic device includes entering a multi-channel video recording mode, detecting a shooting operation of a user, simultaneously recording, after detecting the shooting operation, a first video image and a second video image using a first camera and a second camera, and recording audio of a plurality of sound channels, where the audio includes panoramic audio, first audio corresponding to the first video image, and second audio corresponding to the second video image. The electronic device further records the first audio based on a feature value such as a zoom magnification corresponding to the first display area.

A microphone is provided comprising a microphone capsule adapted to capture an audio signal, an AD converter configured to convert the captured audio signal into a digital signal, a block generator configured to divide the digital audio signal into a plurality of audio blocks, a private key signature unit configured to generate at least one signature based on at least one of the plurality of audio blocks with a private key associated to the microphone and an audio output configured to output the at least one audio block and the generated at least one signature.

An electronic device according to various embodiments disclosed herein may include: a housing, a camera module comprising a camera disposed in the housing, a speaker module comprising a speaker positioned in a first direction with respect to the camera module, a receiver module comprising a receiver positioned in a second direction opposite to the first direction with respect to the camera module, first sound holes including a connection hole positioned in the first direction with respect to the camera module and a hole positioned in the second direction with respect to the camera module, the first sound holes being positioned on a side surface of the electronic device, a first sound conduit interconnecting the connection hole and the speaker module, a second sound hole positioned on a front surface of the electronic device, a second sound conduit including a connection conduit disposed in a third direction perpendicular to the first direction with respect to the first sound conduit, the second sound conduit interconnecting the second sound hole and the receiver module, and a connection portion interconnecting the first sound conduit and the connection conduit of the second sound conduit.

In one embodiment, an eyeglass frame includes a lens holder, a first temple with a first end close to the lens holder and a second end, a second temple, an electrical connector and a printed circuit board. The printed circuit board with at least one electrical component can be in the first temple. The connector can be close to the first end of the first temple, facing downward, and electrically connected to the at least one electrical component. In an embodiment, a pair of glasses can perform hearing enhanced functions to enhance audio signals for the user to hear. One embodiment is configured to be a headset. One embodiment has audio and / or textual output capabilities for a user to communicate in different ways depending on configuration, user preferences, prior history, etc. In one embodiment, the communication between users is achieved by short audio or textual messages.

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.

An acoustic device is provided which comprises an audio system including: a microphone having a microphone output, and a processing circuit having a wakeup input, an audio input, and an audio output, the audio input coupled to the microphone output. In at least one example, the acoustic device further comprises an acoustic sensor separate from the microphone, the acoustic sensor having a sensor output. In at least one example, the acoustic device further comprises a wakeup circuit having a sensor input and a wakeup output, the sensor input coupled to the sensor output, wherein the wakeup output is coupled to the wakeup input.

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.

Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

The invention belongs to the technical field of audio equipment design, and relates to a microphone for AI voice cloning based on pseudo tone of acoustic shielding, and the microphone comprises an audio collection module which is used for collecting an environment audio signal in real time; the analog-to-digital converter is used for converting the environment audio signal into a digital signal; the feature extraction module is used for performing frequency domain conversion on the digital signals and extracting frequency spectrum features and tone features; the acoustic shielding calculation module is used for calculating a shielding threshold value of the environment audio signal and determining a frequency band which cannot be heard by human ears in the environment audio signal according to the shielding threshold value; the pseudo tone generation module is used for generating pseudo tones based on the extracted tone features; the synthesis module is used for replacing a corresponding frequency band of the environment audio signal with the generated pseudo tone to form a synthesized audio; and the audio output module is used for outputting the synthesized audio. According to the invention, the pseudo tone can be generated in real time in the audio recording process, and it is ensured that a user is always protected during audio recording.

An electronic device includes a chassis, a voice receiving unit, an acoustic membrane, and a sealing mechanism. The chassis has a through hole. The sealing mechanism includes a sealing body and a sealing ring. The sealing body has an accommodation portion and a channel. One of two ends of the channel is in communication with the accommodation portion, and the other end has a first opening. The accommodation portion accommodates the voice receiving unit. The sealing ring is between the sealing body and the acoustic membrane, and has a penetration hole. The penetration hole, the through hole and the first opening correspond to each other. A ring hardness of the sealing ring is less than a body hardness of the sealing body.

The sound acquisition device comprises a sound acquisition element and further comprises a shell, a sound transmission channel is defined on the shell, the sound transmission channel comprises a first channel section and a second channel section, the first channel section and the second channel section are sequentially connected, and an included angle is formed between the first channel section and the second channel section. The sound collecting element is located in the first channel section, the second channel section is provided with a second sound inlet for external sound to enter, and the shell is further provided with a section of extending wall extending in the opening direction of the second sound inlet at the second sound inlet. The sound acquisition device has the advantages that the position of pressure fluctuation caused by airflow change can be far away from the position of the sound acquisition element due to the arrangement of the extension wall on the shell, so that airflow pressure fluctuation at the acquisition position of the sound acquisition element is effectively reduced, and the sound acquisition accuracy is improved.

The present invention discloses an audio recording and conversion method for a Bluetooth headset, specifically related to the field of audio recording and conversion. The present invention pairs and connects with the Bluetooth headset by turning on the Bluetooth function of the sound source device; subsequently, the recording process is started by pressing the "Recording" button on the headset, and the system guides the user to select the audio format and confirm the recording settings; during recording, the built-in microphone of the headset collects audio signals in real time, denoises and amplifies the collected audio signals, and then converts the processed audio signals into digital signals; the processed digital signals are embedded with watermarks, and the watermark evaluation value is calculated, and the watermark is verified by comparing the watermark evaluation value; then, the digital signals are converted into analog signals and reconstructed into audio signals, and finally, the system performs watermark detection on the reconstructed audio signals, and ensures the integrity and authenticity of the audio signal data by comparing the detected watermark information.

A microphone assembly for exterior use on vehicles having a housing base including at least one compartment, a PCBA (printed circuit board assembly) with at least one acoustic membrane, a sealing cover to at least partially overlap with the compartment is disclosed. The sealing cover has at least one opening with said opening at least partially aligned with said acoustic membrane. The assembly has a housing cover including a first grid at least partially aligned with said opening in the sealing cover, and a shielding cover. The shielding cover has a second grid at least partially aligned with said opening in the sealing cover. The first grid and the second grid are aligned in offset in order to provide a labyrinth path without straight passthrough passages from the exterior of the microphone assembly to the acoustic membrane.

Eyewear includes at least a temple having a cavity open to the outside through first and second holes positioned opposite each other along a transversal direction. The eyewear includes a speaker assembly mounted inside the cavity, which has a speaker operable to reproduce a sound, a speaker enclosure that houses the speaker and a processing unit configured to control the operation of the speaker. The speaker includes a diaphragm designed to move in order to produce the sound, the diaphragm dividing an internal volume of the speaker enclosure into front and back cavities that are open to the outside of the speaker enclosure respectively through front and back openings. The speaker assembly is mounted inside the cavity to create an air communication between the front opening and the first hole and between the back opening and the second hole. The back opening is positioned to directly face the second hole.

The invention relates to the technical field of wireless communication, in particular to an intelligent frequency band control method and system for a wireless microphone. The method comprises the following steps: acquiring audio signal data of a wireless microphone and environment sensing data, and carrying out preliminary calibration on audio signals to obtain a preliminary calibration environment audio data set; performing environment interference source identification on the initial calibration environment audio data set to obtain a spectrum interference identification report; acquiring real-time wireless spectrum resource allocation data; performing frequency band dynamic adjustment on the real-time wireless spectrum resource allocation data to obtain a dynamic optimization frequency band adjustment data set; constructing a frequency response compensation model based on the dynamic optimization frequency band adjustment data set; and adaptive allocation of wireless microphone communication frequency band resources is carried out through the frequency response compensation model, so that wireless microphone communication frequency band resource data is obtained and transmitted to a wireless microphone frequency band control platform. According to the invention, the signal quality of the wireless microphone and the effective utilization rate of frequency band resources can be obviously improved.

The present description relates to methods and systems for a medical dictation. In one example, a stethoscope includes a first microphone positioned to capture physiological sounds of a patient, a second microphone positioned to capture ambient sounds, one or more processors, and memory storing instructions executable by the one or more processors to: during a stethoscope mode, obtain a first signal from the first microphone and a second signal from the second microphone, process the first signal to capture a physiological sound signal, including performing noise cancellation on the first signal based on the second signal, and transmit the physiological sound signal to an external computing device and / or a speaker of the stethoscope; and during a dictation mode, obtain the second signal from the second microphone, process the second signal to capture a voice signal, and transmit the voice signal to the external computing device.