3133results about "Microphones" patented technology

An auditory operating system designed to facilitate context-aware, audio-based user interactions, particularly with artificial intelligence agents or applications. An auditory operating system shell serves as the primary interface, managing and coordinating multiple specialized agents that handle specific domains like music streaming, scheduling, or home automation. Using natural language processing, the auditory operating system shell identifies the appropriate agent or application for a user's command or query and ensures task execution and context preservation across interactions. The auditory operating system shell enforces privacy and stability by controlling agents' and applications' access to data and system privileges. The auditory operating system shell also supports dynamic context management, enabling seamless handoffs between agents when user requests span multiple domains. This auditory operating system shell may reduce the need for users to memorize specific wake words or commands, as the auditory operating system shell may determine the user's intent from speech or contextual cues.

A device capable of using image data for purposes of determining a location of a user and audio beam selection to isolate audio in the direction of the user. The beamforming / beam-steering may occur after determining the user's location in order to conserve computing resources that would otherwise have been spent determining beams for non-desired directions. The beamformed audio may be used for speech processing, a communication session involving the device, or other purposes.

A controller configured to control a sound producing module includes a volume controlling unit configured to determine a demodulation amplitude and a modulation amplitude corresponding to a target volume. The sound producing module comprises a driving circuit and an air-pulse generating device. The driving circuit generates a demodulation driving signal according to the demodulation amplitude and generates a modulation driving signal according to the modulation amplitude, so as to drive the air-pulse generating device. The air-pulse generating device produces sound via generating a plurality of air pulses at an ultrasonic pulse rate.

The invention discloses a tumble detection method, wearable equipment and a storage medium, relates to the technical field of tumble detection, and discloses a tumble detection method applied to the wearable equipment, the wearable equipment comprises an acceleration sensor and at least one other sensor except the acceleration sensor, the tumble detection method comprises the following steps: acquiring acceleration data acquired by an acceleration sensor based on a time sequence; if the acceleration data accord with the free falling acceleration performance in the first time period and accord with the impact rebound acceleration performance in the second time period, it is judged that a falling impact event is detected; in response to the falling impact event, obtaining a falling detection result determined based on detection data collected by other sensors; and if the tumble detection result is that the tumble event exists, triggering a tumble event response process. On the basis, when falling impact is detected, detection and identification are carried out based on other sensor parameters, so that the falling detection accuracy of impact caused by falling is improved.

The invention relates to the technical field of earphone noise reduction processing, and discloses a motorcycle riding earphone noise reduction method combining different noise reduction algorithms, comprising the following steps: acquiring an original audio signal, and distributing the original audio signal to each processing module; a preprocessing signal is generated, voice activity information is determined, and environmental noise energy information is calculated; according to the environmental noise energy or the auxiliary information, adaptively adjusting a high-low noise energy threshold value; the original signals are input into the traditional and AI noise reduction module in parallel to generate two paths of noise reduction signals; based on the environment noise, the voice information and the threshold value, performing weighted fusion to generate an output signal; and carrying out equalization and compression processing on the output signal, and driving the loudspeaker to output. According to the invention, comprehensive judgment on environmental noise energy information, voice activity information and riding speed is introduced, an output signal of a traditional noise reduction module is set to be zero in a high-speed voice-free environment, and AI noise reduction is combined, so that tone quality distortion of a traditional noise reduction algorithm in a noise environment is effectively avoided.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

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.

The invention relates to the field of aviation voice signal processing, and discloses a voice recognition compensation pilot earphone hearing protection method and system, and the method comprises the following steps: collecting multi-modal data, and separating a sound source through tensor decomposition; inferring a pilot state by using a dynamic network; performing context recognition and semantic evaluation on the attention target voice; and finally, dynamically modulating the sound field based on deep reinforcement learning, enhancing the voice in a personalized manner, and outputting after noise suppression. The system comprises a multi-mode perception data acquisition module, a sound source decoupling module, a pilot state inference module, a voice processing and semantic evaluation module and a sound field modulation and output module. According to the invention, high-fidelity speech extraction is realized through multi-modal perception and tensor decomposition; evaluating priority key information in combination with attention and semantics; and deep reinforcement learning and model prediction control are adopted to dynamically optimize the sound field, so that the voice recognition accuracy and the pilot information acquisition efficiency are improved.

The technology described herein is directed towards a metasurface arranged with unit cells for narrowband sound absorption, in which the unit cells are based on Helmholtz resonators that can have their resonant frequencies adjusted via MEMS actuators. A sound absorbing unit-cell is designed and constructed based on a general resonance frequency, and includes a neck portion and air chamber dimensioned to resonate close to the desired resonance frequency and thereby inverse phase cancel corresponding narrowband frequencies of incoming sound waves. A bimorph MEMS actuator and / or moveable part in the resonators, controlled by a controller, facilitates changing of the airflow the unit cells to adjust the resonant frequencies thereof, to cancel acoustic waves of different frequencies corresponding to noise, which can change over time. The unit cells can be distributed as part of a metasurface, which can be positioned proximate to a noise source to phase cancel the noise.

A hearing aid system comprises a hearing aid, and a portable auxiliary device’ adapted to establish a communication link between them. The hearing aid comprises a microphone providing an electric input signal, a signal processor, and an output unit. The auxiliary device comprises a microphone providing an auxiliary electric input signal, and a user control interface allowing a user to initiate a specific calibration mode of operation of the hearing aid system. The signal processor of the hearing aid is configured to receive corresponding time segments of said electric input signal and said auxiliary electric input signal to provide an estimate of an acoustic transfer function from said microphone of said auxiliary device to said microphone of said hearing aid. A method of operating a hearing aid system is further disclosed. The invention may e.g. be used in various applications related to own voice detection and estimation.

The invention discloses an acoustic system, an ear wearable device and a head-mounted intelligent equipment, the acoustic system comprises an audio device, a sound guide part and a sound output part, the sound guide part and the audio device are separately arranged, the input end of the sound guide part is arranged at the output end of the audio device, and the output end of the sound output part is arranged at the output end of the audio device. The sound output part is arranged at the output end of the sound guide part, the sound guide part is communicated with the audio device through a first sound hole, the cross section area of one end, close to the first sound hole, of the sound guide part is larger than that of the first sound hole, the sound output part is formed by at least one second sound hole, and the second sound hole is communicated with the first sound hole. The area of the cross section of one end, close to the sound output component, of the sound guide component is larger than that of the cross section of each second sound hole. The method can effectively improve the audio quality, and is applied to the technical field of head-mounted intelligent equipment such as intelligent glasses.

The embodiment of the invention discloses an ear clip type earphone stable to wear and an earphone device, and the earphone comprises a first housing which is provided with an audio cavity; the sound production assembly is arranged in the audio cavity; the second shell and the first shell are arranged at an interval; the connecting piece comprises a first end, a second end and a matching part; the first end and the second end are oppositely arranged; the matching part is positioned between the first shell and the second shell; the first end of the connecting piece is connected with the first shell; the second end of the connecting piece is connected with the second shell; the curvature of a first connecting section, located between the first end and the matching part, of the connecting piece is smaller than that of a second connecting section, located between the second end and the matching part, of the connecting piece.

A calibration method comprises adjusting an operating frequency of an air-pulse generating device, such that a sound pressure level (SPL) of the air-pulse generating device is within a specific range. The sound producing module comprises the air-pulse generating device configured to produce sound via generating a plurality of air pulses at a pulse rate corresponding to the operating frequency.

The invention relates to the technical field of earphone noise reduction, and discloses a bone conduction earphone intelligent noise reduction method and system based on multi-mode interactive fusion, and the method comprises the steps: collecting a multi-mode signal when a user uses a bone conduction earphone, carrying out the motion artifact elimination of the multi-mode signal, and obtaining a pure multi-mode signal; identifying a vocal cord vibration fundamental frequency and a formant of the pure multi-mode signal to perform noise fusion characterization processing on the bone conduction earphone to obtain a noise characterization matrix, and identifying a noise type of the bone conduction earphone by using the noise characterization matrix; analyzing a physiological fatigue index of the user, constructing noise reduction parameters of the bone conduction earphone based on the physiological fatigue index and the noise type, and constructing a noise reduction propagation link of the bone conduction earphone; and carrying out anti-howling adaptive filtering on the noise reduction propagation link by using the noise representation matrix to obtain a reverse sound wave, and carrying out noise reduction on the bone conduction earphone to obtain a noise reduction audio. According to the invention, the bone conduction earphone wearing experience of the user can be improved.

A sound system is shown in the form of a speaker with or without a speaker box. A front member may be used with a speaker support, with the speaker mounted to the speaker support, thus comprising a speaker assembly. The front member of the speaker assembly may include an elongate open portion with a top end and a bottom end and the speaker positioned adjacent to the elongate open portion between the top end and the bottom end. The speaker assembly may be coupled to a portion of the vehicle seat which is not obstructive to the positioning of a user in the vehicle seat. This combination may be used in a vehicle seat but also in a backpack, a chair, personal gaming seat, arcade gaming seat, gambling machine seat, bed, massage table or any other similar device.

The invention discloses a self-adaptive music playing intelligent headphone and a system thereof, and relates to the technical field of headphones, the headphone comprises a head frame and earmuffs, the head frame comprises an outer shell, a fixing frame is fixedly mounted in the outer shell, and slideways are fixedly mounted at two ends of the fixing frame; a movable connecting rod is arranged in the slide way, and a limiting groove and a limiting block are arranged at one end of the connecting rod; the earmuff comprises an earmuff shell, a connecting piece used for being connected with the connecting rod is fixedly installed in the earmuff shell, and a touch screen control module is installed on one side of the outer portion of the earmuff shell. According to the system, the mood state of the user is automatically recognized and the emotional state of the user is judged by integrating various sensors (such as a heart rate sensor and a skin electric reaction sensor), so that the accuracy of emotion recognition is improved, subtle emotion changes can be captured, more appropriate music recommendation is provided, and the use effect of equipment is improved.

The invention relates to the technical field of audio signal processing, and discloses a DRC control method for improving indoor riding reducibility of a motorcycle riding earphone, and the method comprises the steps: judging a macroscopic acoustic scene; analyzing a microcosmic acoustic state; and according to the scene, the state and the feedback correction vector, generating an audio processing parameter and applying the audio processing parameter to the audio signal. The method further comprises the steps of analyzing the output signal through an artifact monitor to detect and process artifacts, generating the correction vector and feeding the correction vector back to the parameter synthesizer, and forming closed-loop control. The system comprises a scene judgment module, a joint acoustic state encoder, a parameter synthesizer, a parameterized audio processing engine and an artifact monitor. Through combination of macroscopic scene judgment and microcosmic closed-loop self-adaption, the contradiction between noise reduction performance under high noise and high sound reducibility under low noise is solved, and the audio quality under different scenes is improved.

A system that performs wall direction estimation to determine a position of an acoustically reflective surface relative to a device. For example, the device may detect a direct sound received from an active sound source and a first reflection reflected from a nearby wall. Based on a time delay, the device can estimate a direction and / or distance to the wall. For example, the device determines a time differential of arrival (TDOA) between the direct sound and the first reflection for each microphone in a microphone array. The device combines TDOA values of the entire microphone array into a distance vector, which the device can compare to reference distance data to determine the best wall direction estimate. For example, reference distance values are fixed based on a geometry of the microphone array and the device can identify an azimuth value that is most similar to the distance vector.

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.

A system for hearing assistance includes an array of microphones, which are configured be mounted in proximity to a head of a user of the system and to output electrical signals in response to first acoustic waves that are incident on the microphones. A speaker is configured for mounting in proximity to an ear of the user and configured to output second acoustic waves in response to a drive signal applied to the speaker. Processing circuitry is configured to generate the drive signal by amplifying and filtering the electrical signals using a beamforming filter that emphasizes first sounds that impinge on the array of microphones within a selected angular range while suppressing second sounds that are spoken by the user.

The application provides a sound amplification sound column with adjustable vertical direction. The sound column comprises multiple coaxial cylindrical wave mid-high sound boxes arranged vertically, a sound box connecting mechanism, and a sound column direction adjusting mechanism. The sound box comprises a box with a trapezoidal vertical section and a coaxial cylindrical wave mid-high module installed on the panel of the box. The coaxial cylindrical wave mid-high module comprises a midrange horn, a high-frequency module, and a midrange phase plate. The high-frequency module is embedded in the midrange phase plate, and the midrange phase plate is placed on the midrange horn. The sound column is composed of multiple coaxial cylindrical wave mid-high sound boxes arranged vertically, and the curvature of the cylindrical wave is accurately controlled through the connecting mechanism and the adjusting mechanism to realize continuous adjustment of the vertical direction.

The invention discloses an ear wearable device and a head-mounted intelligent device, the ear wearable device is used for being worn on the auricle of a human body, the ear wearable device comprises a first body, a second body, a bending part and an audio device, the bending part is correspondingly located at the upper position of the auricle of the human body, and the audio device is arranged on the first body. The first body is connected to the first end, located on the front side of the auricle of the human body, of the bending part, the second body is connected to the second end, located on the rear side of the auricle of the human body, of the bending part, a first cavity is formed in the second body, the audio device is arranged in the first cavity, an audio output port is formed in the first body, and the audio output port is connected with the first body. The audio output port is communicated with the first cavity. According to the invention, the wearing comfort of the intelligent glasses can be improved, and the method is applied to head-mounted intelligent equipment such as the intelligent glasses.

The invention discloses a system capable of switching audio output between an earphone and a charging bin, and the system automatically switches audio output to corresponding equipment through detecting the state of the earphone in real time, thereby achieving seamless audio experience. According to the system, the connection sensing module is adopted, and the double Hall sensors and the three-contact PIN detection unit are combined, so that the position of the earphone is accurately sensed. And the control module rapidly switches the audio output path according to the detection result to ensure that the switching response time is short. The system realizes extremely low delay in the audio switching process by pre-loading audio data and quickly adjusting parameters, and a user hardly feels interruption. In addition, the system adopts dynamic link management and anti-interference design, so that the reliability of seamless experience is improved. According to the invention, seamless connection of audio output between the earphone and the charging bin is realized, the audio can be automatically switched to the corresponding equipment with extremely low delay no matter what a user takes out or puts in the earphone, the playing process is smooth and unobstructed, and intelligent and efficient use experience is brought to the user.

The present disclosure generally relates to interfaces and techniques for media playback on one or more devices. In accordance with some embodiments, an electronic device includes a display, one or more processors, and memory. The electronic device receives user input and, in response to receiving the user input, displays, on the display, a multi-device interface that includes: one or more indicators associated with a plurality of available playback devices that are connected to the device and available to initiate playback of media from the device, and a media playback status of the plurality of available playback devices.

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.

An airflow generating device includes a first cell and a second cell. The first cell is disposed within a first region and generates a first air pressure with a first polarity. The second cell is disposed within a second region and generates a second air pressure with a second polarity. The second polarity is opposite to the first polarity.

An image capture device includes a housing that defines a cavity and an aperture positioned within the cavity and an audio device disposed within the housing at a location of the aperture. The image capture device includes stanchions that extend outward from the cavity and a cover coupled to the stanchions. The cover is free of contact with the housing.

A loudspeaker includes a housing having a plurality of walls that together define an acoustic cavity. An electro-acoustic transducer is mounted to a front wall of the housing, and an axis of motion of the electro-acoustic transducer is offset from a centroid of the front wall. The loudspeaker is configured to be stacked with another loudspeaker having the same configuration in a first configuration such that the stacked loudspeakers radiate acoustic energy to produce an omnidirectional radiation pattern; and the loudspeaker is configured to be stacked with the other loudspeaker in a second configuration such that the stacked loudspeakers radiate acoustic energy to produce a cardioid radiation pattern. The loudspeaker includes a keying feature that does not interlock and thereby prevent stacking when the loudspeaker is in a first orientation relative to the other loudspeaker, and that interlocks to allow stacking when the loudspeaker is in a second orientation relative to the other loudspeaker.

An assembly is provided which includes a package and a printed circuit board. The package includes a housing bounding a region and an acoustic sensor within the region. The housing includes a base with a first hole. The sensor is configured to generate signals indicative of sound received by the sensor through the first hole. The printed circuit board is in mechanical communication with the base and includes a second hole aligned with the first hole such that sound received by the second hole propagates through the first hole to the sensor. The printed circuit board further includes an electrically conductive layer, at least a portion of which extends across the second hole and is configured to allow the sound to propagate through the second hole and to at least partially shield the region containing the sensor from electromagnetic interference.