2819results about "Microphones" patented technology

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 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.

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.

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 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.

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.

A first electrical signal is received from a first cardioid microphone capsule of a microphone assembly. The first cardioid microphone capsule is oriented to face in a first direction. A second electrical signal is received from a second cardioid microphone capsule of the microphone assembly. The second microphone capsule is oriented to face in a second direction that is opposite the first direction. A polar pattern is formed for the microphone assembly by converting the first electrical signal into a first audio signal and converting the second electrical signal into a second audio signal. A first weight is applied to the first audio signal and a second weight is applied to the second audio signal. The polar pattern is generated for the microphone assembly by summing the first audio signal and the second audio signal.

The invention provides an earphone, and relates to the technical field of wearable equipment. The earphone provided by the invention comprises a sound production part arranged at the front side of an ear part, a main body part arranged at the rear side of the ear part, and an ear hanging part connected with the sound production part and the main body part, wherein the sound production part comprises a first shell, and a loudspeaker unit, a microphone unit, a camera device and a voice processing unit which are arranged in an inner cavity of the first shell, a view hole is formed in the first shell, and the camera device is arranged at the view hole; the camera device is configured to be capable of shooting an area on the front side of a human face when the earphone is in a wearing state, and the voice processing unit is used for recognizing and processing voice of an earphone wearer. According to the earphone provided by the invention, more use requirements of a wearer can be realized, and the overall performance and the user experience of the earphone are improved.

An audio-enhanced furniture system comprises: a furniture assembly having a partially acoustically transparent material, such as an upholstery fabric or a perforated, rigid structure that is perforated in order to be more acoustically transparent; and a speaker system positioned within the furniture assembly, the speaker system including a speaker covered by the partially acoustically transparent material. The speaker is configured to be tuned to compensate for sound being emitted from the speaker through the perforated partially acoustically transparent material.

MEMS device (10) with a movable displacement element (12) and a drive device (14) for driving the movable displacement element (12), wherein the drive device (14) has at least one drive electrode (16), characterized in that the drive electrode (16) is segmented and at least one functional electrode (18) is provided for measuring the deflection of the displacement element (12).

The present disclosure provides an audio system, method, and system for facilitating machine operation. The machine includes an actuator that assists the tool in performing the task. In an example, an audio system is configured to receive an audio mixed signal of a signal generated by an audio source that includes at least one of a tool or an actuator that is performing a task. An audio source forming an audio mixed signal is identified by a relative position to each microphone of a microphone array that measures the audio mixed signal. The audio system is configured to extract an audio signal from an audio mixed signal generated by the identified audio source based on a correlation of spectral features in a multi-channel spectrogram of the audio mixed signal and directional information indicative of a relative position of the identified audio source. The audio system outputs the extracted audio signal to facilitate operation of the machine.

This disclosure provides methods, devices, and systems for audio signal mixing. The present implementations more specifically relate to mixing audio signals from a microphone array by performing fixed beamforming to generate beams, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and noise reduction units (NRUs) to reduce a noise component of each audio beam. The system also includes logic to calculate a signal characteristic of each reduced noise audio beam to determine, based on the signal characteristics, the reduced noise audio beams that include a speech component. The logic also generates a gain for each audio beam based on the selection, with the gains used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

Embodiments of this disclosure provide a device operation method and apparatus, and an electronic device. In the device operation method, after obtaining a vibration signal collected by a bone conduction microphone connected to the electronic device, the electronic device extracts a vibration signal feature of the vibration signal, and then compares the extracted vibration signal feature with a control feature stored in the electronic device. When a comparison result is that the extracted vibration signal feature matches the control feature, the electronic device performs an operation corresponding to the control feature. The vibration signal collected by the bone conduction microphone may be a bone conduction audio signal, and the bone conduction audio signal is skin vibration triggered when a nasal cavity and / or a throat of a user who uses the electronic device make / makes a specific sound.

Aspects of transducers with feedback transduction are described. One aspect is a transducer system comprising an operational amplifier having an inverting input, a non-inverting input, and an output. The transducer system also includes a piezoelectric microelectromechanical system (MEMS) transducer having a first node and a second node, wherein the first node is coupled to the inverting input of the operational amplifier, and wherein the piezoelectric MEMS transducer is configured to generate an electrical signal across the first node and the second node in response to a signal incident upon the piezoelectric MEMS transducer. The transducer system also includes an attenuator having an input and an output, wherein the input of the attenuator is coupled to the output of the operational amplifier, and wherein the output of the attenuator is coupled to the second node of the piezoelectric MEMS transducer.

Systems can include an acoustic sensor array configured to receive acoustic signals, an illuminator configured to emit electromagnetic radiation, an electromagnetic imaging tool configured to receive electromagnetic radiation, a distance measuring tool, and a processor. The processor can illuminate the target scene via the illuminator, receive electromagnetic image data from the electromagnetic imaging tool representative of the illuminated scene, receive acoustic data from the acoustic sensor array, and receive distance information from the distance measuring tool. The processor can be further configured to generate acoustic image data of the scene based on the received acoustic data and received distance information and generate a display image comprising combined acoustic image data and electromagnetic image data. The processor can determine depths of various acoustic signals within a scene and generate a representation of the scene the shows the determined depths, including floorplan and volumetric representations.

A MEMS audio device includes a first wafer having a top with a first cavity and a bottom with a vent hole coupled to the first cavity, wherein the bottom having first contacts, a second wafer disposed upon the first wafer having a flexible material layer disposed above the first cavity, a third wafer disposed upon the second wafer having physical contacts coupled to the second wafer, wherein the third wafer includes a second cavity disposed above the flexible material layer, a wiring wafer disposed below the first wafer having a second vent hole coupled to the first cavity, wherein the wiring wafer having second contacts coupled to the first contacts, and wherein the flexible material layer forms a diaphragm for the MEMS audio device.