5359results about "Frequency/directions obtaining arrangements" patented technology

The invention discloses a motor fault detection method and system based on voiceprint recognition. According to the method, an annular microphone array is adopted to collect motor sound signals in a non-contact mode, a three-channel time-frequency data set is constructed through empirical mode decomposition (EMD) and a Mel-frequency cepstral coefficient (MFCC), fault diagnosis is carried out in combination with a CNN + ResNet network, and dynamic time warping (DTW) and CNN fusion matching is supported. The system comprises a preprocessing module, a fault template library and a matching algorithm, integrates wavelet denoising and multi-beam acquisition technologies, covers a frequency band of 50Hz-20kHz, can display a fault type and trend analysis in real time, and triggers secondary verification when the confidence coefficient is insufficient. According to the scheme, the anti-interference capability is improved through array signal processing, model parameters are optimized in combination with transfer learning, non-contact detection is achieved, the real-time performance and accuracy of fault diagnosis are remarkably improved, and the method is suitable for industrial motor health monitoring.

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.

A core module and an electronic device are provided. The core module comprises a housing component, a transducer device, a vibration panel, and a face attachment component. The housing component includes a core housing and a core cover plate covering an open end of the core housing. The core cover plate and the core housing cooperate to form a first cavity. The face attachment component and the housing component cooperate to form a second cavity. The transducer device is at least partially located in the first cavity. The vibration panel is at least partially located in the second cavity. The core cover plate is provided with a first avoidance hole allowing the vibration panel to be connected with the transducer device. The core module is provided with a channel connecting the second cavity with the outside of the core module.

A speaker apparatus including a housing; a slot loading loudspeaker inside the housing, the slot loading loudspeaker including a speaker unit, and a cover covering the speaker unit, and including an opening through which an acoustic signal from the speaker unit may be output; and a panel member disposed in a forward direction of the slot loading loudspeaker so that a gap is formed between the cover and the panel member, and an exposure area is formed between an edge of the housing and an edge of the panel member to expose the opening through the exposure area so that an acoustic signal outputted through the opening is transferred to an outside, wherein at least one of the cover or the panel member includes at least one blocking member for preventing at least a portion of the acoustic signal outputted through the opening from flowing into the gap.

A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

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.

An air-pulse generating device, a wearable sound device, bladeless fan, and an airflow producing method are disclosed. The air-pulse generating device includes a film structure, configured to be actuated to generate a plurality of air pulses at an ultrasonic pulse rate. The plurality of air pulses produces a net airflow toward one single direction.

The present invention provides a coaxial speaker including a fixing member, a magnetic circuit system, a voice coil, a first vibration system connected with the voice coil, and a second vibration system connected with the voice coil and arranged surrounding the first vibration system. The second vibration system locates coaxially with the first vibration system. The first vibration system includes a first diaphragm and a first support connecting the first diaphragm and the voice coil. The second vibration system includes a second diaphragm and a second support connecting the second diaphragm and the voice coil. The first diaphragm is used for generating treble sound. The second diaphragm is used for generating bass sound. Bass sound and treble sound form a full-range speaker, providing high-quality sound. Moreover, the assembly process is simplified by sharing voice coil, which improves assembly efficiency and reduces costs.

An air-pulse generating device includes a film structure including a first flap and a second flap opposite to each other. The film structure is actuated to operate at an ultrasonic frequency, and the air-pulse generating device produces a plurality of air pulses at an ultrasonic pulse rate. The first flap and the second flap are actuated to perform a differential movement to form an opening or a virtual valve. A slit is formed between the first flap and the second flap, and the opening or the virtual valve is formed because of the slit. The slit is formed as a zigzagging pattern on the film structure.

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 provides a noise reduction processing method and system for Bluetooth audio equipment, and the method comprises the steps: obtaining an audio signal, and generating spectrum distribution data based on the audio signal; according to the intensity value of each frequency band in the spectrum distribution data, extracting a frequency characteristic set of a multi-source interference component; for the frequency characteristic set, determining a dynamic fluctuation parameter corresponding to the frequency characteristic; adjusting the configuration of the filter according to the corresponding relation between the dynamic fluctuation parameter and the preset noise width; synchronously updating the time domain response parameter of the filter based on the amplitude change rate of the dynamic fluctuation parameter; calculating dynamic threshold upper and lower limits according to the time domain response parameters, and generating a dynamic threshold sequence; and performing gain compensation processing on the environment audio signal based on the dynamic threshold sequence to generate a noise reduction audio signal. Adaptive time domain response parameters and a dynamic threshold sequence are adopted, so that filter configuration can be adjusted in real time according to noise characteristics.

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.

Examples described herein relate to transitioning a playback session between portable playback devices such as “smart” headphones, earbuds, and handheld speakers with playback devices of a zone-based media playback system. Exemplary techniques facilitate continuity of playback when transitioning between locations (e.g., from at home to on-the-go or vice versa) or between listening paradigms (e.g., personal or out-loud playback of audio content). An example implementation includes detecting a swap trigger, determining the source playback device(s) and target playback device(s), and performing a playback session swap between the source playback device(s) and target playback device(s).

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.

The air-pulse generating device comprises a film structure, operating at an ultrasonic operating frequency, and a resonant chamber, formed on a side of the film structure. A resonance is formed within the resonance chamber. Due to the resonant chamber, the APG device has a peak on a frequency response of an acoustic property of the APG device at the ultrasonic operating frequency.

Provided is a speaker device, including a first frame, a first vibration system, a magnetic circuit system, a second frame, a second vibration system, a second voice coil, and a conductive member electrically connected to the second voice coil. The magnetic circuit system includes a lower splint, a first main magnetic steel, a first sub-magnetic steel, a pole core, a second main magnetic steel, and a second sub-magnetic steel. A magnetization direction of the first sub-magnetic steel is the same as that of the second sub-magnetic steel, a magnetization direction of the first main magnetic steel is the same as that of the second main magnetic steel, and the magnetization direction of the second main magnetic steel is opposite to that of the second sub-magnetic steel. Compared with the related art, the speaker device of the present invention has good acoustic performance.

A wall-mounted audio device is provided, including: a display component; a wall-mounted box body, wherein a front panel is provided with a rim region and a display region; the display component is arranged in the display region; a rear panel of the wall-mounted box body is provided with a wall connector; a speaker module, arranged inside the wall-mounted box body a sound output end of the speaker module faces the sound outlet hole; and an mesh. The mesh is detachably connected to the front panel of the wall-mounted box body. The display component is a liquid crystal display screen. The speaker module includes a speaker module housing and one or more transducers designed on the speaker module housing. The speaker module housing is provided with a bass enhancement port. The rear panel of the wall-mounted box body is provided with a wire receiving slot.

The invention relates to the technical field of sensor identification, in particular to a sound parameter determination method and system. The method comprises the following steps: obtaining sound multi-modal sensing data, and carrying out multi-dimensional feature vector combination to obtain a high-dimensional feature data set; performing feature mapping on the high-dimensional feature data set to obtain a sound response feature map; performing environment dynamic parameter adjustment strategy analysis based on the sound response feature map to obtain an environment dynamic parameter adjustment strategy; obtaining user sound control data, and performing user auditory preference analysis to obtain user auditory preference data; and according to the user auditory preference data, performing user personalized sound adjustment feedback optimization on the environment dynamic parameter adjustment strategy to obtain an optimized feedback sound dynamic adjustment strategy, and transmitting the optimized feedback sound dynamic adjustment strategy to the sound control platform to execute a parameter adjustment task. According to the invention, the problems of subjective errors and untimely response to environmental changes in traditional manual adjustment can be effectively avoided.

The invention discloses a method, a system and equipment for voice interaction inside and outside a vehicle and a storage medium, and relates to the technical field of intelligent cabins and human-computer interaction. Identifying an interaction trigger type (in-vehicle active request, out-of-vehicle passive request, or system active trigger) based on the perceived data and / or the vehicle event; determining a corresponding response permission strategy according to the trigger type, and determining whether to allow the system to respond based on the response permission strategy; and if so, calling a large language model to generate a voice text, synthesizing voice by combining external perception characteristics, and broadcasting the voice through a loudspeaker outside the vehicle or a sound box in the vehicle. According to the scheme, three types of interaction intentions including the in-vehicle active request, the out-vehicle passive request or the system active triggering are recognized, the large language model is called to generate the scene-adaptive voice text, and personalized voice synthesis and broadcasting are performed in combination with the external perception characteristics, so that the response efficiency, the expression naturalness and the object adaptability of the in-vehicle and out-vehicle voice interaction are improved.

Disclosed are a method, system, and apparatus of adaptive mission strategy optimization through ai-generated ambient recommendations for tactical command. According to one embodiment, the method includes simultaneously listening to multiple communications surrounding a wearer, amplifying a human speech detected in the multiple communications through a wearable microphone, and using an artificial intelligence model to generate a tactical recommendation to the wearer through a mobile device accessible to the wearer.

The invention relates to the technical field of sound beam control, and provides a sound beam forming pointing control method based on crowd recognition and related equipment thereof. The method comprises the following steps: performing time-space synchronous acquisition and calibration on multi-modal sensor data to obtain a time-space aligned original data set, and performing layered signal enhancement on the original data set to obtain a visual feature tensor and an audio feature tensor; dense crowd detection and tracking are carried out according to the visual feature tensor to obtain a target crowd space-time trajectory matrix, and sound source localization and beam weight calculation are carried out in combination with the audio feature tensor to obtain a beam forming weight vector; performing multi-constraint dynamic optimization in combination with environmental parameters in the multi-modal sensor data to obtain an adaptive control parameter set, and finally performing real-time beam forming in combination with a loudspeaker sequence to obtain a directional sound field output signal. According to the invention, target detection, sound source localization and dynamic adaptive control are combined on the multi-source data, so that the beam direction is dynamically adjusted.

The invention provides an audio processing method, communication equipment, a communication system and a readable storage medium, and the method comprises the steps: obtaining a to-be-processed audio signal; inputting the audio signal and a preset target attenuation coefficient into a first filter to obtain a first output audio signal with an echoing sound effect; inputting the first output audio signal and the target attenuation coefficient into a second filter to obtain a second output audio signal with a sound trailing phenomenon; respectively inputting the second output audio signal into a plurality of third filters to obtain a third output audio signal corresponding to each third filter; and superposing the plurality of third output audio signals to obtain a target output audio signal with a reverberation sound effect. According to the method, the reverberation time of each frequency band can be accurately controlled, the reverberation reality sense of the output audio is effectively improved, the reverberation effect is closer to the convolution reverberation effect, meanwhile, the requirement for the computing power is greatly reduced, the calculation amount of the algorithm is effectively reduced, and the application channel of the algorithm is effectively widened.

The invention relates to the technical field of audio signal processing, and discloses a sound box and sound effect intelligent adjusting method and system based on data analysis and a storage medium, and the sound box and sound effect intelligent adjusting method based on data analysis comprises the steps: constructing a user feature modeling engine, and extracting user auditory characteristic data; constructing an environment characteristic analysis engine, and collecting environment acoustic characteristic data; constructing a content feature extraction engine, and analyzing audio content semantic data; constructing a three-dimensional fusion optimizer, inputting the three feature vectors into an auditory scene fusion model, calculating an auditory experience score through tensor fusion operation, and solving an optimal sound effect parameter by applying a multi-objective optimization algorithm; constructing a parameter generation controller, and converting the optimal sound effect parameter into a specific audio processing parameter; according to the invention, the problem of mutual interference caused by traditional separated processing is solved, and the accuracy of sound effect adjustment and the user satisfaction are improved.

An audio system receives sounds from a local area with a set of sensors. The audio system analyzes the sounds to identify an amount of background noise in the environment. The audio system may also identify target sound in the environment, such as voice from a nearby speaker, to reinforce. The amount of background noise is then used to adaptively set a level of reinforcement for the target sound. In low-noise environments, the target sound may be minimally, or not reinforced, while in high-noise environments a reinforcement sound signal may be generated with the target sound so that a transducer array may generate reinforcing sounds for the target sound and increase its contrast with the noise in the environment. By applying the reinforcement dynamically, the system automatically adjusts for different sounds and environments.

------ Acoustic Enclosure and Method for Leveling an Acoustic Enclosure The present invention relates to an acoustic enclosure (20) comprising a loudspeaker (21) mounted flush on a front face of the enclosure (20), characterized in that at least one Helmholtz resonator (26) is formed on the front face of the enclosure (20) between each lateral edge of the front face and the loudspeaker (21), the at least one Helmholtz resonator (26) being dimensioned to have a band gap to block the propagation, in the plane of the front face of the enclosure (20), of acoustic waves in the useful frequency band of the loudspeaker (21). Figure to be published with the abstract: Figure 6a

An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

A loudspeaker assembly and a headphone are provided. The headphone includes a housing assembly, a bone-conduction core module, and an air-conduction core module, and an accommodation space. The bone-conduction core module is provided in the accommodation space and vibrates in a first vibration direction, the air-conduction core module is provided in the accommodation space, and the bone-conduction core module and the air-conduction core module are arranged along the first vibration direction and face each other. In the above manner, the present disclosure reduces the influence of the air-conduction core module on the vibration effect of the bone-conduction core module, increases the sound volume, and improves the bone-conduction effect of the loudspeaker assembly, thereby improving the sound quality effect of the loudspeaker assembly.

The present disclosure may provide a sound-producing device comprising a diaphragm and a housing. The housing may include a first sound guide hole and a second sound guide hole. The diaphragm may be disposed in the housing. A first acoustic cavity and a second acoustic cavity are located on both sides of the diaphragm. The first sound guide hole is acoustically coupled to the first acoustic cavity, and the second sound guide hole is acoustically coupled to the second acoustic cavity. The first sound guide hole includes a first hole part and a second hole part. The first hole part and the second hole part are connected, the first hole part is located at a first side wall of the housing, the second hole part is located at a second side wall of the housing, and the first side wall and the second side wall are two adjacent side walls of the housing.

A wearable audio output device (e.g., headphones) having an open design that allows ambient noise to pass to a listener without physically isolating the listener from a surrounding environment. The device may include an open earcup design that may partially or completely surround the listener's ear, and in some examples a portion of the listener's head may be uncovered by the open earcup. To improve comfort, the device includes a floating audio component configured to generate output audio in a direction of the listener's ear without contacting the listener's ear. To reduce audio leakage into the environment, the floating audio component may include two audio transducers and perform beamforming to target the output audio toward the listener's ear and cancel the output audio in other directions. For example, the beamforming may cause constructive interference in one direction and destructive interference in all other directions.