224 results about "Audio equipment" patented technology

In some aspects, a user equipment (UE) may send, to an audio device, audio data associated with an audio playback. The UE may send, to a media player user interface associated with a wearable device, an update associated with the audio playback. The UE may receive, from the wearable device, a message indicating that a display screen associated with the wearable device is turned off. The UE may send, to the audio device, an instruction associated with a remote action, wherein the remote action is associated with the audio playback. The UE may refrain from sending, to the media player user interface associated with the wearable device, an update associated with the remote action based at least in part on the message indicating that the display screen is turned off. Numerous other aspects are described.

Apparatus and methods digital audio bus reliability are disclosed. In certain embodiments, a digital audio system includes a plurality of audio devices connected by a first digital audio chain in a clockwise direction and a second digital audio chain in a counterclockwise direction. The first digital audio chain and the second digital audio chain run concurrently, and a controller selects which audio chain to operate at a given time for audio connectivity. For example, the controller can initially select the first digital audio chain to provide audio connectivity, but transition selection from the first digital audio chain to the second digital audio chain in response to detecting a node failure in the first digital audio chain. Thus, the system is tolerant to node failures while maintaining system connectivity.

A wearable audio device is provided. The wearable audio device includes a first capacitive proximity sensor. The first capacitive proximity sensor is configured to generate a raw proximity signal. The wearable audio device further includes a second capacitive proximity sensor. The second capacitive proximity sensor is configured to generate a raw reference signal. The wearable audio device further includes a controller. The controller is configured to calculate a proximity threshold. The proximity threshold is calculated based on a product adjustment ratio, a bias value, and the raw reference signal. The controller is further configured to determine an on-head status. The on-head status is determined based on the raw proximity signal and the proximity threshold.

The application discloses an audio effect evaluation method and device, equipment and a storage medium. The application inputs a preset reference sound source into an audio device to be tested for playing, wherein the preset reference sound source is a sound source determined by statistical analysis of sound sources, in combination with a parameter corrected according to a preset evaluation result and a peak factor corresponding to a music playing platform; collects audio to be tested output by the audio device to be tested when playing the preset reference sound source; determines a relative frequency response and an estimated loudness value according to a spectrum corresponding to the audio to be tested and a spectrum corresponding to the preset reference sound source; and generates an audio effect evaluation result of the audio device to be tested according to the relative frequency response and the estimated loudness value. The application evaluates the audio effect of the audio device to be tested by using the reference sound source constructed in advance, solves the problem of traditional sweep frequency signal tuning deviation, greatly reduces the number of repeated trial and error, and significantly improves the efficiency, accuracy and reliability of audio product tuning work.

This disclosure provides a control device, method, and audio equipment for an animated sound grille. The control device controls the movement of N sound grilles to achieve the animation effect. The N sound grilles are grouped into M groups, each group comprising m sound grilles, where N≥4, M≥2, and m≥1. The control device includes: N motors, each sound grille controlled by one motor; M node controllers, each node controller controlling one of the M groups of sound grilles; and a main controller, communicating with the M node controllers via i buses, each bus connecting at least one node controller, where i≥2. The main controller is configured to: send action commands to designated node controllers of designated groups of sound grilles within the M groups of sound grilles using a time-division multiplexing strategy based on the start time of the M groups of sound grilles. The designated node controllers control the motors of the designated group of sound grilles according to the action parameters of the action commands.

1. Name of the product in this design: Audio System (tide Yun). 2. Purpose of this design: This design is intended for use in audio equipment. 3. The key design features of this product are the combination of shape and pattern. 4. Images or photos that best illustrate the design key points: composite state diagrams.

1. Name of the product in this design: Audio System (K901). 2. Purpose of this design: This design is used for the output of audio signals. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The present invention relates to an audio device with a housing, a microphone arm and a detector for determining an orientation of the microphone arm relative to the housing. The invention may advantageously be applied in audio devices having a microphone arm, such as headsets, earphones, hearing protectors and other audio devices used for sound pickup.The audio device comprises a housing, a microphone arm that is mechanically and movably connected to the housing, a microphone unit arranged at or in the microphone arm and configured to pick up sound from the environment of the audio device and provide a corresponding microphone audio signal, a signal processor arranged at or in the housing and configured to provide electric power to the microphone unit, and a flexible substrate extending in a length direction between a first end mechanically connected to the housing and a second opposite end mechanically connected to the microphone arm.The flexible substrate is configured to bend in dependence on the microphone arm moving relative to the housing and comprises one or more electrical conductors electrically connecting the microphone unit and the signal processor to convey the electric power from the signal processor to the microphone unit and / or the microphone audio signal from the microphone unit to the signal processor.The flexible substrate further comprises a bend sensor that is configured to change the value of an electric property in dependence on a bending of a first portion of the flexible substrate, and the audio device further comprises a bend detector electrically connected to the bend sensor and configured to estimate an orientation of the microphone arm relative to the housing in dependence on the value of the electric property of the bend sensor.Arranging the bend sensor at or in the flexible substrate may provide a both reliable and cost-effective estimation of the orientation of the microphone arm.

A noise injury relief system is provided in the form of a portable handheld device configured for private therapeutic audio playback. The device includes an enclosure sized for pocket carry and having a front face with user controls and a cartridge insertion opening configured to receive a vaping cartridge. The front face includes an eleven-position slide switch for independently selecting internally stored recordings corresponding to eleven bells and a twelve-position slide switch positioned adjacent thereto for independently selecting twelve musical notes. An additional switch permits selection among different musical instruments for playback of the musical notes, and a volume control knob enables user adjustment of audio output. Internal audio signals are processed through a signal generator and transmitted wirelessly to a user audio device for private playback. The system is powered by an internal power supply and is configured for repeated use.

This utility model discloses a speaker stand with easily adjustable counterweight. This speaker stand aims to solve the technical problem in existing technologies where increasing the weight of the speaker stand is inconvenient, causing additional vibrations through resonance, leading to sound distortion and affecting sound quality. It includes a base, a hollow column fixedly connected to the top surface of the base, a placement plate detachably connected to the top surface of the hollow column, a cable winding assembly inside the hollow column, and two symmetrically arranged extension support assemblies on the bottom surface of the placement plate. The cable winding assembly includes a partition fixedly connected to the inner wall of the hollow column. This utility model increases the weight of the speaker stand by injecting sand and gravel particles into the counterweight cavity. Appropriate counterweight can change the natural frequency of the stand, making it less likely to resonate with the vibration frequency generated by the audio equipment.

The application discloses a speech recognition method, device, equipment and medium, and relates to the technical field of speech processing. The method is applied to an audio device, the audio device comprises at least one microphone, and the method comprises the following steps: acquiring an audio signal collected by the microphone in a first sampling period; determining a far-field probability value and a near-field probability value of an audio sub-signal corresponding to the audio signal in each second sampling period according to the audio signal, wherein all the second sampling periods constitute the first sampling period; determining a far-near field recognition result of the audio signal according to the far-field probability value and the near-field probability value of the audio sub-signal corresponding to the audio signal in each second sampling period; and performing speech recognition processing matched with the far-near field recognition result on the audio signal. The method can not only realize speech recognition on the audio signal collected by the microphone, but also realize accurate distinction between the far field and the near field of the audio signal.

1. Name of the product in this design: Audio System (BH004). 2. Purpose of this design: For playing sound. 3. The key design features of this product are the combination of shape and pattern. 4. The image or photograph that best illustrates the design's key points: a 3D model.

The audio device includes a receiver 101 configured to receive an audio signal from which an audio beamformer 103 generates a beamformed output audio signal. The beamformer 103 includes a signal processor 201 that applies signal processing to the audio signal. Signal processing includes spatial beamforming and spatial decorrelation, which is often adapted spatial decorrelation. A decorrelation adapter 203 adapts decorrelation parameters depending on the audio signal, and a beamform adapter 205 adapts beamform parameters depending on the beamformed output audio signal. An equalizer 105 applies frequency equalization to the beamformed output audio signal to generate an output signal, and an equalization adapter 107 adapts frequency equalization depending on the beamform parameters, typically also depending on the decorrelation parameters. This approach provides improved fidelity to the desired sound source and attenuates interfering sound sources.

The application discloses a Bluetooth earphone system, which comprises a charging bin and at least one Bluetooth earphone body accommodated in the charging bin, and a separable electric connection structure is arranged between the charging bin and the Bluetooth earphone body; the charging bin comprises a Bluetooth audio module, an analog audio interface module and a charging interface module; the Bluetooth audio module is integrated with an analog-digital conversion module and a wireless communication module; the charging interface module is integrated with a digital audio interface module connected with the Bluetooth audio module; the analog audio interface module and the digital audio interface module are connected with a non-Bluetooth audio device through signal lines; analog audio signals output by the non-Bluetooth audio device are converted into digital signals, and / or digital audio signals output by the non-Bluetooth audio device are processed by the digital audio interface module, and then the analog audio signals and the digital audio signals are wirelessly transmitted to the Bluetooth earphone body by the Bluetooth audio module. Thus, a user wearing the Bluetooth earphone body can realize wireless monitoring of the non-Bluetooth audio device, the operation is simple, and user experience is improved.

This utility model discloses an in-vehicle audio transmission device and an in-vehicle audio equipment based on an A2B bus. The in-vehicle audio transmission device based on the A2B bus includes a slave module, which comprises multiple slaves corresponding to multiple audio channels. Each slave is configured to receive signals from a master module via the A2B bus. The multiple slaves include at least a first slave and a second slave. The first slave includes a first A2B chip and a first DSP chip, and the second slave includes a second A2B chip and a second DSP chip. The first A2B chip includes a clock signal output port, and the first DSP chip, the second A2B chip, and the second DSP chip each include a clock signal receiving port. Each clock signal receiving port is electrically connected to the clock signal output port of the first A2B chip. This utility model expands the number of audio channels while ensuring data synchronization across multiple channels.

1. Name of the product in this design: Audio System (BH003). 2. Purpose of this design: For playing sound. 3. The key design features of this product are the combination of shape and pattern. 4. The image or photograph that best illustrates the design's key points: a 3D model.

This application provides example device networking and audio channel configuration methods, applied to a system including a first electronic device and at least one audio device. One example method includes sending, by an audio device of the at least one audio device, a networking packet in response to detecting a first gesture. The first electronic device receives the networking packet. The first electronic device establishes a correspondence between the audio device corresponding to the networking packet and an audio channel indicated by a direction corresponding to the first gesture, the established correspondence allowing the audio device corresponding to the networking packet to be configured to play audio of the audio channel indicated by the direction corresponding to the first gesture. The first electronic device establishes a networking connection to the audio device.

Embodiments herein relate to ear-wearable devices. In an embodiment, an ear-wearable device is included having a housing is included having an aperture end; and an ear canal end defining an acoustic outlet; a receiver within the housing; an acoustic tube between the receiver and the acoustic outlet; a foreign material barrier disposed over a portion of the acoustic tube; and a tip portion disposed at the ear canal end, wherein the tip portion is configured to move between a first position and a second position; wherein a movement of the tip portion from the first position to the second position is configured to move foreign material in a direction away from the acoustic tube. Other embodiments are also included herein.

In one embodiment, a method includes, while a first user wears a vocational mask to perform a task, receiving, at a processing device, a first data feed; transmitting, via network interfaces of the vocational mask, the first data feed to processing devices of a computing device, and the first data feeds are presented via a display, emitted by an audio device, or produced via a haptic device; receiving, from the computing device, second data feeds pertaining to instructions for performing the task, wherein the second data feeds are received by the processing device, and the second data feeds are presented via a virtual retinal display, emitted by an audio device, or produced via a peripheral haptic device; and storing, via memory devices communicatively coupled to the processing device, the first data feeds and / or the second data feeds.

1. The name of the design product: sports bone conduction earphone (L10). 2. The use of the design product: as an earphone to play audio equipment. 3. The design points of the design product: in shape. 4. The picture or photo that best indicates the design points: perspective view 1.

The application provides a smart networking method of a Bluetooth audio device, which comprises the following steps: S100, one or more Bluetooth audio devices enter a pairable state; S200, each Bluetooth audio device starts BLE broadcasting to send a BLE broadcasting frame, starts BLE scanning, and determines the number of same type BLE broadcasting frames in the environment according to the content of the scanned BLE broadcasting frame; S300, the number of same type BLE broadcasting frames is used to determine the pairable mode of the Bluetooth audio device; and S400, the Bluetooth audio device is paired and networked according to the pairable mode determined in step S300. The application also provides a Bluetooth audio device and a storage medium. The method, the Bluetooth audio device and the storage medium have the technical effects of improving user experience.

An audio synchronization control method, apparatus, system, chip, and medium are disclosed. The method includes: in an audio playback state where audio data frames sent by an audio host are received, acquiring sampled data, i.e., the clock jitter of the starting frame of the audio data frame; determining the total clock jitter in the current audio playback state for each sampled data of the starting frame; obtaining a corresponding predicted value for the sampled data of the next starting frame based on the sampled data and filter coefficients corresponding to the current adaptive filter; updating the filter coefficients based on the prediction error corresponding to the predicted value; and determining the clock jitter prediction correction amount of the phase-locked loop (PLL) based on the prediction error and the total clock jitter, and adjusting the PLL clock. Through the learning of the adaptive filter during operation, the clock jitter of the audio host at the next moment can be tracked and predicted in real time, and the clock of the audio device can be dynamically adjusted, resulting in high accuracy of clock synchronization after correction and effectively avoiding audio transmission asynchrony.

This application provides an open-type audio device, a sound field control method, and related apparatus, relating to the field of acoustic technology. The device includes an acoustic cavity and multiple sound emitters, sensing components, and control components fixed within the acoustic cavity. The acoustic cavity has a first sound leakage hole and a second sound leakage hole, used to determine the initial distribution and control direction of the external sound field, respectively. The sensing components are used to acquire the target location of an external person. The control components adjust the driving voltage of the sound emitters according to the target location to adjust the external sound field distribution, ensuring the target location is at the lowest possible sound leakage position. By fixing multiple sound emitters within the acoustic cavity and providing the first and second sound leakage holes, combined with the synergistic effect of the sensing and control components, this application dynamically adjusts the external sound field distribution according to the target location of an external person, significantly reducing sound leakage at the target location, thereby minimizing interference with external personnel and protecting user privacy.

The present invention relates to a vehicle-internal audio device (100; 100'; 100''), comprising: - at least one microphone, in particular analog or digital (10), for recording -- engine noise (220) caused by at least one engine (210) associated with the vehicle (200), in particular the drive engine, as well as -- from non-engine noise caused by at least one source other than the engine (210), in particular independent of the engine (210), especially driver and / or passenger voices, tire noise, wind noise and other ambient noise; - at least one ANC unit (40) designed for adaptive noise suppression, by means of which the engine noises (220) detected by the microphone (10) can be processed, in particular suppressed; at least one autoencoder unit (30) upstream of the ANC unit (40), by means of which the motor noises (220) can be distinguished and / or separated from the non-motor noises.

This application discloses a waveguide optimization method, waveguide optimization device, and medium for audio devices, relating to the field of audio processing technology. The disclosed waveguide optimization method for audio devices includes: integrating a verified physical model of the waveguide to be optimized into a full-device simulation model of the target audio product; determining the energy ratio between reflected sound and direct sound at the receiving point when the full-device simulation model simulates sound wave propagation; and outputting the physical structure information of the waveguide if the energy ratio is greater than or equal to a preset energy ratio. This simulation optimization method ensures the compatibility of the sky or surround waveguide structure with the target audio product, improves the effectiveness and reliability of waveguide development and testing, eliminates the need for frequent reliance on costly anechoic chamber environments, and provides precise and economical technical support for optimizing the channel performance of audio devices.

1. The name of the design product: sound (M3808). 2. The use of the design product: for playing audio equipment. 3. The design features of the design product: in shape. 4. The picture or photo that best indicates the design features: perspective view 1.

Audio equipment (20) comprising: - a printed circuit board (23) on which at least one electronic component (24) is mounted; - a loudspeaker (27); - a shielding cover (31) arranged to protect the electronic component, the shielding cover having an outer face (32) positioned opposite the diaphragm (43) of the loudspeaker (27); the outer face (32) comprising at least one curved surface (37) extending from a central portion (38) of the outer face (32) to a contour (39) of the outer face, the shielding cover (31) being thus arranged to diffract acoustic waves produced by the loudspeaker diaphragm during operation so as to reduce unwanted vibrations of the printed circuit board (23) and the shielding cover. Figure 6

Some disclosed methods involve causing one or more loudspeakers in an audio environment to emit sound and receiving microphone signals from one or more microphones in the audio environment corresponding to an acoustic response of the audio environment to the emitted sound. Some disclosed methods involve detecting, by the control system, a change of the acoustic response of the audio environment. Some disclosed methods involve changing one or more aspects of media processing for media played back by one or more devices in the audio environment based, at least in part, on the change of the acoustic response of the audio environment. Alternatively, or additionally, some disclosed methods may involve changing the lighting in the environment, changing a gain of one or more microphone signals, locking or unlocking one or more devices, or combinations thereof, based at least in part on the change of the acoustic response.

This invention provides an audio dynamic range control circuit and an audio device. The audio dynamic range control circuit includes an audio zero-crossing detection circuit, a preprocessing circuit, a first control circuit, a first gain selection circuit, a gain multiplication processing circuit, a second control circuit, a third control circuit, and a second gain selection circuit. The audio dynamic range control circuit provided by this invention achieves an effect similar to the existing RMS algorithm by performing absolute value processing on the linear domain sampling data of the audio signal, while simplifying the hardware circuit implementation. Simultaneously, by detecting the zero-crossing position of the audio signal through the audio zero-crossing detection circuit, and updating the gain value at subsequent zero-crossing positions when a sampling point with high energy is detected, the nonlinearity of the output signal can be greatly improved and auditory discomfort can be eliminated, effectively improving audio quality.

1. The name of the design product: sound (S2873). 2. The use of the design product: for playing audio equipment. 3. The design features of the design product: in shape. 4. The picture or photo that best indicates the design features: perspective view 1.