1206 results about "Audio equipment" patented technology

The invention discloses an intelligent predictive maintenance system for an audio equipment fault, and the system comprises a data collection module which collects the internal sensor data during the operation of audio equipment, and outputs an audio signal feature parameter, an external environment parameter, and a historical operation log; the feature preprocessing module is used for carrying out standardization and noise reduction processing on the multi-source data based on a multi-modal feature fusion algorithm; the health state evaluation module outputs a health state evaluation result of the audio equipment in real time through a hybrid analysis model combining a convolutional neural network, a long and short-term memory network and an attention mechanism; the fault risk prediction module is used for performing real-time fault risk prediction according to the evaluation result and generating predictive maintenance decision parameters; and the maintenance decision and early warning module is used for outputting fault early warning information according to the prediction parameters and automatically generating maintenance operation suggestions when the early warning level reaches a preset condition. According to the invention, the operation reliability of the audio equipment can be effectively improved, and intelligent prediction and advanced maintenance of faults are realized.

The invention discloses an Internet of Things alarm audio call scheduling method and system, and relates to the technical field of Internet of Things intelligent decision, and the method comprises the steps: receiving an alarm request message from an Internet of Things sensing terminal, and carrying out the alarm audio call scheduling according to a device identifier and a region code in the alarm request message; collecting and generating a multi-dimensional information report of real-time environment noise data, audio playing equipment state information and user state information; according to the multi-dimensional information report, generating a dynamic priority score of the alarm request message by querying a preset strategy library; and sorting the dynamic priority scores of the alarm request messages to be processed, and generating a preliminary scheduling strategy for suggesting playing equipment and broadcasting parameters for the alarm request message with the highest priority. According to the method, high-fidelity simulation is performed through the acoustic digital twin model to generate the intelligibility index report, the optimal strategy is selected based on the simulation result to generate the execution instruction, and the execution instruction is issued to the target audio equipment, so that full-process intelligent scheduling from multi-dimensional dynamic perception to acoustic effect optimization is realized.

Various implementations include approaches for voice control in audio devices. In some cases, a method includes: listening, using at least one audio capture device, for user input to control at least one attribute of an audio device; routing the user input through a machine learning (ML) model to determine a control action for the at least one attribute based on the user input; and causing the determined control action to be performed, wherein the ML model need not have been pre-trained with the user input to determine the control action for the at least one attribute of the audio device.

The present disclosure generally relates to user interface for audio routing. In some examples, a computer system detects an audio device and displays a selectable option in accordance with a determination that a set of criteria is met. The selectable option, when selected, enables the computer system to connect to the audio device and transmit audio data to the audio device.

An interactive doll can include an audio device. The interactive doll can be associated with a computing device. In some embodiments, the computing device includes an interactive doll app associated with the interactive doll. In some embodiments, the interactive doll is customizable. In some embodiments, the doll utilizes AI and / or machine-learning.

An audio device is provided. The audio device comprises a controller and an acoustic transducer. The controller is configured to receive an input audio signal. The controller is further configured to generate, via a loudness detector, a loudness level of a first portion of the input audio signal. The controller is further configured to identify a compression curve of a plurality of compression curves corresponding to a user volume setting. The controller is further configured to adjust, via an audio compressor, a second portion of the input audio signal based on the loudness level and the compression curve to generate an output audio signal. The loudness level is determined according to LUFS model.

The invention relates to the technical field of audio equipment testing, and discloses a parallel burning and Bluetooth testing frequency calibration method of a special chip for audio equipment. The method comprises the following steps of: establishing an independent serial communication channel with a plurality of chip chips through a computer control unit, and synchronously transmitting a firmware data packet to each chip storage unit; the high-precision radio frequency detection device is started to scan Bluetooth signal frequency offset values of all chips, and detection data is fed back to the computer control unit in real time; screening unqualified chips according to a preset frequency offset threshold, starting a dynamic frequency calibration program for the unqualified chips, and modifying internal capacitance parameters of the chips through a serial communication channel; repeatedly executing the frequency offset detection and capacitance parameter adjustment step until the frequency offset values of all the chips reach a standard range; and after completing the frequency offset calibration, executing a multi-channel Bluetooth performance test and outputting a test result report. According to the method, the chip firmware burning and Bluetooth test frequency calibration efficiency can be improved, the chip performance is guaranteed, and the method is adaptive to the batch production requirement.

The utility model relates to the technical field of audio equipment, in particular to an anti-eavesdropping external device and system. Comprising a shell, an audio control module, a first connection module, a quick charge control module and a second connection module, the audio control module comprises a plurality of built-in audios, the audio control module is connected with an external terminal through the first connection module so as to obtain the audio control authority of the external terminal, the second connection module is connected with the input end of the fast charging control module, and the output end of the fast charging control module is connected with the first connection module so as to fast charge the terminal; the audio control module is used for controlling the input of the audio signal of the terminal equipment, so that the terminal equipment cannot record the sound around the terminal, the terminal equipment can be effectively prevented from malicious eavesdropping and collecting the sound and conversation content around the terminal, and the privacy of a user is effectively protected; the anti-eavesdropping external device not only can prevent terminal eavesdropping, but also can quickly charge the terminal, and is higher in practicability.

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.

The invention relates to the cross technical field of computer-aided engineering and audio equipment design, discloses a circuit design simulation-based disc player design optimization method and system, and aims to solve the problem that the existing disc player design depends on physical prototype trial-manufacture and subjective listening evaluation and lacks multi-physical domain co-simulation capability. The method comprises the following steps: constructing a unified multi-physical domain model comprising a mechanical transducer, an electromechanical transducer and an electronic circuit sub-model; bidirectional coupling co-simulation based on the time sequence is executed, and an output electric signal is generated; extracting objective audio performance indexes such as total harmonic distortion and signal-to-noise ratio; and carrying out multi-objective optimization on the key design parameters by utilizing a genetic algorithm, and outputting a Pareto optimal scheme. The system comprises a model construction module, a co-simulation module, a performance index extraction module and an optimization module. Through high-fidelity full-link simulation and automatic optimization, the design efficiency and the audio fidelity are remarkably improved, and the research and development cost is reduced.

The invention relates to the technical field of integrated development debugging, and discloses a multi-protocol debugging communication module initialization method and system, a storage medium and equipment, and the method comprises the steps: scanning all connected debugging target equipment, carrying out the weighted evaluation of the scanning priority of the calculation equipment, and constructing a protocol capability matrix of each equipment; pre-distributing communication channels by adopting an optimization algorithm according to the protocol capability matrix, monitoring network environment indexes, and controlling a dynamic decision protocol switching opportunity through fuzzy logic; hot plug detection and equipment health degree evaluation are implemented, and when connection abnormity is detected, a multi-stage recovery strategy is triggered step by step to process the abnormity; monitoring an interface calling success rate and response time, analyzing and predicting an interface state based on a time sequence of historical data, and executing a self-adaptive repairing action of the interface; according to the method, the hardware capability characteristics of the audio equipment are detected, initialization parameters are optimized based on equipment capability scores, and meanwhile, a DHCP address allocation strategy is dynamically managed, so that the unified management of adaptive initialization of various communication protocols is realized.

Various implementations include approaches for device control and / or sound source selection in audio devices. In some implementations, an audio device includes: an electro-acoustic transducer for providing an audio output; a set of microphones for detecting ambient sounds; and a processor coupled with the electro-acoustic transducer and the set of microphones, the processor configured to: evaluate microphone signals from the set of microphones to identify classes of sound sources in the ambient sounds; and adjust output of at least one class of the ambient sounds relative to another class of ambient sounds based on a user input.

The invention provides a loudspeaker assembly, a vehicle-mounted loudspeaker system and a vehicle, and relates to the technical field of vehicle-mounted sound equipment, the loudspeaker assembly comprises a shell, a base, a rotation driving ring, a lifting driving ring and a mesh enclosure part, the base is detachably installed in the shell, and a placement cavity is reserved in the base; the lifting driving ring is vertically and slidably connected with the base, a first directional groove and a second directional groove which are communicated are formed in the outer ring surface of the lifting driving ring, the first directional groove obliquely extends in the vertical direction, and the second directional groove vertically extends and penetrates through one end of the lifting driving ring; the rotary driving ring is rotationally connected to the base, a protrusion and a gear ring are arranged on the rotary driving ring, and the protrusion is slidably connected into the first directional groove or the second directional groove; the mesh enclosure part is vertically connected with the rotary driving ring in a sliding mode, and the mesh enclosure part further vertically abuts against the lifting driving ring and is rotationally connected with the lifting driving ring. The problems that an existing lifting horn device is still large in occupied vertical space, complex in structure and not easy to install are solved.

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.

An apparatus supports smart assistant services with a plurality of smart service providers. The apparatus includes an audio device that receives a speech signal having a user utterance, captures the user utterance when the user utterance includes a user wake word, and sends the captured utterance to a backend computing device. The backend computing device replaces the user wake word with specific wake words associated with different smart service providers. The processed utterances are then sent to selected smart service providers. The backend computing device subsequently constructs feedback to the user utterance based on voice responses from the different smart service providers. The backend computing device then passes a digital representation of the feedback to the audio device, and the audio device converts the digital representation to an audio reply to the user utterance.

An audio device includes a plurality of reflective components that reflect sound emitted from a loudspeaker unit. The audio device forms a reflective space surrounded by the plurality of reflective components. The audio device includes an acoustic channel that is tube-shaped and includes an opening on one end and a terminating portion on an other end, the one end being connected to a communication hole provided in a reflective component among the plurality of reflective components, and a sound absorbing material disposed at a central portion of a sound path formed by the acoustic channel.

Disclosed are systems, apparatuses, processes, and computer-readable media to. According to some aspects, a method of processing audio data may include obtaining, at a computing device, sensing information from an audio device outputting a spatial audio stream for a user, wherein the audio device includes a first audio output device and a second audio output device; determining, based on the sensing information, that the second audio output device is not in use; modifying the spatial audio stream based on determining that the second audio output device is not in use and a head pose of the user to create a modified spatial audio stream; and providing the modified spatial audio stream to the first audio output device.

Aspects of the disclosure relate to an audio hub that may be connected to a computing device. The audio hub may comprise multiple wired and wireless communication interfaces that may be used to interface multiple peripheral audio devices with the computing device. The audio hub may comprise a built-in reference microphone that may be used for noise cancelation, automatic gain control, and / or acoustic echo cancelation. The audio hub may additionally be configured for audio dosimetry measurements based on output audio signals as sent to an audio output device connected to the audio hub.

The invention relates to the technical field of audio playing, and discloses a multi-channel audio linkage playing device based on Bluetooth broadcast and an implementation method, and the device comprises an audio input module, an audio channel switching module, a Bluetooth audio processing broadcast module, a modal perception cooperative control adjustment module, a DSP audio personalized processing module and an audio output module. According to the invention, core limitations of traditional audio equipment in input adaptation, multi-equipment cooperation and experience dimensions can be effectively broken through, flexible access and stable processing of multiple types of audio sources are realized, and synchronous cooperative playing of multiple Bluetooth equipment is achieved to construct an immersive audio environment; and meanwhile, audio experience is extended from single hearing to double sensory linkage of hearing and touch, so that the adaptability, the collaboration and the experience richness of the audio equipment in multiple scenes are greatly improved on the whole, and the diversified and high-quality audio use requirements of users are comprehensively met.

The invention relates to the technical field of USB audio equipment optimization, and discloses a USB audio equipment Device mode implementation method and system, a storage medium and equipment, and the method comprises the steps: dynamically recognizing a connection type through detecting a USB data line voltage threshold value, and dynamically loading a corresponding protocol stack module based on the connection type; in a Device mode, a multi-stage buffer mechanism is adopted to manage audio data streams, the number and size of buffer areas are dynamically allocated according to target delay time, and clock synchronization with a host is realized based on an adaptive clock recovery algorithm; the power consumption of the equipment is dynamically adjusted through a multi-stage power consumption state machine, switching is carried out among working states according to an audio activity state and a data transmission requirement, and a remote wakeup function is supported; according to the method, a compatibility strategy library is matched based on a device fingerprint recognition result, protocol characteristics and audio formats of different host devices are automatically adapted, meanwhile, audio playing is controlled through audio quality monitoring and self-optimization, and Device mode support is added to the USB audio device through the method.

The invention provides an audio processing method and device, and the method comprises the steps: collecting to-be-detected audio data related to a target audio based on to-be-detected audio equipment, and determining reference audio data related to the target audio; determining a to-be-tested audio index corresponding to the to-be-tested audio data in an audio consistency dimension, determining a reference audio index corresponding to the reference audio data, and generating an index difference score based on the to-be-tested audio index and the reference audio index; and under the condition that the to-be-tested audio data is determined to be abnormal based on the index difference score, generating difference processing prompt information corresponding to the to-be-tested audio data based on a difference mode rule and the index difference score. The audio processing method can be widely applied to the field of digital music in the digital creative industry.

A method and electronic consumer device for creating an individual spatial listening zone for a user provided. The method includes receiving an input audio signal and processing it to generate first and second ultrasonic electric audio signals. The ultrasonic signals are provided to first and second ultrasonic transducers which convert them into directional ultrasonic acoustic signals. These signals are directed to a spatial listening zone where the user is located. Through non-linear interaction of the ultrasonic signals in air within the spatial listening zone, an audible sound signal is produced for the user that corresponds to the desired audio based on the input signal. The audible sound is substantially confined to the spatial listening zone and inaudible outside it. Various techniques for dynamically adjusting the spatial listening zone based on user position are disclosed. The system allows for creation of personalized audio experiences while minimizing disturbance to others nearby.

A high-security pet and child confinement gate designed to prevent unauthorized access. The gate incorporates advanced control mechanisms, such as a sensory circuit with a camera, capable of wireless communication with smart devices. The camera can be integrated or removable, providing image data to smart devices. The gate may include a free-spinning mechanism to deter climbing attempts, equipped with sensors and audio devices to produce deterring sounds. Additionally, the gate may include a pressure sensor and an audio device that emits ultrasonic sounds upon detecting crossing attempts. The communication module is adaptive, allowing interaction with cellular networks, Wi-Fi, or Bluetooth, and can receive commands from smart devices to enhance security measures, including sound playback, vibrations, and increased spinning speed.

The invention discloses a restaurant system and method based on Internet real-time audio and video interconnection, and the system comprises interconnection terminals of local and remote restaurants, and each terminal comprises a visual display device, an audio playing device, a visual collection device, an audio collection device, an Internet communication module and a dining table. The visual display equipment adopts a high-resolution display device and is deployed on a decorated wall surface, a curtain projection, or in front of a dining table, or embedded into the dining table and the like; the audio equipment comprises a high-fidelity microphone and a multi-channel sound box and is used for accurately collecting and playing sound; the visual acquisition device is a high-definition camera and is installed by being matched with a suspended ceiling or the side edge of a screen. The internet communication module guarantees low-delay sound and picture transmission based on the high-speed transmission characteristic of the internet, and two-way synchronization is formed. The system supports three interconnection modes of mirroring, complementation and single point, is respectively adapted to single-remote, multi-remote and master-slave interconnection scenes, can effectively optimize the problems of asynchronous sound and picture, poor experience and the like in remote dinner party, and improves the immersion and flexibility of remote interaction.

The invention relates to the technical field of embedded equipment control, and discloses a multi-event cooperative control method and device for audio equipment, equipment and a storage medium, and the method comprises the steps: constructing a queue structure body of multi-event circulation according to hardware parameters of the audio equipment, and achieving the efficient cooperative processing of events through a classification rule and a time scheduling method. The method has the advantages of effectively managing event queues, reducing high-priority event delay, avoiding equipment downtime and improving user experience and equipment performance.

Various implementations include audio devices and methods for spatializing audio output based on content. Certain implementations include at least one audio output device for providing an audio output based on data, and at least one controller coupled with the at least one audio output device, the controller configured to, use the data to determine a content type for the audio output from a group of content types, using a trained content classifier, and take an action based on the determined content type.

A wearable audio device may comprise a first directional loudspeaker having a first directionality, a second directional loudspeaker having a second directionality different from the first directionality, a communication interface configured to receive a communication audio signal, and a processor configured to operatively control the first directional loudspeaker, the second directional loudspeaker, and the communication interface. The processor may be configured to perform operations comprising detecting a relative mount position of wearable communication device, selecting a proximal loudspeaker to emit the communication audio signal in accordance with the relative mount position, wherein the proximal loudspeaker comprises one of the first directional loudspeaker and the second directional loudspeaker; and transferring the communication audio signal from the communication interface to the proximal loudspeaker in accordance with the selecting.