1748 results about "Audio equipment" patented technology

A device environment can include a plurality of devices, including an audio device, in communication over a network. An audio device (such as a soundbar) can comprise a speaker, a camera, and other sensors. The audio device can receive information from other devices in the environment including image data from other cameras and physiological data from wearable devices, etc. The audio device can monitor and control aspects of the environment. The audio device can provide information to the user including audible notifications, and can monitor the user including the user's position or physiological condition, alone or in combination with other devices in the environment.

The invention relates to the technical field of audio signal processing, and discloses a room sound correction method, device and equipment for audio equipment and a storage medium, which are used for improving the accuracy and correction effect of correction parameters and effectively improving the acoustic effect of the audio equipment in a room. The room sound correction method of the audio equipment comprises the following steps: acquiring response data of a room to a multi-band preset test signal, and extracting room acoustic data by adopting a time-frequency conjoint analysis method, the room acoustic data comprising frequency attenuation, reflection paths, phase shifts and reverberation time information of different bands; based on the room acoustic data, performing compensation processing by using preset microphone calibration data to obtain a calibrated frequency response curve; performing fitting calculation on a preset ideal frequency curve and the calibrated frequency response curve according to an optimization algorithm to obtain specific correction parameters of the room; and correspondingly adjusting the output of the audio equipment according to the correction parameter.

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.

A method of audio processing combining WiFi and BLE technologies includes: obtaining target audio data including multi-channel information; establishing connections among audio devices corresponding to multiple rooms through WiFi and BLE Auracast™ protocols; dynamically configuring distribution paths according to intelligent distance detection results, and obtaining a configured audio distribution network; adjusting, using a synchronization algorithm in the BLE Auracast™ protocol, a timing sequence of the target audio data, and obtaining synchronized audio data; and transmitting, according to the configured audio distribution network, the synchronized audio data via WiFi or the BLE Auracast™ protocol to each audio device for playback.

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.

A game controller can provide input to a video game (e.g., to control an object or character in the video game) and, optionally, to interact with a graphical user interface (e.g., to select the video game and / or perform other functions). The game controller can be connected with a mobile computing device, such as a mobile phone, and the game can be played locally on the mobile computing device or remotely via the mobile computing device. In one embodiment discussed herein, a platform-customized mobile game controller is provided. In another embodiment discussed herein, a mobile game controller and method are provided for connecting to a wireless audio device.

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 multi-room multi-channel audio synchronization method, apparatus, and device and a storage medium are provided. In this method, an audio signal to be played in a primary audio device is determined, the audio signal is encoded according to channels by using a preset low-latency audio encoding strategy and audio encoding data is obtained. Next, the audio encoding data is optimized based on a User Datagram Protocol and a Negative Acknowledgment NACK retransmission mechanism to obtain audio coded sequence data packets. The audio encoding sequence data packets are controlled to be synchronously played in at least one secondary audio device by using a preset room synchronization control strategy and a channel synchronization control strategy.

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.

An audio device comprising, an interface, one or more first output transducers, one or more input transducers, and a processor. The audio device is configured to emit a first ultrasound signal, receive a first reflection indicative of the first ultrasound signal, process the first reflection to determine movement, and in response to movement being determined, adjust one or more audio processing parameters of the audio device.

The invention belongs to the technical field of loudspeakers, and discloses a bidirectional loudspeaker group, a sound box and audio equipment, and the bidirectional loudspeaker group comprises a common magnetic circuit module, a first loudspeaker and a second loudspeaker. The common magnetic circuit module comprises a magnetic conduction bowl, a first magnetic gap and a second magnetic gap are formed in the inner side and the outer side of the magnetic conduction bowl in the first direction respectively, a first through hole communicated with the first magnetic gap and the second magnetic gap is formed in the magnetic conduction bowl, and the first loudspeaker and the second loudspeaker are arranged on the two sides of the common magnetic circuit module in the second direction respectively. The first loudspeaker comprises an annular first voice coil, the second loudspeaker comprises an annular second voice coil, the first voice coil and the first magnetic gap are correspondingly arranged, the first voice coil can vibrate in the first magnetic gap in the second direction, the second voice coil and the second magnetic gap are correspondingly arranged, and the second voice coil can vibrate in the second magnetic gap in the second direction. Through the above arrangement, the bidirectional loudspeaker group provided by the invention can achieve the balanced damping, sound quality improvement and heat dissipation requirements.

Embodiments are directed to a wearable audio device, such as an earbud. The earbud may be configured to detect input using various sensors and structures. For example, the earbud may be configured to detect gestures, physical manipulations, and so forth performed along or on the earbud. In response to the detected inputs, the earbud may be configured to change various outputs, such as an audio output or a haptic output of the device. The earbud may also include a microphone to register voice commands. In some cases, the microphone may be used to control the earbud using the registered voice command in response to one or more detected gestures or physical manipulations.

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.

A system, method, and computer-readable medium for performing a smart audio operation. The smart audio operation includes detecting a plurality of audio devices available for use by an information handling system; monitoring information handling system audio related information; analyzing the information handling system audio related information using a trained smart audio artificial intelligence model; and, automatically connecting the information handling system to a particular audio device of the plurality of audio devices based upon the information handling system audio related information.

The invention relates to the technical field of signal processing of intelligent voice equipment, and discloses an equalization method and system for a Chirp modulation signal of audio equipment, and the equalization method for the Chirp modulation signal of the audio equipment comprises the steps: analyzing the non-stationary characteristic of the Chirp modulation signal through an information geometric mapping algorithm, and generating non-stationary complexity mapping; processing the Chirp modulation signal by using a multi-scale entropy analysis algorithm to generate a multi-scale complexity spectrum; extracting long-range correlation characteristics of the Chirp modulation signals by adopting a fractal analysis algorithm, and outputting Hurst index characteristic mapping; constructing a complexity-perceived adaptive equalization system; a time-frequency joint optimization algorithm is constructed, and global optimization and adaptive adjustment of equalization parameters are realized; signal complexity characteristics are quantized by using a multi-scale entropy analysis algorithm, and the adaptability of the equalization system to signals with different complexities is enhanced.

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.

1. The name of this design product: Sports bone conduction headphones (X500). 2. The purpose of this design product: as a headphone audio playback device. 3. The key point of the design of this product lies in its shape. 4. The picture or photo that best illustrates the design points: three-dimensional picture.

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.

Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first calibration signals, generating first modified audio playback signals by inserting the first calibration signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth calibration signals) played back by second through Nth audio devices, extracting second through Nth calibration signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth calibration signals.

The invention discloses intelligent sound equipment, and relates to the technical field of sound equipment, the intelligent sound equipment comprises a shell, a first partition plate, a second partition plate, a high-frequency loudspeaker, a full-frequency loudspeaker and a radiation unit, and the shell is provided with an accommodating cavity; the first partition plate and the second partition plate are arranged in the containing cavity at intervals so as to divide the containing cavity into a high-frequency cavity, an intermediate-frequency cavity and a low-frequency cavity which are sequentially arranged, the size of the high-frequency cavity is smaller than that of the intermediate-frequency cavity, the size of the intermediate-frequency cavity is smaller than that of the low-frequency cavity, and the high-frequency loudspeaker is arranged in the high-frequency cavity. The full-frequency loudspeaker is arranged in the intermediate-frequency cavity, the second partition plate is provided with a frequency division unit communicated with the intermediate-frequency cavity and the low-frequency cavity, the frequency division unit is used for scattering and absorbing medium-high frequency sound waves and keeping propagation of low-frequency sound waves, and the radiation unit is arranged in the low-frequency cavity; the radiation unit is used for receiving low-frequency sound waves which are emitted by the full-frequency loudspeaker and are transmitted to the low-frequency cavity through the frequency division unit so as to enhance low-frequency response.

The invention discloses a display device and an audio control method, and relates to the field of display devices, the audio control method is applied to the display device, the display device comprises a display, an audio output interface and a controller, the controller is in communication connection with the display and the audio output interface, and the audio output interface is configured to be in communication connection with a plurality of target audio devices. The target audio equipment is used for outputting audio data; the method further comprises the following steps: under multi-window display, firstly stopping sound mixing processing on each piece of audio data; and according to the determined target relationship, establishing an audio output channel for audio data transmission between each target window and the target audio equipment, so as to realize transmission of the audio data of different windows to different audio equipment, thereby improving the audio-visual experience of a user under multi-window display.

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 invention discloses a short-distance wireless audio multi-standard compatible communication protocol method, which comprises the following steps of: intelligent protocol selection: automatically selecting a most suitable wireless communication standard to carry out audio data transmission according to information such as communication capability, network environment, bandwidth requirement, signal strength, equipment type and current network load of audio equipment; the communication standard comprises but is not limited to Bluetooth, Wi-Fi, Zigbee, NFC and the like, and multi-protocol cooperative scheduling: under the condition that a plurality of protocols coexist, the load, the bandwidth use condition, the signal quality and the like of each protocol are monitored in real time. Through protocol intelligent selection and multi-protocol cooperative scheduling, the equipment can automatically select the most suitable communication standard according to the actual situation, the problem that different protocols are incompatible is solved, the interconnection and intercommunication between the equipment are improved, the low-delay processing technology is adopted, the data packet transmission sequence and the protocol conversion process are optimized, and the communication efficiency is improved. The audio data can be transmitted and played in real time, delay is reduced, and audio experience is improved.

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.

In some embodiments, a wireless network architecture can include a network protocol configured to allow communication between a host device and each of an audio device and a human interface device. The network protocol can include a transaction having a time duration, with the transaction including a plurality of sub-transactions arranged in sequence. Each sub-transaction can have a time duration and a frame for the human interface device such that a latency period associated with the human interface device is greater than or equal to the time duration of the sub-transaction but less than the time duration of the transaction.

Audio / video base stations may include a plurality of audio / video input ports. The base stations may include an audio / video output port. The base stations may include a communication interface that is coupleable with an audio transceiver. The base stations may include one or more processors. The base stations may include a memory device having instructions stored thereon that, when executed by the one or more processors, cause the audio / video base station to detect that the audio transceiver has been undocked from the audio / video base station. The instructions may further cause the base station to, in response to detecting that the audio transceiver has been undocked from the audio / video base station, automatically switching an audio output signal from the audio / video output port to the communication interface while continuing to transmit a video output signal to the audio / video output port.

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.

1. The name of this design product: Sports bone conduction headphones (K08). 2. The purpose of this design product: as a headphone audio playback device. 3. The key point of the design of this product lies in its shape. 4. The picture or photo that best illustrates the design points: three-dimensional picture.

Aspects of the present disclosure provide techniques, including devices and systems implementing the techniques, to output an adjusted audio mix to a wearable audio device in real-time using at least one pre-trained machined learned (ML) models. The pre-trained ML models enable adjustment of audio output with ultra-low latency. One example technique for adjusting audio output generally includes detecting, by one or more feedforward microphones, one or more spatialized sounds proximate a user, separating, using at least one pre-trained ML model, at least a portion of the spatialized sounds, wherein the separation is based on content of the sound, processing at least a portion of the spatialized sounds, and outputting, by the audio output device, an adaptive mix of the processed spatialized sounds. Such techniques may enable a user to filter out or reduce unwanted noises based on their sound content.