2786 results about "Audio equipment" patented technology

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.

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, automatically select a spatialization mode for the audio output from a plurality of spatialization modes based on the determined content type, and apply the selected spatialization mode to the audio output.

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.

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.

The invention discloses a digital intelligent teaching recording and broadcasting system and method, and relates to the technical field of data encryption, and the method comprises the steps: collecting teaching videos and audios of a teacher through a high-definition camera and an audio device, carrying out the preprocessing, obtaining the course content, recognizing the key information of the course content through a natural language processing NLP and computer vision technology, and carrying out the recording and broadcasting. The method comprises the following steps: segmenting a video into single knowledge point fragments, performing classified storage, forming a learning resource library, collecting basic information and learning habit data of students, constructing a user portrait, matching learning resources, extracting the learning habit data in the user portrait, performing feature extraction, and obtaining a comprehensive feature vector; constructing a learning state evaluation model by using a convolutional neural network to comprehensively analyze the comprehensive feature vector to obtain a learning state comprehensive score, formulating a playing strategy according to the learning state comprehensive score and matched learning resources, regularly testing knowledge points of students, evaluating the mastering degree of the students, and updating a user portrait according to a result.

Various implementations include audio devices and methods for noise reduction control in wearable audio devices and / or vehicle audio systems. Certain implementations include a non-occluding wearable audio device having: at least one electro-acoustic transducer; at least one microphone; and a control system coupled with the at least one electro-acoustic transducer and the at least one microphone, the control system programmed to: adjust an active noise reduction (ANR) setting for audio output to the at least one electro-acoustic transducer in response to detecting use of the non-occluding wearable audio device in a vehicle.

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.

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.

The invention relates to the technical field of audio processing, and discloses a multi-room multi-channel audio synchronization method, device and equipment and a storage medium, and the method comprises the steps: determining an audio signal to be played in main audio equipment, and carrying out the coding of the audio signal according to a channel through employing a preset low-delay audio coding strategy, and obtaining audio coding data; based on a user datagram protocol and an NACK retransmission mechanism, optimizing the audio coding data to obtain an audio coding sequence data packet; and controlling the audio coding sequence data packet to be synchronously played in the at least one slave audio device by using a preset room synchronization control strategy and a sound channel synchronization control strategy. In the invention, the audio signal is coded through the low-delay LC3 coding, so that the transmission delay is reduced to the greatest extent, and the coded audio signal is transmitted by using the UDP protocol and the NACK retransmission mechanism, so that the network bandwidth occupation is reduced, and the additional delay caused by transmission or retransmission is avoided.

The invention provides a rising and falling control method and system for a PCM audio sampling rate, and belongs to the technical field of audio processing, and the method comprises the following steps: obtaining audio signal features of pulse code modulation PCM audio data, the audio signal features comprising spectrum energy distribution, instantaneous power, a signal-to-noise ratio and a peak-to-average ratio; monitoring hardware state parameters of the Bluetooth audio equipment in real time, wherein the hardware state parameters comprise a current representative processor load rate, a bandwidth occupancy rate and a current coding and decoding mode parameter; according to the predicted value of the target sampling rate, performing resampling processing on the PCM audio data by adopting a filtering processing method and an interpolation algorithm to obtain PCM audio data matched with the target sampling rate; the resampling processing comprises up-sampling processing or down-sampling processing; and inputting the PCM audio data matched with the target sampling rate to an audio output module of the Bluetooth audio equipment for playing. The audio quality, the computing resources and the transmission efficiency are optimized through self-adaptive resampling processing.

To provide an alternative audio transducer device that works somewhat to address some of the shortcomings of existing technologies, a manufacturing method for the same, and a personal audio device.SOLUTION: An audio transducer A100 includes a diaphragm A101, a diaphragm suspension, and a transducer base structure. The diaphragm suspension includes multiple hinge mounts A107a, A107b. The transducer base structure includes a body A110 and a conductive coil A106 of a conversion mechanism. The diaphragm suspension rotatably mounts the diaphragm A101 to the transducer base structure. The diaphragm suspension is made of soft material and / or damping material. The position of the rotational axis of the diaphragm A101 is determined on the basis of a nodal axis of the diaphragm A101. The conversion mechanism of the audio transducer converts sound in cooperation with the moving diaphragm A101 and includes a design in which a magnet moves or a design in which a coil moves.SELECTED DRAWING: Figure 1F

A user scenario-based audio playback method includes: obtaining a target user scenario that a user is currently in and user historical data; determining a plurality of pieces of candidate preset information corresponding to the target user scenario from a preset device configuration information table; calculating, based on the user historical data, a degree of scenario relevance, the quantity of playbacks and the quantity of selected times of a corresponding preset state of each candidate preset information; calculating a recommendation coefficient of each candidate preset information based on the degree of scenario relevance, the quantity of playbacks and the quantity of selected times; determining target preset information from the plurality of pieces of candidate preset information according to the recommendation coefficient; and adjusting a playback parameter of an audio device based on the target preset information and playing an audio.

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.

The invention relates to the technical field of wireless communication, and discloses a low-power-consumption communication method and device for wireless audio equipment, an SoC chip and a storage medium. The method comprises the following steps: sampling acoustic characteristics of heterogeneous audio data of a plurality of target audio devices, calculating a voice spectrum corresponding to device characteristics, and extracting multi-device acoustic characteristics, performing multi-frequency sub-band speech enhancement, heterogeneous segmentation and grouping of multi-target audio equipment, verification value calculation and differential coding on each extracted acoustic feature sequence to obtain a multi-source coding speech segment; performing transmission unit division and cooperative signal modulation, cooperative resource allocation and data stream transmission of each transmission unit on the multi-source coding voice segment to obtain a transmission voice data stream; and obtaining a transmission voice data stream transmitted with low power consumption and performing audio recovery on the obtained data to obtain a target audio data stream transmitted by the multi-target audio equipment. The communication power consumption of the multiple target audio devices is reduced, and the audio transmission quality is ensured.

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 intelligent sound channel switching, and provides an intelligent sound channel switching method for audio equipment, which comprises the following steps: S1, converting an audio signal to a preset standard sampling rate, and carrying out amplitude normalization and framing processing; extracting a time domain feature and a frequency domain feature of the audio signal, and identifying a content type of the audio signal by using a deep learning model; s2, acquiring environmental acoustic characteristics through a microphone of the audio equipment; the audio equipment performs playing time period statistics of the user, monitors a volume adjustment mode of the user, monitors a sound channel switching frequency of the user, and constructs a user behavior model based on the Bayesian theorem; s3, constructing a sound channel switching decision model, and performing weighted calculation to obtain a final decision score; when the difference value between the decision score and the switching threshold value is greater than a preset value, performing sound channel switching; and S4, realizing sound channel switching by using a high-fidelity down-mixing algorithm and an intelligent sound field reconstruction algorithm. Audio signal characteristics and a playing environment can be detected in real time, and seamless switching between a sound channel 6 and a sound channel 2 can be automatically carried out.

The invention provides outdoor portable sound equipment, and particularly relates to the technical field of sound equipment, the outdoor portable sound equipment comprises a sound equipment main body, a volume knob is arranged on one side of the top surface of the sound equipment main body, two sounding horns are symmetrically arranged on the surface of one side of the sound equipment main body, and four supporting legs are uniformly and fixedly arranged at the bottom of the sound equipment main body. Through the cooperation of components such as the gear and the gear ring, after the sound equipment main body is used, no matter how many angles a user rotates the volume knob clockwise to increase the volume of the sound equipment main body, after the sound equipment main body is used, the volume knob is driven to rotate anticlockwise through the rotation of the gear ring, so that the volume of the sound equipment main body is increased. Therefore, the sound volume is prevented from being in the maximum state when the sound main body is turned off, and when the sound main body is turned on and used next time, the sound production element in the sound main body is not suddenly in the full-load state, so that the sound production element in the sound main body is protected, and the service life of the sound main body is prolonged.

The present disclosure relates an acoustic signal output equipment, and in particular to an audio equipment. The audio equipment comprises a sound emit device, the sound emit device is used to emit sound signals, the audio equipment also comprises a sound signals amplification structure and a connection structure. The sound signals amplification structure has an axis and a sound signals amplification surface, the sound signals amplification surface is formed by rotating the axis, the sound signals amplification surface faces the sound emit device directly, with a distance between the sound signals amplification surface and the sound emit device, in a direction of away from the sound emit device, a cross sectional area of the sound signals amplification surface is increased gradually.

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.

The invention relates to the technical field of audio processing, and discloses an audio processing method and device combining WiFi and BLE technologies, equipment and a storage medium, which are used for realizing high-quality and low-delay multi-room multi-channel synchronous playing. The audio processing method combining the WiFi and the BLE technology comprises the steps of obtaining target audio data containing multichannel information, establishing a multi-room audio equipment connection through the WiFi and a BLE Auracast protocol, dynamically configuring a distribution path according to intelligent distance detection, obtaining a configured audio distribution network, adjusting a time sequence of the target audio data by using a BLE Auracast synchronization algorithm, and sending the target audio data to the audio distribution network through the BLE Auracast synchronization algorithm. And according to the configured audio distribution network, sending the synchronized audio data to each device for playing through WiFi or BLE Auracast.

The invention provides a method for improving the audio quality of a voice conference of multiple terminals at the same place. The method comprises the steps of time calibration, detection of equipment in the same room, audio quality calculation, delay detection, data collection and issuing, adjustment and playing of a dynamic buffer area, speaking button control and speaking sound processing. Time is unified through a time service server or a chip, equipment in the same room is detected, audio quality is calculated, delay is detected, equipment information is collected and issued, synchronous playing of a buffer area is adjusted, mutual exclusion of speaking buttons is controlled, speaking sound is processed to reduce howling and acoustic feedback, and use of the audio equipment is optimized. According to the invention, the audio synchronization and quality of the multi-terminal voice conference can be improved, howling and acoustic feedback are reduced, and the user experience is optimized.

Some disclosed methods involve receiving, by a control system, location control data from a sound source as the sound source emits sound in a plurality of sound source locations within an audio environment. Some such methods involve receiving, by the control system, direction of arrival data from each audio device of a plurality of audio devices in the audio environment. In some examples, each audio device of the plurality of audio devices includes a microphone array and the direction of arrival data corresponding to microphone signals from microphone arrays responsive to sound emitted by the sound source in the plurality of sound source locations. Some such methods involve estimating, by the control system, sound source locations and audio device locations based, at least in part, on the location control data and the direction of arrival data.

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.

The invention discloses a call recording method based on dynamic sampling rate adjustment and a Bluetooth earphone, and aims to improve the quality and stability of call recording in a wireless transmission environment. According to the method, uplink audio data and downlink audio data are collected at the same time through the Bluetooth headset, and the transmission rate of a downlink data packet is monitored in real time. When the rate is lower than the preset threshold value, the audio sampling rate is automatically reduced from 16 kHz to 8 kHz to adapt to bandwidth limitation, and recording interruption or tone quality reduction is avoided. The adjusted audio data is cached through a double-layer cache mechanism after being coded and compressed, the first cache temporarily stores the adjusted audio data, and the second cache manages data packets to be sent and retransmitted, so that the continuity and integrity of data transmission are ensured. The method is suitable for wireless audio equipment such as a Bluetooth earphone, and the reliability, adaptability and user experience of call recording are effectively improved.

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 relates to the technical field of audio equipment configuration, and discloses a parameter configuration method and device of audio equipment, equipment and a storage medium, which are used for improving the accuracy of parameter configuration of the audio equipment. The parameter configuration method of the audio equipment comprises the following steps: acquiring the type and audio format information of a current input signal source; based on the type of the input signal source and the audio format information, identifying an audio content type and determining an optimal playing mode; adjusting a sound box combination and sound channel configuration of the audio equipment based on the optimal playing mode; and matching a volume level according to the audio content type, and carrying out adaptive volume adjustment on the audio equipment in combination with user preference and environment analysis.

The invention relates to an out-ear audio device comprising a bone conductor transducer (1) configured to transfer an audio signal by means of bone conduction to an ear of a user, the device comprising a housing (2) accommodating the bone conductor transducer (1), the device further comprising holding means (3) configured to releasably attach the housing (2) to the tragus of the user, wherein:the holding means (3) together with a portion (4) of the outer surface of the housing (2) of the audio device forms a substantially U-shaped structure, where the holding means (3) comprises a contact member (10, 28) formed for contact with an inner surface of the tragus that faces towards the head of the user; andwherein a contact region that is formed for contact with an outer surface of the tragus corresponding to said inner surface of the tragus is provided by a portion (4) of the housing (2), which portion (4) is in vibrational contact with the bone conduction transducer (1), such that vibrations generated by the bone conductor transducer (1) is transmitted to the contact region;where the contact member and the contact region form a gap (27) between the contact member (10, 28) and the contact region (4); andwherein the holding means (3) is displaceably or pivotally connected to the bone conductor transducer (1) either directly or via the housing (2), such that the distance between the contact member (10) and contact region (4) can be varied, andwherein the housing (2) and / or the holding means (3) is provided with biasing means, such as a spring member 5, 6, 7; 15, 18), that is configured to force the contact member (10, 28) and the contact region (4) against each other, whereby the contact member (10, 28) and the contact region (4), respectively, presses against said inner and outer surfaces of the tragus, when the audio device is provided on the tragus.

The invention relates to the technical field of audio equipment correction, and discloses a sound channel correction method and device, equipment and a storage medium, which are used for improving the accuracy of sound channel correction of audio equipment. The sound channel correction method comprises the following steps: acquiring an audio signal of audio equipment and user historical data; identifying a current use scene based on the audio signal; performing sound channel characteristic analysis on the left and right sound channels based on the audio signal; performing user habit analysis and scene adaptability analysis according to the user historical data and the current use scene; and correcting the parameters of the left and right sound channels of the audio equipment according to the sound channel characteristic analysis result, the user habit analysis result and the scene adaptability analysis result.