2101 results about "Audio equipment" patented technology

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.

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 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.

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.

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 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.

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 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.

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 invention belongs to the technical field of audio equipment design, and relates to a microphone for AI voice cloning based on pseudo tone of acoustic shielding, and the microphone comprises an audio collection module which is used for collecting an environment audio signal in real time; the analog-to-digital converter is used for converting the environment audio signal into a digital signal; the feature extraction module is used for performing frequency domain conversion on the digital signals and extracting frequency spectrum features and tone features; the acoustic shielding calculation module is used for calculating a shielding threshold value of the environment audio signal and determining a frequency band which cannot be heard by human ears in the environment audio signal according to the shielding threshold value; the pseudo tone generation module is used for generating pseudo tones based on the extracted tone features; the synthesis module is used for replacing a corresponding frequency band of the environment audio signal with the generated pseudo tone to form a synthesized audio; and the audio output module is used for outputting the synthesized audio. According to the invention, the pseudo tone can be generated in real time in the audio recording process, and it is ensured that a user is always protected during audio recording.

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.

The utility model provides an earphone, and relates to the technical field of audio equipment. The earphone comprises a main body part, an extension part and a sound production part which are connected in sequence, the earphone further comprises a shell, and the shell is provided with a first pickup hole and a first microphone cavity. The first microphone is arranged in a first microphone cavity of the shell, the first pickup hole is communicated with the first microphone cavity, so that the outside of the shell can be communicated with the first microphone cavity, sound waves outside the shell can enter the first microphone cavity through the first pickup hole, and the sound waves can be received by the first microphone and converted into electric signals. The first pickup hole is opposite to at least part of the area of the side, facing the auricular conchae, of the tragus, so that at least part of the tragus can shield the first pickup hole, and therefore, when a user faces the wind, the tragus can prevent wind noise waves from entering the first pickup hole to a certain extent, and the wind noise generated when the earphone is used is reduced.

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.

According to the sound system control method provided by the invention, the environmental acoustic characteristic parameters and the positions of the plurality of sound devices are detected, whether the recommended placement positions generated according to the environmental acoustic characteristic parameters are met is judged, and if not, the position adjustment information is output; a user can adjust the position of the sound equipment according to the position adjustment information so as to obtain a better sound effect, and the user can enjoy high-quality audio experience without professional acoustic knowledge; if the user does not perform position adjustment or the position adjustment does not meet the requirement, the sound system generates a second audio processing algorithm according to the real-time position of the sound equipment, so that the audio experience is ensured as far as possible. By adopting the method, the user obtains a large option, the product use experience is improved, and if the sound effect is poor in the subsequent product use process, the user can autonomously call out the position adjustment information and adjust the position of the sound equipment according to the position adjustment information.

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.