2818results about "Transducer circuits" patented technology

The invention relates to the technical field of earphone noise reduction processing, and discloses a motorcycle riding earphone noise reduction method combining different noise reduction algorithms, comprising the following steps: acquiring an original audio signal, and distributing the original audio signal to each processing module; a preprocessing signal is generated, voice activity information is determined, and environmental noise energy information is calculated; according to the environmental noise energy or the auxiliary information, adaptively adjusting a high-low noise energy threshold value; the original signals are input into the traditional and AI noise reduction module in parallel to generate two paths of noise reduction signals; based on the environment noise, the voice information and the threshold value, performing weighted fusion to generate an output signal; and carrying out equalization and compression processing on the output signal, and driving the loudspeaker to output. According to the invention, comprehensive judgment on environmental noise energy information, voice activity information and riding speed is introduced, an output signal of a traditional noise reduction module is set to be zero in a high-speed voice-free environment, and AI noise reduction is combined, so that tone quality distortion of a traditional noise reduction algorithm in a noise environment is effectively avoided.

The invention relates to the technical field of audio signal processing, and discloses a sound box and sound effect intelligent adjusting method and system based on data analysis and a storage medium, and the sound box and sound effect intelligent adjusting method based on data analysis comprises the steps: constructing a user feature modeling engine, and extracting user auditory characteristic data; constructing an environment characteristic analysis engine, and collecting environment acoustic characteristic data; constructing a content feature extraction engine, and analyzing audio content semantic data; constructing a three-dimensional fusion optimizer, inputting the three feature vectors into an auditory scene fusion model, calculating an auditory experience score through tensor fusion operation, and solving an optimal sound effect parameter by applying a multi-objective optimization algorithm; constructing a parameter generation controller, and converting the optimal sound effect parameter into a specific audio processing parameter; according to the invention, the problem of mutual interference caused by traditional separated processing is solved, and the accuracy of sound effect adjustment and the user satisfaction are improved.

Disclosed in the present invention is a high-dynamic capacitive digital microphone system based on electrostatic force feedback. The system comprises an MEMS microphone and an interface circuit chip thereof, wherein the interface circuit chip comprises a charge pump, an analog front end, an analog-to-digital conversion circuit, an amplitude detection module, an amplitude adjustment control module and a pulse density modulation module. In the present invention, the characteristic of the sensitivity being able to be adjusted by means of changing a bias voltage by a capacitive sensor is used, and a charge pump with an adjustable output voltage is used to provide electrostatic force feedback, thereby further expanding the adjustable range of a front-end gain, and also reducing requirements for the noise performance of a readout circuit. Therefore, the system in the present application has the characteristics of high dynamic range and high-precision quantization.

The invention relates to the field of aviation voice signal processing, and discloses a voice recognition compensation pilot earphone hearing protection method and system, and the method comprises the following steps: collecting multi-modal data, and separating a sound source through tensor decomposition; inferring a pilot state by using a dynamic network; performing context recognition and semantic evaluation on the attention target voice; and finally, dynamically modulating the sound field based on deep reinforcement learning, enhancing the voice in a personalized manner, and outputting after noise suppression. The system comprises a multi-mode perception data acquisition module, a sound source decoupling module, a pilot state inference module, a voice processing and semantic evaluation module and a sound field modulation and output module. According to the invention, high-fidelity speech extraction is realized through multi-modal perception and tensor decomposition; evaluating priority key information in combination with attention and semantics; and deep reinforcement learning and model prediction control are adopted to dynamically optimize the sound field, so that the voice recognition accuracy and the pilot information acquisition efficiency are improved.

The invention discloses a Bluetooth hearing-aid earphone optimization method and system for adaptive hearing compensation. The method comprises the following steps: S1, acquiring a personalized hearing curve of a user and background audio data of left and right ears in each typical scene, and respectively preprocessing the background audio data; s2, extracting time-frequency fusion features of each typical scene, generating scene feature vectors, and constructing a scene feature vector library; s3, calculating a scene perception conversion vector, an abrupt change risk vector and a smooth gain vector; s4, calculating a binaural gain change vector, a binaural synchronous entropy vector, a phase alignment factor vector and a left and right ear cooperative gain vector; and S5, according to the left and right ear cooperative gain vectors, processing the background audio data after left and right ear preprocessing in the current scene, and outputting the processed background audio data to the Bluetooth hearing-aid earphone. According to the invention, the problem of gain abrupt change generated during scene switching of the existing Bluetooth hearing-aid earphone and the problem of spatial perception distortion caused by binaural coordination imbalance can be solved.

A logarithmic amplifier system is disclosed. The system includes first and second amplifiers respectively having first and second gains; a first multiplier having an input connected to an output of the first amplifier, and being configured to output a first multiplied signal based on an output signal of the first amplifier and based on a received first multiplication factor equal to a value K; a second multiplier having an input connected to an output of the second amplifier, and being configured to output a second multiplied signal based on an output signal of the second amplifier and based on a received second multiplication factor equal to 1 minus the value K; a transition shaping circuit configured to change the value K from 0 to 1 with a filtered transfer function; and a summing circuit having inputs coupled to outputs of the first multiplier and the second multiplier.

In aspects of audio broadcast management based on audio disruption, a wireless device receives an audio broadcast from an audio source device, and initiates a connection to the audio broadcast. The wireless device detects an audio disrupt event of the audio broadcast based on audio frequency data, audio intensity data, or a predefined time limit associated with the audio broadcast. In response to this, the wireless device initiates a disconnection from the audio broadcast, and initiates a connection to a new in-range audio broadcast.

Disclosed in the present invention are an automatic delay calibration method for a smart speaker system, a device, and a storage medium. The method comprises: initializing a smart speaker system and performing environmental testing to preliminarily configure system parameters; exciting a target sound channel and playing a target test signal; locating a first test signal on the basis of a voice activity detection (VAD) algorithm, determining the start and end of the first test signal, and identifying a time difference of the first test signal to obtain a delay of main channels; locating a time of arrival of a second test signal on the basis of a fast peak search algorithm, and identifying a delay of a subwoofer; and performing delay calibration on the subwoofer and the main channels, and dynamically adjusting delay calibration parameters of the channels on the basis of a delay calibration amount, thereby achieving accurate time alignment between the subwoofer and the main channels. The present invention solves the problem of low-frequency signal delays that are difficult to process in conventional technologies, and particularly makes a breakthrough progress in the coordination of a subwoofer and other sound channels.

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.

The invention relates to the technical field of wireless communication, in particular to an OTA upgrading method and system for Bluetooth sound equipment. The method comprises the following steps: acquiring to-be-upgraded firmware data and currently played audio data, and acquiring power amplifier current feedback data of a sound speaker by sending a double-frequency detection signal containing 21kHz and 22kHz; acquiring equipment transmission capability confirmation data according to a silent response mode of the sound equipment to the preset handshake signal; and calculating a nonlinear distortion coefficient of the loudspeaker according to the intensity of the 1kHz difference frequency component in the power amplifier current feedback data, and obtaining real-time channel evaluation data based on current waveform deviation analysis. According to the method, the nonlinear distortion degree of the loudspeaker is accurately evaluated through the 1kHz difference frequency component generated by the 21kHz and 22kHz dual-frequency detection signals, so that the reliability of data transmission is more accurately predicted.

The utility model relates to the technical field of eavesdropping prevention, in particular to a terminal anti-eavesdropping device and a terminal anti-eavesdropping system. The device comprises a first connection module, an audio management module, a storage module and an access management module, and the first connection module is used for establishing communication connection with an external terminal to obtain the audio control authority of the external terminal and obtain to-be-stored data; the audio management module is connected with the access management module, and a preset audio signal is built in the audio management module and is transmitted to the access management module; the access management module is connected with the first connection module, the storage module and the audio management module, and the access management module is used for performing data transmission with the storage module, performing data transmission with an external terminal through the first connection module and transmitting a preset audio signal to the external terminal through the first connection module. On the basis of anti-eavesdropping, the system also has data storage and reading functions, and has richer functions.

The invention discloses a parallel constant-voltage digital audio power amplifier and a broadcasting system. Comprising a line input module, a line output module, a first analog-to-digital conversion circuit, a second analog-to-digital conversion circuit, a third analog-to-digital conversion circuit, a digital signal processor, a single-chip microcomputer, a digital-to-analog conversion circuit, an error comparator, a modulator, an integrating circuit, a pulse width modulation input half-bridge output stage, and a first carrier frequency selection and gain adjustment module. A second carrier frequency selection and gain adjustment module, a first LC filter, a high-frequency compensation network, a communication bus and a power switch. The technical problems that an existing multi-channel parallel digital constant-voltage power amplifier is only limited to power amplifier channels in the same power amplifier, the parallel connection mode only uses the first channel as a main power amplifier and only uses the other channels as slave power amplifiers, and certain limitation exists are solved.

A zonal audio system for a vehicle may include a microcontroller unit (MCU), a non-volatile storage communicatively coupled with the MCU that stores audio files for different types of audio functions, and a digital signal processing (DSP) module communicatively coupled with the MCU. The DSP module may process multiple types of audio functions simultaneously through a layered architecture that consolidates and prioritizes different audio streams including emergency calls, acoustic vehicle alerts, chimes, and infotainment.

The invention discloses an ultra-low power consumption control method and system for an earphone end, and the method comprises the steps: monitoring power utilization parameters, and carrying out the analysis to obtain a power consumption state; predicting a power consumption trend based on the connection history and the signal environment; and setting a state regulation and control plan based on the power consumption state and the power consumption trend. Therefore, according to the monitored electric quantity and the use environment, the subsequent power consumption condition is predicted, then according to the prediction result, the state regulation and control plan is dynamically set, the standby state is automatically converted, a user does not need to manually start a shipping mode, it is ensured that the earphone state and power consumption are maintained in a reasonable interval, and the standby endurance is greatly improved.

The invention discloses a conference sound reinforcement system based on an AI intelligent algorithm and 360-degree omni-directional noise reduction, and relates to the technical field of audio signal processing, the conference sound reinforcement system comprises a conference management center, the conference management center is in communication connection with the following modules: a multi-sound-source sensing module used for capturing 360-degree omni-directional sound field information in a conference environment and constructing a sound source space distribution model; according to the invention, the omnidirectional microphone array unit covers all directions of a conference space, synchronously collects audio data streams, eliminates the limitation of a conventional unidirectional microphone, combines the sound source positioning space mapping unit, constructs a sound source space distribution model based on the time difference of arrival and the phase difference through a deep learning model, and improves the sound source positioning accuracy. The azimuth angle, pitch angle and distance parameters of the sound source are accurately analyzed, the position of the sound source is mapped to a virtual space coordinate system, a dynamically updated 3D sound source distribution diagram is generated, the position change and intensity distribution of the sound source are reflected in real time, and the positioning precision in a complex acoustic environment is remarkably improved.

The invention relates to an audio processing method and device, electronic equipment and a storage medium, and the method comprises the steps: obtaining a gain amplitude value of a current audio signal and a signal flow loudness value of the current audio signal when the set volume of the audio signal is detected to be changed; determining the loudness amplitude value of the current audio signal according to the gain amplitude value and the signal flow loudness value; acquiring an audio compensation strategy corresponding to the loudness amplitude value; and performing loudness compensation on the audio signal according to the audio compensation strategy to obtain an audio signal meeting a preset loudness condition. According to the method, the user hearing change caused by the amplitude change of the input audio stream and the set volume change is comprehensively considered, the amplitude change of the input audio stream is calculated by adopting a loudness analysis algorithm, the loudness change is reflected more comprehensively and accurately, and the user experience is improved by calling corresponding compensation parameters. Therefore, the sound experience of the user is greatly improved, and a better audio playing effect is realized.

The invention relates to an effector parameter adaptive adjustment method based on machine learning, and the method comprises the steps: constructing a parameter association network model to obtain relation data among parameters in a sound equipment effector, and generating a topological structure description of a parameter association network; carrying out simulation analysis on the displacement chain reaction, and determining a propagation direction and intensity distribution of an influence path; analyzing the action weight of the key node, positioning the abnormal fluctuation according to the action weight of the key node, and determining the specific position and the influence range of the abnormal fluctuation; extracting a subtle change mode of the infinitesimal displacement, judging whether a chain reaction is triggered or not, evaluating a potential risk level, and determining a priority sequence of an influence range; and generating a parameter adjustment scheme according to the priority ranking, and obtaining effector configuration parameters of the optimal compensation scheme.

The invention relates to a sound box built-in power amplifier remote audio transmission and control method based on 5G communication, and the method comprises the steps: obtaining real-time monitoring data in an audio data transmission process, and generating an initial delay distribution data set for each audio channel; analyzing network delay fluctuation of each audio channel through a timestamp calibration method, and determining a delay reference value; setting a delay deviation threshold range according to the delay reference value, identifying an abnormal channel and recording a delay offset; analyzing a phase relation of the audio data stream after time correction, calculating a phase deviation angle and determining a phase adjustment parameter; performing phase correction on the audio data stream through a digital signal processing module to generate audio signals with consistent phases; detecting tone quality consistency difference among multiple devices, adjusting audio output feature data and optimizing a time window configuration scheme; and dynamically updating the audio channel coordination strategy, and storing correction parameters to a system database.

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 audio data transmission method and apparatus for a wireless earphone, and a storage medium and a terminal are provided. A play mode adopted by the wireless earphone includes a first play mode and a second play mode, and the method includes: determining whether the first play mode currently adopted by the wireless earphone satisfies a transmission bandwidth required by to-be-played audio data at least based on parameter information of the to-be-played audio data; and based on the first play mode not satisfying the transmission bandwidth required by the to-be-played audio data, transmitting the to-be-played audio data through a connection channel corresponding to the second play mode.

The present disclosure relates to the technical field of audio control. Provided are a computer-implemented method for audio control, and an electronic device and a storage medium. The computer-implemented method for audio control comprises: on the basis of a specified measurement signal, performing acoustic effect testing on a target space, in order to obtain environmental response information, wherein the environmental response information comprises impulse response information and post-separation impulse response information; analyzing the environmental response information, in order to obtain acoustic characteristic information of the target space; on the basis of the acoustic characteristic information, performing compensation processing on audio to be processed, in order to obtain regulated audio; regulating gains of multiple channels corresponding to the regulated audio, in order to obtain re-regulated audio; and performing multi-channel collaborative optimization processing on the re-regulated audio, in order to obtain audio having a target effect. The method achieves precise compensation for spatial acoustic defects, resolves inter-channel crosstalk, improves the acoustic image localization and spatial perception, significantly improves the perceptual audio quality, and optimizes an output.

The invention discloses a volume adjusting method and system of a vehicle-mounted sound system, electronic equipment and a storage medium, and relates to the field of vehicle control, and the method comprises the steps: obtaining vehicle operation information, and starting a passenger detection system according to a vehicle motion state; passenger vision related information is obtained by starting a passenger detection system; through a multi-modal fusion algorithm, generating judgment information of the state of the passenger in the vehicle, and judging whether the output volume of the vehicle-mounted sound equipment is greater than a preset playing volume threshold value or not; if yes, judging whether the judgment information of the passenger state in the vehicle confirms that the passenger is in a sleep state; judging whether the in-car entertainment system has an active media focus or not according to the sleep state; if yes, a dialogue manager is started, and inquiry voice is directionally projected to the position of the driver for voice interaction; judging whether a multimedia volume reduction instruction is received or not through voice interaction; according to the scheme, the sleep comfort level of passengers is improved, and the distracting risk of drivers is reduced.

The application discloses a loudspeaker phase equalization signal processing system and a pole parameter determination method thereof, which equalizes the loudspeaker phase response in a set equalization frequency range to linear phase, and the system comprises a phase controller and a group delay equalizer in cascade; the phase controller is a first-order IIR all-pass filter; the group delay equalizer comprises S low-order IIR all-pass filters in cascade, wherein the first N1 low-order IIR all-pass filters are low-order causal IIR all-pass filters, and the last N2 low-order IIR all-pass filters are low-order non-causal IIR all-pass filters. The application can equalize the loudspeaker phase through the all-pass filter with low operation complexity and parameters, can significantly reduce the calculation amount required by low-frequency phase equalization, can improve the linear degree of the loudspeaker phase response, and can improve the sound quality and listening experience.

Devices, systems, and methods are disclosed for customizing audio experiences associated with operation of a ridable device. An electronic device having a processor detects contextual information such as speed, acceleration, ambient noise level, environment type, or location of the ridable device. Sensor data from accelerometers, gyroscopes, microphones, cameras, or positioning systems may be processed to identify these conditions. Based on the detected context, the electronic device determines audio configuration parameters, including selection of audio content, adjustment of volume, blending of tracks, or modification of tempo or pitch. The determined parameters are applied to output audio through one or more speakers of the electronic device, the ridable device, or associated accessories. In some implementations, playback is synchronized with other riders using a common time reference, geo-fence triggers are employed to provide location-specific content, or warning sounds are generated when nearby objects are detected, enhancing both rider enjoyment and environmental awareness.

Embodiments of this disclosure provide a device operation method and apparatus, and an electronic device. In the device operation method, after obtaining a vibration signal collected by a bone conduction microphone connected to the electronic device, the electronic device extracts a vibration signal feature of the vibration signal, and then compares the extracted vibration signal feature with a control feature stored in the electronic device. When a comparison result is that the extracted vibration signal feature matches the control feature, the electronic device performs an operation corresponding to the control feature. The vibration signal collected by the bone conduction microphone may be a bone conduction audio signal, and the bone conduction audio signal is skin vibration triggered when a nasal cavity and / or a throat of a user who uses the electronic device make / makes a specific sound.

The invention belongs to the field of artificial intelligence, particularly relates to an intelligent sound field adaptive system of digital professional sound equipment, and aims to solve the problems of inaccurate sound field regulation and control, response lag, dependence on manual tuning and the like in a complex acoustic environment. The system comprises a sound field sensing module, an acoustic modeling and analysis module, a self-adaptive sound field regulation and control engine, a multi-channel digital audio processing unit and a feedback optimization module, and dynamic sound field modeling, real-time audio processing and environment self-adaptive regulation and control are realized through distributed sensing, hybrid modeling, multi-target optimization and closed-loop feedback. The system supports rapid re-calibration, multi-scene memory and user preference learning, ensures voice clarity and music fidelity, balances full-field hearing consistency, and significantly reduces manual intervention requirements.

Aspects of transducers with feedback transduction are described. One aspect is a transducer system comprising an operational amplifier having an inverting input, a non-inverting input, and an output. The transducer system also includes a piezoelectric microelectromechanical system (MEMS) transducer having a first node and a second node, wherein the first node is coupled to the inverting input of the operational amplifier, and wherein the piezoelectric MEMS transducer is configured to generate an electrical signal across the first node and the second node in response to a signal incident upon the piezoelectric MEMS transducer. The transducer system also includes an attenuator having an input and an output, wherein the input of the attenuator is coupled to the output of the operational amplifier, and wherein the output of the attenuator is coupled to the second node of the piezoelectric MEMS transducer.

A therapeutic vibration device includes a body with at least a first vibration element connected to the body, and at least a second vibration element connected to the body. Additionally, at least one audio transducer may be connected to the body. The first vibration element and second vibration element may have vibration characteristics which are different from each other when compared. Additionally, the device may include a controller that is connected to a user input interface which allows the user to control the vibration elements.

The proposed invention relates to the field of computer engineering and can be used to control electronic devices. The technical result achieved by implementing the claimed invention, in addition to the invention realizing its purpose, is the elimination of the disadvantages of analogues and thus ensuring the rapid disconnection of the bracket and the device placed on it, while ensuring the usual reliability of placing the device on a vertical surface. Another technical result achieved by implementing the claimed invention, in addition to the invention realizing its purpose, is also ensuring precise positioning of the device in space and ensuring a rapid change of its operating mode and / or settings.

A MEMS audio device includes a first wafer having a top with a first cavity and a bottom with a vent hole coupled to the first cavity, wherein the bottom having first contacts, a second wafer disposed upon the first wafer having a flexible material layer disposed above the first cavity, a third wafer disposed upon the second wafer having physical contacts coupled to the second wafer, wherein the third wafer includes a second cavity disposed above the flexible material layer, a wiring wafer disposed below the first wafer having a second vent hole coupled to the first cavity, wherein the wiring wafer having second contacts coupled to the first contacts, and wherein the flexible material layer forms a diaphragm for the MEMS audio device.