4376results about "Transducer circuits" patented technology

The invention discloses an intelligent control method and system for tunnel loudspeakers, and relates to the technical field of tunnel audio control, environmental parameters in a tunnel are collected by adopting a mode of deploying sampling equipment in a distributed manner, and data preprocessing is performed in a targeted manner for different environmental parameters; a sound propagation model is established, and attenuation and delay of sound in different environments are simulated. According to the intelligent control method and system for the tunnel loudspeakers, various temperature and humidity sensors are arranged in the tunnel, and the absolute humidity is calculated in combination with the air pressure data, so that the sound velocity is accurately corrected, and the phase difference of the multiple loudspeakers is reduced; an adaptive Kalman filtering algorithm is adopted to process wind speed data, and reliable input is provided for a sound propagation model; a deep reinforcement learning algorithm is used to carry out collaborative optimization on parameters such as amplitudes and directional angles of multiple loudspeakers, a Bayesian network is used to detect loudspeaker faults, and Delaunay triangulation and a distributed consistency algorithm are combined to realize rapid fault reconstruction.

The invention relates to an intelligent microphone pickup and speech enhancement method, system and device, and the method comprises the steps: obtaining an audio signal collected by a dual-silicon microphone array, and carrying out the cross-correlation analysis of the audio signal, and obtaining a corresponding human voice signal correlation feature; performing sound source positioning analysis on the audio signal based on the human voice signal correlation feature to obtain a corresponding target human voice signal; performing frequency characteristic analysis on the target human voice signal, and performing segmented dynamic gain processing on the signal according to a preset frequency band range to obtain a corresponding voice enhancement signal; performing noise component adaptive filtering processing on the target human voice signal to obtain a corresponding noise suppression parameter; and inputting the speech enhancement signal and the noise suppression parameter into a preset echo cancellation model for joint optimization to obtain a corresponding output human voice signal. According to the invention, the coupling problem of multiple acoustic interferences can be effectively solved.

The invention discloses an intelligent sound effect processing system for an automobile DSP (Digital Signal Processor) power amplifier, particularly relates to the field of optimization of a vehicle-mounted sound system, is used for solving the problem of insufficient sound effect consistency and intelligent adaptation in a dynamic sound field, and is characterized in that acoustic data are acquired in real time through a microphone array, and a three-dimensional acoustic geometric model is constructed by combining ray tracing and an image source method; sound wave propagation paths and dynamic changes are accurately captured; frequency bands of low-frequency, intermediate-frequency and high-frequency signals are divided, spectrum distribution differences and gain fluctuation centers of all seats are evaluated, and a sound field consistency adjustment coefficient is generated and used for guiding design and optimization of a directional beam forming filter, adjusting amplitudes and phases of all sound channels, forming directional sound beams and suppressing noise interference; therefore, the method can dynamically adapt to the acoustic environment change in the vehicle, improves the sound effect balance and high fidelity, and enhances the auditory experience and satisfaction of passengers.

The invention provides an audio processing method, electronic equipment and a storage medium, belongs to the technical field of audio processing, and is particularly suitable for a Bluetooth headset. The method comprises the following steps: acquiring an audio signal and preprocessing the audio signal through a multi-channel filtering and dynamic noise suppression algorithm; performing spectrum decomposition on the preprocessed audio signal by using short-time Fourier transform based on dynamic time window length, and extracting spectrum characteristics and power distribution of a plurality of frequency bands; an external interference source is detected in real time, an adaptive filtering algorithm is used for interference signal suppression, and an audio signal after interference suppression is obtained; according to the real-time change data of the environmental conditions and the multi-modal state data of the user, dynamically adjusting a plurality of output parameters, and optimizing the audio signal; and generating a final optimized audio signal, and outputting the final optimized audio signal to a loudspeaker of the Bluetooth earphone for playing. According to the method, the audio quality of the Bluetooth headset can be effectively improved, external noise interference is reduced, and the user experience is optimized.

Disclosed are an audio parameter adjustment method, an audio parameter adjustment apparatus, an audio parameter adjustment device and a storage medium. The method includes: obtaining a target audio effect of an audio system, and analyzing the target audio effect to obtain a corresponding audio metric parameter; determining a target environment where the audio system is currently located, and calculating an environmental optimization parameter of the target environment; calculating, based on the audio metric parameter and the environmental optimization parameter, an optimization parameter of the audio system; and performing adaptive adjustment on the audio system based on the optimization parameter of the audio system.

A logarithmic amplifier includes programmable gain amplifiers each having a different gain, wherein an input of each of the programmable gain amplifiers is coupled to an input of the logarithmic amplifier; and a summing circuit having inputs coupled to a corresponding output of each of the programmable gain amplifiers and an output coupled to an output of the logarithmic amplifier, wherein the summing circuit generates a logarithmic transfer function having piecewise linear segments.

Provided is a sound reproduction device which reduces air noise. The sound reproduction device includes a casing in which an opening section communicating with an inside and an outside is formed, and a microphone that is disposed in an internal space communicating with the opening section, in which a value obtained by dividing a volume of the internal space from the opening section to the microphone by an area of the opening section is 2.9 or more.

A communications system for a restaurant includes multiple field communications units, a portable communications device, and processing circuitry. The field communications units include a speaker and a microphone. The portable communications device is transportable by an order taker of the restaurant. The processing circuitry obtains a user input to transition the portable communications device between multiple different channels. The multiple different channels correspond to one of the multiple field communications units. The processing circuitry transitions the portable communications device to one of the multiple different channels according to the user input. The processing circuitry facilitates end-to-end bi-directional audio communication between the order taker and a customer by operating the portable communications device and the one of the multiple field communications units to exchange audio data over the one of the multiple different channels.

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 provides an intelligent video sound box regulation and control system based on AI. The intelligent video sound box regulation and control system comprises a video data processing module, a user behavior analysis module, an environment perception module, a computing power scheduling module, a parameter optimization module, a real-time interaction module and a data storage module. The video data processing module analyzes the video stream and extracts multi-dimensional features; the user behavior analysis module generates behavior tags by analyzing user actions and voice instructions; the environment sensing module collects environment light intensity, space size and sound field parameters; the computing power scheduling module dynamically allocates GPU and TPU computing power resources; the parameter optimization module generates sound box regulation and control parameters based on the user behavior label and the environment parameters; the real-time interaction module adjusts the volume, the frequency response and the video playing frame rate of the sound box according to the regulation and control parameters; and the data storage module synchronizes video streams and regulation and control parameter historical data through distributed storage nodes. The system operation efficiency and the resource utilization rate can be improved.

A method of operating a microelectromechanical system (MEMS) includes, in a first operational mode, converting an analog output of the MEMS into a first internal data stream and a first external data stream having a first sampling rate; transitioning from the first operational mode to a second operation mode without restarting the MEMS; and in the second operational mode, converting the analog output of the MEMS into a second internal data stream having a second sampling rate different from the first sampling rate, and performing a sampling rate conversion of the second internal data stream to generate a second external data stream.

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 relates to the technical field of sensor identification, in particular to a sound parameter determination method and system. The method comprises the following steps: obtaining sound multi-modal sensing data, and carrying out multi-dimensional feature vector combination to obtain a high-dimensional feature data set; performing feature mapping on the high-dimensional feature data set to obtain a sound response feature map; performing environment dynamic parameter adjustment strategy analysis based on the sound response feature map to obtain an environment dynamic parameter adjustment strategy; obtaining user sound control data, and performing user auditory preference analysis to obtain user auditory preference data; and according to the user auditory preference data, performing user personalized sound adjustment feedback optimization on the environment dynamic parameter adjustment strategy to obtain an optimized feedback sound dynamic adjustment strategy, and transmitting the optimized feedback sound dynamic adjustment strategy to the sound control platform to execute a parameter adjustment task. According to the invention, the problems of subjective errors and untimely response to environmental changes in traditional manual adjustment can be 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.

The invention relates to the technical field of sound equipment, and discloses a light follow-up sound equipment based on data analysis and a control system thereof, and the system comprises an audio signal processing module which is used for carrying out the noise filtering of audio data, and extracting the real-time audio signal intensity; the volume regulation and control module is connected with the audio signal processing module and used for regulating and controlling the volume according to the intensity of the real-time audio signal after noise filtering; the frequency analysis module is used for analyzing frequency characteristics of the audio data so as to determine corresponding light color parameters; according to the system, the audio is processed by adopting dynamic filtering and weighting algorithms, and the signal intensity is accurately extracted to regulate and control the volume. High synchronization of light and music is realized through delay compensation and adaptive threshold analysis frequency characteristics in combination with multi-stage dimming, and meanwhile, environment interference correction and system self-inspection are realized, so that the audio-visual experience of a user is 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.

The invention discloses a double-earphone sound effect adjusting method and system, a storage medium and a program product, and relates to the field of loudspeakers, and the method comprises the steps: collecting an environment audio signal through a binaural microphone, and executing spectrum analysis to obtain a frequency distribution characteristic of the environment audio signal; based on a preset human voice frequency range, extracting an audio signal conforming to the human voice feature from the frequency distribution feature as a target audio; calculating the time difference and the intensity difference of the target audio between the left ear and the right ear to obtain the human sound source direction and the relative distance in the environment; acquiring head attitude sensor data, and calculating a matching included angle between the head orientation of the user and the human sound source direction based on the head attitude sensor data; when the matching included angle is smaller than a preset threshold value, increasing the audio signal gain of the head facing the direction; and when the conversation voice of the user is detected, the audio signal gain is kept unchanged. By implementing the application, the interaction efficiency when the double earphones are used can be improved.

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.

The present invention discloses an audio recording and conversion method for a Bluetooth headset, specifically related to the field of audio recording and conversion. The present invention pairs and connects with the Bluetooth headset by turning on the Bluetooth function of the sound source device; subsequently, the recording process is started by pressing the "Recording" button on the headset, and the system guides the user to select the audio format and confirm the recording settings; during recording, the built-in microphone of the headset collects audio signals in real time, denoises and amplifies the collected audio signals, and then converts the processed audio signals into digital signals; the processed digital signals are embedded with watermarks, and the watermark evaluation value is calculated, and the watermark is verified by comparing the watermark evaluation value; then, the digital signals are converted into analog signals and reconstructed into audio signals, and finally, the system performs watermark detection on the reconstructed audio signals, and ensures the integrity and authenticity of the audio signal data by comparing the detected watermark information.

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.

A method and apparatus are described for synchronization between Bluetooth audio transmitters. In one example, an audio source includes a synchronization bus. A first audio transmitter is coupled to the audio source and to the synchronization bus. The first audio transmitter is configured to assess available radio channels, select a set of radio channels, send transmit commands to a second audio transmitter through the synchronization bus, and to transmit a first set of audio streams from the audio source to a first set of one or more audio sinks using the selected radio channels. The second audio transmitter is configured to receive the transmit commands from the first audio transmitter through the synchronization bus and to transmit a second set of audio streams from the audio source to a second set of one or more audio sinks in accordance with the transmit commands.

A signal processing method for selectively removing noise from a digital input signal is provided. The method includes the steps of, for example, receiving, by a signal input, a digital input signal of a device under test, determining, by an analysis circuit, a DUT noise contribution to noise of the digital input signal, wherein the DUT noise contribution is associated with noise originating from the device under test, reducing, by the analysis circuit, a noise level of the digital input signal to a noise level of the DUT noise contribution determined, thereby obtaining a noise-reduced output signal, and providing the noise-reduced output signal to a measurement application for further processing. A signal processing system for carrying out the method or others is also provided.

In an example, the present invention provides a micro-speaker device. The device has a movable diaphragm device comprising a thickness of silicon or graphene material which has a first surface and a second surface opposite of the first surface and sensor to track position of the diaphragm. The device has a housing enclosing the movable diaphragm device, the electrode device and an encapsulation device. The electrode device can be part of a CMOS device with electronics integrated on to the device that converts input audio signal in to signal that electrostatically actuates the micro-speaker from one or more surfaces.

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.

A playback device comprising a first spectrum radio for communication via a first wireless spectrum and a second spectrum radio for communication via a second wireless spectrum operates in a first mode in which the first spectrum radio is enabled and the second spectrum radio is disabled. While operating in the first mode, the playback device receives a command to join a synchrony group with a display device. Based on the command, the playback device begins to operate in a second mode in which the second spectrum radio is enabled. The playback device receives audio content for playback, wherein the audio content corresponds to video content being displayed at a display device. Using the second spectrum radio, the playback device coordinates with the display device for playing back the audio content in synchrony with the video content.

A pop-up sensor head with a camera transitions between a retracted and extended state. While retracted, the camera lens is covered. While extended, the camera lens is exposed and available for use. A motor maintains the sensor head at a setpoint position, such as in the retracted or extended state. A position data, such as from a Hall effect sensor associated with the motor, may be used to determine a displacement of the sensor head due to an external force being applied, such as a user pressing on the sensor head. Responsive to this input, the sensor head may transition between the retracted and extended state.

An exciter-equipped glass diaphragm, includes: a glass plate structure; and a first exciter and a second exciter, which are attached to the glass plate structure, in which, when a lowest resonant frequency of the first exciter is F1(0) (Hz), and a lowest resonant frequency of the second exciter is F2(0) (Hz), the exciter-equipped glass diaphragm satisfies: 3≤|F1(0)−F2(0)|≤100 (Hz).