6689results about "Transducer circuits" patented technology

The invention discloses a low-delay adaptive wireless audio transmission method, and relates to the technical field of audio data transmission. The invention relates to a low-delay self-adaptive wireless audio transmission method, which comprises the steps of audio signal acquisition and coding, priority scheduling and time slot allocation, wireless channel transmission and interference resistance, congestion control algorithm optimization, decoding and reconstruction of a receiving end, and dynamic buffer management and delay optimization. According to the method, low-delay transmission of audio signals is ensured through a plurality of technologies such as priority scheduling, time slot allocation, dynamic buffer management and adaptive coding, the stability of audio transmission is remarkably improved by combining congestion control, forward error correction (FEC) and adaptive interference elimination technologies, and even in a complex wireless environment, the stability of the audio transmission is greatly improved. And the audio signal can keep high quality, and the transmission stability is ensured by quickly responding to channel change, eliminating interference, correcting errors, reducing packet loss and retransmission.

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.

The invention relates to the technical field of audio control, and provides an audio control method and device, electronic equipment and a storage medium. The audio control method comprises the steps that a sound effect test is carried out on a target space based on a specified measurement signal to obtain environment response information, and the environment response information comprises pulse response information and information obtained after pulse response is separated; analyzing the environmental response information to obtain acoustic characteristic information of the target space; performing compensation processing on the to-be-processed audio based on the acoustic characteristic information to obtain a regulated audio; regulating and controlling the gains of the multiple sound channels corresponding to the regulated and controlled audio to obtain audio regulated and controlled again; and carrying out multi-channel collaborative optimization processing on the audio after being regulated and controlled again to obtain the audio with a target effect. According to the method, accurate compensation of the acoustic defect of the target space is realized, the problem of crosstalk between sound channels is effectively solved, the positioning and space sense of the sound image is remarkably improved, the overall hearing sense of the audio is greatly improved, and the audio output is continuously optimized.

The invention discloses an automatic delay calibration method, device and equipment for an intelligent loudspeaker box system and a storage medium. The method comprises the steps that the intelligent loudspeaker box system is initialized, and environment detection is carried out to preliminarily configure system parameters; exciting the target sound channel and playing the target test signal; positioning a first test signal based on a VAD voice activity detection algorithm, judging the start and end of the first test signal, and identifying the time difference of the first test signal to obtain the delay time of the main sound channel; positioning the arrival time of the second test signal based on a peak velocity search algorithm, and identifying the delay time of the subwoofer; and performing delay time calibration on the subwoofer and each main sound channel, and dynamically adjusting the delay calibration parameter of each sound channel according to the delay time calibration amount, thereby realizing accurate time alignment of the subwoofer and the main sound channel. The problem of low-frequency signal delay which is difficult to process in the prior art is solved, and a breakthrough progress is achieved especially in coordination of the subwoofer and other sound channels.

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.

The invention provides an open earphone audio optimization method based on a self-adaptive compensation algorithm, and aims to improve the audio auditory experience of a user. The method comprises the following steps: firstly, acquiring an environmental noise signal through an earphone microphone, receiving a target audio signal, analyzing auditory perception of a user in a current environment by combining an ISO226 loudness curve and the like, and determining auditory sensitivity of the user to different frequency bands (including high frequency, intermediate frequency and low frequency); and then, enhancing the high-frequency part, deepening the low-frequency part and performing smooth transition compensation on the intermediate-frequency part by using a self-adaptive compensation algorithm. And then, dynamically adjusting the low-frequency part by using a hyperbolic filter, generating an adjusted audio signal, and generating a harmonic signal based on the amplitude and frequency characteristics to further optimize the tone quality. Finally, the optimized audio signal and the generated harmonic signal are provided for a user together, and the effects of automatically optimizing audio output in different environments and enhancing tone quality performance are achieved.

The invention discloses an earphone noise reduction method based on an improved Transform model, an earphone and a storage medium. The earphone noise reduction method comprises the following steps: acquiring a current environment audio signal, position information and attitude information of the earphone; extracting an audio feature sequence from the environment audio signal; extracting an auxiliary feature sequence from the position information and the attitude information; inputting the audio feature sequence and the auxiliary feature sequence into an improved Transform model configured with a proxy attention mechanism to obtain a matching value of each scene type; and obtaining a target scene type according to the size of the matching value, and obtaining a noise reduction parameter corresponding to the target scene type. By improving the Transform model and the multi-source feature information, the recognition and classification functions of the complex audio scene where the earphone is located are achieved, the efficiency and accuracy of recognizing the complex audio scene by the earphone are improved while low calculation complexity is kept, then the matching degree of adaptive noise reduction of the earphone is improved, the adaptive noise reduction effect of the earphone is improved, and the user experience is improved. And the user experience is improved.

An audio cable connector with selectable impedance includes low impedance interface circuitry and high impedance interface circuitry, and a user-operable switch to allow a user to select either of the two impedances to connect to a headset microphone. With the audio cable connector in communication between a headset and equipment, the user may select the appropriate impedance interface circuitry to allow the headset to operate properly with the equipment.

The invention relates to the technical field of audio processing, in particular to an intelligent power amplifier calibration method, system and device. The method comprises the following steps: monitoring corresponding voltage and current waveforms of a power amplifier in real time and carrying out load impedance response analysis to obtain power amplifier load impedance; acquiring a power amplifier output audio signal, and performing spectrum analysis and audio distortion detection to obtain a power amplifier output audio distortion degree; performing gain self-adaptive adjustment on the power amplifier based on the power amplifier output audio distortion degree to obtain power amplifier output self-adaptive adjustment gain; performing frequency response optimization corresponding to different frequency bands on the power amplifier output audio frequency spectrum to generate power amplifier response optimization output frequencies corresponding to different frequency bands; and performing intelligent feedback calibration on the power amplifier based on the power amplifier output adaptive adjustment gain and the power amplifier response optimization output frequency corresponding to different frequency bands to generate an intelligent calibration power amplifier. The overall performance and the use experience of the power amplifier can be improved.

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.

According to the DMA controller applied to audio data carrying, the control method and the I2S system provided by the invention, the bottleneck that multi-channel audio data and 24-bit audio data cannot be transmitted through general DMA is effectively solved by introducing the DMA controller special for audio. The method supports various audio data formats, and can intelligently and automatically select the optimal DMA data transmission strategy according to specific configuration information such as the number of sound channels, the width of the audio data and the width of a groove. According to the innovation, the audio data transmission efficiency and flexibility are remarkably improved, the burden of a CPU in the data carrying process is greatly relieved, the CPU can process more complex tasks in a centralized computing power mode, and therefore the overall system performance is improved.

The invention relates to the technical field of music sharing, and particularly discloses a music sharing method and system based on an intelligent Bluetooth headset and the Bluetooth headset. Comparing the working state signal interference evaluation parameters of each Bluetooth earphone with a working state signal interference evaluation parameter threshold value of the Bluetooth earphone, and if the working state signal interference evaluation parameter of a certain Bluetooth earphone is greater than the working state signal interference evaluation parameter threshold value of the Bluetooth earphone, switching a working channel of the Bluetooth earphone; according to the method, the quality of each channel is evaluated, the to-be-adjusted Bluetooth headset is subjected to channel switching according to each channel switching reference index of the to-be-adjusted Bluetooth headset, and the working channels of the Bluetooth headset are dynamically switched, so that the Bluetooth headset can be helped to maintain stable connection, disconnection or unstable signal conditions are reduced, and the service life of the Bluetooth headset is prolonged. And the data transmission rate can be improved, and congestion is reduced.

A flexible display device is disclosed. The flexible display device may comprise a first body, a second body, a flexible display, a first microphone, a first cavity and a first variable cavity. The first variable cavity is formed to extend from the first cavity and have a volume that varies when the second body moves with respect to the first body. When the second body moves with respect to the first body, the volume of the first variable cavity changes so that resonance frequencies in the first cavity and the first variable cavity can vary. By using the first variable cavity, the sensitivity of the first microphone can be improved and the state of the flexible display device can be identified.

The invention belongs to the technical field of sound equipment control, and particularly relates to an intelligent sound equipment control method and system and a storage medium, and the method enables the interaction control of the intelligent sound equipment on home equipment to be based on the control effect, and can screen out the interaction control record with the abnormal control effect based on the historical interaction control record. According to the method and the device, the voice control instruction is subjected to abnormal traceability and voice control instruction complexity correlation analysis, so that the targeted adjustment direction of the sound interaction control is obtained, feedback closed-loop adjustment of the sound interaction control is realized, the blank of the existing sound interaction control on feedback control is effectively filled, and the user experience is improved. Continuous optimization and improvement of sound interaction control are accelerated and promoted to a certain extent, so that the user experience is improved continuously, and sustainable development of the intelligent sound is facilitated.

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 present application discloses an adaptive noise elimination method, device and equipment for brain-computer interface headphones, the method comprising: obtaining an electroencephalogram (EEG) signal and a time-frequency component set collected by the brain-computer interface headphones, and obtaining the complexity corresponding to each time-frequency component in the time-frequency component set; determining a characteristic noise component in the corresponding time-frequency component according to multiple complexities; obtaining environmental influencing variables, performing nonlinear regression analysis according to the EEG signals and the environmental influencing variables, and obtaining environmental control factors; obtaining a variable prediction model corresponding to the environmental influencing variables, and determining an EEG control signal according to the environmental control factors and the variable prediction model; obtaining an EEG noise sequence and a corresponding noise steady-state factor and a noise dynamic factor according to the characteristic noise component and the EEG control signal to construct a noise characteristic vector; determining a control parameter of the EEG signal according to the noise characteristic vector, and adjusting the signal processing unit of the brain-computer interface headphones according to the control parameters to achieve adaptive noise elimination.

According to the sound box directional broadcasting method and device, the equipment and the storage medium provided by the invention, the initial sound box parameters and the sound field data of the target area are acquired, and the sound field data comprise the sound box array position information and the site characteristic information; constructing an acoustic model of the target area according to the site characteristic information and the initial sound box parameters, and simulating a coverage effect of sound in the target area; according to a simulation result, optimizing the initial sound box parameter to obtain a target sound box parameter; and according to the target loudspeaker box parameter, setting a loudspeaker box array position and a beam forming parameter of the loudspeaker box in the target area. According to the invention, by accurately controlling the position of the loudspeaker box array and the beam forming parameters, the sound can be ensured to uniformly cover the target area, and the common problems of non-uniform sound, interference and the like in the traditional broadcasting system can be avoided at the same time. According to the directional broadcasting method, sound can be effectively concentrated to a target area, and interference to a non-target area is reduced.

The invention provides a sound effect calibration method, device and system of an audio playing system and a storage medium. In the mode, a subwoofer and a main sound box are controlled to play test audio, an audio acquisition device is utilized to acquire transmission audio at a main sound listening position, sound time delay and volume parameter difference are analyzed, and the subwoofer is calibrated in combination with an audio response curve of a scene. Therefore, the problem that in the prior art, the accuracy of sound effect calibration between the subwoofer and the main loudspeaker box is low, and consequently the hearing experience of a user is low is solved.

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 method of making an acoustic sensor (e.g., a piezoelectric sensor for a piezoelectric microelectromechanical systems microphone) includes forming or depositing one or more piezoelectric layers to define a beam extending between a proximal portion and a distal tip (e.g., unsupported free end), the beam having a width in plan view that is greater at a location distal of the proximal portion than at the proximal portion. The method also comprises attaching the beam to a substrate in cantilever form so that the proximal portion of the beam is anchored to the substrate and the distal tip is a free unsupported end of the beam. One or more electrodes are disposed on or in the proximal portion of the beam.

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 present utility model discloses a Bluetooth speaker table lamp with essential oil aromatherapy, including a base, an aromatherapy mechanism arranged on the base, a lamp panel arranged around the aromatherapy mechanism, a lampshade sleeved on the lamp panel and abutting against a top of the base, and an aromatherapy control board arranged at a top of the aromatherapy mechanism, where the aromatherapy mechanism includes an essential oil bottle, a microporous tube B communicating with the essential oil bottle, a silicone pad of the essential oil bottle that is arranged on a bottle mouth of the essential oil bottle, an aromatherapy chamber detachably connected to the bottle mouth of the essential oil bottle, an air pump motor arranged beside the essential oil bottle, and a microporous tube A with one end communicating with the air pump motor.

The present invention provides a wireless audio data transmission method and related device. A master device transmits a control packet to N slave devices when the master device fails to receive an audio data packet sent by a first target device being one of N slave devices, or fails to send the audio data packet to a second target device being one of the N slave devices in a first sub-event of one isochronous interval. The control packet indicates at least one third time slot within a second sub-event of the one isochronous interval in which the first target device is permitted to transmit the audio data packet to the master device, and / or at least one fourth time slot in the second sub-event in which the second target device is permitted to receive the audio data packet transmitted by the master device. The time slots can be used for both receiving audio data packets and transmitting audio data packets, so that agile multiplexing of time slot resources between audio input slave devices and audio output slave devices can be realized, and link efficiency and transmission reliability during transmission of audio data streams can be improved.