1615results about "Transducer circuits" patented technology

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 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 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.

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.

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.

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.

Disclosed is an electronic device which includes at least one camera module, at least one microphone, a communication module, a memory, and a processor operatively connected to the at least one camera module, the at least one microphone, the communication module, or the memory, the processor being configured to detect whether a user has taken a vehicle based on detection of the vehicle through the communication module, determine an existence or nonexistence of a passenger in the vehicle based on at least one of an external electronic device searched for through the communication module, a sound collected by the at least one microphone, or an image obtained from the at least one camera module, identify relationship information on the passenger, analyze an attribute of audio of which an output has been requested, and control an audio output through the vehicle based on the relationship information and the attribute of the audio.

Aspects of the disclosure relate to an audio hub that may be connected to a computing device. The audio hub may comprise multiple wired and wireless communication interfaces that may be used to interface multiple peripheral audio devices with the computing device. The audio hub may comprise a built-in reference microphone that may be used for noise cancelation, automatic gain control, and / or acoustic echo cancelation. The audio hub may additionally be configured for audio dosimetry measurements based on output audio signals as sent to an audio output device connected to the audio hub.

The invention relates to the technical field of audio processing, and discloses a wireless earphone multi-device cooperation method and system based on spatial audio rendering, and the method comprises the steps: extracting the multi-dimensional cooperation features of a wireless earphone, and generating a spatial audio rendering control instruction and a multi-device connection switching instruction; setting a cooperative control strategy set of a spatial audio rendering control instruction and a multi-device connection switching instruction in combination with the scene type, the visual focus coordinate and the noise interference index; calculating a sound field distortion value of the spatial audio rendering control instruction, and determining a dynamic bandwidth allocation value of an equipment audio priority; setting a multi-device audio output threshold of the wireless earphone through the dynamic bandwidth allocation value and the noise reduction intensity adjustment coefficient; and generating a multi-device audio switching scheme of the wireless earphone based on the cooperative control strategy set and the multi-device audio output threshold. The fracture risk of the space sound field in the switching process can be reduced, and the auditory experience of a user is improved.

In one embodiment, a computer-implemented method includes receiving an input audio signal for playback by at least one speaker and at least one tactile transducer, receiving a control signal associated with a user-specified setting for the at least one tactile transducer, and generating a processed audio signal to drive the at least one tactile transducer, wherein the processed audio signal is adjusted based on the control signal independent of a speaker output signal driving the at least one speaker.

The invention provides a microphone configuration method, an earphone assembly, a storage medium and a program product. The method comprises the following steps: acquiring voice signals collected by a plurality of microphones within a preset time period when a call begins, the voice signals collected by the plurality of microphones being voice signals synchronously collected in the same environment; determining a plurality of voice parameters in intensity difference, phase difference and time delay based on the voice signals collected by the plurality of microphones; based on the plurality of voice parameters, obtaining a determination result, the determination result being at least used for indicating a distance relation of the plurality of microphones relative to a mouth of an earphone assembly wearer; and according to the determination result, setting the microphone closest to the mouth of the earphone assembly wearer in the plurality of microphones as a main microphone. Through the scheme of the invention, the call quality of reverse wearing of the earphone assembly can be improved.

The invention belongs to the technical field of vehicle-mounted sound equipment, and provides a sound effect adjustment method and system for self-adaption of scenes inside and outside a vehicle and application, and the method comprises the steps: collecting multi-channel audio data of a portable sound equipment in the vehicle; performing acoustic signal processing on the multi-channel audio data, and positioning the position of the portable sound equipment relative to the vehicle, including the position inside the vehicle or the position outside the vehicle; and executing a corresponding scene audio routing strategy inside and outside the vehicle based on a positioning result. According to the invention, the in-vehicle sound quality experience can be improved through the portable sound equipment of the user; and the system can also be used as a supplement of an intelligent cabin, and the portable sound equipment is distinguished from scenes inside and outside the vehicle, so that the interaction experience of a user is improved.

The invention provides an AI-based loudspeaker nonlinear compensator parameter determination method, system and device, and the method comprises the steps: carrying out the real-time processing of an input audio sequence corresponding to a calibration audio based on a nonlinear compensator, and carrying out the sound amplification output; performing compensator parameter updating according to a preset updating beat until a set convergence condition is met, and determining a target compensator parameter; wherein the compensator parameter updating comprises the following steps: acquiring a reference digital sequence corresponding to the same time window, a current compensator parameter and an acoustic feedback sequence corresponding to the output of a loudspeaker to be calibrated acquired by a microphone, and processing the acoustic feedback sequence, the reference digital sequence and the current compensator parameter corresponding to the same time window through a preset AI updating network to obtain an iterative compensator parameter. The method inputs the original time domain waveform data, does not need to carry out explicit time delay alignment, synchronous compensation and other feature extraction steps, and has the characteristics of flow automation, individual difference adaptation, fast convergence, strong engineering landing and the like.

The invention provides a differential microphone array speech enhancement device and method based on reference point optimization, and relates to the technical field of speech signal processing. The device comprises a microphone array, a pre-processing module, a reference point selection module and a signal processing module. According to the invention, the reference point of the microphone array is improved from the first microphone in the prior art to the geometric center of the invention, the maximum value of the distance rm from the microphones in the microphone array to the reference point is reduced by half, the high frequency band is significantly reduced, and the approximation precision of Jacobi-Anger expansion is ensured. Through reference point optimization, the invention aims to realize a beam pattern with invariable frequency in a broadband range and combined optimization of DF and WNG. While the beam pattern approximation requirement and no distortion constraint are met, the white noise gain can be maximized, and the robustness is improved.

The invention relates to the field of microphone sound reception, in particular to a precise sound reception system and method based on double microphone arrays and 3D visual identification, and the system comprises a sound source positioning module, a visual identification module, an audio processing module, a sound reception control module and a precise tracking module. The visual identification module is used for calculating the listening angle of the microphone array, the audio processing module is used for obtaining the sound arrival time difference of each microphone, the sound reception control module is used for outputting the position of a target sound source, and the precise tracking module is used for enabling the camera to track the target sound source. The method is advantaged in that requirements of real-time interaction scenes are satisfied, sound reception capability of the microphone array in complex scenes is improved, good robustness and accuracy are realized, sound source positioning precision and dynamic tracking capability are improved, and environment adaptability and real-time performance of microphone sound reception are improved.

The invention provides a sound amplification device based on an M-H metamaterial and a sound transmission method using the sound amplification device, the sound amplification device at least comprises an upper shell, a lower shell and a waveguide located between the upper shell and the lower shell, and the waveguide is made of the Mie metamaterial; the waveguide comprises an outer wall, a plurality of sound wave inlets are formed in the outer wall, and the upper shell, the lower shell and the outer wall enclose a waveguide space; a through hole penetrating through the waveguide and the upper shell is formed in the waveguide space, and the through hole is a sound wave outlet in the sound amplification device; the waveguide comprises a plurality of fan parts which are arranged around the through hole in a circumferential radiation manner; a waveguide channel penetrating through the sound wave outlet and the sound wave inlet is formed between the adjacent fan parts; the control current of the sound source is determined by the size of the sound amplifying device and the scattering distribution condition of the sound amplifying device in different resonance modes, the sound at the sound wave outlet is larger than the sound at the sound source in energy level, and an included angle exists in the propagation direction. By determining the control current at the sound source which is most matched with the size of the sound amplifying device, the sound propagation control with low cost, high accuracy and wide application range is realized.

In an embodiment, a device includes: phase frequency detector (PFD) circuitry having an input and an output; an oscillator having an input and an output, the input of the oscillator coupled to the output of the PFD circuitry; clock divider circuitry having a first input, a second input, and an output, the first input of the clock divider circuitry coupled to the output of the oscillator, the output of the clock divider circuitry coupled to the input of the PFD circuitry; and a controller having an output coupled to the second input of the clock divider circuitry.

The invention relates to an audio decoding output stage circuit and audio equipment, the audio decoding output stage circuit comprises an I / V conversion module and a low-pass filtering module, and the I / V conversion module comprises a conversion circuit, a crossed diamond amplification circuit, an overcurrent protection circuit and a Roche network circuit; the conversion circuit comprises a JFET (Junction Field Effect Transistor) audio operational amplifier chip and an I / V (Input / Output) conversion resistor and is used for converting an input current into an analog voltage signal; the crossed diamond amplifying circuit can reduce the input offset voltage; the over-current protection circuit can carry out over-current protection; the rowbell network circuit can prevent high-frequency self-oscillation of the power output stage; and the low-pass filtering module comprises a filtering circuit and an ultrahigh frequency filtering circuit arranged at the output end of the audio decoding output stage circuit. The scheme can bring high-performance sound performance, and has the advantages of lower distortion, higher conversion rate, low output impedance and large pushing current.

The invention discloses a sonic boom suppression method, electronic equipment, a storage medium and a computer program product, and relates to the technical field of audio signal processing, the sonic boom suppression method is applied to a target vehicle, the target vehicle at least comprises an automobile machine and a power amplifier module, and the power amplifier module comprises an audio access. A first control signal is sent to the audio channel, and the first control signal is used for setting the audio channel to be in a mute state; in the upgrading process of the automobile machine, an audio channel is kept in a mute state; under the condition that the automobile machine is upgraded and restarted, if initialization of the power amplifier module is completed, a second control signal is sent to the audio channel, and the second control signal is used for setting the audio channel to be in a non-mute state. According to the invention, the technical problem that the sonic boom cannot be effectively suppressed after the vehicle machine system is upgraded and restarted is solved.

An audio amplifier that implements current mode control without the use of an explicit or separate current mode sensor is disclosed. The audio amplifier may include a pair of feedback loops that provide current from a node located before an inductor of an output filter and current from a node located after the inductor of the output filter to an integrator circuit. The integrator circuit may be formed from existing circuitry of the audio amplifier controller. Thus, current mode control can be implemented without a separate current mode sensor. Further, the audio amplifier can be implemented as a half bridge or as a full bridge design without increasing the number of operational amplifiers compared to the half bridge.

The invention discloses an audio channel verification method and device, equipment, a storage medium and a program product, and relates to the technical field of communication. According to the method, under the condition that a data issuing interface receives audio algorithm data sent by an upper-layer application, the audio algorithm data is packaged into an RTAC format to obtain real-time audio calibration data; the audio link state of the digital audio processor is obtained through the real-time audio calibration module, under the condition that the audio link state is a creation state, a data issuing thread is started, and real-time audio calibration data are sent to the digital audio processor through the real-time audio calibration module. The digital audio processor is used for processing the real-time audio calibration data to obtain a return value and sending the return value to the real-time audio calibration module; and starting a result callback thread, obtaining a return value from the real-time audio calibration module through a result callback interface, and performing audio channel verification according to the real-time audio calibration data and the return value to obtain an audio channel verification result. According to the embodiment of the invention, the audio channel verification flexibility is improved.

The application discloses a KTV system rear surround setting method and device, a terminal and a medium. The method comprises the following steps: acquiring rear surround device static parameters, box environment data, box acoustic data and customer preference data; determining a target position of the rear surround device in the box based on the rear surround device static parameters, the box environment data and the box acoustic data; determining a target adjustment parameter combination corresponding to the rear surround device based on the rear surround device static parameters, the box acoustic data and the customer preference data; processing the rear surround device based on the target adjustment parameter combination, and placing the rear surround device at the target position. The application aims to achieve the optimal acoustic effect of the rear surround device of the KTV system in the box, and ensure that the acoustic effect of the rear surround device meets the preferences of customers in the box.

Apparatus and techniques for adaptively adjusting an input current limit of a boost converter that supplies power to a load, such as an amplifier. An exemplary circuit for supplying power generally includes a boost converter having an output coupled to the load, and logic configured to adaptively adjust the input current limit of the boost converter and apply the input current limit to the boost converter based on an estimated output power of the boost converter. One exemplary method for supplying power generally includes converting an input voltage to an output voltage by a boost converter to power a load for the boost converter, adaptively adjusting the input current limit of the boost converter based on the estimated output power of the boost converter, and applying the input current limit to the boost converter during conversion.

A method of managing audio in a multi-speaker environment, performed by a wearable audio device may be provided. The method may include generating, based on a binaural audio signal captured by the wearable audio device, a virtual sound source map indicating a localized position of one or more sources of sound. The method may include estimating, based on the virtual sound source map, one or more target sources indicating sources of sound of interest to a user of the wearable audio device. The method may include transmitting metadata associated with the wearable audio device, to an electronic device coupled to the wearable audio device, to cause the electronic device to refine the one or more target sources based on the metadata. The method may include receiving, from the electronic device, a processed audio signal associated with at least one refined target source.

The application discloses a kind of adaptive hearing compensation Bluetooth hearing aid optimization method and system, comprising: S1: collecting user individual hearing curve and the background audio data of left and right ear under each typical scene and respectively pre-processing;S2: extracting the time-frequency fusion feature of each typical scene, generate scene feature vector and construct scene feature vector library;S3: calculate scene perception conversion vector, mutation risk vector, smooth gain vector;S4: calculate binaural gain change vector, binaural synchronization entropy vector, phase alignment factor vector and left and right ear collaborative gain vector;S5: according to left and right ear collaborative gain vector, the background audio data of left and right ear pre-processed under current scene is handled, and output to Bluetooth hearing aid.The application can solve the gain mutation problem generated when existing Bluetooth hearing aid switches scene and the problem of spatial perception distortion caused by binaural collaborative disorder.

Example techniques may involve multiple calibrations for one or more playback devices. An example implementation may involve detecting, via a microphone, calibration sounds as emitted by one or more playback devices during a calibration sequence, perhaps by recording first samples while the microphone is in motion through a given environment and recording second samples while the microphone is stationary at one or more particular locations. The implementation may also include determining a first calibration for the one or more playback devices based on at least the first samples of the calibrations sounds and determining a second calibration for the one or more playback devices based on at least the second samples of the calibrations sounds. The implementation may further include applying at least one of (a) the first calibration or (b) the second calibration to playback by the one or more playback devices.

The application relates to a machine learning-based effecter parameter self-adaptive adjustment method, a parameter correlation network model is constructed to obtain relationship data among parameters in an acoustic equipment effecter, and a topological structure description of the parameter correlation network is generated; a displacement chain reaction is simulated and analyzed to determine the propagation direction and intensity distribution of an influence path; then, the action weight of a key node is analyzed, abnormal fluctuations are positioned according to the action weight of the key node, and the specific position and influence range of the abnormal fluctuations are determined; a subtle change mode of a slight displacement is extracted, whether a chain reaction is triggered is judged, and a potential risk level is evaluated, and the priority of the influence range is sorted; a parameter adjustment scheme is generated according to the priority sorting, and effecter configuration parameters of an optimal compensation scheme are obtained.