3251results about "Signal processing" patented technology

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.

A device capable of using image data for purposes of determining a location of a user and audio beam selection to isolate audio in the direction of the user. The beamforming / beam-steering may occur after determining the user's location in order to conserve computing resources that would otherwise have been spent determining beams for non-desired directions. The beamformed audio may be used for speech processing, a communication session involving the device, or other purposes.

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 controller configured to control a sound producing module includes a volume controlling unit configured to determine a demodulation amplitude and a modulation amplitude corresponding to a target volume. The sound producing module comprises a driving circuit and an air-pulse generating device. The driving circuit generates a demodulation driving signal according to the demodulation amplitude and generates a modulation driving signal according to the modulation amplitude, so as to drive the air-pulse generating device. The air-pulse generating device produces sound via generating a plurality of air pulses at an ultrasonic pulse rate.

An ear-worn device is provided that operates to isolate and individually treat the received speech of a target speaker or multiple target speakers from an audio input signal detected in a multi-speaker environment. The ear-worn device uses a machine learning model that receives a voice signature of each of one or more target speakers as input signals, to identify and isolate the component of the audio input signal attributable to the target speaker(s). Once isolated, the target speaker's speech may be enhanced, de-emphasized, or otherwise processed in a manner desired by the wearer of the ear-worn device. The wearer may use an external electronic device, e.g., a phone, to select one or more target speakers in a conversation and / or configure various settings associated with processing the speech on the ear-worn device.

Disclosed herein is an audio system that can be customized by the user (including the developer) and that may allow the user to control more than just the sound levels and may allow the user to select more than one of a few pre-determined settings. Disclosed herein is a development platform that allows manufacturers (including the developers) to test various different parameters, settings, algorithms, functions, processes, accessories, and the like. The development platform allows the manufacturer to simulate the performances from possible options for the audio system, and then load the selected option from the development platform to the audio system. Additionally, the development platform that enables the end user to have the same level of control and ability to access, upload, and control plugins on the end user's device(s).

A mixing device is provided. The mixing device includes: an input unit, inputting sounds; multiple output units, outputting the sounds; a sound setting operator, setting a sound setting value that is a setting value relating to the sounds; a selection operator, selecting the output unit of a target for setting the sound setting value; a setting value acquisition means, acquiring the sound setting value set by the sound setting operator with respect to the output unit selected by the selection operator; and a mixing means, for each of the output units, mixing the sounds input at the input unit based on the sound setting value corresponding to the output unit and being the sound setting value acquired by the setting value acquisition means.

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.

An audio system is provided that includes a loudspeaker, at least one microphone, and at least one controller. The loudspeaker transmits an audio signal into a listening environment defined by at least one wall in a room. The at least one microphone is positioned on the loudspeaker and is configured to capture a reverberated audio signal including a plurality of reverberations and a plurality of peaks. The reverberated audio signal is indicative of the audio signal being reflected from the at least one wall. The at least one controller is programmed to apply a confidence score to the plurality of peaks to obtain a maximum score which is indicative of a maximum peak of the audio signal that is reflected from the at least one wall and determine a distance between the loudspeaker and the wall based at least on the maximum score.

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.

A system for presenting audio content to a user. The system comprises one or more microphones coupled to a frame of a headset. The one or more microphones capture sound from a local area. The system further comprises an audio controller integrated into the headset and communicatively coupled to an in-ear device worn by a user. The audio controller identifies one or more sound sources in the local area based on the captured sound. The audio controller further determines a target sound source of the one or more sound sources and determines one or more filters to apply to a sound signal associated with the target sound source in the captured sound. The audio controller further generates an augmented sound signal by applying the one or more filters to the sound signal and provides the augmented sound signal to the in-ear device for presentation to a user.

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.

A conference host output system disclosed by the present application comprises a main conference room device and a plurality of secondary conference room devices, the main conference room device comprises a conference host, a PoE switch, a plurality of microphone units, a plurality of power amplifiers and a plurality of sound boxes, the secondary conference room devices do not comprise the conference host, and the conference host communicates with each secondary conference room device through a network. The conference host comprises a host control module and an audio control module, each partition output by the conference host is connected with a power amplifier and a sound box, when the partition further comprises at least one microphone unit, the audio control module is connected with the microphone unit through an interface of the PoE switch, and the audio control module is connected with the sound box through the power amplifier. Therefore, the conference host distributes and outputs the plurality of audio input signals to different partitions, an audio processor does not need to be additionally added, an n * n audio matrix function is realized, and a plurality of conference rooms share one conference host, so that the cost of audio equipment is reduced, and the cost performance of a conference audio system is improved.

The present invention introduces a method and system for monitoring an area based on acoustic data acquired from at least one acoustic sensor array. The proposed method initiates by acquiring (10) acoustic data and environmental data including at least temperature. This data is processed (20) to generate a global acoustic image by employing a delay and sum based beamformer for adjusting the phase and amplitude of the acoustic data. The method then determines (25) the presence of objects of interest within the monitored area based on the global acoustic image. Upon identifying an object of interest, a location identification procedure is initiated (30). This procedure includes estimating the initial location of the object then refining the acquired acoustic data focusing on sections which are estimated to pertain to the identified objects, thereby creating a locally refined acoustic image. Utilizing this refined image, the method generates a location data set which indicates the location of the object(s) of interest. Additionally, a categorization procedure (40) may be initiated subsequently to the location identification procedure (30), the categorization procedure (40) results in a categorization data set indicating characteristics of the identified object(s).

The invention relates to the technical field of audio equipment configuration, and discloses a parameter configuration method and device of audio equipment, equipment and a storage medium, which are used for improving the accuracy of parameter configuration of the audio equipment. The parameter configuration method of the audio equipment comprises the following steps: acquiring the type and audio format information of a current input signal source; based on the type of the input signal source and the audio format information, identifying an audio content type and determining an optimal playing mode; adjusting a sound box combination and sound channel configuration of the audio equipment based on the optimal playing mode; and matching a volume level according to the audio content type, and carrying out adaptive volume adjustment on the audio equipment in combination with user preference and environment analysis.

An AI-driven wireless lubricator remote control system includes multiple grease injection lubrication devices, a cloud server host and an artificial intelligence (AI) system. Each grease injection lubrication device includes a central control processing unit, a grease injection lubrication unit, a parameter setting unit, an operating status information unit, a detection unit, a warning unit, a wireless communication unit and a remaining oil volume information unit. The grease injection lubrication device can independently monitor the oil supply status and abnormal operating status information of individual lubrication points, and through wireless remote transmission (network or satellite network), the grease injection lubrication device information distributed on the mechanical equipment scattered around the factory is sent to the cloud server host. Furthermore, the mobile device can view the information of each grease injection lubrication device and issue commands directly through APP and the Internet (connected to the cloud server host).

An acoustic control method according to an embodiment includes: acquiring sensor data from two or more sensors mounted on a moving body that moves in a three-dimensional space; acquiring a position of the moving body; specifying a sound source and a position of the sound source outside the moving body on a basis of output of acoustic event information acquisition processing using the sensor data as input; and displaying, on a display, a moving body icon corresponding to the moving body, in which the display further displays metadata of the sound source that has been specified in a visually identifiable manner reflecting a relative positional relationship between the position of the moving body and the position of the sound source that has been specified.

The invention provides a control method and device of a vehicle-mounted external loudspeaker and a vehicle-mounted external loudspeaker system. The control method comprises the following steps that in response to starting of a vehicle-mounted loudspeaker box mode, the relative position relation between a user and a vehicle is obtained at a preset time interval, and the vehicle is provided with a plurality of vehicle-mounted external loudspeakers making sounds outwards; determining a target loudspeaker and a corresponding sound channel from the plurality of vehicle-mounted external loudspeakers at least based on the relative position relationship; and controlling each target loudspeaker to play the audio data of the corresponding sound channel. The invention provides a mode of taking the vehicle as a large mobile sound box capable of providing a surround sound field, and the provided surround sound field can be adjusted according to the relative position relationship between the user and the vehicle, so that the hearing effect of the user when using the vehicle-mounted sound box is improved, the application scene of the vehicle is expanded, and the user experience is improved.

An audio device comprising, an interface, one or more first output transducers, one or more input transducers, and a processor. The audio device is configured to emit a first ultrasound signal, receive a first reflection indicative of the first ultrasound signal, process the first reflection to determine movement, and in response to movement being determined, adjust one or more audio processing parameters of the audio device.

A concentric circular microphone array (CCMA) may include a number of omnidirectional microphones and an equal number of directional microphones, wherein the omnidirectional microphones and the directional microphones form a plurality of concentric rings on a substantially planar platform. Each of the plurality of concentric rings includes a subset of the omnidirectional microphones and a subset of the directional microphones (e.g., arranged in mixed pairs of microphones). Responsive to a sound source, the omnidirectional microphones and the directional microphones may respectively generate first and second electronic signals. A target beampattern of Nth order may be specified for the CCMA. An Nth order beamformer for the CCMA, that is steerable in a three-dimensional space including the sound source, may be determined based on the specified target beampattern. The beamformer may be executed to calculate an estimate of the sound source based on the first electronic signals and the second electronic signals.

A calibration method comprises adjusting an operating frequency of an air-pulse generating device, such that a sound pressure level (SPL) of the air-pulse generating device is within a specific range. The sound producing module comprises the air-pulse generating device configured to produce sound via generating a plurality of air pulses at a pulse rate corresponding to the operating frequency.

An electronic device including at least one speaker, a communication interface, and at least one processor that is connected with the at least one speaker and the communication interface and controls the electronic device. The processor, when the electronic device is in a first mode, may output one or more first channel signals included in a sound via at least one speaker and control the communication interface to transmit one or more second channel signals in the sound to at least one external speaker, and when the electronic device is in a second mode, mix at least one of the one or more first channel signals and the one or more second channel signals to thereby produce a mixed signal and control the communication interface to transmit the mixed signal to the at least one external speaker.

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.

A sound control system includes a sound generating device disposed at a vehicle ceiling corresponding to a seat region of the vehicle and at a sound space formed within the vehicle to correspond to the seat region, and a sound processing circuit for providing a vibration driving signal to the sound generating device. The sound generating device vibrates an interior material of the vehicle ceiling based on the vibration driving signal.

This disclosure provides methods, devices, and systems for signal processing. The present implementations relate more specifically to a spatio-temporal beamformer. In some aspects, a beamforming system may receive an audio signal via a plurality of microphones, the audio signal including a number (B) of frames for each of the plurality of microphones, each of the B frames for each of the plurality of microphones including a number (N) of time-domain samples. For a first microphone, the beamforming system may transform the B*N time-domain samples into B*N / 2 first frequency-domain samples; transform the B*N / 2 first frequency-domain samples into B*N / 2 second frequency-domain samples; and determine a probability of speech associated with the B*N / 2 second frequency-domain samples based on a neural network model. The beamformer system may determine a minimum variance distortionless response (MVDR) beamforming filter based at least in part on the probability of speech for the first microphone.

A method for acquiring a vehicle-mounted audio signal includes obtaining a target sampling position of an in-vehicle audio signal, and determining, based on the target sampling position, a target microphone set from a candidate microphone collection; and obtaining, by performing enhancement processing on an audio signal acquired by the target microphone set, a target audio signal corresponding to the target sampling position.

The invention discloses an audio signal switching method and device, a medium and a program product. The method comprises the steps that fade-out processing is carried out on N first sampling points with sampling points as granularity, fade-in processing is carried out on M second sampling points, and N processed first sampling points and M processed second sampling points are obtained; and switching the first audio signal into a second audio signal based on the processed N first sampling points and the processed M second sampling points. A transition interval is introduced, fade-in and fade-out processing is carried out in the transition interval, so that the amplitude of the first audio signal is smoothly reduced, the amplitude of the second audio signal is smoothly increased, the two audio signals are naturally transited in the transition interval, and violent amplitude jump caused by direct switching is avoided. Therefore, the noise problem in the audio switching process is reduced.

This application provides a system for generating a drive signal and a method for generating a drive signal for driving a two-in-one device integrating a first component capable of generating sound and a second component capable of generating vibration. The system includes an audio signal pre-processing module, configured to perform low-frequency filtering and acoustic effect processing on an input signal according to a high-pass filtering parameter to obtain an audio signal; a vibration signal generation module, configured to generate a vibration signal based on an acoustic characteristic of the audio signal or a preset vibration library; a signal synthesis module, configured to superpose the audio signal with the vibration signal to output the drive signal; and an amplitude-limiting control module, configured to determine whether the audio signal and the vibration signal exceeds a maximum allowable range when superposed, and attenuate the audio signal and / or the vibration signal.

In aspects of adjusting audio output in a vehicle, a vehicle audio system implements an audio playback manager that detects a location of a first person in the vehicle, the first person located in proximity of a first speaker device configured for audio output. The audio playback manager also detects an additional location of a second person in the vehicle, the second person located in proximity of a second speaker device configured for the audio output. The audio playback manager determines a preference for the first person related to the audio output and an additional preference for the second person related to the audio output. Based on the preference of the first person or the additional preference of the second person, the audio playback manager adjusts a volume of at least one of the first speaker device or the second speaker device.