4273results about "Signal processing" patented technology

A device capable of using data from multiple sensors to determine an estimated position / direction of a user with respect to the device. The device may use estimated position data, along with confidence data, that originated from a plurality of sensors to fuse the data to determine the user's estimated position and comprehensive confidence of the estimated position. The system may use the location information to perform beamforming / beam steering and / or other downstream operations using the comprehensive estimated position.

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.

This disclosure describes an invention that that mutes specific talkers using at least one beamforming microphone array 102 that is configured to generate N audio signals 108 where each audio signal is associated with a spatial pickup pattern 130, the microphone array(s) 102 are located in a room 200; a processor 104 and memory 105 operably coupled to the microphone array 102, the processor 104 configured to execute the following steps: (a) selectively mute or unmute an individual talker the room with a mute function 106 that controls whether to mute or unmute the individual talker T1-T7 that is picked up by one or more of the individual audio signals, the mute function 106 includes speech learning that learns to identify different talkers in real time to allow the mute function 106 to identify transitions from one talker to another talker in the room 200, (b) output an audio signal 110 based on the selective muting of the talkers T1-T7 in the room 200.

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.

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.

Various implementations include a method of training a road noise cancelation (RNC) system for a vehicle, including: providing inputs to RNC system, the inputs obtained from: a set of ear-mounted microphones on a user, at least one transducer, an accelerometer, a set of cabin microphones in the vehicle, and a controller area network (CAN) bus, the inputs from the set of ear-mounted microphones on the user approximating a signal detected by the ears of the user; adapting a set of parameters in the RNC system defining an estimated signal detected at respective ears of the user based on the inputs; and generating at least one of the following for input during an operating mode of the RNC system: estimated ear microphone signals based on the adapted set of parameters, or a set of projection filters for use in determining an estimated ear signal at the respective ears of the user.

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 hearing system comprises a hearing device, e.g. a hearing aid or a headset, configured to be worn by a user. The hearing device comprises at least one input transducer for providing at least one electric input signal representative of sound in the environment of the hearing device, wherein said at least one electric input signal comprises a target signal component assumed to be of current interest to the user, and a noise component. The hearing device further comprises a noise control system configured to provide an estimate of said target signal component and an estimate of said noise component and to apply a statistical structure to said noise component to thereby provide a modified noise component comprising said statistical structure; and to determine a modified estimate of said target signal component in dependence of said modified noise component. Thereby an improved segregation of sound sources may be provided.

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 discloses a sound field optimization method and device for a multichannel audio system, equipment and a storage medium. The sound field optimization method comprises the following steps: initializing the system and acquiring acoustic characteristic data; simulating a propagation path of sound waves based on a sound ray tracking algorithm and identifying measurement data, and constructing a 3D sound field model; calculating a TDOA result of a sound channel signal between different microphone positions based on a GCC-PHAT algorithm, obtaining a listening position, combining the TDOA result to calculate an optimal delay value, and performing corresponding compensation; a target mathematical model is formed, a sound field optimization problem of a plurality of listening positions is converted into a multi-target optimization problem, a Pareto optimal solution is generated based on an NSGA-II algorithm, a solution for balancing performance of different positions is selected, the sound quality of other positions is considered while the optimal effect of the main listening position is ensured, the listening experience of a single position is remarkably improved, and the listening experience of the single position is improved. And sound field optimization in a whole space range is realized, and immersive audio experience is brought to the user.

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.

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.

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.

Examples described herein relate to transitioning a playback session between portable playback devices such as “smart” headphones, earbuds, and handheld speakers with playback devices of a zone-based media playback system. Exemplary techniques facilitate continuity of playback when transitioning between locations (e.g., from at home to on-the-go or vice versa) or between listening paradigms (e.g., personal or out-loud playback of audio content). An example implementation includes detecting a swap trigger, determining the source playback device(s) and target playback device(s), and performing a playback session swap between the source playback device(s) and target playback device(s).

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.

An ear-worn device includes two or more microphones and noise reduction circuitry including neural network circuitry. The neural network circuitry is configured to: receive multiple audio signals wherein at least two of the multiple audio signals each originate from a different one of the two or more microphones and / or at least one of the multiple audio signals is a beamformed audio signal originating from the two or more microphones; and implement one or more neural network layers trained to perform background noise modification and spatial focusing based on the multiple audio signals, such that the neural network circuitry generates, based on the multiple audio signals, one or more neural network outputs. The noise reduction circuitry is configured to output, based on the one or more neural network outputs, an output audio signal comprising a background noise-modified and spatially-focused version of a first audio signal of the multiple audio signals.

Systems and methods for contactless monitoring of cardiac activities include a vibration sensor operable to collect audio input signals of a user and a processor. The processor is operable to extract, using an audio model, cardiac sound data from the audio input signals, and transfer, using a trained neural network, the cardiac sound data into the simulated electrocardiogram (ECG) data based on a weak alignment between cardiac sounds and ECG.

A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.

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.

An audio system receives sounds from a local area with a set of sensors. The audio system analyzes the sounds to identify an amount of background noise in the environment. The audio system may also identify target sound in the environment, such as voice from a nearby speaker, to reinforce. The amount of background noise is then used to adaptively set a level of reinforcement for the target sound. In low-noise environments, the target sound may be minimally, or not reinforced, while in high-noise environments a reinforcement sound signal may be generated with the target sound so that a transducer array may generate reinforcing sounds for the target sound and increase its contrast with the noise in the environment. By applying the reinforcement dynamically, the system automatically adjusts for different sounds and environments.

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).

Systems and methods for controlling a speaker. In one example, a method includes determining, based on a request to operate a speaker, a type of sound to be produced by the speaker, the request being indicative of whether operation of the speaker is for a communication session or to produce a system sound, and adjusting one or more parameters of the speaker to output (i) speech, based on the request indicating that operation of the speaker is to be for the communication session, or (ii) the system sound, based on the request indicating that the operation of the speaker is to produce the system sound. An example of the method further includes initiating operation of the speaker using the one or more parameters to produce the type of sound.

The invention relates to the technical field of sound effect optimization, and discloses a sound effect optimization method and device based on a coaxial loudspeaker echo wall sound box, and the method comprises the steps: carrying out the acoustic response measurement and acoustic feature analysis of the coaxial loudspeaker echo wall sound box, and obtaining a frequency domain acoustic response characteristic curve; performing acoustic partition calculation on the internal space of the coaxial loudspeaker echo wall sound box to obtain a spatial arrangement parameter of an acoustic processing unit; carrying out acoustic center positioning calculation on the coaxial loudspeaker to obtain a space coordinate parameter and an axial inclination angle parameter of the coaxial loudspeaker; carrying out sound reflection angle calculation and material combination optimization on the inner wall structure of the echo wall sound box to obtain an inner wall geometric curvature parameter and a box body material lamination parameter; sound pressure level frequency response curve analysis and sound phase positioning precision measurement are carried out on the inner wall geometric curvature parameters and the box body material lamination parameters, sound box acoustic performance verification data are obtained, the uniformity of the sound field in the sound box is guaranteed, and then the acoustic performance of the sound box is improved.