2266results about "Frequency response correction" patented technology

A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

An air-pulse generating device includes first and second cells. The first cell includes first flap and second flaps configured to perform a first differential movement to form a first opening. The second cell includes third and fourth flaps configured to perform a second differential movement to form a second opening. The first flap and the second flap are opposite to each other and actuated to move toward opposite directions to perform the first differential movement. The third flap and the fourth flap are opposite to each other and actuated to move toward opposite directions to perform the second differential movement. The second flap of the first cell and the third flap of the second cell are actuated to move toward opposite directions and disposed adjacent to each other.

An air-pulse generating device includes a film structure including a first flap and a second flap opposite to each other. The film structure is actuated to operate at an ultrasonic frequency, and the air-pulse generating device produces a plurality of air pulses at an ultrasonic pulse rate. The first flap and the second flap are actuated to perform a differential movement to form an opening or a virtual valve. A slit is formed between the first flap and the second flap, and the opening or the virtual valve is formed because of the slit. The slit is formed as a zigzagging pattern on the film structure.

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 air-pulse generating device includes a film structure including a first flap and a second flap opposite to each other. The film structure is actuated to operate at an ultrasonic frequency, and the air-pulse generating device produces a plurality of air pulses at an ultrasonic pulse rate. The first flap and the second flap are actuated to perform a differential movement to form an opening or a virtual valve. A slit is formed between the first flap and the second flap, and the opening or the virtual valve is formed because of the slit. The slit is formed as a zigzagging pattern on the film structure.

The invention relates to the technical field of audio signal processing, and discloses a room sound correction method, device and equipment for audio equipment and a storage medium, which are used for improving the accuracy and correction effect of correction parameters and effectively improving the acoustic effect of the audio equipment in a room. The room sound correction method of the audio equipment comprises the following steps: acquiring response data of a room to a multi-band preset test signal, and extracting room acoustic data by adopting a time-frequency conjoint analysis method, the room acoustic data comprising frequency attenuation, reflection paths, phase shifts and reverberation time information of different bands; based on the room acoustic data, performing compensation processing by using preset microphone calibration data to obtain a calibrated frequency response curve; performing fitting calculation on a preset ideal frequency curve and the calibrated frequency response curve according to an optimization algorithm to obtain specific correction parameters of the room; and correspondingly adjusting the output of the audio equipment according to the correction parameter.

The air-pulse generating device comprises a film structure, operating at an ultrasonic operating frequency, and a resonant chamber, formed on a side of the film structure. A resonance is formed within the resonance chamber. Due to the resonant chamber, the APG device has a peak on a frequency response of an acoustic property of the APG device at the ultrasonic operating frequency.

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.

A method for providing navigation assistance using a head-worn device that has a camera, several microphones, and several speakers. The method captures sound in an environment as a plurality of microphone audio signals, captures, using the camera, a scene of the environment as a digital image, and processes the digital image to detect an object therein. The method selects, in response to the detection of the object, one of several navigation audio rendering modes. The several navigation audio rendering modes include a first mode that activates an acoustic transparency function to cause the speakers to reproduce the sound of the environment, a second mode that sonifies the object and activates the acoustic transparency function, and a third mode that sonifies the object, partially activates the acoustic transparency function, and an activates active noise cancellation function.

The invention relates to a hearing aid intelligent noise reduction and human voice enhancement system based on electroencephalogram signals, and belongs to the field of biomedical engineering and acoustic signal processing. The system comprises an electroencephalogram signal acquisition module, a multi-channel acoustic sensor array, an embedded neural signal processor, an adaptive beam forming module, a dynamic speech enhancement engine and a dual-mode output device, and constructs electroencephalogram-acoustics joint features by extracting an alpha / theta wave power ratio, a P300 component and auditory cortical Gamma phase synchronism. A deep network is driven to separate target voice, a wave beam direction and a frequency response curve are dynamically adjusted based on neural feedback, a closed-loop calibration unit is innovatively adopted, gain is reversely adjusted according to N1-P2 wave amplitude, heart rate variability and eye movement data are fused to optimize decisions, and when the signal-to-noise ratio is-5dB, the voice recognition rate reaches 89%, the auditory fatigue is reduced by 37%, and the decision conflict rate is smaller than 6%. The defects of attention blind area, noise separation failure and physiological adaptation of a traditional hearing aid are overcome. The system is suitable for the fields of hearing impairment rehabilitation, special communication and intelligent cabins.

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.

An active noise cancellation (ANC) method applied to a headset includes obtaining a first group of filtering parameters, and performing noise cancellation using the first group of filtering parameters. The first group of filtering parameters is one of N1 groups of filtering parameters prestored in the headset. The N1 groups of filtering parameters are respectively used to perform noise cancellation on ambient sound in N1 leakage states. The N1 leakage states are formed by the headset and N1 different ear canal environments. N1 is a positive integer greater than or equal to two.

The invention relates to the technical field of audio processing, in particular to a method and a system for compensating frequency response of a microphone. The method comprises the following steps: obtaining microphone output signal data and multi-dimensional environment data, and carrying out frequency response multi-level feature integration to obtain coupling frequency response data; frequency response-environment parameter correlation analysis is carried out based on the coupling frequency response data, and a frequency characteristic matrix is obtained; acquiring environmental noise monitoring data, and performing frequency band environmental noise calibration to obtain a noise calibration matrix; performing Monte Carlo multi-dimensional parameter optimization based on the frequency characteristic matrix and the noise calibration matrix so as to obtain an optimization compensation strategy; acquiring real-time microphone output signal data, and performing dynamic compensation feedback to obtain dynamic compensation feedback data; and performing user personalized compensation strategy selection according to the dynamic compensation feedback data to obtain a user microphone personalized compensation strategy. According to the invention, the accuracy and quality of audio processing can be improved.

The invention discloses a phase synchronization compensation algorithm for a bone conduction-air conduction mixed sound field of an earphone, and the algorithm comprises the steps: enabling a bone conduction vibration sensor signal # imgabs0 # and an air conduction microphone signal # imgabs1 # to pass through a wavelet packet decomposition module, obtaining a multi-resolution sub-band coefficient group # imgabs2 # and a multi-resolution sub-band coefficient group # imgabs3 #, and enabling the # imgabs4 # to be a sub-band index;

Techniques and apparatuses are described that perform active acoustic sensing. Provided according to one or more preferred embodiments is a hearable, such as an earbud, that is capable of performing a novel physiological monitoring process termed herein audioplethysmography, an active acoustic method capable of sensing subtle physiologically-related changes observable at a user's outer and middle ear. Instead of relying on other auxiliary sensors, such as optical or electrical sensors, audioplethysmography involves transmitting and receiving acoustic signals to monitor a user's biometrics, recognize facial behaviors, and / or sense an environment. In addition to being relatively unobtrusive, some hearables can be configured to support audioplethysmography without the need for additional hardware. As such, the size, cost, and power usage of the hearable can help make health monitoring accessible to a larger group of people and improve the user experience with hearables.

A wearable audio output device (e.g., headphones) having an open design that allows ambient noise to pass to a listener without physically isolating the listener from a surrounding environment. The device may include an open earcup design that may partially or completely surround the listener's ear, and in some examples a portion of the listener's head may be uncovered by the open earcup. To improve comfort, the device includes a floating audio component configured to generate output audio in a direction of the listener's ear without contacting the listener's ear. To reduce audio leakage into the environment, the floating audio component may include two audio transducers and perform beamforming to target the output audio toward the listener's ear and cancel the output audio in other directions. For example, the beamforming may cause constructive interference in one direction and destructive interference in all other directions.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

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.

A device includes a memory configured to store audio data. The device also includes one or more processors configured to obtain, from first audio data, first subband audio data and second subband audio data. The first subband audio data is associated with a first frequency subband and the second subband audio data is associated with a second frequency subband. The one or more processors are also configured to use a first machine-learning model to process the first subband audio data to generate first subband noise suppressed audio data. The one or more processors are further configured to use a second machine-learning model to process the second subband audio data to generate second subband noise suppressed audio data. The one or more processors are also configured to generate output data based on the first subband noise suppressed audio data and the second subband noise suppressed audio data.

The invention discloses an earphone sound effect personalized balanced automatic tuning method and system, and particularly relates to the technical field of earphone tuning, and the method comprises the steps: collecting the hearing feature data of a user in an initialization stage, constructing an individual compensation baseline, and carrying out the self-tuning of the earphone sound effect; solidifying a safety boundary containing the maximum allowable gain, the maximum output, the compression parameter upper limit and the parameter change rate upper limit as a hard constraint; in daily use, multi-source data is collected by taking a wearing session as granularity, the comprehensive quantity of comfort evidence is calculated, and a comfort label sequence is generated; based on the scene feature sequence, the comfort level label and the current parameter, constructing an individualized discomfort risk model, performing multi-objective decision making by combining the definition income, the comfort level risk cost and the security boundary hard constraint, and outputting an updated parameter set; smooth taking-into-effect and safe output of parameters are achieved through gradual issuing and protection state control, and evolution updating and version backspacing are carried out based on long-term logs.

The embodiment of the invention provides an acoustic device and a signal processing method, a bone conduction sounding assembly of the acoustic device generates a first time delay when converting a first driving signal into a bone conduction sound wave, and an air conduction sounding assembly generates a second time delay when converting a second driving signal into an air conduction sound wave. The absolute value of the difference between the second time delay and the first time delay is larger than 100 microseconds, the signal processing circuit obtains an audio signal during operation, generates a first driving signal based on a first component of the audio signal and sends the first driving signal to the bone conduction sounding assembly, and generates a second driving signal based on a second component of the audio signal and sends the second driving signal to the air conduction sounding assembly, one of the first driving signal and the second driving signal is transmitted in a delayed manner relative to the other, so that the time difference between the generation time of the bone-guided sound wave and the generation time of the air-guided sound wave at at least partial frequency is less than or equal to 100 microseconds. According to the scheme, the problem of partial sound can be avoided.

Disclosed is a howling suppression method, including: determining, according to an audio signal collected by a microphone, whether a howling event occurs; if a howling event occurs, determining a howling type corresponding to the howling event; if the howling type is impulse wave howling, reducing an amplitude gain of a frequency band to which the howling event belongs, and performing a filtering process on the audio signal the amplitude gain reduction of which is reduced, to obtain a target audio signal; and controlling a speaker to play the target audio signal. According to the present disclosure, howling event suppression effect can be improved, thereby reducing the damage of audio playing to the hearing of a user. Further disclosed in the present disclosure are a howling suppression apparatus, an in-ear earphone, and a storage medium, to which the howling suppression method is applied.

An air-pulse generating device includes a film structure. The film structure is driven to form an amplitude-modulated ultrasonic air pressure variation with an ultrasonic carrier frequency. The film structure is driven to form an opening at a rate synchronous with the ultrasonic carrier frequency. The air-pulse generating device produces a plurality of air pulses according to the amplitude-modulated ultrasonic air pressure variation, and the plurality of air pulses is asymmetric.

A class D amplifier circuit receives an analog audio signal with a first reference voltage as its center level, and outputs an output pulse signal having a duty cycle that corresponds to the analog audio signal. A bias circuit generates a second reference voltage having a voltage level obtained as a division of the first reference voltage and the power supply voltage. A periodic voltage generating circuit of the class D amplifier circuit generates a periodic voltage having a triangle waveform or otherwise a sawtooth waveform having an amplitude that corresponds to the second reference voltage.

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.

A digital microphone includes at least one integrator; a state detection and parameter control component directly coupled to an output of the integrator; and a signal processing component coupled to an output of the state detection and parameter control component, wherein a parameter of the signal processing component includes a first value in a first operational mode and a second value in a second operational mode different from the first operational mode.

The embodiment of the invention discloses a sound field correction method and device and audio playing equipment. According to the embodiment of the invention, audio collection is carried out on the currently played content to obtain a current original audio file and a current echo cancellation audio file, and a current original frequency response curve corresponding to the current original audio file and a current echo cancellation frequency response curve corresponding to the current echo cancellation audio file are determined; and adjusting the current echo cancellation frequency response curve according to the reference equalization curve to obtain a target frequency response curve, determining a dynamic equalization curve according to the current original frequency response curve and the target frequency response curve, and further controlling audio signal output according to the dynamic equalization curve. Wherein the reference equilibrium curve is used for representing the difference between the reference original frequency response curve and the reference echo cancellation frequency response curve. Therefore, according to the embodiment of the invention, the audio playing equipment can realize sound field self-adaptive correction on the premise of not needing manual intervention, so that the user experience is improved.

To provide a speaker system and the like with a wider playback range than before.SOLUTION: A speaker system 1 includes: a first vibration actuator 130 that is fixed to a first vibration surface to give vibration to a first vibration surface to generate sound; and a first tweeter 230 that outputs high-pitched sounds.SELECTED DRAWING: Figure 1

The present description relates to an acoustic output device, comprising: a housing; a bone conduction vibrator, which is used for generating bone-conducted sound waves, the bone-conducted sound waves being transmitted to the cochlea by means of the housing to generate sound; an air conduction vibrator, which is used for generating air-conducted sound waves, the air-conducted sound waves being transmitted to the ear of a user by means of sound guide holes in the housing; and a processing circuit, which is configured to provide a first audio signal for the bone conduction vibrator, and provide a second audio signal for the air conduction vibrator, wherein the processing circuit is further configured to adjust a low-frequency component in the first audio signal or adjust a high-frequency component in the second audio signal. The low-frequency component in the first audio signal or the high-frequency component in the second audio signal are adjusted by means of the processing circuit, such that the ratio of bone-conducted components to air-conducted components in the output of the acoustic output device is variable, so as to adjust high and low frequency output effects of the acoustic output device. Thus, the acoustic output device is suitable for different scenarios, thereby enhancing the use experience of the acoustic output device.

An audio device is provided. The audio device comprises a controller and an acoustic transducer. The controller is configured to receive an input audio signal. The controller is further configured to generate, via a loudness detector, a loudness level of a first portion of the input audio signal. The controller is further configured to identify a compression curve of a plurality of compression curves corresponding to a user volume setting. The controller is further configured to adjust, via an audio compressor, a second portion of the input audio signal based on the loudness level and the compression curve to generate an output audio signal. The loudness level is determined according to LUFS model.