3029results about "Frequency response correction" patented technology

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).

Disclosed is an earphone including: two speaker assemblies, a connection member, and a processing circuit. The connection member is configured to connect the two speaker assemblies, and the connection member provides, through a bending deformation, a clamping force for placing the two speaker assemblies on a head of a user, and the connection member includes a housing with an accommodation cavity. A bending sensor is disposed in the accommodating cavity, and the bending sensor is configured to generate a bending signal based on a bending state of the connection member. The processing circuit is configured to determine a placement state of the earphone based on the bending signal. The placement state includes one of a normal wearing state, an abnormal wearing state, or a free placement state.

A phase shifter includes an active matrix transistor array. Each transistor of the active matrix transistor array is configured to control a voltage bias across a corresponding voltage-controlled capacitor of voltage-controlled capacitors to control a capacitance of the corresponding voltage-controlled capacitor. The phase shifter further includes signal channels. Each signal channel of the signal channels has a first end and a second end and is AC coupled at a point between the first end and the second end to one or more of the voltage-controlled capacitors. The phase shifter is configured to control a phase shift applied to each corresponding signal channel of the signal channels by setting the voltage bias across each of the one or more of the voltage-controlled capacitors coupled to the corresponding signal channel.

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.

In some embodiments, a first audio signal is received via a first microphone, and a first probability of voice activity is determined based on the first audio signal. A second audio signal is received via a second microphone, and a second probability of voice activity is determined based on the first and second audio signals. Whether a first threshold of voice activity is met is determined based on the first and second probabilities of voice activity. In accordance with a determination that a first threshold of voice activity is met, it is determined that a voice onset has occurred, and an alert is transmitted to a processor based on the determination that the voice onset has occurred. In accordance with a determination that a first threshold of voice activity is not met, it is not determined that a voice onset has occurred.

Disclosed herein are systems and methods for processing speech signals in mixed reality applications. A method may include receiving an audio signal; determining, via first processors, whether the audio signal comprises a voice onset event; in accordance with a determination that the audio signal comprises the voice onset event: waking a second one or more processors; determining, via the second processors, that the audio signal comprises a predetermined trigger signal; in accordance with a determination that the audio signal comprises the predetermined trigger signal: waking third processors; performing, via the third processors, automatic speech recognition based on the audio signal; and in accordance with a determination that the audio signal does not comprise the predetermined trigger signal: forgoing waking the third processors; and in accordance with a determination that the audio signal does not comprise the voice onset event: forgoing waking the second processors.

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.

A haptic driver for driving a haptic actuator, the haptic driver configured to: provide a haptic drive signal and an audio drive signal, the haptic drive signal for causing the haptic actuator to generate a haptic vibration output and the audio drive signal for causing the haptic actuator to generate an audio output; and drive the haptic actuator with the haptic drive signal and the audio drive signal so that the haptic actuator generates the haptic vibration output and the audio output.

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.

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 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 present disclosure generally relates to user interfaces and techniques for managing audio exposure using a computer system (e.g., an electronic device). In accordance with some embodiments, the electronic device displays a graphical indication of a noise exposure level over a first period of time with an area of the graphical indication that is colored to represent the noise exposure level, the color of the area transitioning from a first color to a second color when the noise exposure level exceeds a first threshold. In accordance with some embodiments, the electronic device displays noise exposure levels attributable to a first output device type and a second output device type and, in response to selecting a filtering affordance, visually distinguishes a set of noise exposure levels attributable to the second output device type.

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.

According to an embodiment, a digital microphone includes an analog-to-digital converter (ADC) for receiving an analog input signal; a DC blocker component coupled to the ADC; a digital low pass filter coupled to the DC block component; and a nonlinear compensation component coupled to the digital low pass filter for providing a digital output signal.

Apparatuses, systems, and techniques are presented to reduce noise in audio. In at least one embodiment, one or more neural networks are used to determine a noise signal in one or more speech signals.

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.