1576results 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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

This application relates to methods and apparatus for switched mode drivers. A BTL driver has a switch network operable in different switch states, wherein, in each switch state, each of first and second output nodes is connected to a respective one of a first switching voltage, a second switching voltage or an intermediate switching voltage between the first and second switching voltages which is provided by a driver capacitance. A controller is configured to control the switch network to operate in a sequence of switch states to generate a differential drive signal based on an input signal. The controller is configured such that operation of driver switch network in said sequence of switch states to generate said differential drive signal provides voltage regulation of the intermediate voltage provided by said driver capacitance.

A computer-implemented sound calibration method, a system and a storage medium are provided. A test audio is played by a subwoofer and a main speaker, transmitted audio is collected by an audio collection device at a primary listening position, a sound generation delay and a volume parameter difference are analyzed, and the subwoofer is calibrated based on of a frequency response curve of a scenario, so as to solve the problem in the related art where the user experience is adversely affected due to low sound calibration accuracy between the subwoofer and the main speaker.

A sound generation apparatus comprises a transducer, a driving apparatus, and a control circuit. The driving apparatus is connected to the transducer. The control circuit is electrically connected to the transducer and the driving apparatus. The control circuit is configured to drive a vibration member of the transducer to vibrate, and the control circuit is further configured to control the driving apparatus to drive the transducer to perform periodic motion. The transducer is disposed to emit a first sound wave to the outside while performing periodic motion.

The invention relates to the technical field of audio driving, in particular to a driving circuit and method and electronic equipment. The method comprises the following steps: arranging two boosting units at the output end of the power amplifier, respectively receiving two paths of signals with opposite phases output by the power amplifier so as to realize that boosted signals output by the two boosting units are also signals with opposite phases, arranging one boosting unit to be connected with the positive electrode of the piezoelectric ceramic module, and arranging the other boosting unit to be connected with the negative electrode of the piezoelectric ceramic module; and the other boosting unit is connected with the negative electrode of the piezoelectric ceramic module, so that the driving voltage of the piezoelectric ceramic module is increased. The boost unit is simple in structure, occupies a small area of a circuit board, and can be effectively applied to miniaturized electronic equipment.

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 provides a sound effect control method and electronic equipment, and relates to the technical field of terminals. The audios played by the electronic equipment have similar sound effects in different use states, so that the sound effect stability of the electronic equipment is ensured. The method is applied to the folding screen electronic equipment, and comprises the following steps: the electronic equipment plays a first audio with a first sound effect through a first loudspeaker and a second loudspeaker in a first use state, the first loudspeaker performs playing by adopting a first sound effect parameter, and the second loudspeaker performs playing by adopting a second sound effect parameter. And in response to user operation, switching to a second use state, playing the first audio with a second sound effect through the first loudspeaker and the second loudspeaker, the first loudspeaker playing with the third sound effect parameter, and the second loudspeaker playing with the fourth sound effect parameter. The first sound effect parameter is different from the third sound effect parameter, the deviation between the second sound effect and the first sound effect is smaller than a first threshold value, and the first use state and the second use state are related to the folding state of the electronic equipment.

Methods, apparatus, systems and articles of manufacture are disclosed to determine audio quality. Example apparatus disclosed herein include an equalization (EQ) model query generator to generate a query to a neural network, the query including a representation of a sample of an audio signal. Example apparatus disclosed herein also include an EQ analyzer to access a plurality of equalization settings determined by the neural network based on the query; and compare the equalization settings to an equalization threshold to determine if the audio signal is to be removed from subsequent processing.

A balanced armature receiver is disclosed including a multicoil motor located in a housing and comprising a movable armature connected to a diaphragm separating an interior of the housing into front and back volumes. The multicoil motor is electrically connected to contacts of a terminal board having an integrated capacitor component, wherein electrical connections of the multicoil motor and the capacitor component to the terminal board are configured to change an electrical impedance of the balanced armature receiver based on a frequency of an electrical signal applied to the multicoil motor.

Embodiments are provided for receiving a request to output audio at a first speaker and a second speaker of an electronic device, determining that the electronic device is oriented in a portrait orientation or a landscape orientation, identifying, based on the determined orientation, a first equalization setting for the first speaker and a second equalization setting for the second speaker, providing, for output at the first speaker, a first audio signal with the first equalization setting, and providing, for output at the second speaker, a second audio signal with the second equalization setting.

For example, one object is to realize more excellent sound field correction. An information processing apparatus includes a control unit that converts a transfer characteristic of a sound pressure from a speaker device to a listening position into an energy characteristic, and calculates a correction parameter for correcting a sound field generated by an output sound of the speaker device using the converted energy characteristic.

The invention discloses an environmental sound classification and noise reduction method and system for an intelligent Bluetooth hearing-aid earphone. The method comprises the following steps: S1, collecting background noise original audio signals and real-time binaural sound field signals in a typical scene; s2, Mel spectrum features and auditory perception features are extracted from each typical scene, fusion features are calculated, dimension compression is carried out, and a scene feature vector library is obtained; s3, extracting a live sound field feature, and extracting a scene feature vector matched with the live sound field in the scene feature vector library; s4, calculating a sound source direction correction coefficient, and calculating a corrected sound source direction; s5, calculating cross-modal interaction characteristics, then calculating noise suppression intensity and sound source gain intensity, and generating an audio after noise reduction; and S6, generating a synchronous stereo audio signal. The method can solve the problem that the traditional method is difficult to distinguish different types of background noise, so that the voice of a dialogue is inhibited by mistake, and the voice enhancement effect is poor.

An example apparatus includes: equalizer circuitry having first and second inputs and an output, the equalizer circuitry configured to adjust an audio signal at the first input responsive to coefficient values at the second input; amplifier circuitry having an input coupled to the output of the equalizer circuitry, and having an output; control circuitry having a first input coupled to the first input of the equalizer circuitry, having a second input coupled to the output of the amplifier circuitry, and having an output coupled to the second input of the equalizer circuitry, the control circuitry configured to determine a resonant frequency of a speaker device responsive to the audio signal and a signal at the output of the amplifier circuitry, and to provide the coefficient values at the output of the control circuitry responsive to the resonant frequency of the speaker.

A method of operating a PMUT electro-acoustical transducer, the method comprising: applying over an excitation interval to the transducer an excitation signal which is configured to emit corresponding ultrasound pulses towards a surrounding space, acquiring at a receiver reflected ultrasound pulses as reflected in said surrounding space, generating a reference echo signal, performing a cross-correlation of said acquired received ultrasound pulses with said reference echo signal, performing a measurement based on the cross-correlation results, in particular a measurement of the time of flight of the ultrasound pulses, wherein said reference echo is obtained by finding an oscillation frequency of the transmitter on the basis of a transmitter ringdown signal, finding an oscillation frequency of the receiver on the basis of a receiver ringdown signal, performing a frequency tuning respectively on the transmitter and the receiver on the basis of said respective oscillation frequencies, then sweeping an input frequency of the transmitter to find a frequency of the maximum displacement in the ringdown signal, performing a frequency tuning of the receiver at said frequency of the maximum displacement in the ringdown signal of the transmitter.

The invention relates to the technical field of stereo devices, in particular to a sound effect adjusting method for a high-performance audio loudspeaker. The method comprises the following steps: firstly, inputting physical parameters of a loudspeaker unit, a cavity structure and a passive diaphragm, establishing a coupling acoustic model by adopting finite element simulation, and generating a theoretical frequency response curve; actual frequency response data of the sound box are acquired through analog-to-digital conversion and a sensor, and a core frequency band which has the greatest influence on sound quality is extracted by combining a Gaussian process regression method; extracting acoustic characteristic parameters based on a frequency response deviation function in the core frequency band, inputting the acoustic characteristic parameters to a PESQ model, and driving an LMS adaptive filtering algorithm to complete preliminary adjustment; and then carrying out secondary optimization on the full-band parameter space by adopting a particle swarm optimization algorithm, and finally obtaining an optimal sound effect adjustment result. The method has the advantages of being high in adjustment efficiency, high in parameter adaptive capacity, accurate in sound effect evaluation and the like, and is suitable for automatic sound effect optimization scenes of high-fidelity Bluetooth sound boxes, intelligent sound boxes and other products.

A method includes determining whether an audio signal processed in a class-D amplifier is clipping. Responsive to clipping of the audio signal, a speaker voltage is estimated as a ratio of a power supply voltage of the class-D amplifier to a resistance of a speaker driven by the class-D amplifier. An amplitude of the audio signal is controlled based on the speaker voltage.

The present disclosure relates to a method and an apparatus for adding reverberation, a device, and a product. The method includes: separating reverberant audio based on wet audio, where the wet audio includes dry audio and the reverberant audio corresponding to the dry audio. The method further includes: determining, based on the reverberant audio, reverberation parameters for indicating reverberation effects of the reverberant audio. In addition, the method further includes: adding reverberation to synthesized dry audio based on the reverberation parameters.

A localized sound projection system can coordinate the sounds of speakers to simulate the placement of an auditory cue in a 3D space. The system can include a plurality of speakers configured to output auditory signals and a sound localization controller configured to control the plurality of speakers to coordinate the auditory signals to simulate an origination location of a patient alarm. The sound localization controller can determine adjusted auditory signals and control a plurality of speakers to output the plurality of adjusted auditory signals. A method for dynamically controlling speaker settings in a medical environment can include determining volume settings corresponding to a speaker, monitoring a level of ambient noise corresponding to a room of a patient, controlling the volume settings of the speaker to reduce or increase a sound level output of a speaker. A patient monitoring system can be configured to physically manipulate medical devices in response to audible commands. The system can receive a plurality of vocal commands from a user and can manipulate various settings after confirmation from a user.

Signal processing circuitry configured to receive an input signal and to output a processed output signal, wherein the signal processing circuitry is configured to: receive an indication of a temporal location of a transient in the input signal; and provide, in the processed output signal, a masking signal bridging the temporal location of the transient to mask the transient.