931results 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 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 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 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.

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.

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.

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.

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.

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.

A system configured to reduce loudspeaker distortion by performing nonlinear signal processing is provided. A device may include preprocessing component(s) that apply nonlinear signal correction prior to sending a playback audio signal to a driver in order to compensate for a nonlinear response of the driver. While the driver response may be nonlinear, a combination of the preprocessing and the nonlinear driver response results in a combined response that is linear and / or compensates for the nonlinear driver response. For example, applying the nonlinear driver response to a processed audio signal may result in output audio generated by the driver accurately reproducing the playback audio signal input to the preprocessing components. To train the preprocessing components to apply the nonlinear signal correction, a deep neural network (DNN) is trained to model the driver response.

A device to perform location-based audio signal compensation includes a memory configured to store instructions and one or more processors. The one or more processors are configured to execute the instructions to receive an audio input signal corresponding to sound received from a second device, determine position data indicative of a position of the device, and generate, based on the audio input signal, a compensation filter to be applied to an audio playback signal prior to playout from the second device, to at least partially compensate for distortion associated with sound propagation from the second device to the position of the device.

A method performed by an audio system that includes a headset for playing back audio, the method includes receiving an audio signal having audio content, determining a characteristic of the audio content, determining a parameter of the headset, producing a filtered audio signal by applying an equalization (EQ) filter to the audio signal based on the characteristic of the audio content and the parameter of the headset, and driving one or more speakers of the headset with the filtered audio signal.

Features described herein generally relate to detecting and correcting audible distortion due to speaker port occlusion. Particularly, an audio signal is output, an impedance curve for the audio signal is calculated, a change in a resonance frequency for the speaker based on an impedance curve model for the speaker and the impedance curve for the speaker is detected, and a gain of the amplifier is set based on the detected change in the resonance frequency for the speaker.

The invention relates to a method, a device, equipment and a product for adding reverberation. The method includes separating reverberation audio based on wet sound audio, where the wet sound audio includes dry sound audio and reverberation audio corresponding to the dry sound audio. The method further includes determining a reverberation parameter indicative of a reverberation effect of the reverberation audio based on the reverberation audio. Further, the method includes adding reverberation to the synthesized dry sound audio based on the reverberation parameter. Through the method for flexibly adding reverberation, reverberation effects can be adaptively added to different synthetic dry sound audios, the added reverberation effects can be ensured to be close to original wet sound audios, and the user experience is improved.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones.

A method for adjusting audio frequency includes steps of: obtaining initial frequency and temporary data storage of audio data input to a buffering unit; performing a first adjustment procedure on initial frequency to obtain a first variation of temporary data storage corresponding to a first change of initial frequency; calculating a first frequency correction amount according to first variation and a first period of first adjustment procedure; adjusting initial frequency into first frequency according to first frequency correction amount; inputting first frequency into buffering unit; performing a second adjustment procedure on first frequency to obtain a second variation of temporary data storage corresponding to a second change of first frequency; calculating a second frequency correction amount according to second variation and a second period of second adjustment procedure, which first period is less than second period; and adjusting first frequency into target frequency according to second frequency correction amount.

A sound pickup device includes a processor with hardware. The processor calculates a first relational expression between acoustic filter factors of acoustic filters applied to microphone acquisition signals acquired by two or more microphones, based on a multiplication factor at a sensitization control point with high sound pressure sensitivity by the two or more microphones and a transfer function between each of the two or more microphones and the sensitization control point. The processor calculates a second relational expression between the acoustic filter factors, based on information of a frequency of a sound source and an interval between the two or more microphones. The processor calculates each of the acoustic filter factors, based on the first relational expression and the second relational expression.

Operating an electrostatic acoustic device simultaneously as a speaker and as a microphone. The electrostatic acoustic device includes a membrane and an electrode disposed proximate to the membrane. An input varying audio signal is input to the electrostatic acoustic device. The membrane is configured to respond mechanically to a varying electric field responsive to the varying audio signal input. A portion of the input varying audio signal is tapped to produce a reference signal. A signal is detected responsive to motion of the membrane, to convert the signal to an output varying voltage signal. The output varying voltage signal is compared to the reference signal to produce a microphone signal. The microphone signal is responsive to motion of the membrane induced by air pressure variations of ambient sound.

An electronic device according to an embodiment of this document includes: a first housing; a second housing coupled to the first housing and movable relative to the first housing; a speaker disposed in the first housing and outputting an audio signal; a flexible display disposed at least partially in the first and second housings, wherein the size of an area visually exposed by the front surface of the electronic device can be expanded according to movement of the second housing; a first conduit formed in the first housing such that the audio signal is output to the outside of the first housing; a second conduit formed at a position corresponding to the first conduit when the movement distance of the second housing is within a first range, and formed in the second housing such that the audio signal is output to the outside of the electronic device via the first conduit; a third conduit connected to the first conduit when the movement distance of the second housing is within a second range, and formed in the second housing such that the audio signal is output to the outside of the electronic device; and a processor operatively connected to the speaker and the flexible display, wherein the processor can sense the movement distance of the second housing relative to the first housing and correct the characteristics of the audio signal generated by the speaker based on the sensed movement distance. According to the various embodiments described herein, the shape of the conduit for sound emission from an audio module such as a loudspeaker changes in response to changes in the shape of the electronic device, thus providing the user with optimal acoustic service through a structure that adapts to changes in shape.

Embodiments of this application provide a frequency response consistency calibration method and an electronic device, to solve a problem that when a screen sound production component of an electronic device plays a sound signal, a frequency response curve of the sound signal greatly fluctuates. The frequency response consistency calibration method includes: An electronic device plays a test audio signal. The electronic device stores a calibration parameter. The calibration parameter is a running parameter of an equalizer calibration module. The calibration parameter is determined based on a frequency response of a first audio signal and a standard frequency response of the test audio signal. The first audio signal is obtained by a calibration device by acquiring the test audio signal played by the electronic device. The calibration parameter is used to adjust, through the equalizer calibration module when the electronic device plays a second audio signal, a frequency response of the second audio signal played by the electronic device.

The present invention addresses the problem of providing an "audio signal processing device" that achieves a good sense of localization with respect to a sound having a narrow frequency band. [Solution] The present invention is provided with signal processing units provided so as to correspond to respective channels of audio signals of a plurality of channels, and each signal processing unit has: a target sound separation unit (411) that separates a target sound signal indicating a target sound, which is a sound to be subjected to localization enhancement, from a source sound signal, which is an audio signal of the corresponding channel; a sense-of-localization-enhanced sound generation unit that generates a sense-of-localization-enhanced sound signal in which the frequency of the target sound signal separated by the target sound separation unit (411) is multiplied by k (k is an integer of 2 or more); and a synthesis unit that synthesizes and outputs the localization-enhanced sound signal generated by the localization-enhanced sound generation unit and the source sound signal.

An electronic furniture assembly of the present invention includes: (i) a furniture assembly comprising: (A) a base (e.g., a seat portion), (B) at least one transverse or upright member (e.g., a side, armrest or backrest), and (ii) a speaker system mounted within one or more portions of the furniture assembly. A method of tuning a speaker of an audio-enhanced furniture system includes presenting a user with predetermined tuning profiles corresponding to a plurality of configuration footprint shapes in which the furniture assembly can be assembled and in response to a user selection of a given configuration footprint shape, tuning the speaker system according to the tuning profile selected by the user.

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.

Various embodiments disclose a computer-implemented method that can include driving an audio output device to reproduce a stimulus signal when a wearable device is placed along an ear canal of a user, receiving a sound signal from a microphone based on the stimulus signal, and determining, based on the sound signal and a calculated response of the microphone retrieved from a memory, one or more characteristics of the ear canal of the user, wherein the one or more characteristics of the ear canal comprises an ear canal impedance, and applying, based on the one or more characteristics of the ear canal, an ear canal response correction to an output signal played back by the audio output device.

A computer having a system for mitigating mechanical rattle arising due to a mechanical coupling of an audio output transducer to a part of the computer device, the system comprising: a processing subsystem configured to receive an input audio signal and to output a drive signal for driving the audio output transducer to produce a transducer output, wherein the processing subsystem is configured to selectively apply an attenuation function to the input audio signal to attenuate a signal component of the input audio signal at a frequency that causes mechanical rattle in the computer; and an analyser subsystem configured to receive the input audio signal and to output a control signal to the processing subsystem to control application of the attenuation function to the input audio signal by the processing subsystem based on a spectral content of the received input audio signal; wherein the attenuation function is based on characteristic acoustic behaviour of the computer.

The application relates to the technical field of audio processing, in particular to an audio processing method, an electronic device and a readable storage medium. In the method, a sound signal is received by a device that can be worn on the ear of a user, then the sound signal is divided into a low-frequency sound signal and a high-frequency sound signal by using a filter, and a steady-state sound signal and a transient sound signal in the low-frequency sound signal and the high-frequency sound signal are extracted. The steady-state sound signal and the transient sound signal are processed based on the individual calibration parameters corresponding to the user, so that the calibrated sound signal finally heard by the user conforms to the target psychoacoustic loudness of the user. In this way, the hearing protection of the user can be realized.