A method for user recognition and emotion monitoring based on a smart headset is provided. —The smart headset includes an earplug part and a main body, wherein the earplug part is provided with a first microphone and a wearing detection sensor, and a housing of the main body is provided with a signal amplification circuit, a communication module, and a microcontroller. The wearing detection sensor is to detect whether a user wears the smart headset, and the first microphone is to obtain a sound signal in an ear canal. The sound signal is amplified, and then is outputted to the microcontroller. The amplified sound signal is transmitted by the microcontroller to a smart terminal paired with the smart headset to extract a heart sound signal characteristic, and legality of—identity of the user is validated and emotional state of the user is inferred according to the heart sound signal characteristic.

Systems and methods for enhancing a headset user's own voice include at least two outside microphones, an inside microphone, audio input components operable to receive and process the microphone signals, a voice activity detector operable to detect speech presence and absence in the received and/or processed signals, and a cross-over module configured to generate an enhanced voice signal. The audio processing components includes a low frequency branch comprising low pass filter banks, a low frequency spatial filter, a low frequency spectral filter and an equalizer, and a high frequency branch comprising highpass filter banks, a high frequency spatial filter, and a high frequency spectral filter.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

The present disclosure generally relates to audio synchronization. An example method includes, at a first device with a communication device: performing an audio timing synchronization process that includes: transmitting, via the communication device, to a second device, a request for the second device to participate in the audio timing synchronization process; subsequent to transmitting the request to the second device, causing an output of an audio tone; subsequent to causing the output of the audio tone, receiving data from the second device based on the audio tone; and adjusting an audio timing synchronization setting of a third device based at least in part on the data received from the second device.

Techniques used to selectively amplify audio signals are described in connection with audio amplification devices, such as hearing aids. A device and its operation are described to facilitate setting low and high tone/volume controls separately, using at least two selection mechanisms. In one aspect, a first selection mechanism includes a pitch frequency control rocker switch and the second selection mechanism includes a bass frequency control rocker switch disposed separately. In one aspect, the bass frequency control rocker switch causes a processor to bias the frequency response of the sound amplifier for frequencies below 1 kHz. In another aspect, the pitch frequency control rocker switch causes a processor to bias the frequency response of the hearing for frequencies above 1 kHz. In another aspect, the selection mechanism involves the separate attenuation of treble and bass adjustments in response to a user selection of a rocker switch setting for each adjustment.

A method includes converting an audio signal of a current time period to a frequency domain to produce a set of frequency coefficients. For each of the frequency coefficients, the method includes computing a gradient of that frequency coefficient's magnitude relative to a magnitude of the same frequency coefficient from a previous time period. The method then includes summing the gradients computed for the set of frequency coefficients to produce a sum value, and then generating a haptic signal based on the sum value.

A method for acoustic source separation comprises inputting acoustic data from a plurality of acoustic sensors, combined from a plurality of acoustic sources, converting the acoustic data to time-frequency domain data comprising time-frequency data frames, and constructing a multichannel filter for the time-frequency data frames to separate signals from the acoustic sources. The constructing comprises determining a set of de-mixing matrices (W.sub.f) to apply to each time-frequency data frame to determine a vector of separated outputs (y.sub.ft) by modifying each of the de-mixing matrices by a respective gradient value (G;G′) for a frequency dependent upon a gradient of a cost function measuring a separation of the sources by the respective de-mixing matrix. The respective gradient values for each frequency are each calculated from a stochastic selection of the time-frequency data frames.

The present disclosure relates to a waterproof open earphone. The waterproof open earphone may include a housing, at least one button, at least one elastic pad, and at least one pair of speaker units. The housing may be placed on a head or at least one ear of a user while not blocking an ear canal of the user. The at least one button may be set on the housing, wherein each of the at least one button corresponds to a button hole. The at least one elastic pad may correspond to the at least one button, respectively, wherein each elastic pad prevents the corresponding button from moving relative to the button hole. Each pair of the at least one pair of speaker units may generate sound within a frequency range from two sound guiding holes through two sound guiding tubes.

An image capture device includes a sensor, a microphone array, and a processor. The microphone array may include a first microphone, a second microphone, a third microphone, or any combination thereof. The first microphone may be configured to face a first direction. The second microphone may be configured to face a second direction. The second direction may be diametrically opposed to the first direction. The third microphone may be configured to face a third direction. The third direction may be substantially perpendicular to the first direction, the second direction, or both. The processor may be configured to determine a microphone capture pattern and detect wind noise. The microphone capture pattern may be determined based on data obtained from the sensor. The sensor data may include image data, audio data, image capture device orientation data, location data, accelerometer data, or any combination thereof.

An apparatus for generating at least one distance estimate to at least one sound source within a sound scene comprising the least one sound source, the apparatus configured to: receive at least two audio signals from a microphone array located within the sound scene; receive at least one further audio signal associated with the at least one sound source; determine at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source; determine a distance estimate to the at least one sound source based on the at least one portion of the at least two audio signals from a microphone array corresponding to the at least one further audio signal associated with the at least one sound source.