A hearing device includes at least one microphone configured to capture sound signals within an overall frequency range and to convert them into an input signal. A signal processor is provided for processing the input signal within a lower frequency range that is part of an overall frequency range. A detector is provided for detecting a noise that has frequency components both inside and outside the lower frequency range, namely in an upper frequency range above the lower frequency range. The hearing device is configured in such a way that the detector detects the noise based on its frequency component in the upper frequency range. A method for operating a hearing device is also provided.

Environmental sound is recorded using one or more microphones. A source of the recorded environmental sound is classified. The recorded environmental sound is weighted based on the classification of the source and the source media sound using a weighting mode to determine whether to mix the recorded environmental. The recorded environmental sound is mixed with source media sound to produce a mixed sound based on the determination. The mixed sound is played over one or more speakers.

A first input signal captured by one or more sensors associated with an ANR headphone is received. A frequency domain representation of the first input signal is computed for a set of discrete frequencies, based on which a set of parameters is generated for a digital filter disposed in an ANR signal flow path of the ANR headphone, the set of parameters being such that a loop gain of the ANR signal flow path substantially matches a target loop gain. Generating the set of parameters comprises: adjusting a response of the digital filter at frequencies (e.g., spanning between 200 Hz-5 kHz). A response of at least 3 second order sections of the digital filter is adjusted. A second input signal in the ANR signal flow path is processed using the generated set of parameters to generate an output signal for driving the electroacoustic transducer of the ANR headphone.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

The present invention provides a method for processing audio data, comprising the steps of providing input audio data containing a mixture of audio data including first audio data of a first musical timbre and second audio data of a second musical timbre different from said first musical timbre, decomposing the input audio data to provide decomposed data representative of the first audio data, transforming the decomposed data to obtain third audio data.

The present disclosure relates to a pair of glasses. The pair of glasses may include a frame, one or more lenses, and one or more temples. The pair of glasses may further include at least one low-frequency acoustic driver, at least one high-frequency acoustic driver, and a controller. The at least one low-frequency acoustic driver may be configured to output sounds from at least two first guiding holes. The at least one high-frequency acoustic driver may be configured to output sounds from at least two second guiding holes. The controller may be configured to direct the low-frequency acoustic driver to output the sounds in a first frequency range and direct the high-frequency acoustic driver to output the sounds in a second frequency range. The second frequency range may include one or more frequencies higher than one or more frequencies in the first frequency range.

A cochlear implant system for processing polyphonic pitch includes an electrode array for implanting in a cochlea of a patient. The electrode array includes a first set of electrodes, where each electrode of the first set is for implanting on a first region of the cochlea. The electrode array also includes a second set of electrodes, where each electrode of the second set is for implanting on a second region of the cochlea. The system also includes a sound processor configured to capture a sound signal having polyphonic pitch. For each electrode of the first set and second set, the speech processor generates at least two different modulated frequency signals from the sound signal, such that each of the modulated frequency signals corresponds to a different pitch in the sound signal. The speech processor stimulates the electrode by simultaneously applying the at least two different modulated frequency signals.

Provided is an echo cancellation apparatus capable of calculating an acoustic coupling amount with high accuracy regardless of the magnitude of the near-end speaker component and without using a double talk detector. The echo cancellation apparatus cancels an echo included in a sound pickup signal picked up by a microphone placed at a near-end and includes an acoustic coupling amount calculation unit that updates and calculates an acoustic coupling amount estimated value of a component of a reproduction signal, which is a signal picked up by a microphone placed at a far-end included in the sound pickup signal, such that an update amount is decreased the greater a magnitude of a component other than an echo component is in the sound pickup signal; a gain calculation unit that calculates a gain coefficient on the basis of the acoustic coupling amount estimated value; and an integration unit that integrates the gain coefficient with the sound pickup signal and generates an echo cancellation signal.

A data processing system for identifying an audio signal includes an audio sensor, a receiver module, a signal recognition module, and a receiver device. The receiver module receives audio data from the audio sensor. The receiver module transmits the audio data to the signal recognition module. The signal recognition module calculates time-varying vector arrays of octave band energies, and/or of fractional octave band energies, and calculates time-varying vector arrays of Mel-Frequency Cepstral Coefficients (MFCC) values based on the received audio data. The signal recognition module generates audio feature image data based on the vector arrays. The signal recognition module includes binary classifier machine learning models and inference models to identify the audio signal based on the generated audio feature image data. The signal recognition module transmits a notification message to the receiver device.

Proposed is an acoustic output device that obtains enough distance attenuation to achieve localization of a sound field by controlling the driving of a loudspeaker array by use of a more compact shape. The acoustic output device 10 comprises: a loudspeaker array 20 that includes a plurality of loudspeakers 20-1, 20-2, . . . , 20-N arranged in a two-dimensional plane; and an amplifier array 40 that includes a plurality of amplifiers 40-1, 40-2, . . . , 40-N that control the amplitude and phase of the drive signals for each loudspeaker according to the eigenvectors of the predetermined radiation mode of the loudspeaker array 20.