A system for masking audio signals includes a microphone for generating an ambient audio signal representing ambient noise, a speaker for rendering masking audio, and a processor in communication with the microphone and the speaker. The processor performs spectral analysis on the ambient audio signal from the microphone to determine a spectral envelope of the ambient noise, adjusts a frequency response of an optimizing filter based on the spectral envelope, applies the optimizing filter to a baseline masking waveform, producing an output waveform with relative spectral distribution matching the ambient noise, and provides the output waveform to the speaker.

Examples disclosed herein relate to controlling volume on an immersive display device. One example provides a near-eye display device comprising a sensor subsystem, a logic subsystem, and a storage subsystem storing instructions executable by the logic subsystem to receive image sensor data from the sensor subsystem, present content comprising a visual component and an auditory component, while presenting the content, detect via the image sensor data that speech is likely being directed at a wearer of the near-eye display device, and in response to detecting that speech is likely being directed at the wearer, attenuate an aspect of the auditory component.

Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

A signal processing apparatus includes: an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; and a calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount.

The present disclosure is a method for optimizing a working state of a bone conduction earphone. The bone conduction earphone includes an earphone core and at least one vibration sensor. The method includes: obtaining a vibration signal through the at least one vibration sensor, the vibration signal being at least partially derived from vibration generated by the earphone core in response to an audio signal, and the vibration of the earphone core being transmitted to a user wearing the bone conduction earphone through bone conduction; determining a vibration response feature of the earphone core based on the vibration signal and the audio signal; and feeding back a working state of the bone conduction earphone based on the vibration response feature of the earphone core.

An amplifier circuit includes a continuous-time linear equalizer, an adjustable gain circuit and a filter circuit. The continuous-time linear equalizer includes a first high-pass path, a first low-pass path, a second high-pass path, and a second low-pass path. The first high-pass path is used to increase a gain of a high-frequency part of a first signal source, and the second high-pass path is used to increase a gain of a high-frequency part of a second signal source. The filter circuit is used to amplify and filter the first signal source and the second signal source, and includes a fully-differential operational amplifier, a first filter network, and a second filter network.

A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

A multi-path audio amplification system that provides an output drive signal to electromechanical output transducers provides improved undistorted headroom, reduced path switching noise, and/or improved frequency response performance. Multiple signal amplification paths receive an audio input signal and have corresponding multiple output stages that have differing output impedances. A mode selector selects an active one of the multiple signal amplification paths is selected to supply the output drive signal. Outputs of the multiple output stages are coupled to the electromechanical transducer to provide the output drive signal and at least one of the multiple signal amplification paths includes an equalization filter for filtering the audio input signal to compensate for phase or gain differences referenced from the input to the outputs of the multiple output stages due to interaction between the differing output impedances and an impedance of the electromechanical transducer.

A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.