Extracting a common signal from multiple audio signals may include summing a first signal and a second signal to obtain a first+second signal; subtracting the second signal from the first signal to obtain a first−second signal; transforming the first+second signal and the first−second signal to frequency domain representations; calculating absolute value of the frequency domain representations of the first+second signal and the first−second signal; subtracting the absolute value of the frequency domain representation of the first−second signal from the absolute value of the frequency domain representation of the first+second signal to obtain a difference signal; multiplying the difference signal by the frequency domain representation of the first+second signal to obtain a product signal; dividing the product signal by the absolute value of the frequency domain representation of the first+second signal to obtain a frequency domain representation of the common signal; and transforming the frequency domain representation to the common signal.

There is disclosed a portable programmable device including a battery, a memory and a terminal connectable to earpieces, the device including in the memory a calibration file, parameter or parameters relating to audio sensitivity of the earpieces, the device being configured to play media data including audio, and to provide audio output to the earpieces, the device being further configured to, using the calibration file, parameter or parameters, calculate a noise dose relating to a sound exposure of a user resulting from audio output provided to the earpieces, and to record the noise dose on the device, wherein the device is configured to adjust audio output level in response to: (a) audio content included in played media data; (b) the calibration file, parameter or parameters, and (c) noise dose data of the user recorded on the device. A related method and computer program product are also disclosed.

A mixing apparatus comprises a weighting circuit configured to, within a particular control target band, perform weighting amplitude spectra of a priority audio signal and a non-priority audio signal taking frequency masking and time masking into account, an amplitude changer configured to relatively amplify an amplitude spectrum of the priority audio signal within the control target band based on the amplitude spectrum to which the weighting has been applied, a phase changer configured to make a phase spectrum of the non-priority audio signal approach a phase spectrum of the priority audio signal, within the control target band, based on the amplitude spectrum to which the weighting has been applied, and a mixer configured to mix the priority audio signal and the non-priority audio signal after the amplitude spectrum and the phase spectrum are changed.

Disclosed are examples of systems, apparatus, methods and computer-readable storage media for dynamically adjusting thresholds of a compressor. An input audio signal having a number of frequency band components is processed. Time-varying thresholds can be determined. A compressor performs, on each frequency band component, a compression operation having a corresponding time-varying threshold to produce gains. Each gain is applied to a delayed corresponding frequency band component to produce processed band components, which are summed to produce an output signal. In some implementations, a time-varying estimate of a perceived spectrum of the output signal and a time-varying estimate of a distortion spectrum induced by the perceived spectrum estimate are determined, for example, using a distortion audibility model. An audibility measure of the distortion spectrum estimate in the presence of the perceived spectrum estimate can be predicted and used to adjust the time-varying thresholds.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

An apparatus and method are provided for controlling the volume of a content. In one embodiment, sound associated with a content is received as well as a request for a volume change associated with the content. An equalization ratio is then obtained based on an amplification and a compression parameter. It is then analyzed whether a volume change will cause a coordinate change in the amplifier or compression levels associated with the content. If a volume change will cause a coordinate change in the amplifier or compression levels associated with the content, the volume change is limited.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

A method comprises: detecting noise in an environment with a microphone to produce a noise signal; receiving a voice signal to be played into the environment through a loudspeaker; performing multiband correction of the noise signal based on a microphone transfer function of the microphone, to produce a corrected noise signal; performing multiband correction of the voice signal based on a loudspeaker transfer function of the loudspeaker to produce a corrected voice signal; and computing multiband voice intelligibility results based on the corrected noise signal and the corrected voice signal.

A speaker has a magnetic angle sensor that detects the displacement of the vibration system of the speaker. A non-linearity compensation filter compensates for output distortion due to a non-linear parameter of the speaker. A transfer function of a linear inverse filter is applied so that output distortion is eliminated from the displacement, detected by the magnetic angle sensor, of the vibration system. A controller drives the speaker by using a test signal, and measures a response from the detected displacement of the vibration system. If error between the measured response and the theoretical value, theoretically determined from the design specifications, of the response is small, the controller uses the theoretical value, theoretically determined from the design specifications, of the non-linear parameter as an active non-linear parameter and sets, in the non-linearity compensation filter, a transfer function that compensates for distortion in an output due to the active non-linear parameter.

A signal processing method comprises comparing a first frequency domain representation of a sequence of power values for respective windows of source input samples of a source input signal with a second frequency domain representation of a sequence of power values for respective windows of target input samples of a target input signal so as to generate a frequency domain difference representation; inverse-frequency-transforming the frequency domain difference representation to generate a modification indication; and applying the modification indication to the source input samples to generate respective output samples of an output signal.