Abstract: The present disclosure relates to a system for providing substantially synchronised haptic and audio outputs. The system includes a signal processor which is configured to receive an audio signal from a main processor of the system and to receive a haptic signal, which may be received from the main processor, or may be retrieved from memory, or else may be generated in real-time by the signal processor. The signal processor calculates a delay to be applied to the haptic signal or the audio signal and outputs a delayed version of the audio signal and the haptic signal, or a delayed version of the haptic signal and the audio signal, to appropriate output stages.

Abstract: A method and device for detecting whether a headset is on ear. Microphone signals from a plurality of microphones are used to derive a plurality of signal feature measures, which are normalized to a common reference scale. The signal feature measures are weighted based upon detected signal conditions in the microphone signals. The normalized and variably weighted signal feature measures are then combined to produce an output indication of whether a headset is on ear.

Abstract: There is provided a method in a peripheral device comprising one or more microphones. The peripheral device is connectable to a host device via a digital connection. The method comprises: receiving, from the one or more microphones, an audio data stream relating to speech from a user, the audio data stream comprising a stream of data segments; and, responsive to detection of a trigger phrase in one or more first data segments of the audio data stream: effecting activation of the digital connection; and transmitting one or more biometric features extracted from the one or more first data segments to the host device via the digital connection for use in a voice biometric authentication process.

Abstract: A method and device for detecting whether a headset is on ear. Microphone signals from a plurality of microphones are used to derive a plurality of signal feature measures, which are normalized to a common reference scale. The signal feature measures are weighted based upon detected signal conditions in the microphone signals. The normalized and variably weighted signal feature measures are then combined to produce an output indication of whether a headset is on ear.

Abstract: A method and device for detecting whether a headset is on ear. Microphone signals from a plurality of microphones are used to derive a plurality of signal feature measures, which are normalized to a common reference scale. The signal feature measures are weighted based upon detected signal conditions in the microphone signals. The normalized and variably weighted signal feature measures are then combined to produce an output indication of whether a headset is on ear.

Abstract: A method and device for detecting whether a headset is on ear. A probe signal is generated for acoustic playback from a speaker. A microphone signal from a microphone is received, the microphone signal comprising at least a portion of the probe signal as received at the microphone. The microphone signal is passed to a state estimator, to produce an estimate of at least one parameter of the portion of the probe signal contained in the microphone signal. The estimate of the at least one parameter is processed to determine whether the headset is on ear.

Abstract: This application relates to transfer of microphone data from a microphone (101, 102, 102a, 201) to a processing module, such as voice biometric authentication module (111) in a secure manner, such that the receiving module can trust that the received audio is genuine. An authentication module (203) is configured to receive microphone data (DM) representative of an audio signal received at the microphone (201), and generate from the microphone data, authentication data (DA) for certifying that the microphone data did pass via the authentication module. The first authentication data (DA) comprises information relating to distinguishing characteristics of the audio content of the microphone data and may, for instance be an acoustic fingerprint of the audio content. The authentication data, may be cryptographically signed or encrypted and sent with the microphone audio to allow a receiver to verify that the audio is genuine and the content has not been substantially altered.

Abstract: Detection of a blocked microphone involves receiving microphone signals from a plurality of microphones. A plurality of signal feature measures are derived from the microphone signals. The signal feature measures are normalised. The normalised signal feature measures are variably weighted in response to detected environmental conditions in the microphone signals. The variably weighted normalised signal feature measures are combined to produce an output indication of whether a microphone is blocked.

Abstract: Embodiments of the invention determine a speech estimate using a bone conduction sensor or accelerometer, without employing voice activity detection gating of speech estimation. Speech estimation is based either exclusively on the bone conduction signal, or is performed in combination with a microphone signal. The speech estimate is then used to condition an output signal of the microphone. There are multiple use cases for speech processing in audio devices.

Abstract: A signal processing module is configured to receive left and right channels of stereo input audio data and generate first and second channels of output audio data for first and second loudspeakers where the first and second loudspeakers have different frequency responses to one another. The signal processing module comprises an impulse emphasis block configured to emphasize impulsive sounds in the received audio in at least one of the first and second channels of output audio data.

Abstract: Wind noise reduction is provided by obtaining a first signal from a first microphone and a contemporaneous second signal from a second microphone. A level of the first signal is compared to a level of the second signal, within a short or substantially instantaneous time frame. If the level of the first signal exceeds the level of the second signal by greater than a predefined difference threshold, a suppression is applied to the first signal.

Abstract: A signal processing module is configured to receive left and right channels of stereo input audio data and generate first and second channels of output audio data for first and second loudspeakers where the first and second loudspeakers have different frequency responses to one another. The signal processing module comprises an impulse emphasis block configured to emphasise impulsive sounds in the received audio in at least one of the first and second channels of output audio data.

Abstract: A signal processing module is configured to receive left and right channels of stereo input audio data and generate first and second channels of output audio data for first and second loudspeakers where the first and second loudspeakers have different frequency responses to one another. The signal processing module comprises an impulse emphasis block configured to emphasize impulsive sounds in the received audio in at least one of the first and second channels of output audio data.

Abstract: A method of wind noise reduction. Left side and right side microphone signals are obtained. In a first stage wind noise reduction is applied to a first sub-band of one of the signals, below a spectral threshold NA. In a second stage the wind noise reduced first side signal, and the second side signal, are both split into a sub-band below a threshold NB less than NA. The sub-band of the first side signal is mixed with the sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise, which is recombined with the respective sub-bands above the threshold NB to produce output first and second side signals.

Abstract: A signal processing module is configured to receive left and right channels of stereo input audio data and generate first and second channels of output audio data for first and second loudspeakers where the first and second loudspeakers have different frequency responses to one another. The signal processing module comprises an impulse emphasis block configured to emphasise impulsive sounds in the received audio in at least one of the first and second channels of output audio data.

Abstract: A method of adaptively mapping a plurality of microphone signals to a multi-channel audio signal, for example to capture stereo audio from a device held in variable orientations. At least first and second microphone signals, from respective first and second spaced apart microphones, are obtained. A device orientation signal is also obtained from a device orientation sensor. The microphone signals are adaptively mapped, based on the device orientation signal, to produce a first audio signal channel of a multi-channel audio signal. The first and second microphone signals are also adaptively mapped, based on the device orientation signal, to produce a second audio signal channel of a multi-channel audio signal.

Abstract: A method of wind noise reduction. Left side and right side microphone signals are obtained. In a first stage wind noise reduction is applied to a first sub-band of one of the signals, below a spectral threshold NA. In a second stage the wind noise reduced first side signal, and the second side signal, are both split into a sub-band below a threshold NB less than NA. The sub-band of the first side signal is mixed with the sub-band of the second side signal to produce an aggregate third sub-band signal having reduced wind noise, which is recombined with the respective sub-bands above the threshold NB to produce output first and second side signals.