Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.

Abstract: This disclosure describes, in part, techniques to process audio signals to lessen the impact that wind and/or other environmental noise has upon the resulting quality of these audio signals. For example, the techniques may determine a level of wind and/or other noise in an environment and may determine how best to process the signals to lessen the impact of the noise, such that one or more users that hear audio based on output of the signals hear higher-quality audio.

Abstract: A device that determines an echo latency estimate by subsampling reference audio data. The device may determine the echo latency corresponding to an amount of time between sending reference audio data to loudspeaker(s) and microphone audio data corresponding to the reference audio data being received. The device may generate subsampled reference audio data by selecting only portions of the reference audio data that have a magnitude above a desired percentile. For example, the device may compare a magnitude of an individual reference audio sample to a percentile estimate value and sample only the reference audio samples that exceed the percentile estimate value. The device generate cross-correlation data between the subsampled reference audio data and the microphone audio data and may estimate the echo latency based on an earliest significant peak represented in the cross-correlation data.

Abstract: In a method of adaptive buffering in a mobile device having a host processor and a sensor processor coupled with the host processor, the sensor processor is used to buffer data received from a sensor that is operated by the sensor processor. The data is buffered by the sensor processor into a circular data buffer. Responsive to the sensor processor detecting triggering data within the received data: a first adaptive data buffering action is initiated with respect to the data received from the sensor operated by the sensor processor; a second adaptive data buffering action is initiated with respect to second data received from a second sensor of the mobile device; and a command is sent from the sensor processor to a second processor.

Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.

Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.

Abstract: In a method of adaptive buffering in a mobile device having a host processor and a sensor processor coupled with the host processor, the sensor processor is used to buffer data received from a sensor that is operated by the sensor processor, wherein the data is buffered by the sensor processor into a circular data buffer. In response to the sensor processor detecting triggering data within the received data: the sensor processor sets a start-end marker in the circular data buffer; and a command is sent from the sensor processor to a second processor.

Abstract: A speech-capturing device that can modulate its output audio data volume based on environmental sound conditions at the location of a user speaking to the device. The device detects the sound pressure of a spoken utterance at the device location and determines the distance of the user from the device. The device also detects the sound pressure of noise at the device and uses information about the location of the noise source and user to determine the sound pressure of noise at the location of the talker. The device can then adjust the gain for output audio (such as a spoken response to the utterance) to ensure that the output audio is at a certain desired sound pressure when it reaches the location of the user.

Abstract: In a method of adaptive buffering in a mobile device having a host processor and a sensor processor coupled with the host processor, the sensor processor is used to buffer data received from a sensor that is operated by the sensor processor. The data is buffered by the sensor processor into a circular data buffer. Responsive to the sensor processor detecting triggering data within the received data: a first adaptive data buffering action is initiated with respect to the data received from the sensor operated by the sensor processor; a second adaptive data buffering action is initiated with respect to second data received from a second sensor of the mobile device; and a command is sent from the sensor processor to a second processor.

Abstract: Systems and methods for improving performance of a directional audio capture system are provided. An example method includes correlating phase plots of at least two audio inputs, with the audio inputs being captured by at least two microphones. The method can further include generating, based on the correlation, estimates of salience at different directional angles to localize a direction of a source of sound. The method can allow providing cues to the directional audio capture system based on the estimates. The cues include attenuation levels. A rate of change of the levels of attenuation is controlled by attack and release time constants to avoid sound artifacts. The method also includes determining a mode based on an absence or presence of one or more peaks in the estimates of salience. The method also provides for configuring the directional audio capture system based on the determined mode.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: In a method of adaptive buffering in a mobile device having a host processor and a sensor processor coupled with the host processor, the sensor processor is used to buffer data received from a sensor that is operated by the sensor processor. The data is buffered by the sensor processor into a circular data buffer. Responsive to the sensor processor detecting triggering data within the received data: a first adaptive data buffering action is initiated with respect to the data received from the sensor operated by the sensor processor; a second adaptive data buffering action is initiated with respect to second data received from a second sensor of the mobile device; and a command is sent from the sensor processor to a second processor.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to frequency domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. A time-domain acoustic signal may be received and be transformed to frequency-domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: A robust noise suppression system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: Systems and methods for improving performance of a directional audio capture system are provided. An example method includes correlating phase plots of at least two audio inputs, with the audio inputs being captured by at least two microphones. The method can further include generating, based on the correlation, estimates of salience at different directional angles to localize a direction of a source of sound. The method can allow providing cues to the directional audio capture system based on the estimates. The cues include attenuation levels. A rate of change of the levels of attenuation is controlled by attack and release time constants to avoid sound artifacts. The method also includes determining a mode based on an absence or presence of one or more peaks in the estimates of salience. The method also provides for configuring the directional audio capture system based on the determined mode.

Abstract: Systems and methods for joint noise and echo suppression using noise and echo subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary acoustic signal, a secondary acoustic signal, and an echo reference acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtain a noise component signal. An adjusting coefficient for the noise component signal and an adjusting coefficient for the echo reference signal may be determined and applied. The noise component signal and echo reference signal may be subtracted from the primary acoustic signal to generate a noise and echo subtracted signal. An additional echo cancellation in the noise and echo subtracted signal may be carried out by applying a nonlinear processor. The non-linear processor is driven by at least a ratio of the noise and echo subtracted signal energy and the primary acoustic signal energy.

Abstract: Systems and methods for noise suppression using noise subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary and a secondary acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtain a noise component signal. A determination may be made of a reference energy ratio and a prediction energy ratio. A determination may be made as to whether to adjust the noise component signal based partially on the reference energy ratio and partially on the prediction energy ratio. The noise component signal may be adjusted or frozen based on the determination. The noise component signal may then be removed from the primary acoustic signal to generate a noise subtracted signal which may be outputted.

Abstract: An audio equivalent of a video zoom feature for video recording and communication applications, as well as video post production processes. The audio zoom may operate in conjunction with a video zoom feature or independently. The audio zoom may be achieved by controlling reverberation effects of a signal, controlling a gain of the signal, as well as controlling the width of a directional beam which is used to select the particular audio component to focus on. The audio zoom may operate in response to user input, such as a user selection of a particular direction, or automatically based a current environment or other factors.

Abstract: A system for audio processing of an acoustic signal captured by at least one microphone based on analysis of a video signal associated with the acoustic signal is provided. The video signal may be dynamically processed and analyzed in real time by the system to locate one or more sound sources and to determine distances between the sound sources and the at least one microphone. This information may be used to perform selective noise suppression, noise cancellation, or selective adjustment of acoustic signal components or energy levels acoustic signal components such that noise or unrelated acoustic signal components associated with sound sources not associated with located sound sources are suppressed or filtered. The video signal analysis may track sound source movement for dynamic steering of an acoustic beam towards the sound source.