Abstract: Sound is banked laterally over an array of microphones arranged on a rear surface of a device. Sound enters a duct behind the device from different directions via inlets along the sides of the device. The duct directs the sound waves across the microphone array. An effective direction from which the banked sounds originated is determined, relative to a front of the device. Based on the determined effective direction, the device applies spatial filtering to isolate the received sound waves, selectively increasing a signal-to-noise ratio of sound from the selected source and at least partially occluding sounds from other sources.

Abstract: A speech-based audio device may be configured to detect a user-uttered wake expression and to respond by interpreting subsequent words or phrases as commands. In order to distinguish between utterance of the wake expression by the user and generation of the wake expression by the device itself, directional audio signals may by analyzed to detect whether the wake expression has been received from multiple directions. If the wake expression has been received from many directions, it is declared as being generated by the audio device and ignored. Otherwise, if the wake expression is received from a single direction or a limited number of directions, the wake expression is declared as being uttered by the user and subsequent words or phrase are interpreted and acted upon by the audio device.

Abstract: An echo cancellation system that performs audio beamforming to separate audio input into multiple directions and determines a target signal and a reference signal from the multiple directions. For example, the system may detect a strong signal associated with a speaker and select the strong signal as a reference signal, selecting another direction as a target signal. The system may determine a speech position and may select the speech position as a target signal and an opposite direction as a reference signal. The system may create pairwise combinations of opposite directions, with an individual direction being selected as a target signal and a reference signal. The system may select a fixed beamformer output for the target signal and an adaptive beamformer output for the reference signal, or vice versa. The system may remove the reference signal (e.g., audio output by the loudspeaker) to isolate speech included in the target signal.

Abstract: A speech-based audio device may be configured to detect a user-uttered wake expression and to respond by interpreting subsequent words or phrases as commands. In order to distinguish between utterance of the wake expression by the user and generation of the wake expression by the device itself, directional audio signals may by analyzed to detect whether the wake expression has been received from multiple directions. If the wake expression has been received from many directions, it is declared as being generated by the audio device and ignored. Otherwise, if the wake expression is received from a single direction or a limited number of directions, the wake expression is declared as being uttered by the user and subsequent words or phrase are interpreted and acted upon by the audio device.

Abstract: An echo cancellation system that performs audio beamforming to separate audio input into multiple directions and determines a target signal and a reference signal from the multiple directions. For example, the system may detect a strong signal associated with a speaker and select the strong signal as a reference signal, selecting another direction as a target signal. The system may determine a speech position and may select the speech position as a target signal and an opposite direction as a reference signal. The system may create pairwise combinations of opposite directions, with an individual direction being selected as a target signal and a reference signal. The system may select a fixed beamformer output for the target signal and an adaptive beamformer output for the reference signal, or vice versa. The system may remove the reference signal (e.g., audio output by the loudspeaker) to isolate speech included in the target signal.

Abstract: An echo cancellation system performs audio beamforming to separate audio input into multiple directions (e.g., target signals) and generates multiple audio outputs using two acoustic echo cancellation (AEC) circuits. A first AEC removes a playback reference signal (generated from a signal sent a loudspeaker) to isolate speech included in the target signals. A second AEC removes an adaptive reference signal (generated from microphone inputs corresponding to audio received from the loudspeaker) to isolate speech included in the target signals. A beam selector receives the multiple audio outputs and selects the first AEC or the second AEC based on a linearity of the system. When linear (e.g., no distortion or variable delay between microphone input and playback signal), the beam selector selects an output from the first AEC based on signal to noise (SNR) ratios. When nonlinear, the beam selector selects an output from the second AEC.

Abstract: A limiter for an audio system prevents loud audio signals that exceed a threshold from being output. Output of the audio signals are delayed. When a loud signal exceeds the threshold, the gain applied to the delayed signals is gradually reduced so that by the time the loud signal reaches the output, the gain is at a level that reduces the loud audio signal to be within the threshold. Thereafter the gain is gradually restored to normal over a longer period of time than the audio signals are delayed.

Abstract: An echo cancellation system that uses a combined reference signal using a playback reference signal and an adaptive reference signal. The playback reference signal is generated from a playback signal sent to a loudspeaker and the adaptive reference signal is generated using beamforming on microphone inputs corresponding to audio received from the loudspeaker. The system applies a low pass filter to the playback reference signal and applies a high pass filter to the adaptive reference signal to generate the combined reference signal. The system may remove the combined reference signal from target signals associated with the microphone inputs to isolate speech included in the target signals.

Abstract: A gain control system is responsive to a user gain setting and to an effective signal level to calculate and adaptively vary an output gain to avoid output clipping. The effective signal level is calculated by smoothing the energy of an input audio signal in accordance with selected attenuation factors. In times during which fast transient sounds are occurring, attenuation factors corresponding to relatively low levels of attenuation are selected. In times during which fast transient sounds are not detected, attenuation factors corresponding to relatively higher levels of attenuation are selected. The effective signal level is held at a constant level during periods of silence in order to avoid increasing the output gain during these periods. The output gain is calculated based on a comparison of the effective signal level to a compression threshold.

Abstract: An echo cancellation system that detects and compensates for differences in sample rates between the echo cancellation system and a set of wireless speakers based on a frequency-domain analysis. The system generates Fourier transforms for a microphone signal and a reference signal and determines a series of angles for individual frames. For each tone in the Fourier transforms, the system determines the angles and uses linear regression to determine an individual frequency offset associated with the tone. Using the individual frequency offsets associated with the tones, the system uses linear regression to determine an overall frequency offset between the audio sent to the speakers and the audio received from a microphone. Based on the overall frequency offset, samples of the audio are added or dropped when echo cancellation is performed, compensating for the frequency offset.

Abstract: An echo cancellation system that detects and compensations for differences in sample rates between the echo cancellation system and a set of wireless speakers based on a frequency-domain analysis of estimated impulse response coefficients. The system tracks the real and imaginary number components of the coefficients, and determines a “rotation” of the coefficients over time caused by a frequency offset between the audio sent to the speakers and the audio received from a microphone. Based on the rotation, samples of the audio are added or dropped when echo cancellation is performed, compensating for the frequency offset.

Abstract: A wearable computer is configured to use beamforming techniques to isolate a user's speech from extraneous audio signals occurring within a physical environment. A microphone array of the wearable computer may generate audio signal data from an utterance from a user's mouth. A motion sensor(s) of the wearable computer may generate motion data from movement of the wearable computer. This motion data may be used to determine a direction vector pointing from the wearable computer to the user's mouth, and a beampattern may be defined that has a beampattern direction in substantial alignment with the determined direction vector to focus the microphone array on the user's mouth for speech isolation.

Abstract: A loudness learning and balancing system for one or more audio channels. The system learns user volume preferences from manual interventions, providing effective and robust loudness learning to provide a consistent and balanced sonic experience. The system also reduces and/or eliminates incorrect attenuation or over-amplification of quiet interludes in the audio material, thereby eliminating artifacts such as noise pumping or breathing. In addition, the system reduces low level noise in the audio signals while preserving a wide dynamic range at most listening levels.

Abstract: Embodiments of systems and methods are described for determining which of a plurality of beamformed audio signals to select for signal processing. In some embodiments, a plurality of audio input signals are received from a microphone array comprising a plurality of microphones. A plurality of beamformed audio signals are determined based on the plurality of input audio signals, the beamformed audio signals comprising a direction. A plurality of signal features may be determined for each beamformed audio signal. Smoothed features may be determined for each beamformed audio signal based on at least a portion of the plurality of signal features. The beamformed audio signal corresponding to the maximum smoothed feature may be selected for further processing.

Abstract: A test system for testing directional capabilities of an audio device includes a horizontal linear actuator and a vertical linear actuator. The horizontal linear actuator supports a rotary actuator upon which the audio device is placed. The vertical linear actuator supports a sound source. To test the device, the actuators are controlled to establish multiple relative positions of the audio device and the sound source. A test sound is emitted at each of the relative positions. The audio device is configured to provide data generated in response to the test sound at each of the relative positions. The test system receives and records the data along with coordinates indicating the relative positions to which the data corresponds.

Abstract: An audio device may be configured to produce output audio and to capture input audio for speech recognition. In some cases, a second device may also be used to capture input audio to improve isolation of input audio with respect to the output audio. In addition, acoustic echo cancellation (AEC) may be used to remove components of output audio from input signals of the first and second devices. AEC may be implemented by an adaptive filter based on dynamically optimized filter coefficients. The filter coefficients may be analyzed to detect situations in which the first and second devices are too close to each other, and the user may then be prompted to increase the distance between the two devices.

Abstract: Audio of electronic audio devices may be synchronized by a signal synchronization component that receives one or more signals corresponding to elements of the audio output by the electronic audio devices. The signal synchronization component may perform calculations to align signals corresponding to the output audio of the electronic audio devices and then determine a delay for the output audio transmitted from the electronic audio devices with respect to each other. Additionally, the signal synchronization component may operate in conjunction with audio sources of the electronic audio devices to modify the timing for transmitting output audio by one or more of the electronic audio devices based, at least in part, on the delay. In this way, the output audio transmitted by the electronic audio devices may be synchronized.

Abstract: A speech-based audio device may be configured to detect a user-uttered wake expression and to respond by interpreting subsequent words or phrases as commands. In order to distinguish between utterance of the wake expression by the user and generation of the wake expression by the device itself, directional audio signals may by analyzed to detect whether the wake expression has been received from multiple directions. If the wake expression has been received from many directions, it is declared as being generated by the audio device and ignored. Otherwise, if the wake expression is received from a single direction or a limited number of directions, the wake expression is declared as being uttered by the user and subsequent words or phrase are interpreted and acted upon by the audio device.

Abstract: A method of determining a prothrombin time is disclosed. A mediator is applied to a stratum corneum. The stratum corneum is penetrated to allow the mediator to enter a region containing interstitial fluid and interact with at least one capillary, causing blood and/or blood components to leak from the at least one capillary into the region containing interstitial fluid. A characteristic affected by the blood and/or blood components is measured in the region containing interstitial fluid which correlates to the prothrombin time. A system for measuring prothrombin time is also disclosed. The system has a mediator, one or more microneedles, and a processor directly or indirectly coupled to the one or more microneedles and configured to determine a coagulation change in blood or blood components in a region around the one or more microneedles after the one or more microneedles penetrate a stratum corneum.

Abstract: A method of determining a prothrombin time is disclosed. A mediator is applied to a stratum corneum. The stratum corneum is penetrated to allow the mediator to enter a region containing interstitial fluid and interact with at least one capillary, causing blood and/or blood components to leak from the at least one capillary into the region containing interstitial fluid. A characteristic affected by the blood and/or blood components is measured in the region containing interstitial fluid which correlates to the prothrombin time. A system for measuring prothrombin time is also disclosed. The system has a mediator, one or more microneedles, and a processor directly or indirectly coupled to the one or more microneedles and configured to determine a coagulation change in blood or blood components in a region around the one or more microneedles after the one or more microneedles penetrate a stratum corneum.