Abstract: A device that includes a microphone may be worn in or on an ear of a user. A microphone signal generated by the microphone may be processed to determine a heart activity of a user. An indication of a heart pathology may be detected by applying a predictive algorithm to at least the heart activity. Other aspects are described.

Abstract: An appliance can include a microphone transducer, a processor, and a memory storing instructions. The appliance is configured to receive an audio signal at the microphone transducer and to detect an utterance in the audio signal. The appliance is further configured to classify a speech mode based on the utterance. The appliance is further configured to determine conditions of an environment of the appliance. The appliance is further configured to select at least one of a playback volume or a speech output mode from a plurality of speech output modes based on the classification, and the conditions of the environment of the appliance. The appliance is further configured to adapt the playback volume and/or mode of played-back speech according to the speech output mode. The appliance may be configured to synthesize speech according to the speech output mode, or to modify synthesized speech according to the speech output mode.

Abstract: A device may include microphones worn on a head of a user. The device may include a processor, configured to obtain microphone signals from the plurality of microphones. The processor may attenuate breathing sound from the user by processing the microphone signals, resulting in attenuated microphone signals. The processor may render one or more output audio channels based on the plurality of attenuated microphone signals.

Abstract: A device that includes a microphone may be worn in or on an ear of a user. A microphone signal generated by the microphone may be processed to determine a heart activity of a user. An indication of a heart pathology may be detected by applying a predictive algorithm to at least the heart activity. Other aspects are described.

Abstract: A plurality of microphone signals can be obtained. In the plurality of microphone signals, speech of a user can be detected. A gaze of a user can be determined based on the plurality of microphone signals. A voice activated response of the computing device can be performed in response to the gaze of the user being directed at the computing device. Other aspects are described and claimed.

Abstract: A device implementing a system for processing speech in an audio signal includes at least one processor configured to receive an audio signal corresponding to at least one microphone of a device, and to determine, using a first model, a first probability that a speech source is present in the audio signal. The at least one processor is further configured to determine, using a second model, a second probability that an estimated location of a source of the audio signal corresponds to an expected position of a user of the device, and to determine a likelihood that the audio signal corresponds to the user of the device based on the first and second probabilities.

Abstract: An appliance can include a microphone transducer, a processor, and a memory storing instructions. The appliance is configured to receive an audio signal at the microphone transducer and to detect an utterance in the audio signal. The appliance is further configured to classify a speech mode based on the utterance. The appliance is further configured to determine conditions of an environment of the appliance. The appliance is further configured to select at least one of a playback volume or a speech output mode from a plurality of speech output modes based on the classification, and the conditions of the environment of the appliance. The appliance is further configured to adapt the playback volume and/or mode of played-back speech according to the speech output mode. The appliance may be configured to synthesize speech according to the speech output mode, or to modify synthesized speech according to the speech output mode.

Abstract: An echo canceller is disclosed in which audio signals of the playback content received by one or more of the microphones from a loudspeaker of the device may be used as the playback reference signals to estimate the echo signals of the playback content received by a target microphone for echo cancellation. The echo canceller may estimate the transfer function between a reference microphone and the target microphone based on the playback reference signal of the reference microphone and the signal of the target microphone. To mitigate near-end speech cancellation at the target microphone, the echo canceller may compute a mask to distinguish between target microphone audio signals that are echo-signal dominant and near-end speech dominant. The echo canceller may use the mask to adaptively update the transfer function or to modify the playback reference signal used by the transfer function to estimate the echo signals of the playback content.

Abstract: An appliance can include a microphone transducer, a processor, and a memory storing instructions. The appliance is configured to receive an audio signal at the microphone transducer and to detect an utterance in the audio signal. The appliance is further configured to classify a speech mode based on the utterance. The appliance is further configured to determine conditions of an environment of the appliance. The appliance is further configured to select at least one of a playback volume or a speech output mode from a plurality of speech output modes based on the classification, and the conditions of the environment of the appliance. The appliance is further configured to adapt the playback volume and/or mode of played-back speech according to the speech output mode. The appliance may be configured to synthesize speech according to the speech output mode, or to modify synthesized speech according to the speech output mode.

Abstract: An echo canceller is disclosed in which audio signals of the playback content received by one or more of the microphones from a loudspeaker of the device may be used as the playback reference signals to estimate the echo signals of the playback content received by a target microphone for echo cancellation. The echo canceller may estimate the transfer function between a reference microphone and the target microphone based on the playback reference signal of the reference microphone and the signal of the target microphone. To mitigate near-end speech cancellation at the target microphone, the echo canceller may compute a mask to distinguish between target microphone audio signals that are echo-signal dominant and near-end speech dominant. The echo canceller may use the mask to adaptively update the transfer function or to modify the playback reference signal used by the transfer function to estimate the echo signals of the playback content.

Abstract: Systems, methods, and computer readable media to improve the operation of an electronic device having multiple microphones organized in an arbitrary, but known, arrangement in the device are described (i.e., having a specific form-factor). In general, techniques are disclosed for using a priori knowledge of an electronic device's spatial acoustic transfer functions to recreate or reconstitute a prior recorded three-dimensional (3D) audio field or environment. More particularly, techniques disclosed herein enable the efficient recording of a 3D audio field. That audio field may later be reconstituted using an acoustic characterization based on the device's form-factor. In addition, sensor data may be used to rotate the audio field so as to enable generating an output audio field that takes into account the listener's head position.

Abstract: An audio appliance can include a microphone transducer configured to receive sound from an environment and to convert the received sound into an audio signal and a display. The audio appliance can include an audio analytics module configured to detect an audio-input impairment by analyzing the audio signal and output a detection signal identifying the audio-input impairment in real-time. The audio-input impairment can include, for example, a poor-intelligibility impairment, a microphone-occlusion impairment, a handling-noise impairment, a wind-noise impairment, or a distortion impairment. The audio appliance can also include an impairment module configured to identify and emit a user-perceptible alert corresponding to the identified audio-input impairment in real-time; and an interactive guidance module configured to present a suggested action to address the audio-input impairment in real-time. Related aspects also are described.

Abstract: An audio appliance can include a microphone transducer configured to receive sound from an environment and to convert the received sound into an audio signal and a display. The audio appliance can include an audio analytics module configured to detect an audio-input impairment by analyzing the audio signal and output a detection signal identifying the audio-input impairment in real-time. The audio-input impairment can include, for example, a poor-intelligibility impairment, a microphone-occlusion impairment, a handling-noise impairment, a wind-noise impairment, or a distortion impairment. The audio appliance can also include an impairment module configured to identify and emit a user-perceptible alert corresponding to the identified audio-input impairment in real-time; and an interactive guidance module configured to present a suggested action to address the audio-input impairment in real-time. Related aspects also are described.

Abstract: Systems and methods for reducing effects of time-division multiplexing noise in mobile communications devices. When cellular communication with time-division multiplexing is detected, such as Global System for Mobiles (GSM) communication with Time Division Multiple Access (TDMA) protocol, total energy and energy at a repetition frequency of the time division multiplexing is measured in audio signals received from several microphones located in the device. A control signal indicating microphones affected by TDMA noise is provided to signal processing subsystems that receive audio signals from the microphones. A beam former circuit may combine two or more audio signals to produce beam formed signals. The control signal may further indicate beam formed signals affected by TDMA noise based on a ratio of the energy from the repetition frequency to the total energy in the beam formed signals.

Abstract: Systems and methods for reducing effects of time-division multiplexing noise in mobile communications devices. When cellular communication with time-division multiplexing is detected, such as Global System for Mobiles (GSM) communication with Time Division Multiple Access (TDMA) protocol, total energy and energy at a repetition frequency of the time division multiplexing is measured in audio signals received from several microphones located in the device. A control signal indicating microphones affected by TDMA noise is provided to signal processing subsystems that receive audio signals from the microphones. A beam former circuit may combine two or more audio signals to produce beam formed signals. The control signal may further indicate beam formed signals affected by TDMA noise based on a ratio of the energy from the repetition frequency to the total energy in the beam formed signals.

Abstract: An audio system has a housing in which are integrated a number of microphones. A programmed processor accesses the microphone signals and produces a number of acoustic pick up beams. A number of separation values are computed, each being a measure of the difference between strength of a respective beam and strength of a noise reference input signal. One of the beams is selected whose separation value is the largest, and the selected beam is applied to a first input of a two-channel noise suppression process, while the noise reference input signal is applied to the second input of the noise suppression process. Other embodiments are also described and claimed.

Abstract: An orientation detector can have a first microphone, a second microphone, and a reference microphone spaced from the first microphone and the second microphone. An orientation processor can be configured to determine an orientation of the first microphone, the second microphone, or both, relative to a user's mouth based on a comparison of a relative strength of a first signal associated with the first microphone to a relative strength of a second signal associated with the second microphone. A channel selector in a speech enhancer can select one signal from among several signals based at least in part on the orientation determined by the orientation processor. A mobile communication handset can include a microphone-based orientation detector of the type disclosed herein.

Abstract: An audio device may use the audio detected at two opposite facing, front and rear omnidirectional microphones to determine the angular directional location of a user's voice while the device in speaker mode or audio command input mode. The angular directional location may be determined to be at front, side and rear locations of the device during the period of time by calculating an energy ratio of audio signals output by the front and rear microphones during the period. Comparing the ratio to experimental data for sound received from different directions around the device may provide the location of the user's voice. Based on the determination, audio beamforming input settings may be adjusted for user voice beamforming. As a result, the device can perform better beamforming to combine the signals captured by the microphones and generate a single output that isolates the user's voice from background noise.

Abstract: An orientation detector can have a first microphone, a second microphone, and a reference microphone spaced from the first microphone and the second microphone. An orientation processor can be configured to determine an orientation of the first microphone, the second microphone, or both, relative to a user's mouth based on a comparison of a relative strength of a first signal associated with the first microphone to a relative strength of a second signal associated with the second microphone. A channel selector in a speech enhancer can select one signal from among several signals based at least in part on the orientation determined by the orientation processor. A mobile communication handset can include a microphone-based orientation detector of the type disclosed herein.

Abstract: Systems and methods for improving the audio quality in a living-room type of environment having multimedia equipment are described. The audio signals emanating from various speakers may be modified/altered, so as to capture audio signals with the highest signal-to-noise ratio. In the context of a video/audio call, where the positions of the microphone and speakers are generally fixed, the speaker outputs may be controlled/modified to provide an improved signal-to-noise ratio. The audio emanating from the central speaker may be subdued or eliminated to reduce significantly the problems that occur due to the audio from the center speaker being captured by the microphone, which is typically in close proximity to the center speaker. The audio may be provided to the speakers by various ways, such as HDMI, RCA, and so on.