Abstract: In a particular aspect, an apparatus includes an audio sensor configured to receive an input audio signal. The apparatus also includes speech generative circuitry configured to generate a synthesized audio signal based at least partly on automatic speech recognition (ASR) data associated with the input audio signal and based on one or more parameters indicative of state information associated with the input audio signal.

Abstract: An electronic device includes a display, wherein the display is configured to present a user interface, wherein the user interface comprises a coordinate system. The coordinate system corresponds to physical coordinates. The display is configured to present a sector selection feature that allows selection of at least one sector of the coordinate system. The at least one sector corresponds to captured audio from multiple microphones. The sector selection may also include an audio signal indicator. The electronic device includes operation circuitry coupled to the display. The operation circuitry is configured to perform an audio operation on the captured audio corresponding to the audio signal indicator based on the sector selection.

Abstract: An apparatus includes one or more sensor units configured to detect non-audible sensor data associated with a user. The apparatus also includes a processor, including an action determination unit, coupled to the one or more sensors units. The processor is configured to generate a descriptive text-based input based on the non-audible sensor data. The processor is also configured to determine an action to be performed based on the descriptive text-based input.

Abstract: Various embodiments provide systems and methods which disclose a command device which can be used to establish a wireless connection, through one or more wireless channels, between the command device and a remote device. An intention code may be generated, prior to, or after, the establishment of the wireless connection, and the remote device may be selected based on the intention code. The command device may initiate a wireless transfer, through one or more wireless channels of the established wireless connection, of an intention code, and receive acknowledgement that the intention code was successfully transferred to the remote device. The command device may then control the remote device, based on the intention code sent to the remote device, through the one or more wireless channels of the established wireless connection between the command device and the remote device.

Abstract: A method for evaluating strength of an audio password by an electronic device is described. The method includes obtaining an audio signal captured by one or more microphones. The audio signal includes an audio password. The method also includes evaluating the strength of the audio password based on measuring one or more unique characteristics of the audio signal. The method further includes informing a user that the audio password is weak based on the evaluation of the strength of the audio password.

Abstract: An electronic device includes a classifier circuit, a ranking circuit, and a data generator circuit. The classifier circuit is configured to determine, based on first data indicating samples of sounds detected at a plurality of geographic locations, a plurality of acoustic event classifications associated with the plurality of geographic locations. The ranking circuit is configured to determine a plurality of index scores associated with the plurality of geographic locations by ranking each of the plurality of geographic locations based on the plurality of acoustic event classifications. The data generator circuit is configured to generate, based on the plurality of index scores, second data indicating a geographic map corresponding to the plurality of geographic locations. The second data further indicates the plurality of index scores and a prompt to enable a search for a particular type of acoustic event.

Abstract: A method for displaying a user interface on an electronic device is described. The method includes presenting a user interface. The user interface includes a coordinate system. The coordinate system corresponds to physical coordinates based on sensor data. The method also includes providing a sector selection feature that allows selection of at least one sector of the coordinate system. The method further includes providing a sector editing feature that allows editing the at least one sector.

Abstract: Disclosed is an application interface that takes into account the user's gaze direction relative to who is speaking in an interactive multi-participant environment where audio-based contextual information and/or visual-based semantic information is being presented. Among these various implementations, two different types of microphone array devices (MADs) may be used. The first type of MAD is a steerable microphone array (a.k.a. a steerable array) which is worn by a user in a known orientation with regard to the user's eyes, and wherein multiple users may each wear a steerable array. The second type of MAD is a fixed-location microphone array (a.k.a. a fixed array) which is placed in the same acoustic space as the users (one or more of which are using steerable arrays).

Abstract: A method of processing audio may include receiving, by a computing device, a plurality of real-time audio signals outputted by a plurality of microphones communicatively coupled to the computing device. The computing device may output to a display a graphical user interface (GUI) that presents audio information associated with the received audio signals. The one or more received audio signals may be processed based on a user input associated with the audio information presented via the GUI to generate one or more processed audio signals. The one or more processed audio signals may be output to, for example, one or more output devices such as speakers, headsets, and the like.

Abstract: In a particular aspect, an apparatus includes an audio sensor configured to receive an input audio signal. The apparatus also includes speech generative circuitry configured to generate a synthesized audio signal based at least partly on automatic speech recognition (ASR) data associated with the input audio signal and based on one or more parameters indicative of state information associated with the input audio signal.

Abstract: A method of processing audio may include receiving, by a computing device, a plurality of real-time audio signals outputted by a plurality of microphones communicatively coupled to the computing device. The computing device may output to a display a graphical user interface (GUI) that presents audio information associated with the received audio signals. The one or more received audio signals may be processed based on a user input associated with the audio information presented via the GUI to generate one or more processed audio signals. The one or more processed audio signals may be output to, for example, one or more output devices such as speakers, headsets, and the like.

Abstract: A method of performing noise reduction includes capturing a first audio signal at a first microphone of a first device. The method also includes receiving, at the first device, audio data representative of a second audio signal from a second device. The second audio signal is captured by a second microphone of the second device. The method further includes performing noise reduction on the first audio signal based at least in part on the audio data representative of the second audio signal.

Abstract: A method for speech modeling by an electronic device is described. The method includes obtaining a real-time noise reference based on a noisy speech signal. The method also includes obtaining a real-time noise dictionary based on the real-time noise reference. The method further includes obtaining a first speech dictionary and a second speech dictionary. The method additionally includes reducing residual noise based on the real-time noise dictionary and the first speech dictionary to produce a residual noise-suppressed speech signal at a first modeling stage. The method also includes generating a reconstructed speech signal based on the residual noise-suppressed speech signal and the second speech dictionary at a second modeling stage.

Abstract: A system includes a memory configured to store data associated with a service that is available. The system also includes a microphone associated with an acoustic space and configured to receive an audio input produced by a person. The system further includes a sensor located within the acoustic space and configured to detect vibrations produced by the person. The system includes a processor coupled to the memory, to the microphone, and to the sensor. The processor is configured to conditionally authorize execution of the service requested by the person, the service conditionally authorized based on the audio input and the vibrations.

Abstract: A multichannel acoustic system (MAS) comprises an arrangement of microphones and loudspeakers and a multichannel acoustic processor (MAP) to together enhance conversational speech between two or more persons in a shared acoustic space such as an automobile. The enhancements are achieved by receiving sound signals substantially originating from relatively near sound sources; filtering the sound signals to cancel at least one echo signal detected for at least one microphone from among the plurality of microphones; filtering the sound signals received by the plurality of microphones to cancel at least one feedback signal detected for at least one microphone from among the plurality of microphones; and reproducing the filtered sound signals for each microphone from among the plurality of microphones on a subset of loudspeakers corresponding that are relatively far from the source microphone.

Abstract: A method for multi-channel echo cancellation and noise suppression is described. One of multiple echo estimates is selected for non-linear echo cancellation. Echo notch masking is performed on a noise-suppressed signal based on an echo direction of arrival (DOA) to produce an echo-suppressed signal. Non-linear echo cancellation is performed on the echo-suppressed signal based, at least in part, on the selected echo estimate.

Abstract: A method for outputting virtual sound includes detecting an audio signal in an environment at one or more microphones. The method also includes determining, at a processor, a location of a sound source of the audio signal and estimating one or more acoustical characteristics of the environment based on the audio signal. The method further includes inserting a virtual sound into the environment based on the one or more acoustical characteristics. The virtual sound has one or more audio properties of a sound generated from the location of the sound source.

Abstract: An apparatus includes multiple microphones to generate audio signals based on sound of a far-field acoustic environment. The apparatus also includes a signal processing system to process the audio signals to generate at least one processed audio signal. The signal processing system is configured to update one or more processing parameters while operating in a first operational mode and is configured to use a static version of the one or more processing parameters while operating in the second operational mode. The apparatus further includes a keyword detection system to perform keyword detection based on the at least one processed audio signal to determine whether the sound includes an utterance corresponding to a keyword and, based on a result of the keyword detection, to send a control signal to the signal processing system to change an operational mode of the signal processing system.

Abstract: An apparatus for detecting a sound in an acoustical environment includes a microphone array configured to detect an audio signal in the acoustical environment. The apparatus also includes a processor configured to determine an angular location of a sound source of the audio signal. The angular location is relative to the microphone array. The processor is also configured to determine at least one reverberation characteristic of the audio signal. The processor is further configured to determine a distance, relative to the microphone array, of the sound source along an axis associated with the angular location based on the at least one reverberation characteristic.

Abstract: A method for outputting virtual sound includes detecting an audio signal in an environment at one or more microphones. The method also includes determining, at a processor, a location of a sound source of the audio signal and estimating one or more acoustical characteristics of the environment based on the audio signal. The method further includes inserting a virtual sound into the environment based on the one or more acoustical characteristics. The virtual sound has one or more audio properties of a sound generated from the location of the sound source.