Abstract: A first digital signal processor (DSP) associated with a first user outputs audio to the first user via first headphones, and captures speech from the first user via a first microphone. Similarly, a second DSP associated with a second user outputs audio to the second user via second headphones, and captures speech from the second user via a second microphone. The first DSP is coupled to the second DSP in order to allow the first and second users to share music and communicate with one another. The first user may speak into the first microphone, and the first and second DSPs may then interoperate to output that speech to the second user without substantially disrupting audio output to the second user. Each of the first and second users may also select between first and second audio sources that may be coupled to the first and second DSPs, respectively.

Abstract: A personal assistant device may include a microphone configured to receive an audio command from a user and a processor. The processor may be configured to receive a microphone output signal from the microphone based on the received audio command, receive at least one other microphone output signal from another personal assistant device, and autocorrelate the microphone output signals. The processor may also be configured to determine a reverberation of each of the microphone output signals, determine whether the microphone output signal from the microphone has a lower reverberation than the at least one other microphone output signal, and transmit the microphone output signal to at least one other processor for processing of the audio command in response to the microphone output signal having a lower reverberation than the at least one other microphone output signal.

Abstract: One or more embodiments set forth an audio processing system for a personal listening device that includes a set of microphones, a noise reduction module, an audio ducker, and a mixer. The set of microphones is configured to receive a first set of audio signals from an environment. The noise reduction module is configured to detect when a signal of interest is present in the first plurality of audio signals, and, upon detecting a signal of interest, transmit a ducking control signal. The audio ducker is configured to receive the ducking control signal, and receive a second plurality of audio signals via a playback device. The audio ducker is further configured to reduce an amplitude of a second plurality of audio signals relative to the signal of interest based on the ducking control signal. The mixer combines the first plurality of audio signals and second plurality of audio signals.

Abstract: Systems and methods for detecting objects in imaging systems are provided. An example method includes, during an image composition process for capturing an image of an object using an image sensor of the imaging system, and responsive to an indication that an intensity of glare light interfering with the imaging system is higher than a threshold, acquiring a first image using the image sensor, converting the first image into at least a second image, the second image including an intensity-adjusted version of the first image, intermittently displaying images including the first image and at least the second image on a display of the imaging system, and ceasing the intermittent displaying of the images.

Abstract: In one embodiment, earphones with bimodally fitting earbuds enable accurate reproduction of music while exposing a wearer to environmental noises on one side of the wearer. More specifically, an open earbud is designed to create a substantially open (i.e., unsealed) sound chamber around one ear while a closed earbud is designed to create a substantially closed sound chamber around the other ear. Because the open earbud is associated with sound leakage that alters both the intended level and the bass content of the music, the earphones include a signal redistribution subsystem that perspicaciously replaces at least a portion of this lost signal energy. Advantageously, such earphones expose the wearer to useful localized sounds, such as noises from an auto lane adjacent to a bike lane, while conveying the intended mix of music more accurately than conventional earphones with two open earbuds.

Abstract: A personal assistant device configured to determine a location of the personal assistant device and apply settings specific to the location may include at least one speaker configured to emit a stimulus noise, at least one microphone configured to acquire a room sample based on the stimulus noise, the room sample including a sample room impulse response, and a memory configured to maintain a plurality of room profiles, each room profile including a room location, a stored room impulse response, and a room specific response. A processor may be configured to receive the room sample, compare the sample room impulse response with at least one of the stored room impulse responses, determine whether the sample room impulse response matches one of the stored room impulse responses, apply the room specific response associated with the room profile of the matched stored room impulse response.

Abstract: In one embodiment, a gain redistribution application restructures gains associated with multiple microphones included in a head-worn audio system to minimize near field noise. In response to a sound generated by a sound source, the microphones generate input signals. The gain redistribution application performs mixing operations on the input signals to generate an output signal that mitigates near field noise associated with the same side of the head as the sound source. Subsequently, the gain redistribution application transmits the output signal to a speaker that targets the same side of the head as the sound source. Advantageously, by reducing the gain associated with an input signal received via a microphone located on the same side of the head as the sound source, the gain redistribution application reduces near field noise transmitted to the user during operation in a more comprehensive fashion relative to conventional designs.

Abstract: An audio processing system for a listening device includes an input device, a voice activity detector and a ratio-based attenuator. The input device is configured to receive a first audio signal emanating from an environment and including a signal of interest. The voice activity detector is configured to generate a control signal in response to the first audio signal. The ratio-based attenuator is configured to receive the control signal and determine whether the signal level of the first audio signal exceeds the signal level of an audio signal received from an audio playback device by at least a target difference. If so, then the audio level of the playback audio signal is maintained. Otherwise, the audio level of the playback audio signal is adjusted, where, at the adjusted value, the first signal level exceeds the playback signal level by at least the target difference.

Abstract: An audio processor receives ambient audio input and performs time frequency analysis. A controller receives a first description of a first step of a project, generates a first sound signature according to the time frequency analysis of the ambient audio input during a timeframe corresponding to performance of the first step, adds the first step to a project script including the first description and the first signature for matching to future audio input to determine whether the first step is being performed, receives a second description of a second step of the project, generates a second sound signature based on the time frequency analysis of the ambient audio input during a timeframe corresponding to performance of the second step, and adds the second step to the project script including the second description and the second signature for matching to future audio input to determine whether the second step is performed.

Abstract: A personal assistant device configured to determine a location of the personal assistant device and apply settings specific to the location may include at least one speaker configured to emit a stimulus noise, at least one microphone configured to acquire a room sample based on the stimulus noise, the room sample including a sample room impulse response, and a memory configured to maintain a plurality of room profiles, each room profile including a room location, a stored room impulse response, and a room specific response. A processor may be configured to receive the room sample, compare the sample room impulse response with at least one of the stored room impulse responses, determine whether the sample room impulse response matches one of the stored room impulse responses, apply the room specific response associated with the room profile of the matched stored room impulse response.

Abstract: An audio processor receives ambient audio input and performs time frequency analysis. A controller receives a first description of a first step of a project, generates a first sound signature according to the time frequency analysis of the ambient audio input during a timeframe corresponding to performance of the first step, adds the first step to a project script including the first description and the first signature for matching to future audio input to determine whether the first step is being performed, receives a second description of a second step of the project, generates a second sound signature based on the time frequency analysis of the ambient audio input during a timeframe corresponding to performance of the second step, and adds the second step to the project script including the second description and the second signature for matching to future audio input to determine whether the second step is performed.

Abstract: Systems and methods for detecting objects in imaging systems are provided. An example method includes, during an image composition process for capturing an image of an object using an image sensor of the imaging system, and responsive to an indication that an intensity of glare light interfering with the imaging system is higher than a threshold, acquiring a first image using the image sensor, converting the first image into at least a second image, the second image including an intensity-adjusted version of the first image, intermittently displaying images including the first image and at least the second image on a display of the imaging system, and ceasing the intermittent displaying of the images.

Abstract: A non-transitory storage maintains a project script including a sequence of steps, each step including a description of the step and a sound signature indicative of ambient sounds that occur during performance of the step. An audio processor is programmed to receive ambient audio input and perform a time frequency analysis of the ambient audio input. A controller is programmed to provide a description of a next step of the project script upon completion of a current step identified responsive to the time frequency analysis of the ambient audio input matching a sound signature corresponding to the next step of the project script.

Abstract: One or more embodiments set forth include a personal aural device that includes an expandable eartip. The expandable eartip has a contracted state and an expanded state. The expandable eartip includes an insertion sensor configured to receive an indication that the eartip has been placed in an ear of a user. The expandable eartip further includes a controller configured to, in response, actuate a first device to inflate a first chamber to conform to a first portion of the ear. At least one advantage of the described techniques is that eartips in a set of earphones automatically inflate to provide an optimal fit for any listener's ears.

Abstract: One or more embodiments set forth an audio processing system for a personal listening device that includes a set of microphones, a noise reduction module, an audio ducker, and a mixer. The set of microphones is configured to receive a first set of audio signals from an environment. The noise reduction module is configured to detect when a signal of interest is present in the first plurality of audio signals, and, upon detecting a signal of interest, transmit a ducking control signal. The audio ducker is configured to receive the ducking control signal, and receive a second plurality of audio signals via a playback device. The audio ducker is further configured to reduce an amplitude of a second plurality of audio signals relative to the signal of interest based on the ducking control signal. The mixer combines the first plurality of audio signals and second plurality of audio signals.

Abstract: In one embodiment, a recording optimizer included in a recorder optimizes the fidelity of output signals. In operation, the recorder receives and records two sets of input signals—one set of input signals received via microphones included in the recorder and another set of input signals received as line-inputs from a front of house console mixer. The recording optimizer analyzes the recorded signals to identify discrepancies, such as unamplified instruments that are underrepresented in the line-inputs. The recording optimizer then performs compensation operations that adjust one or more recorded signals to mitigate the discrepancies. Subsequently, the recording optimizer combines the compensated recorded signals, generating output signals that leverage the strengths of both sets of input signals to accurately represent the experienced sound.

Abstract: A multiband ducker is configured to duck a specific range of frequencies within a music signal in proportion to a corresponding range of frequencies within a speech signal, and then combine the ducked music signal with the speech signal for output to a user. In doing so, the multiband ducker separates the music signal into different frequency ranges, which may include low, middle, and high-range frequencies. The multiband ducker then reduces the amplitude of the specific range of frequencies found in the speech signal, typically the mid-range frequencies. When the ducked music signal and the speech signal are combined, the resultant signal includes important frequencies of the original music signal, including low-range and high-range, thereby allowing perception of the music signal to continue in relatively uninterrupted fashion. Additionally, the combined signal also includes the speech signal, allowing for the perception of intelligible speech concordant with the perception of music.

Abstract: An audio processing system includes a group of microphones associated with a dynamic network of microphones and a receiver. The receiver is configured to identify a first signal received by a microphone in the plurality of microphones that is designated as a primary microphone, and identify a subset of microphones included in the plurality of microphones, where each microphone in the subset is associated with a respective signal corresponding to the first signal. The receiver is further configured to calculate a weighting factor for each microphone included in the subset based on the first signal and the respective signal and opportunistically establish a connection with a microphone associated with the dynamic network of microphones that is not included in the plurality of microphones; and, based on a signal received from this microphone, adjust a weighting factor for at least one of the microphones in the subset.

Abstract: In one embodiment, a gain redistribution application restructures gains associated with multiple microphones included in a head-worn audio system to minimize near field noise. In response to a sound generated by a sound source, the microphones generate input signals. The gain redistribution application performs mixing operations on the input signals to generate an output signal that mitigates near field noise associated with the same side of the head as the sound source. Subsequently, the gain redistribution application transmits the output signal to a speaker that targets the same side of the head as the sound source. Advantageously, by reducing the gain associated with an input signal received via a microphone located on the same side of the head as the sound source, the gain redistribution application reduces near field noise transmitted to the user during operation in a more comprehensive fashion relative to conventional designs.

Abstract: An audio processing system for a listening device includes an input device, a voice activity detector and a ratio-based attenuator. The input device is configured to receive a first audio signal emanating from an environment and including a signal of interest. The voice activity detector is configured to generate a control signal in response to the first audio signal. The ratio-based attenuator is configured to receive the control signal and determine whether the signal level of the first audio signal exceeds the signal level of an audio signal received from an audio playback device by at least a target difference. If so, then the audio level of the playback audio signal is maintained. Otherwise, the audio level of the playback audio signal is adjusted, where, at the adjusted value, the first signal level exceeds the playback signal level by at least the target difference.