Abstract: The disclosed computer-implemented method may include (1) establishing a communication channel to indirectly convey a conversation, (2) receiving, via the communication channel, a portion of the conversation, (3) presenting the portion of the conversation to a user, (4) receiving, via the communication channel, an additional portion of the conversation, (5) detecting an additional communication channel capable of conveying the conversation, (6) determining a human-perceivable difference between how the conversation has been conveyed via the communication channel and how the conversation will be conveyed via the additional communication channel, and (7) compensating for the human-perceivable difference when presenting the additional portion of the conversation to the user in order to smoothly transition the conversation from the communication channel to the additional communication channel. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: An eyewear device includes an audio system. In one embodiment, the audio system includes a microphone array that includes a plurality of acoustic sensors. Each acoustic sensor is configured to detect sounds within a local area surrounding the microphone array. For a plurality of the detected sounds, the audio system performs a direction of arrival (DoA) estimation. Based on parameters of the detected sound and/or the DoA estimation, the audio system may then generate or update one or more acoustic transfer functions unique to a user. The audio system may use the one or more acoustic transfer functions to generate audio content for the user.

Abstract: A listening device includes a reference microphone positioned outside a blocked ear canal of a user to receive environmental sounds and generate first signals based on the environmental sounds. A loudspeaker is coupled to the reference microphone and positioned inside the ear canal to generate internal sounds based on the first signals. An internal microphone is positioned inside the ear canal to receive the internal sounds from the loudspeaker and generate second signals based on the internal sounds. A controller is coupled to the internal microphone and the reference microphone to compute a transfer function based on the first signals and the second signals. The transfer function describes a variation between the environmental sounds and the internal sounds caused by the listening device blocking the ear canal. The controller adjusts, based on the transfer function, the internal sounds to mitigate the variation.

Abstract: A method for manufacturing a cartilage conduction audio device is disclosed. A manufacturing system receives data describing a three-dimensional shape of an ear (e.g., the outer ear, behind the ear, the concha bowel, etc.) of a user. The system identifies one or more locations for one or more transducers along a back of an auricle of the ear for the user that vibrate the auricle over a frequency range causing the auricle to create an acoustic pressure wave at an entrance of the ear canal. The system then generates a design for a cartilage conduction audio device for the user based on the one or more identified locations of the transducers at which acoustic pressure waves generated by the one or more transducers satisfy a threshold performance metric for the user. The design may then be used to fabricate the cartilage conduction audio device.

Abstract: Embodiments relate to obtaining head-related transfer function (HRTF) through performing simulation using images of a user's head. The geometry of the user's head is determined based in part on one or more images of the user's head. The simulation of sound propagation from an audio source to the user's head is performed based on the generated geometry. The geometry may be represented in a three-dimensional meshes or principal component analysis (PCA)-based where the user's head is represented as a combination of representative three-dimensional shapes of test subjects' heads.

Abstract: An audio system for providing content to a user. The system includes a first and a second transducer assembly of a plurality of transducer assemblies, an acoustic sensor, and a controller. The first transducer assembly couples to a portion of an auricle of the user's ear and vibrates over a first range of frequencies based on a first set of audio instructions. The vibration causes the portion of the ear to create a first range of acoustic pressure waves. The second transducer assembly is configured to vibrate over a second range of frequencies to produce a second range of acoustic pressure waves based on a second set of audio instructions. The acoustic sensor detects acoustic pressure waves at an entrance of the ear. The controller generates the audio instructions based on audio content to be provided to the user and the detected acoustic pressure waves from the acoustic sensor.

Abstract: Embodiments relate to providing audio by focusing vibrations from an array of a plurality of bone conduction transducers to a cochlea of a user's ear. When bone conduction signals are received, bone conduction transducers of the array transmit vibrations to the cochlea of the user. A bone conduction signal generator generates the bone conduction signals, which may vary in amplitude and phase for different bone conduction transducers to amplify a level of vibrations at the cochlea of one ear while attenuating vibrations at another cochlea of another ear.

Abstract: A system including base stations determines head-related transfer functions (HRTFs) for a user. Each base station is located at a distinct location within a local area and includes a speaker configured to emit a test sound in accordance with calibration instructions. A depth camera assembly determines depth information describing a position of a head-mounted display (HMD) in the local area relative to the locations of the base stations. A microphone is placed in an ear canal of a user wearing the HMD, and generates a respective audio sample from the test sound emitted by the speaker of each base station. A controller determines the relative position of the HMD using the depth information, generates the calibration instructions based on the relative position of the HMD, and determines the HRTFs based on the audio samples.

Abstract: A listening device includes a reference microphone positioned outside a blocked ear canal of a user to receive environmental sounds and generate first signals based on the environmental sounds. A loudspeaker is coupled to the reference microphone and positioned inside the ear canal to generate internal sounds based on the first signals. An internal microphone is positioned inside the ear canal to receive the internal sounds from the loudspeaker and generate second signals based on the internal sounds. A controller is coupled to the internal microphone and the reference microphone to compute a transfer function based on the first signals and the second signals. The transfer function describes a variation between the environmental sounds and the internal sounds caused by the listening device blocking the ear canal. The controller adjusts, based on the transfer function, the internal sounds to mitigate the variation.

Abstract: A technique for auditory masking for a coherence-controlled calibration system includes generating a first auditory masking pattern based on a frequency spectrum of at least a first frame of a first audio signal. The technique continues by generating a first calibration signal based on the first auditory masking pattern. The technique then includes producing a combined input signal based on the first audio signal and the first calibration signal. The technique also includes performing one or more calibration operations based on the combined input signal.

Abstract: An audio system includes a transducer assembly, an audio sensor, and a controller. The transducer assembly is coupled to a back of an auricle of an ear of the user. The transducer assembly vibrates the auricle over a frequency range to cause the auricle to create an acoustic pressure wave in accordance with vibration instructions. The acoustic sensor detects the acoustic pressure wave at an entrance of the ear of the user. The controller dynamically adjusts a frequency response model based in part on the detected acoustic pressure wave, updates the vibration instructions using the adjusted frequency response model, and provides the updated vibration instructions to the transducer assembly.

Abstract: An audio system includes a transducer assembly, an audio sensor, and a controller. The transducer assembly is coupled to a back of an auricle of an ear of the user. The transducer assembly vibrates the auricle over a frequency range to cause the auricle to create an acoustic pressure wave in accordance with vibration instructions. The acoustic sensor detects the acoustic pressure wave at an entrance of the ear of the user. The controller dynamically adjusts a frequency response model based in part on the detected acoustic pressure wave, updates the vibration instructions using the adjusted frequency response model, and provides the updated vibration instructions to the transducer assembly.

Abstract: Embodiments relate to cancelling crosstalk vibrations due to the use of multiple bone conduction transducers by modifying bone conduction signals sent to the bone conduction transducers. The bone conduction signals are modified using an adaptive filter with its coefficients adjusted based on the vibrations generated at a bone conduction transducer and crosstalk vibrations sensed at a vibration sensor assembly. The bone conduction transducer for generating the calibration vibrations and the vibration sensor assembly for detecting the crosstalk vibrations may be placed at opposite sides of the user's head.

Abstract: An audio system includes a transducer assembly, an audio sensor, and a controller. The transducer assembly is coupled to a back of an auricle of an ear of the user. The transducer assembly vibrates the auricle over a frequency range to cause the auricle to create an acoustic pressure wave in accordance with vibration instructions. The acoustic sensor detects the acoustic pressure wave at an entrance of the ear of the user. The controller dynamically adjusts a frequency response model based in part on the detected acoustic pressure wave, updates the vibration instructions using the adjusted frequency response model, and provides the updated vibration instructions to the transducer assembly.

Abstract: Embodiments relate to crosstalk cancellation to reduce crosstalk vibrations during use of multiple bone conduction transducers. A system, for example, a head-mounted display (HMD) for providing audio to a user, uses vibration sensors to detect vibrations caused by the bone conduction transducers. In particular, a vibration sensor may generate an error signal representing residual vibrations due to crosstalk at a given ear region of the user. The system uses estimated transfer functions for noise propagation paths through the user's head to generate an anti-crosstalk signal. In response to the anti-crosstalk signal, a bone conduction transducer transmits anti-crosstalk vibrations that reduce the error signal at the given ear region.

Abstract: A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate to the speaker to provide an error signal indicative of the effectiveness of the noise cancellation. A secondary path estimating adaptive filter is used to estimate the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. A level of the source audio with respect to the ambient audio is determined to determine whether the system may generate erroneous anti-noise and/or become unstable.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: A speaker impedance may be determined by monitoring a voltage and/or current of the speaker. The calculated impedance may be used to determine whether the mobile device containing the speaker is on- or off-ear. The impedance determination may be assisted by applying a test tone low level signal to the speaker. The test tone may be inaudible to the user, but used to determine an impedance of the speaker at the frequency of the test tone. The impedance at that test tone may be used to determine whether a resonance frequency of the speaker is at a frequency corresponding to an on- or off-ear condition. The measured speaker impedance may be provided as feedback to an adaptive noise cancellation (ANC) algorithm to adjust the output at the speaker. For example, when the mobile device is removed from the user's ear, the ANC algorithm may be disabled.

Abstract: A speaker impedance may be determined by monitoring a voltage and/or current of the speaker. The calculated impedance may be used to determine whether the mobile device containing the speaker is on- or off-ear. The impedance determination may be assisted by applying a test tone low level signal to the speaker. The test tone may be inaudible to the user, but used to determine an impedance of the speaker at the frequency of the test tone. The impedance at that test tone may be used to determine whether a resonance frequency of the speaker is at a frequency corresponding to an on- or off-ear condition. The measured speaker impedance may be provided as feedback to an adaptive noise cancellation (ANC) algorithm to adjust the output at the speaker. For example, when the mobile device is removed from the user's ear, the ANC algorithm may be disabled.

Abstract: A speaker impedance may be determined by monitoring a voltage and/or current of the speaker. The calculated impedance may be used to determine whether the mobile device containing the speaker is on- or off-ear. The impedance determination may be assisted by applying a test tone low level signal to the speaker. The test tone may be inaudible to the user, but used to determine an impedance of the speaker at the frequency of the test tone. The impedance at that test tone may be used to determine whether a resonance frequency of the speaker is at a frequency corresponding to an on- or off-ear condition. The measured speaker impedance may be provided as feedback to an adaptive noise cancellation (ANC) algorithm to adjust the output at the speaker. For example, when the mobile device is removed from the user's ear, the ANC algorithm may be disabled.