Abstract: A system and method for determining an in-ear status of wearable audio devices includes determining that the devices are each proximate a portion of a body of a user. Each device determines its orientation; if both orientations correspond to ears of the user, the devices output corresponding audio.

Abstract: A system includes multiple microphone arrays positioned at different locations on a roof of an autonomous vehicle. Each microphone array includes two or more microphones. Internal clocks of each microphone array are synchronized by a processor and used to generate timestamps indicating when microphones capture a sound. Based on the timestamps, the processor is configured to localize a source of the sound.

Abstract: A computerized microphone-equipped audio playback device comprising a speaker and microphone and being configured to receive digital audio; and play the digital audio on a speaker, in accordance with a playback delay that is derivative of, at least: an arrival time of a first calibration sound at the processor and an arrival time of the first calibration sound at a second microphone-equipped audio playback device, wherein the first calibration sound originated at the listener position; a generation time of a second calibration sound at the second microphone-equipped audio playback device, and an arrival time of the second calibration sound at the processor; and a generation time of a third calibration sound at the processor, and an arrival time of the third calibration sound at the second microphone-equipped audio playback device; thereby synchronizing arrival of sound of the two playback devices at the listener position.

Abstract: A microelectromechanical system (MEMS) microphone includes a cavity to receive an acoustic signal. The acoustic signal causes movement of a diaphragm relative to one or more other surfaces, which in turn results in an electrical signal representative of the received acoustic signal. A light sensor is included within the packaging of the MEMS microphone such that an output of the light sensor is representative of a light signal received with the acoustic signal. The output of the light sensor is used to modify the electrical signal representative of the received acoustic signal in a manner that limits light interference with an acoustical output signal.

Abstract: The disclosed technology generally relates to a microphone device configured to receive sound from different beams, where each beam has a different spatial orientation and is configured to receive sound from different directions. The microphone device is also configured to process the received sound using a generic or specific head related transfer function (HRTF) to generate processed audio and transmit the processed audio to hearing devices worn by a hearing-impaired user. Additionally, the microphone device can use a reference line and/or reference point when processing the received audio.

Abstract: A sensor system includes a triplet element including a first hydrophone, a second hydrophone, and a third hydrophone configured to receive an incoming signal at a first phase, a second phase, and a third phase, respectively, the first to third hydrophones extending along a first direction, and a processor configured to determine an incidence direction of the incoming signal, and to dynamically generate a cardioid null in the incidence direction to reject the incoming signal based on the incoming signal at the first to third phases.

Abstract: Embodiments disclosed herein include networked microphone devices (NMD) determining whether a Do Not Disturb (DND) feature should be activated, in response to determining that the DND feature should be activated, activating the DND feature. In some embodiments, the NMD determines whether to activate the DND feature based on various configuration and operational states. And in some embodiments, activating the DND feature includes activating the DND feature includes activating the DND feature at one or more additional NMDs based on the configuration and operational state of the NMD and the one or more additional NMDs.

Abstract: A steerable speaker array is provided, comprising a plurality of drivers arranged in a concentric, nested configuration formed by arranging the drivers in a plurality of concentric groups and placing the groups at different radial distances from a central point of the configuration. Each group is formed by a subset of the plurality of drivers being positioned at predetermined intervals from each other along a perimeter of the group. Also, the concentric groups are harmonically nested and rotationally offset from each other. An audio system is also provided comprising at least one steerable speaker array and a beamforming system configured to receive one or more input audio signals from an audio source, generate a separate audio output signal for each driver of the speaker array based on at least one of the input signals, and provide the audio output signals to the corresponding drivers to produce a beamformed audio output.

Abstract: A vehicle sound synthesis system is provided with a loudspeaker, a microphone and a controller. The loudspeaker is adapted to project sound indicative of synthesized engine noise (SEN) within a cabin of a vehicle in response to receiving an adjusted SEN signal. The microphone is disposed in the cabin and adapted to provide a microphone signal having a SEN component and a noise component. The controller is programmed to receive the microphone signal, adjust a SEN signal based on the microphone signal, and provide the adjusted SEN signal to the loudspeaker.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer and a parameter determinator. The upmixer is configured to apply temporally variable upmix parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal, wherein the temporally variable upmix parameters include temporally variable smoothened phase values. The parameter determinator is configured to obtain one or more temporally smoothened upmix parameters for usage by the upmixer on the basis of a quantized upmix parameter input information. The parameter determinator is configured to combine a scaled version of a previous smoothened phase value with a scaled version of an input phase information using a phase change limitation algorithm, to determine a current smoothened phase value on the basis of the previous smoothened phase value and the phase input information.

Abstract: A vehicle is provided to minimize the difference in left and right output intensity of a headrest speaker by adjusting an output intensity of based on the position of the passenger's ear. The headrest includes a left-side speaker provided on the left-side of the headrest of a seat inside the vehicle to output sound and a right-side speaker provided on the right-side of the headrest to output sound. A pressure sensor measures a force applied to the headrest and a controller adjusts the output intensity of the left speaker and the right speaker based on an action point of the force measured by the pressure sensor.

Abstract: A computer-assisted method for spatialized sound reproduction based on an array of loudspeakers, for the purpose of producing a selected sound field at a position of a listener. The method includes iteratively and continuously: obtaining a current position of a listener; determining respective acoustic transfer functions of the loudspeakers at a virtual microphone of which the position is defined dynamically as a function of the current position of the listener, estimating a sound pressure at the virtual microphone; calculating an error between the estimated sound pressure and a target sound pressure; calculating and applying respective weights to the control signals of the loudspeakers as a function of the error and of a weight forgetting factor, the forgetting factor being calculated as a function of a movement of the listener, and calculating the sound pressure at the current position of the listener.

Abstract: A method for detecting blocking of a microphone and related products are provided. The method includes the following. Voice data is collected through a microphone of the first wireless earphone. Whether the voice data has a missing voice segment is detected. The microphone (e.g., microphone-hole) of the first wireless earphone is determined to be blocked in response to detecting that the voice data has the missing voice segment.

Abstract: A method for detecting blocking of a microphone and related products are provided. The method includes the following. A first electric quantity of a first wireless earphone and a second electric quantity of a second wireless earphone are determined when both the first wireless earphone and the second wireless earphone are worn. A first operation and a second operation are performed in parallel when the first electric quantity is greater than the second electric quantity, where the first operation is to obtain first audio through the first microphone, and the second operation is to obtain second audio through the second microphone. A reference parameter of the second audio is determined. The first wireless earphone is determined to be blocked in response to detecting that a first parameter of the first audio does not match the reference parameter.

Abstract: Systems, methods, and computer-readable media are provided for enhancing a user's listening experience by adjusting physical attributes of an audio playback system based on detected environmental attributes of the system's environment.

Abstract: There is disclosed a device that can record a sound by selecting a directivity direction as desired by the user at the time of recording and can play back the sound by selecting, at the time of playback, a desired directivity direction by the user, separate from the directivity direction specified at the time of recording. The device has microphones, a touch panel for specifying a directivity direction, and a signal processing unit. A recording module performs directivity control in a direction specified on the touch panel and records audio data subjected to effect processing in a memory as directivity-controlled data, and records audio data that are not subjected to directivity control or effect processing in the memory as directivity-control unprocessed data.

Abstract: For certain application cases, such as e.g., in a sports stadium, a microphone array having a particularly high directivity in the vertical direction and a high, yet in wide limits adjustable directivity in horizontal direction is provided. The microphone array has a plurality of microphones whose output signals are combined into at least one common output signal. The microphones are directional microphones with a preferred direction of high sensitivity and arranged substantially in one plane on a circle or segment of a circle, such that each microphone has a different direction of high directivity. For each of the microphones, the preferred direction of high sensitivity is substantially orthogonal to the circle or segment of the circle. A common output signal of the microphone array is obtained by beamforming. The microphone array has an adjustable preferred direction of high sensitivity, wherein the common output signal comprises the sound recorded from this adjustable direction.

Abstract: An engine harmonic cancellation system includes an accelerometer disposed within a vehicle to detect a harmonic produced by an engine of the vehicle and to produce a harmonic reference signal representative of the harmonic; a controller configured to produce a harmonic cancellation signal that, when transduced into an acoustic signal, cancels the harmonic within at least one cancellation zone within a cabin of the vehicle, wherein the harmonic cancellation signal is based, at least in part, on mixing the harmonic reference signal converted to baseband with a baseband signal output from a look up table; and a speaker disposed within the cabin and configured to receive the harmonic cancellation signal and to transduce the harmonic cancellation signal into an acoustic harmonic cancellation signal, such that the harmonic is cancelled within the cancellation zone.

Abstract: An image capture device with beamforming for wind noise optimized microphone placements is described. The image capture device includes a front facing microphone configured to capture an audio signal. The front facing microphone co-located with at least one optical component. The image capture device further includes at least one non-front facing microphone configured to capture an audio signal. The image capture device further includes a processor configured to generate a forward facing beam using the audio signal captured by the front facing microphone and the audio signal captured by the at least one non-front facing microphone, generate an omni beam using the audio signal captured by the at least one non-front facing microphone, and output an audio signal based on the forward facing beam and the omni beam.

Abstract: A method of determining a position of a sound source is provided which comprises generating a spatially encoded sound-field signal using a sound-field microphone system comprising at least two microphones, wherein the spatially encoded sound-field signal comprises a plurality of components, each component including sound from the sound source. The method further comprises generating a local microphone signal corresponding to sound from the sound source using a local microphone positioned close to the sound source, comparing the local microphone signal with each of the plurality of components to generate a plurality of comparison results and using the plurality of comparison results to determine the position of the sound source relative to the sound-field microphone system.