Abstract: An audio system including a first microphone-equipped playback device and a second microphone-equipped playback device. The audio system is configured to synchronize playing of audio to a listener position by receiving by the first microphone-equipped playback device an audio stream, and playing the audio stream on a speaker of the first microphone-equipped playback device, in accordance with a playback delay ?t. The playback delay ?t is in accordance with a first calibration sound originating at the listening position, a second calibration sound originating at the second microphone-equipped playback device, and a third calibration sound originating at the first microphone-equipped playback device.

Abstract: According to one embodiment, a failure detection apparatus includes processing circuitry. The processing circuitry acquires a time-series signal generated by a sensor module, generates an analysis result including information concerning saturation of the time-series signal by analyzing the time-series signal, and determine a failure of the sensor module based on the analysis result.

Abstract: An audio personalisation method for a user includes: testing the user with an audio test, the audio test comprising: moving a portable device, comprising a position tracking mechanism and a speaker, to a plurality of predetermined test positions relative to the user's head; when within a threshold distance of a predetermined test position, playing a test sound through the speaker of the portable device; and detecting the test sound using a microphone proximate to each of the user's ears, and associating resulting measurement data with the predetermined test position, where the resulting measurement data derived from the detected test sounds is characteristic of the user's head morphology; for a corpus of reference individuals for whom respective HRTFs have been generated, comparing the measurement data from the user's audio test or a head-related transfer function ‘HRTF’ derived from this measurement data with corresponding measurement data of some or all respective reference individuals in the corpus or HRTF

Abstract: The disclosure provides a method and a host for adjusting audio of speakers, and a computer readable medium. The method includes: controlling a far-field speaker to play a first audio signal; controlling an audio receiver to receive the first audio signal from the far-field speaker and accordingly positioning a speaker location of the far-field speaker; establishing a first hearing transfer function related to the far-filed speaker based on the speaker location of the far-filed speaker; controlling the far-field speaker to play a second audio signal based on a second hearing transfer function; controlling the audio receiver to receive the second audio signal and accordingly estimating a first reference hearing volume; obtaining a second reference hearing volume corresponding to a first near-field speaker; and adjusting a first volume of the far-field speaker based on the first reference hearing volume and the second reference hearing volume.

Abstract: Synchronized output of audio on a group of devices can comprise sending audio data from an audio distribution master device to one or more slave devices in the group. Scores can be assigned to respective audio playback devices, the scores being indicative of a performance level of the respective audio playback devices acting as a master device. The device with the highest score is designated as a candidate master device and one or more remaining devices are designated as a candidate slave(s). A throughput test is conducted with the highest scoring device acting as the candidate master device. The results of the throughput test are used to determine a master device for a group of devices. Latency of the throughput test can be reduced by using a prescribed time period for completion of the throughput test, and/or by selecting a first group configuration to passes the throughput test.

Abstract: Embodiments include an array microphone comprising a plurality of microphone sets arranged in a linear pattern relative to a first axis and configured to cover a plurality of frequency bands. Each microphone set comprises a first microphone arranged along the first axis and a second microphone arranged along a second axis orthogonal to the first microphone, wherein a distance between adjacent microphones along the first axis is selected from a first group consisting of whole number multiples of a first value, and within each element, a distance between the first and second microphones along the second axis is selected from a second group consisting of whole number multiples of a second value.

Abstract: An example process may include identifying an area of an environment to emit one or more sound masking signals, determining a noise level of one or more surrounding areas adjacent to the area, selecting one or more frequency ranges and respective amplitudes of the one or more sound masking signals to apply to the area based on the noise level, and emitting the masking signals via one or more emitters disposed in the area.

Abstract: A hearing device includes: a first housing; a second housing; first and second primary microphones for provision of first and second primary microphone input signals; a secondary microphone in the second housing for provision of a secondary microphone input signal; a mixing module for provision of a mixer output based on a primary mixer input and a secondary mixer input, wherein the primary mixer input is based on the first primary microphone input signal and the second primary microphone input signal, and the secondary mixer input is based on the secondary microphone input signal; a processing unit configured to process the mixer output and to provide an electrical output signal; and a receiver for converting the electrical output signal to an audio output signal; wherein the mixing module is configured to mix a first component of the primary mixer input and a first component of the secondary mixer input.

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: A method for directional signal processing for an acoustic system, wherein first and second input transducers generate a first and a second input signal from an ambient sound. First and second intermediate signals are generated from the first and second input signal, wherein a preliminary superposition parameter is obtained for a first superposition of the first intermediate signal and the second intermediate signal in such a way that for the first superposition an attenuation in a first target direction has a maximum. A superposition parameter is formed from the preliminary superposition parameter so that a second superposition of the first and second intermediate signals, formed in the first target direction with the superposition parameter, has a pre-specified first value for a gain that is greater than zero. An output signal of the acoustic system is formed from the second superposition.

Abstract: The present disclosure provides an acoustic device including a microphone array, a processor, and at least one speaker. The microphone array may be configured to acquire an environmental noise. The processor may be configured to estimate a sound field at a target spatial position using the microphone array. The target spatial position may be closer to an ear canal of a user than each microphone in the microphone array. The processor may be configured to generate a noise reduction signal based on the environmental noise and the sound field estimation of the target spatial position. The at least one speaker may be configured to output a target signal based on the noise reduction signal. The target signal may be used to reduce the environmental noise. The microphone array may be arranged in a target area to minimize an interference signal from the at least one speaker to the microphone array.

Abstract: A wearable device can provide an audio module that is operable to provide audio output from a distance away from the ears of the user. For example, the wearable device can be worn on clothing of the user and direct audio waves to the ears of the user. Such audio waves can be focused by a parametric array of speakers that limit audibility by others. Thus, the privacy of the audio directed to the user can be maintained without requiring the user to wear audio headsets on, over, or in the ears of the user. The wearable device can further include microphones and/or connections to other devices that facilitate calibration of the audio module of the wearable device. The wearable device can further include user sensors that are configured to detect, measure, and/or track one or more properties of the user.

Abstract: A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal.

Abstract: A portable calibration system is disclosed that calibrates an audio equipment without using a dedicated sound level meter. The calibration system comprises a coupler configured to couple a transducer to an energy sensor, where an output of the transducer is provided to the energy sensor via the coupler, an analyzer module configured to receive information from the energy sensor regarding the output of the transducer, a processor, in the analyzer module, configured to process the information to provide a result of a calibration for the audio equipment with respect to expected results, and a display configured to display the result of the calibration.

Abstract: A system includes a microphone unit coupled to a roof of an autonomous vehicle. The microphone unit includes a microphone board having a first opening. The microphone unit also includes a first microphone positioned over the first opening and coupled to the microphone board. The microphone unit further includes an accelerometer. The system also includes a processor coupled to the microphone unit.

Abstract: According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

Abstract: Provided is a semiconductor device that does not require an A/D converter circuit and is capable of performing sound source separation without converting an analog signal. The semiconductor device includes a plurality of microphones, a plurality of delay circuits, and a signal processing circuit. The plurality of microphones are electrically connected to the respective delay circuits, and output signals of the plurality of delay circuits are input to the signal processing circuit. The delay circuit includes a plurality of signal retention circuits capable of retaining an analog potential, and has functions of retaining an electric signal output from the microphone as a discrete analog signal in the signal retention circuits and outputting the signal at a time different from the time when the microphone outputs the signal. The signal processing circuit has a function of adding the output signals of the plurality of delay circuits.

Abstract: A headset implements an earloop microphone and includes a housing. An earloop of the headset secures the headset to an ear of a user. A first microphone is acoustically coupled to a first opening in the housing. A second microphone is acoustically coupled to a second opening in the housing. A third microphone is acoustically coupled to a third opening in the earloop.

Abstract: An image capture device, having: a housing, a lens snout, a front microphone, a top microphone, and an audio processor. The housing has a top and front housing surface. The lens snout protrudes from the front housing surface. The front microphone mounted within or on the front housing surface and below the lens snout. The top microphone mounted within or on a top housing surface in a position biased toward the front housing surface. The audio processor comprises a memory that is configured to store instructions that when executed cause the audio processor to generate an output audio signal. The top microphone is located at a position to receive direct freestream air flow when the housing is positioned in a pitched forward orientation at a pitched forward angle relative to a vertical axis. The front microphone receives turbulent air flow from the lens snout when the housing is positioned in the pitched forward orientation.

Abstract: An electronic device localizes an audio source by normalizing an amplitude of an audio signal over a time period. The electronic device receives, from one or more microphones of the electronic device, signal(s) representative of audio emitted by an audio source over a time period. The electronic device estimates amplitudes of the signal(s) at a first time within the time period and at a second time within the time period, where the second time is different from the first time. The electronic device normalizes the amplitudes associated with the first and second times to generate normalized amplitudes. The electronic device determines a combined amplitude representative of the audio emitted by the audio source by combining the normalized amplitudes. The electronic device determines, based at least in part on the combined amplitude and motion of the electronic device, an estimated position of the audio source relative to the electronic device.