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: 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 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: 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: 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: 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: 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: 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 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.

Abstract: An acoustophoretic system is controlled and driven to attain a desired level of performance. An RF controller and a driver provide a frequency and power to an acoustic transducer, which can be implemented as a piezoelectric element, which presents a reactive load or a complex load. A controller implements a control technique for efficient transducer operation. The control technique can locate a frequency for operation that is at a reactance minimum or maximum for the system to produce a modal pattern and to provide efficient operation of the transducer. A method of detecting a minimum or maximum reactance in a acoustophoretic system used to trap, separate, deflect, cluster, fractionate or otherwise process particles or secondary fluids or tertiary fluids in a primary fluid and utilizing the frequency of the detected reactance to operate the acoustophoretic system.

Abstract: A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal.

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: A dual-microphone arrangement (300) provides improve voice performance in a wireless headset (12). A vibration sensor (1130) is used for voice pickup and will add low-frequency voice audio content in windy conditions. An equalizer (810) is used to restore low-frequency voice audio content in wind-free conditions. Depending on the measured wind power, the output will derive more signal from the equalizer (810) or more signal from the vibration sensor (1130).

Abstract: An audio system has an ambient sound enhancement (ASE) function, in which an against-the-ear audio device having a speaker converts a digitally processed version of an input audio signal into amplified sound. The amplification may be in accordance with a stored hearing profile of the user. The audio system also has an acoustic noise cancellation (ANC) function that may be combined in various ways with the ASE function, and that may be responsive to voice activity detection. Other aspects are also described and claimed.

Abstract: Apparatus and methods for bone conduction detection are disclosed herein. An example apparatus includes a first sensor to output a first vibration signal associated with a first bone structure of a user, a second sensor to output a second vibration signal associated with a second bone structure of the user, memory, and at least one processor to execute instructions to determine a sound associated with at least one of the first vibration signal or the second vibration signal originated external to the user; determine a direction from which the sound originated relative to the user; and generate a signal to cause an output device to present an alert to the user, the alert indicative of the direction of the sound.

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: Techniques for estimating a direction of arrival of sound in an environment are discussed. First and second audio data are received from a first pair of audio sensors associated with a vehicle. A first region of ambiguity associated with the first pair of audio sensors is determined based on the first and second audio data. Third and fourth audio data are received from a second pair of audio sensors. A second region of ambiguity associated with the second pair of audio sensors is determined based on the third and fourth audio data. The regions of ambiguity can be further based on confidence levels associated with sensor or audio data. An area of intersection of the first region of ambiguity and the second region of ambiguity can be determined. A direction of arrival of an audio event can be determined based on the area of intersection.

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.