A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

A system and method for differentially locating and modifying audio sources that includes receiving multiple audio inputs from a set of distinct locations; determining a multi-dimensional audio map from the audio inputs; acquiring a set of positional audio control inputs applied to the audio map, each audio control input comprising a location and audio processing property; and generating an audio output according to the audio control inputs and the audio inputs. The audio control inputs capable of configuration through manual, automatic, computer vision analysis, and other configuration modes.

A method includes: receiving multi-channel sound source signals from a microphone array; synchronizing the multi-channel sound source signals based on spatial information of the microphone array; and detecting an abnormal channel of the microphone array by inputting the synchronized sound source signals and first conditional information to a neural network model configured to perform an inverse operation.

A system is described herein. The system includes at least one hardware processor that is configured to identify a pre-determined acoustic barrier filter, wherein the acoustic barrier filter coincides with the physical acoustic barrier and receive an audio signal within a time window at the first microphone and the second microphone. The hardware processor is also configured to calculate a first measure of variability, a second measure of variability, a third measure of variability, and a fourth measure of variability. The hardware processor further concatenates the first measure of variability, the second measure of variability, the third measure of variability, and the fourth measure of variability to form a feature vector, and inputs the feature vector into a location classifier to obtain an audio source location.

At least a system or a method is provided for remote delivery or collection of a device such as an audio collection device. For example, a device comprising an aperture collection and retrieval pin is provided. An apparatus is provided having an aperture receiver, an aperture drive gear and a drive motor. The drive motor is configured to drive the aperture drive gear to open or close the aperture receiver of the apparatus for retrieving or releasing the device comprising the aperture collection pin.

An audio capture device selects between multiple microphones to generate an output audio signal depending on detected conditions. The audio capture device determines whether one or more microphones are wet or dry and selects one or more audio signals from the one or more microphones depending on their respective conditions. The audio capture device generates a mono audio output signal or a stereo output signal depending on the respective conditions of the one or more microphones.

An audio signal processing device includes a sound acquisition unit configured to acquire audio data generated by collecting a sound in a sound collection target space, a selection unit configured to select, based on a priority of each of a plurality of areas in the sound collection target space, one or more of the areas in the sound collection target space, and an output unit configured to output processed data, for which predetermined signal processing for the areas selected by the selection unit is executed on the audio data acquired by the sound acquisition unit, and the predetermined signal processing for an area not selected by the selection unit is not executed on the audio data.

Method, apparatus, and computer-readable media to manage undesired sound sources in microphone pickup/focus zones preferably mitigates one or more of the undesired sound source(s) in a space having a plurality of microphones and at least one desired sound source. Preferably, at least one microphone input receives plural microphone input signals from the plurality of microphones in the space. Preferably, the least one processor is coupled to the at least one microphone input and receives the plural microphone input signals. Preferably, the at least one processor determines plural micro-zones in the space. Preferably, the at least one processor determines a threshold sound field level for each micro-zone based on received plural microphone input signals that correspond to the one or more undesired sound source(s). Preferably, the at least one processor recognizes a desired sound source when received plural microphone input signals exceed one or more threshold sound field level.

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.

One example discloses an apparatus for voice communication, including: a first wireless device including a first pressure sensor having a first acoustical profile and configured to capture a first set of acoustic energy within a time window; wherein the first wireless device includes a near-field magnetic induction (NFMI) signal input; wherein the first wireless device includes a processing element configured to: receive, through the NFMI signal input, a second set of acoustic energy captured by a second pressure sensor, having a second acoustical profile, within a second wireless device and within the time window; apply a signal enhancement technique to the first and second sets of acoustic energy based on the first and second acoustical profiles; and output an enhanced voice signal based on applying the signal enhancement.