Described are systems and methods that utilize nodes distributed at different geographic locations to detect and track the approximate position, trajectory, and/or predicted path of aerial vehicles operating below a defined altitude (e.g., 500 feet). As nodes detect an aerial vehicle, the node determines a bearing toward the aerial vehicle and provides the bearing to an air-traffic system. The air-traffic system processes bearings received from each node and determines one or more of an approximate position, trajectory, and/or predicted path of the detected aerial vehicle. The approximate position, trajectory, and/or predicted path may be provided to one or more subscribing clients and/or used to alter paths of one or more aerial vehicles.

An apparatus and method for monitoring audio output is disclosed. The apparatus may comprise means for providing one or more primary audio signals based on signals from one or more first microphones associated with an audio capture device and providing one or more secondary audio signals based on signals from one or more second microphones associated with an audio monitoring device, the audio monitoring device being separate from the audio capture device and configured for output of the one or more primary audio signals and the one or more secondary audio signals through one or more loudspeakers. The apparatus may comprise means for modifying one or both of the primary and secondary audio signals such that output of the one or more primary audio signals are distinguished over output of the one or more secondary audio signals.

Systems and methods are described for generating a transcript of a legal proceeding or other multi-speaker conversation or performance in real time or near-real time using multi-channel audio capture. Different speakers or participants in a conversation may each be assigned a separate microphone that is placed in proximity to the given speaker, where each audio channel includes audio captured by a different microphone. Filters may be applied to isolate each channel to include speech utterances of a different speaker, and these filtered channels of audio data may then be processed in parallel to generate speech-to-text results that are interleaved to form a generated transcript.

A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

This disclosure describes techniques that include aligning processing of audio samples collected by multiple audio sensors or microphones. In one example, this disclosure describes a method comprising detecting a transition by the second microphone from a disabled state to an enabled state; after detecting the transition, performing phase alignment between audio samples collected by the first microphone and audio samples collected by the second microphone by introducing a delay in starting processing of the audio samples collected by the second microphone; and processing the phase-aligned audio samples.

A device to process speech includes a speech processing network that includes an input configured to receive audio data corresponding to audio captured by one or more microphones. The speech processing network also includes one or more network layers configured to process the audio data to generate a network output. The speech processing network includes an output configured to be coupled to multiple speech application modules to enable the network output to be provided as a common input to each of the multiple speech application modules. A first speech application module corresponds to a speaker verifier, and a second speech application module corresponds to a speech recognition network.

A method, computer program product, and computing system for receiving a signal from each microphone of a plurality of microphones, thus defining a plurality of signals. One or more inter-microphone gain-based augmentations may be performed on the plurality of signals, thus defining one or more inter-microphone gain-augmented signals.

The invention relates to a variable-directivity MEMS microphone. The microphone comprises an acoustic cavity. The following components are provided inside the acoustic cavity: a first acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a first acoustic conversion signal; a first pre-amplifier, connected to the first acoustic transducer, and configured for outputting a first electric signal; a second acoustic transducer for detecting an acoustic signal and converting the acoustic signal into a second acoustic conversion signal; a second pre-amplifier, connected to the second acoustic transducer, and configured for outputting a second electric signal; and a signal processing chip, connected to the first pre-amplifier and the second pre-amplifier, and configured for generating a directional output signal by performing an arithmetic operation on the first electric signal and the second electric signal under the action of a switching control signal.

An audio communication device includes: a sound position determiner that determines sound localization positions for N audio signals in a virtual space having first and second walls; N sound localizers each performing sound localization processing to localize sound in the sound localization position determined by the sound position determiner, and outputting localized sound signals; an adder that sums the N localized sound signals, and outputs a summed localized sound signal. Each sound localizer performs the processing using: a first head-related transfer function (HRTF) assuming that a sound wave emitted from the sound localization position of the sound localizer determined by the sound position determiner directly reaches each ear of a hearer virtually present at the hearer position; and a second HRTF assuming that the sound wave emitted from the sound localization position reaches each ear of the hearer after being reflected by closer one of the first and second walls.

The present application discloses a Method for converting vibration to voice frequency wirelessly and a method thereof. By sensing a first vibration variation data and a voice frequency variation data of a vocal vibration part in a first sensing period, a voice frequency reference data is obtained from the voice frequency variation data and the first vibration result. A second vibration result is obtained at a second sensing period for converting to a voice frequency output signal, and the voice frequency output signal is used to output as a voice signal corresponding to the voice frequency various result. Thus, the present application provides a voice signal close to a human voice.