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 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.

A digital stethoscope includes a stethoscope housing defining a housing edge. The digital stethoscope also includes a surface region secured to the stethoscope housing at the housing edge, and a number of microphones. The digital stethoscope also includes a processing device disposed within the stethoscope housing and in communication with the microphones. The processing device receives the digital audio data from the microphones.

Active noise cancellation systems, components, and methods are provided with single-source forward cancellation using a direction-dependent filter response. One illustrative active sound cancelling device includes: a primary external microphone that produces a primary receive signal; a secondary external microphone that produces a secondary receive signal, the primary and secondary receive signals representing ambient audio that potentially includes sound having a predominate direction of arrival; a speaker that converts an output signal into internal audio to at least partly cancel said sound, the output signal including a forward cancellation signal; a forward filter that operates solely on the primary receive signal to produce the forward cancellation signal; and a direction finder that operates on the primary and secondary receive signals to derive an estimate of said predominate direction of arrival, the direction finder adjusting the forward filter to implement a filter response corresponding to said estimate.

Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.

Systems and methods for optimizing network microphone devices using noise classification are disclosed herein. In one example, individual microphones of a network microphone device (NMD) detect sound. The sound data is analyzed to detect a trigger event such as a wake word. Metadata associated with the sound data is captured in a lookback buffer of the NMD. After detecting the trigger event, the metadata is analyzed to classify noise in the sound data. Based on the classified noise, at least one performance parameter of the NMD is modified.