Shure Acquisition Holdings, Inc.

Michael Ryan Lester

Wenshun Tian

Mathew T Abraham

Dusan S Veselinovic

Avinash K Vaidya

Michelle Michiko Ansai

David Grant Cason

Ryan Jerold Perkofski

Martin K Reiling

Justin Joseph Sconza

Ali Enteshari

Jose Roberto Regalbuto

Matthew Curtis Morrow

David Andrew VerLee

Michael Harrison Prosinski

Ross Lawrence Penniman

Brent Robert Shumard

John Casey Gibbs

Daniel Aaron Law

John Born

David Grundtvig

Gavin Baker

Iris Lorente

William Oakley

Christian Yost

Techniques for isolating audio signals related to audio sources within an audio environment are discussed herein. Examples may include receiving a plurality of audio data objects. Each audio data object includes digitized audio signals captured by a capture device positioned within an audio environment. Examples may also include inputting the audio data objects to a source localizer model that is configured to generate, based on the audio data objects, one or more audio source position estimate objects. Examples may also include inputting the audio data objects and each audio source position estimate object to a source generator model of one or more source generator models. The source generator model is configured to generate, based on the audio source position estimate object, a source isolated audio output component. The source isolated audio output component may include isolated audio signals associated with an audio source within the audio environment.

Audio signal isolation related to audio sources within an audio environment

Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.

Auto focus, auto focus within regions, and auto placement of beamformed microphone lobes with inhibition and voice activity detection functionality

An example immersive audio signal processing system and a computer-implemented method for generating a target arena environment audio stream are provided. The example immersive audio signal processing system includes a plurality of multi-lobe digital sound wave capture devices positioned within the arena environment. The plurality of multi-lobe digital sound wave capture devices is configured to direct first beamformed lobes to a playing region of the arena environment, second beamformed lobes to a spectator region of the arena environment, and third beamformed lobes to a noise source region of the arena environment. A digital signal processor is configured to isolate noise audio components originating from at least the spectator region or the noise source region from the audio signal stream and generate a target arena environment audio stream.

Multi-lobe digital microphone enabled audio capture and spatialization for generating an immersive arena based audio experience

Auto focus, auto focus within regions, and auto placement of beamformed microphone lobes with inhibition functionality

Audio beamforming systems and methods that enable more precise control of lobes and nulls of an array microphone are provided. Optimized beamformer coefficients can be generated to result in beamformed signals associated with one or more lobes steered towards one or more desired sound locations and one or more nulls steered towards one or more undesired sound location. The performance of acoustic echo cancellation can be improved and enhanced.

Audio beamforming with nulling control system and methods

Embodiments allow for an auto-mixer to gate microphones on and off based on speech detection, without losing or discarding the speech received during the speech recognition period. An example method includes receiving and storing an input audio signal. The method also includes determining, based on a first segment of the input audio signal, that the input audio signal comprises speech, and determining a delay between the input audio signal and a corresponding output audio signal provided to a speaker. The method also includes reducing the delay, wherein reducing the delay comprises removing one or more segments of the stored input audio signal to create a time-compressed audio signal and providing the time-compressed audio signal as the corresponding output audio signal. The method also includes determining that the delay is less than a threshold duration, and responsively providing the input audio signal as the corresponding output audio signal.

Systems and methods for intelligent voice activation for auto-mixing

Hybrid audio beamforming systems and methods with narrower beams and improved directivity are provided. The hybrid audio beamforming system includes a time domain beamformer for processing upper frequency band signals of an audio signal using a time domain beamforming technique, and a frequency domain beamformer for processing groups of lower frequency band signals of the audio signal using frequency domain beamforming techniques.

Hybrid audio beamforming system

Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Low latency automixer integrated with voice and noise activity detection

A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.

Time domain spectral bandwidth replication

Systems, apparatus, and methods for processing audio signals associated with conferencing devices communicatively connected in a daisy-chain configuration using local connection ports included on each device are provided. One method involving a first conferencing device comprises receiving auxiliary mixed microphone signal(s) from at least one other conferencing device via at least one local connection port, each auxiliary signal comprising a mix of microphone signals captured by the at least one other conferencing device; determining a gain adjustment value for each auxiliary mixed microphone signal based on a daisy-chain position of the at least one other conferencing device relative to the position of the first conferencing device; adjusting a gain value for each auxiliary mixed microphone signal based on the corresponding gain adjustment value; generating a loudspeaker output signal from the gain-adjusted auxiliary mixed microphone signal(s); and providing the loudspeaker signal to the loudspeaker of the first conferencing device.

System and method for distributed call processing and audio reinforcement in conferencing environments

An audio signal processor includes a digital filter block configured to receive an audio signal and output a first filtered audio signal, and a phase linearization block configured to receive the first filtered audio signal and output a second filtered audio signal with a more linear phase.

Anti-causal filter for audio signal processing

Systems and methods are provided for automatically adjusting a reference gain of an audio mixer having a reference channel for receiving a far end audio signal from a remote location as a reference signal and a plurality of audio input channels for receiving audio signals captured by a plurality of microphone element. An exemplary method includes determining an echo level in an input audio signal received at a given audio input channel, and automatically determining a gain amount for the reference channel based on the echo level. An exemplary system includes a reference gain adjuster configured to automatically determine a gain amount for the reference channel based on an echo level detected in an input audio signal received at a given audio input channel.

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Colorado Springs, CO, United States of America

Michael Ryan Lester