Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: Systems and methods are disclosed for operating a wireless audio network including a plurality of wireless microphone units (e.g., wireless delegate units) and a central access point having a mixer. The wireless microphone units may perform voice detection and level sensing, and make a preliminary gating decision. The central access point may make a final gating decision, determine the granting of wireless communications channels, and generate a final mixed audio output signal.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.