Abstract: Audio-visual systems and methods are configured to determine a first talker location based on a first group of sound locations corresponding to audio detected by the microphone in association with one or more talkers; receive a new sound location for new audio detected by the microphone in association with at least one talker; determine a proximity of the new sound location to the first group of sound locations; based on the new sound location being in close proximity to one or more of the sound locations in the first group, determine a second talker location based on the new sound location and the first group of sound locations; determine a second proximity of the second talker location to the first talker location; provide the second talker location to the camera if the second proximity meets or exceeds a threshold; and otherwise, provide the first talker location the camera.

Abstract: Conferencing systems and methods configured to generate talker coordinates for directing a camera towards talker locations in an environment are disclosed, as well as talker tracking using multiple microphones and multiple cameras. One method includes determining, using a first microphone array and based on audio associated with a talker, a first talker location in a first coordinate system relative to the first microphone array; determining, using a second microphone array and based on the audio associated with the talker, a second talker location in a second coordinate system relative to the second microphone array; determining, based on the first talker location and the second talker location, an estimated talker location in a third coordinate system relative to a camera; and transmitting, to the camera, the estimated talker location in the third coordinate system to cause the camera to point the camera towards the estimated talker location.

Abstract: Various embodiments of the present disclosure provide methods, apparatus, systems, devices, and/or the like for inferring characteristics of a physical enclosure using a plurality of audio signals. The plurality of audio signals may be processed using a feature extraction framework to generate structured audio event data sets, which may be processed using an audio event framework to determine the characteristics of the physical enclosure.

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: Embodiments include a method of reducing echo in an audio system comprising a microphone, an acoustic echo canceller (AEC), and at least one processor, the method comprising receiving, by the at least one processor, an audio signal detected by the microphone; deploying, by the at least one processor, a microphone lobe towards a first location associated with the detected audio signal; obtaining, by the at least one processor, one or more AEC parameters for the first location, the one or more AEC parameters being stored in a memory in communication with the at least one processor; initializing, by the at least one processor, the AEC using the one or more AEC parameters; and generating, by the at least one processor, an echo-cancelled output signal using the initialized AEC and based on the detected audio signal and a reference signal provided to the AEC.

Abstract: Embodiments include an audio system comprising a plurality of microphones disposed in an environment, wherein the plurality of microphones is configured to detect one or more audio sources, and generate location data indicating a location of each of the one or more audio sources relative to the plurality of microphones; and at least one processor communicatively coupled to the plurality of microphones, wherein the at least one processor is configured to receive the location data from the plurality of microphones, and define a plurality of audio pick-up regions in the environment based on the location data, the plurality of audio pick-up regions comprising a first audio pick-up region and a second audio pick-up region, wherein the plurality of microphones are configured to deploy a first lobe within the first audio pick-up region and a second lobe within the second audio pick-up region.

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: 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: 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: 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: A system and a method perform frame interpolation for a compressed video bitstream. The system and the method may combine candidate pictures to generate an interpolated video picture inserted between two original video pictures. The system and the method may generate the candidate pictures from different motion fields. The candidate pictures may be generated partially or wholly from motion vectors extracted from the compressed video bitstream. The system and the method may reduce computation required for interpolation of video frames without a negative impact on visual quality of a video sequence.

Abstract: A system and method for efficiently upsampling and filtering a signal is provided. The system includes a controller 216 and an upsampler 202 capable of upsampling a digitized signal. The upsampler 202 converts a digitized signal having a first sampled frequency into a digitized signal having a second sampled frequency by inserting, for example, at least one zero between the samples of the digitized signal. A plurality of filters 205-208 is coupled to the upsampler. These filters, which may be band-pass filters, would be suitable for front end encoding in audio-based applications. The controller 216 adjusts at least one output of the plurality of the filters to zero, either by deactivation or forcing the filter to zero, based upon a predetermined characteristic of the digitized signal. One example of a predetermined characteristic is a maximum frequency of interest.