Abstract: This disclosure describes a method for video synchronization in video distribution systems that includes a system that re-synchronizes the video playback of a video synchronization device with that of a host device after any change by the host, interruption within the host, or interruption of data flow between a video source device and a video display device.

Abstract: This disclosure describes a method to manufacture a conferencing apparatus for echo cancelation with beamforming microphone arrays. This method provides a beamforming microphone array for developing a plurality of microphone signals, where each microphone of the beamforming microphone array is configured to sense acoustic waves from a direction vector substantially different from other microphones in the beamforming microphone array. Additionally provided is a memory configured for storing computing instructions.

Abstract: This disclosure describes a conferencing apparatus with an automatically adapting beamforming microphone array for teleconferencing applications. The conferencing apparatus includes a processor coupled to memory, storage, and a housing. The apparatus further includes a communication element, a plurality of microphones arranged in the housing and configured as a beamforming microphone array, and an orientation sensor. Additionally, the apparatus includes executing a series of steps including: (a) adapting a signal-processing characteristic of the beamforming microphone array responsive to the orientation signal, (b) beamforming to combine the plurality of microphone signals to a plurality of combined signals, (c) performing an acoustic echo cancellation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals, and (d) selecting one or more of the plurality of combined echo-canceled signals for transmission.

Abstract: The present disclosure discloses a conferencing apparatus that combines a beamforming microphone array (BMA) with an acoustic echo canceller where the BMA further comprises a plurality of microphones that are oriented to develop a corresponding plurality of microphone signals. The apparatus further includes a processor, memory, and storage where the processor is configured to execute program instructions. The processor performs a beamforming operation to create a plurality of combined signals. In addition, the processor performs an acoustic echo cancellation operation to generate a plurality of combined echo cancelled signals. Further, the processor selects one of the combined echo cancelled signals for transmission to the far end where the signal selection module uses the far end signal as information to inhibit the signal selection module from changing the selection of the combined echo cancelled signals while only the far end signal is active.

Abstract: The present disclosure describes a system using an audio mixer for managing endpoint parameters of an input audio device and an output audio device. The system includes a data terminal, multiple endpoints, and the audio mixer, where the data terminal provides multiple data files including configuration parameters of endpoints. The endpoints include a first endpoint and a second endpoint. The audio mixer is configured to receive the data files from the data terminal, process the data files to obtain a first set of configuration parameters corresponding to the first endpoint and a second set of configuration parameters corresponding to the second endpoint, convert the first set and the second set of configuration parameters into a predefined common data format, and organize the predefined common data structure into a generic data structure.

Abstract: This disclosure describes a method to manufacture a conferencing apparatus for echo cancelation with beamforming microphone arrays. This method provides a beamforming microphone array for developing a plurality of microphone signals, where each microphone of the beamforming microphone array is configured to sense acoustic waves from a direction vector substantially different from other microphones in the beamforming microphone array. Additionally provided is a memory configured for storing computing instructions.

Abstract: This disclosure describes a band-limited beamforming microphone array made by augmenting a beamforming microphone array with non-beamforming microphones that includes: a plurality of first microphones configured as a beamforming microphone array to resolve first audio input signals within a first frequency range; one or more additional microphones configured to resolve second audio input signals within a restricted second frequency range; and an augmented beamforming module that includes a processor that executes software program steps to: receive the resolved first audio signals from the beamforming microphone array; receive the resolved and restricted second audio input signals; perform beamforming on the received and resolved first audio input signal; and combine the beamformed first audio input signal with the resolved and restricted second audio input signals to create an audio signal within a band-limited frequency range.

Abstract: The present disclosure describes a system (100) for reducing background noise from a speech audio signal generated by a user. The system (100) includes a user device (102) receiving the speech audio signal, a noise reduction device (118) in communication with a stored data repository (208), where the noise reduction device is configured to convert the speech audio signal to text; generate synthetic speech based on the converted text; optionally determine the user as an actual subscriber based on a comparison between the speech audio signal with the synthetic speech; and selectively transmit the speech audio signal or the synthetic speech based on comparison between the predicted subjective quality of the recorded speech and the synthetic speech.

Abstract: The present disclosure describes a system using an audio mixer for managing endpoint parameters of an input audio device and an output audio device. The system includes a data terminal, multiple endpoints, and the audio mixer, where the data terminal provides multiple data files including configuration parameters of endpoints. The endpoints include a first endpoint and a second endpoint. The audio mixer is configured to receive the data files from the data terminal, process the data files to obtain a first set of configuration parameters corresponding to the first endpoint and a second set of configuration parameters corresponding to the second endpoint, convert the first set and the second set of configuration parameters into a predefined common data format, and organize the predefined common data structure into a generic data structure.

Abstract: This disclosure describes a conferencing apparatus with an automatically adapting beamforming microphone array for teleconferencing applications. The conferencing apparatus includes a processor coupled to memory, storage, and a housing. The apparatus further includes a communication element, a plurality of microphones arranged in the housing and configured as a beamforming microphone array, and an orientation sensor. Additionally, the apparatus includes executing a series of steps including: (a) adapting a signal-processing characteristic of the beamforming microphone array responsive to the orientation signal, (b) beamforming to combine the plurality of microphone signals to a plurality of combined signals, (c) performing an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals, and (d) selecting one or more of the plurality of combined echo-canceled signals for transmission.

Abstract: The present disclosure provides a method for adding realism to synthetic speech. The method includes receiving text (218) that is to be converted into synthetic speech from a mobile device (108). The text (218) may include embedded emoticons indicating a first prosody information and a predefined sound stored in a stored data repository (208). The method also includes identifying a user associated with the text (218) based on a comparison between metadata associated with the text (218) and user profiles stored in the stored data repository (208); retrieving a speech font from a speech data corpus associated with the user stored in the stored data repository (208). The speech font includes a second prosody information and a predefined accent of the user. The method further includes converting the text (218) into synthetic speech based on the retrieved speech font, which is being modulated based on the emoticon.

Abstract: This disclosure describes a method for video synchronization in video distribution systems that includes a system that re-synchronizes the video playback of a video synchronization device with that of a host device after any change by the host, interruption within the host, or interruption of data flow between a video source device and a video display device.

Abstract: The present disclosure describes a system (100) for reducing background noise from a speech audio signal generated by a user. The system (100) includes a user device (102) receiving the speech audio signal, a noise reduction device (118) in communication with a stored data repository (208), where the noise reduction device is configured to convert the speech audio signal to text; generate synthetic speech based on the converted text; optionally determine the user as an actual subscriber based on a comparison between the speech audio signal with the synthetic speech; and selectively transmit the speech audio signal or the synthetic speech based on comparison between the predicted subjective quality of the recorded speech and the synthetic speech.

Abstract: This disclosure describes a system that uses a virtual audio device system for unified communications (UC) applications between a host device and a plurality of audio devices (100, 130), which includes a host device (108), a plurality of audio devices (104, 106), and a virtual audio device system (124). The host device (108) executes a Unified Communications (UC) application (114) to generate a first audio data stream. The host device (108) is in communication with a microphone (110) and a speaker (112). The plurality of audio devices (104, 106) receive the first audio data stream generated by the host device, where each of the plurality of audio devices (104, 106) generates its own audio data stream. Further, the virtual audio device system (124) establishes a communication channel between the UC application (114) and the plurality of audio devices (104, 106) based on the plurality of audio devices (104, 106) connected to the host device (108).

Abstract: This disclosure describes a method and system for encoding an RGB24 video signal (504) having multiple video frames. An RGB video frame is received and is split into red, green, and blue data. The received RGB video frame is converted into a first YUV frame (508-1) having Y data same as the red data and a second YUV frame (508-2) having Y data same as the green data. The blue data is segmented into a first data segment having 0-127 values and a second data segment having 128-255 values of the blue color component. The first data segment and the second data segment are embedded as the UV data of the first YUV frame (508-1) and the second YUV frame (508-2) respectively. The first YUV frame (508-1) and the second YUV frame (508-2) are encoded based on a timestamp same as that associated with the received RGB frame.

Abstract: Embodiments of the present disclosure include an apparatus (116) configured to perform beamforming on multiple audio input signals. The apparatus (116) includes a first plurality of microphones (302, 502) configured to resolve first audio input signals within a first frequency range. The apparatus (116) also includes at least one microphone (504) configured to resolve second audio input signals within a second frequency range. A lowest frequency in the first frequency range is greater than a lowest frequency in the second frequency range.