Abstract: The present disclosure discloses a conferencing apparatus that combines a beamforming microphone array with an acoustic echo canceller. The apparatus includes a beamforming microphone array that further comprises a plurality of microphones wherein each microphone is configured to sense acoustic waves and the plurality of microphones 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 performs a direction of arrival determination to select one of the plurality of combined echo cancelled signals. And, the processor selects one of the combined echo cancelled signals for transmission to the far end.

Abstract: This disclosure describes a system for negotiating video camera and display capabilities in a video conferencing environment. The system includes a local camera; a local communication device; and a session negotiation device where the session negotiation device further comprises a processor, memory, and a storage.

Abstract: This disclosure describes an apparatus (106) for improving performance of a multipoint control unit (210). The apparatus (106) provides a video stream manipulator (208) and a multipoint control unit (210). The video stream manipulator (208) may encode one or more video streams in a predetermined video codec standard separately. Each of the encoded video streams includes at least one encoded video frame made of a plurality of macroblocks, where the macroblocks are segregated into a predetermined number of macroblock lines. The multipoint control unit (210) may assemble predetermined number of macroblock lines from each of the encoded video streams in a predetermined composition.

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: A multi end-point (EP) conferencing system, including a decentralized scalable network of End-Points that includes a source EP being configured to connect to a non-virtual target EP. The source EP being configured to transmit to the non-virtual target EP a conference list thereof, for distributing to non-virtual EPs and virtual EPs associated with the target EP. It is further configured to receive a conference list of the non-virtual target EP, integrate the received conference list with its conference list to obtain an integrated conference list and distribute the integrated conference list or parts thereof to virtual EPs and non-virtual EPs associated with the source EP. The source EP being further configured to receive a positive selection notification being indicative of a requesting EP from EPs associated with the source EP that would like to get data from the source EP.

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: 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.

Abstract: This disclosure describes a computer-implemented method for a control mechanism for video output. Additionally provided is a video stream that is generated from the video source and encoded by the video encoder, where the video stream further comprises one or more video frames with each video frame including a time stamp that includes timing and clocking information. Additionally provided is a video display apparatus that includes a video display apparatus processor, video display apparatus memory, video decoder, and a video frame buffer, where the video decoder decodes the encoded video stream, and the video display apparatus generates a pixel clock. Additionally provided is a desired latency threshold for displaying a first video frame with respect to a second video frame that includes an upper threshold limit and a lower threshold limit. The disclosed method includes the following steps that control jitter in the displayed video stream.

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 and method for negotiating video camera and display capabilities in a multi-camera/multi-display video conferencing environment. The method includes the at least one processor (120, 210, 218) determining negotiation information associated with a group of local devices (112). The negotiation information includes relative spatial information and data stream characteristics. The method also includes the at least one processor (120, 210, 218) defining the determined negotiation information in a predetermined format compatible with at least one remote device (114). The method further includes the at least one processor (120, 210, 218) applying the determined negotiation information for negotiating capability of at least one device (112-1) among the group of local devices (112) and the at last one remote device (114) for data communication.

Abstract: Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of microphones oriented to cover a corresponding plurality of direction vectors and to develop a corresponding plurality of microphone signals. A processor is operably coupled to the plurality of microphones. The processor is configured to perform a beamforming operation to combine the plurality of microphone signals to a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals. The processor is also configured perform an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals and select one of the plurality of combined echo-canceled signals for transmission.

Abstract: A conferencing system may comprise an electronic display configured to display remote video generated within a remote conference room, a speaker configured to reproduce remote audio generated within the remote conference room, and a processor configured to receive local audio generated within a local conference room and picked up by a microphone assembly that is part of a separate device from the electronic display. A related method may include displaying remote video on an electronic display of an all-in-one display, reproducing remote audio through at least one speaker of the all-in-one display; and performing echo cancellation of local audio using the remote audio as an echo cancellation reference. Another method may include receiving, at an all-in-one display, a plurality of local audio signals from a plurality of microphone assemblies that are separate from the all-in-one display, and controlling, within the all-in-one display, gating of the plurality of microphone assemblies.

Abstract: Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of directional microphones oriented to cover a corresponding plurality of direction vectors and disposed in a housing. An orientation sensor is configured to generate an orientation signal indicative of an orientation of the housing. A processor is operably coupled to the plurality of directional microphones and the orientation sensor. The processor is configured to automatically adjust a signal processing characteristic of one or more directional microphones of the plurality of directional microphones responsive to the orientation signal.

Abstract: Embodiments include methods and apparatuses for echo cancelation involving multiple audio channels and the production, sensing, or a combination thereof of multiple audio channels in conferencing systems. A conferencing apparatus with a plurality of speakers configured to generate outgoing acoustic waves responsive to a multi-channel audio signal. One or more microphones are configured to sense incoming acoustic waves from the plurality of speakers and from locally produced acoustic waves from a participant of a conference to generate one or more incoming audio signals. A processor is operably coupled to the plurality of speakers and the one or more microphones. The processor is configured to perform acoustic echo cancelation on the one or more incoming audio signals relative to at least two different channels of the multi-channel audio signal.

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.

Abstract: This disclosure describes a bridge and control proxy for unified communication systems. The bridge transmits and receives data from a unified communication system. In addition, the bridge transmits and receives data from a non-compatible device. The bridge translates the data between the unified communication system and the non-compatible device. The bridge stores selected data such as presence information for later use by the unified communication system and/or the non-compatible device. Further, the bridge uses Web services to discover, promote, and/or advertise non-compatible devices to the unified communication system.