Abstract: A local network coding framework and method including techniques to improve efficiency in a wireless network by reducing overhead. The local network coding method includes exchanging data availability between nodes on the wireless network by sending Bloom filters of lists of packets to neighboring nodes. Based on data availability, optimized mixing of pure packets is performed to form mixture packets for output. A separate acknowledgement buffer keeps track of the pure packets transmitted but not acknowledged. If an acknowledgement does not arrive after a certain time period, the packet is assumed to be lost and is retransmitted. An optimized packet mixing process generates mixture packets and decides which nodes to send the mixture packets. The local network coding framework and method also includes methods for representing the composition of a mixture packet and using mixing at a wireless access point to improve the performance of the wireless local area network.

Abstract: A method and system for collecting and communicating contextual information relating to a VoIP conversation is provided. Structured hierarchies are utilized for efficient communications of various amounts and types of contextual information relating to a VoIP conversation. Information identifying at least one structured hierarchy, which will be used to carry the contextual information, is transmitted during establishment of a conversation between two VoIP enhanced devices. The structured hierarchy is selected from a set of predefined and declared structured hierarchies. Subsequently transmitted contextual information exchanged between two VoIP enhanced devices is represented in accordance with the identified structural hierarchy. VoIP clients, network infrastructure, and various service providers can collect the contextual information based on the identified structured hierarchy, update the contextual information by adding, deleting and/or modifying the contextual data.

Abstract: Content synchronization in a file sharing environment is described. A node in a circle share maintains a version vector that represents that node's knowledge of file synchronization events. Version vector data may be transmitted between nodes of the circle share via a presence service in the form of short notification messages. The version vector data is compared to identify one or more nodes to be synchronized. A resource-aware overlay representing the circle share is computed, and is used to initiate communication to determine which of the available nodes with which to synchronize will provide the most synchronization value to other nodes in the circle share.

Abstract: An energy based technique to estimate the positions of people speaking from an ad hoc network of microphones. The present technique does not require accurate synchronization of the microphones. In addition, a technique to normalize the gains of the microphones based on people's speech is presented, which allows aggregation of various audio channels from the ad hoc microphone network into a single stream for audio conferencing. The technique is invariant of the speaker's volumes thus making the system easy to deploy in practice.

Abstract: A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.

Abstract: Techniques to perform rate matching for multimedia conference calls are described. An apparatus may comprise a conferencing server and a rate matching module. The rate matching module may be arranged to adjust bit rates between media communications channels for client terminals in a conference call, with the rate matching module to remove video frames from a set of video information received on a first media communications channel to reduce a bit rate for the video information. Other embodiments are described and claimed.

Abstract: Gaze tracking or other interest indications are used during a video conference to determine one or more audio sources that are of interest to one or more participants to the video conference, such as by determining a conversation from among multiple conversations that a subset of participants are participating in or listening to, for enhancing the audio experience of one or more of the participants.

Abstract: A template and/or knowledge associated with a synchronous meeting are obtained by a computing device. The computing device then adaptively manages the synchronous meeting based at least in part on the template and/or knowledge.

Abstract: A virtual media device is described for processing one or more input signals from one or more physical media input devices, to thereby generate an output signal for use by a consuming application module. The consuming application module interacts with the virtual media device as if it were a physical media input device. The virtual media device thereby frees the application module and its user from the burden of having to take specific account of the physical media input devices that are connected to a computing environment. The virtual media device can be coupled to one or more microphone devices, one or more video input devices, or a combination of audio and video input devices, etc. The virtual media device can apply any number of processing modules to generate the output signal, each performing a different respective operation.

Abstract: A multimedia conference technique is disclosed that allows physically remote users to participate in an immersive telecollaborative environment by synchronizing multiple data, images and sounds. The multimedia conference implementation provides users with the perception of being in the same room visually as well as acoustically according to an orientation plan which reflects each remote user's position within the multimedia conference environment.

Abstract: A system and method for distributing streaming content data from a server to multiple clients enables the server to handle transient surges of requests by having the clients cooperate with the server and other clients to distribute content, thereby alleviating the load on the server. The server divides the streaming content into multiple sub-streams (e.g., by using multiple description coding), and constructs multiple distribution trees with itself at the root and each client as a node in each of the trees. Each sub-stream is transmitted down a corresponding one of the distribution trees. Clients that receive sub-streams from its parent nodes in the distribution trees in turn forward the sub-streams to their child nodes in the trees.

Abstract: A system and method that can receive, buffer, and asynchronously combine data from various incoming data packets is disclosed. The system and method can do so by, first, receiving incoming packets of one or multiple generations that have incoming data and incoming metadata, with the incoming data of each of the incoming packets that are of a first generation being a combination of a first, original set of data vectors, and the incoming metadata of each of the first generation of incoming packets including an indicator for the first generation. Second, once these incoming packets are received, the system and method can buffer them based on the indicator and, asynchronously from the receiving and buffering, combine the incoming data in the buffered, first generation incoming packets into outgoing data in an outgoing packet.

Abstract: Difficulties associated with choosing advantageous network routes between server and clients are mitigated by a routing system that is devised to use many routing path sets, where respective sets comprise a number of routing paths covering all of the clients, including through other clients. A server may then apportion a data stream among all of the routing path sets. The server may also detect the performance of the computer network while sending the data stream between clients, and may adjust the apportionment of the routing path sets including the route. The clients may also be configured to operate as servers of other data streams, such as in a videoconferencing session, for example, and may be configured to send detected route performance information along with the portions of the various data streams.

Abstract: A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission.

Abstract: Multi-device capture and spatial browsing of conferences is described. In one implementation, a system detects cameras and microphones, such as the webcams on participants' notebook computers, in a conference room, group meeting, or table game, and enlists an ad-hoc array of available devices to capture each participant and the spatial relationships between participants. A video stream composited from the array is browsable by a user to navigate a 3-dimensional representation of the meeting. Each participant may be represented by a video pane, a foreground object, or a 3-D geometric model of the participant's face or body displayed in spatial relation to the other participants in a 3-dimensional arrangement analogous to the spatial arrangement of the meeting.

Abstract: A method and system including an improved generalized reference decoder that operates according to any number of sets of rate and buffer parameters for a given bit stream. Each set characterizes a leaky bucket model and contains three parameters representing the transmission bit rate, buffer size, and initial decoder buffer fullness. An encoder provides at least two sets of these parameters, whereby the decoder selects one or interpolates between them to operate at any desired peak bit rate, buffer size or delay. The generalized reference decoder may select the smallest buffer size and corresponding delay that decodes the bit stream without buffer underflow or overflow, or alternatively may select and operate at the minimum required peak transmission rate, or something between both. In practice, the buffer size, delay and/or the peak transmission rate can be reduced by significant factors, and/or the signal-to-noise ratio (SNR) can be increased.

Abstract: Various embodiments can utilize a distributed solution for scheduling connections between peers in a file sharing environment. In accordance with at least some embodiments, individual nodes in a peer-to-peer system include scheduling software that enables the nodes to make scheduling decisions with regard to how, when and where connections are made with other peers. Scheduling decisions can be based on a number of different parameters. In at least some embodiments, a synchronization negotiation manager is embodied with logic that drives the negotiation and scheduling process. This logic is represented by an innovative state machine that is designed to implement the negotiation and synchronization process.

Abstract: A multimedia conference technique is disclosed that allows physically remote users to participate in an immersive telecollaborative environment by synchronizing multiple data, images and sounds. The multimedia conference implementation provides users with the perception of being in the same room visually as well as acoustically according to an orientation plan which reflects each remote user's position within the multimedia conference environment.

Abstract: Difficulties associated with choosing advantageous network routes between server and clients are mitigated by a routing system that is devised to use many routing path sets, where respective sets comprise a number of routing paths covering all of the clients, including through other clients. A server may then apportion a data stream among all of the routing path sets. The server may also detect the performance of the computer network while sending the data stream between clients, and may adjust the apportionment of the routing path sets including the route. The clients may also be configured to operate as servers of other data streams, such as in a videoconferencing session, for example, and may be configured to send detected route performance information along with the portions of the various data streams.

Abstract: An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.