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 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:
Real-time packet-based audio communications over packet-based networks frequently results in the loss of one or more packets during any given communication session. The real-time nature of such communications precludes retransmission of lost packets due to the unacceptable delays that would result. Consequently, packet loss concealment methods are employed to “hide” lost packets from the listener. Unfortunately, conventional loss concealment methods, such as packet repetition or stretch/overlap methods, do not fully exploit information available from partially received samples. Therefore, when a single frame of N coefficients is lost, 2N samples are only partially reconstructed, thereby degrading the reconstructed signal. To address this problem, an optimized packet loss concealment solution is identified for particular lost packets by solving an underdetermined system of linear equations representing partially received samples while minimizing a computed error based on a model of the signal obtained from neighboring blocks or frames received by the decoder.
Abstract:
A “communications rate controller” provides various techniques for maximizing a quality of real-time communications (RTC) (including audio and/or video broadcasts and conferencing) over multi-hop networks such as, for example, the Internet. Endpoints in such networks generally communicate via a segmented path that extends through one or more routers between each endpoint. Maximization of conferencing quality is generally accomplished by providing in-session bandwidth estimation across segments of the network path between endpoints (i.e., communication/conference participants) in combination with a robust non-oscillating dynamic rate control strategy for maximizing usage of available bandwidth between RTC endpoints. Further, the dynamic rate control techniques provided by the communications rate controller are designed to prevent degradation in end-to-end delay, jitter, and packet loss characteristics of the RTC.
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 overlay network and scheme for building and using the overlay network are described. As the overlay network is built, new nodes joining the network are connected randomly with other nodes which results in a random graph as the network topology. The network is fully scalable, with each new node providing the same number of network connections for output to other nodes as it consumes when it joins the network. In addition, network coding is implemented at each node to mix packets of data entering each node using random linear functions. The network coding at each node generates new packets that are independent combinations of original data packets entering the node. The new coded packets make the distribution of data to other nodes more efficient and robust.
Abstract:
A media recommendation and sharing technique that employs agents on media players/devices to expand the scope of media sharing scenarios. The technique assists a user in discovering media items, such as, for example, music, recordings, play lists, pictures, video games, on nearby media players or devices (devices which are capable of receiving, storing and playing media) which are interesting to the user. The collaborative media recommendation and sharing technique contemporaneously determines a user's media preferences based on media stored on a pair of media devices and recommends media for potential sharing based on these determined user preferences.
Abstract:
A “Media Transmission Optimizer” provides a media transmission optimization framework for lossy or bursty networks such as the Internet. This optimization framework provides a novel form of dynamic Forward Error Correction (FEC) that focuses on the perceived quality of a recovered media signal rather than on the absolute accuracy of the recovered media signal. In general, the Media Transmission Optimizer provides an encoder that optimizes the transmission of redundant frames of electronic media information encoded at different bit rates, and provides optimized playback quality by providing a decoder that automatically selects an optimal path through one or more available representations of each frame as a function of overall rate/distortion criteria.
Abstract:
A system and method that enables broadcasting of data in packets across a network using network coding is described. This system and method enables a network to broadcast information in packets without full knowledge of the network's topology. Further, it enables broadcasting of data in packets with a low probability of failure.