Abstract: Multi-view video that is being streamed to a remote device in real time may be encoded. Frames of a real-world scene captured by respective video cameras are received for compression. A virtual viewpoint, positioned relative to the video cameras, is used to determine expected contributions of individual portions of the frames to a synthesized image of the scene from the viewpoint position using the frames. For each frame, compression rates for individual blocks of a frame are computed based on the determined contributions of the individual portions of the frame. The frames are compressed by compressing the blocks of the frames according to their respective determined compression rates. The frames are transmitted in compressed form via a network to a remote device, which is configured to render the scene using the compressed frames.

Abstract: The claimed subject matter relates to an architecture that can preprocess audio portions of communications in order to enrich multiparty communication sessions or environments. In particular, the architecture can provide both a public channel for public communications that are received by substantially all connected parties and can further provide a private channel for private communications that are received by a selected subset of all connected parties. Most particularly, the architecture can apply an audio transform to communications that occur during the multiparty communication session based upon a target audience of the communication. By way of illustration, the architecture can apply a whisper transform to private communications, an emotion transform based upon relationships, an ambience or spatial transform based upon physical locations, or a pace transform based upon lack of presence.

Abstract: Context-based routing in multi-hop networks involves using a context-based routing metric. In a described implementation, respective path values are calculated for respective ones of multiple paths using the context-based routing metric. A path is selected from the multiple paths responsive to the calculated path values. Data is transmitted over at least one link of the selected path. In an example embodiment, the context-based routing metric is ascertained responsive to an estimated service interval (ESI) of a bottleneck link of each path of the multiple paths. In another example embodiment, the context-based routing metric is ascertained responsive to an expected resource consumption (ERC) metric. In an example embodiment of path selection, the path is selected using a context-based path pruning (CPP) technique that involves maintaining multiple local contexts at each intermediate node, with each local context representing at least one partial path.

Abstract: Context-based routing in multi-hop networks involves using a context-based routing metric. In a described implementation, respective path values are calculated for respective ones of multiple paths using the context-based routing metric. A path is selected from the multiple paths responsive to the calculated path values. Data is transmitted over at least one link of the selected path. In an example embodiment, the context-based routing metric is ascertained responsive to an estimated service interval (ESI) of a bottleneck link of each path of the multiple paths. In another example embodiment, the context-based routing metric is ascertained responsive to an expected resource consumption (ERC) metric. In an example embodiment of path selection, the path is selected using a context-based path pruning (CPP) technique that involves maintaining multiple local contexts at each intermediate node, with each local context representing at least one partial path.

Abstract: A system for protecting against information security breaches comprises a credential module that maintains a list of protected security credentials that are each associated with a known computing system and that detects when a security credential in the list is used. The system also includes a protection module that detects use of the security credential in connection with a computing system other than the known computing system with which the security credential is associated and that reports such use to a security server. Methods for using the system are also provided.

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.

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.

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.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: A method and system for modifying a digital audio signal to vary its playback speed while preserving the signal's pitch and quality. The variable speed playback (VSP) system and method mitigates artifacts remaining after processing by existing techniques. The VSP system and method produces a consistent and pleasing sound to an audio file, even while its speed is varied during playback. The VSP method includes selecting and estimating an input frame, adjusting the frame position, and overlapping and adding the adjust frame to an output signal. The frame position adjustment is achieved using an enhanced correlation technique that finds all local maxima over a cross-correlation function. The local maxima having a highest correlation score is designated as a cut position, where the adjusted frame is cut from the input buffer. The VSP system and method using four input frames to generate one output frame.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: A unique system and method that facilitate automatic tracking of user data and providing on-demand site or merchant reputation information is provided. The system and method involve monitoring user input such as by keystrokes over a period of time and forming a long continuous string of characters. The string of characters can be monitored for repeating substrings or non-repeating substrings that indicate sensitive user information such as SSN or credit card number (CCN). The substrings can be displayed to the user in a digest form and order according to frequency of occurrence, for example. Thus, the user can readily view where and when their sensitive information has been entered. In addition, the typing of certain substrings such as CCNs or other sensitive field data can trigger a reputation service to retrieve the reputation data for the current site or merchant to mitigate dealing with fraudulent sites or merchants.

Abstract: The claimed subject matter provides systems and/or methods that facilitate utilizing a shared secret to obscure a password within a sequence of characters. The sequence of characters can include the password as well as noise. The shared secret can leverage utilizing a set of known images that a user can uniquely distinguish from random images. By employing the imaged based shared secret, the user can login to a server from an untrusted machine suspected to be infected with spyware such as a keylogger that tracks user input.

Abstract: A “Client-Side Echo Canceller” provides a unique system and method for reducing Multipoint Control Unit (MCU) computational overhead in a multi-point audio conference. In general, the local audio input signal of each client is transmitted in real-time to the MCU. The MCU then combines the audio input signals of all clients to create a single composite signal that is transmitted back to all clients in real-time. Each client then locally processes the composite signal to remove each client's local contribution to the composite signal prior to local playback in order to eliminate a local echo of each client's local audio input. In various embodiments, local cancellation of the local audio input from the composite signal is performed on either a time domain or a transform domain representation of the composite signal. Further, since each client receives the same signal, MCU transmission bandwidth can be reduced via multicast transmissions.

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: Context-based routing in multi-hop networks involves using a context-based routing metric. In a described implementation, respective path values are calculated for respective ones of multiple paths using the context-based routing metric. A path is selected from the multiple paths responsive to the calculated path values. Data is transmitted over at least one link of the selected path. In an example embodiment, the context-based routing metric is ascertained responsive to an estimated service interval (ESI) of a bottleneck link of each path of the multiple paths. In another example embodiment, the context-based routing metric is ascertained responsive to an expected resource consumption (ERC) metric. In an example embodiment of path selection, the path is selected using a context-based path pruning (CPP) technique that involves maintaining multiple local contexts at each intermediate node, with each local context representing at least one partial path.

Abstract: Systems and methods facilitate secure one-time-password access to an account in a remote server from an untrusted client. The system consists of an intermediary component whose salient components are a proxy component, a webserver component, and an encryption/decryption component, and it preserves the characteristics of both the server and client. In a man-in-the-middle fashion, the proxy substitutes a one-time password entered at a login interface with a true password, and forwards it to the remote login server. True passwords are encrypted using a seed associated with user identifiers, and a list of one-time passwords is generated/updated and stored on media or transmitted to an electronic device. Substitution takes place by decrypting the one-time password with the seed used for encryption, ensuring the proxy avoids storing the true password.

Abstract: A novel enhanced beamforming technique that improves beamforming operations by incorporating a model for the directional gains of the sensors, such as microphones, and provides means of estimating these gains. The technique forms estimates of the relative magnitude responses of the sensors (e.g., microphones) based on the data received at the array and includes those in the beamforming computations.

Abstract: A decentralized computer network architecture and method that gathers metadata from local and remote clients and, based on that metadata, locally makes a decision whether to send a packet over the network. Each client listens to what other clients are doing, and only sends when the total number of concurrent speakers is below some threshold. In a multi-party voice conferencing embodiment, the threshold is a number of concurrent speakers that is restricted to less than a certain number. Under the decentralized computer network architecture, the type of network topology used to connect the clients is flexible, as long as each client is running a peer-aware system to decide locally whether to send their packets. The decentralized computer network architecture and method is distributed to run on each client, making it suitable for a wide variety of network topologies (such as full-mesh, bridge-based, or a hybrid of the two).

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.