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: The subject disclosure is directed towards an immersive conference, in which participants in separate locations are brought together into a common virtual environment (scene), such that they appear to each other to be in a common space, with geometry, appearance, and real-time natural interaction (e.g., gestures) preserved. In one aspect, depth data and video data are processed to place remote participants in the common scene from the first person point of view of a local participant. Sound data may be spatially controlled, and parallax computed to provide a realistic experience. The scene may be augmented with various data, videos and other effects/animations.

Abstract: A phishing detection client component and method is provided. The component can be employed as part of a system to detect and, optionally, prevent phishing attacks. The phishing detection client component can provide password reuse event report(s), for example, to a phishing detection server component. The client component can further include a credential component that can track use of credentials by a user and determine whether a specific security credential is being used or presented. Due to the malicious nature of phishing in general, the client component can be susceptible to attacks by phishers. For example, phishers can generate false logins in an attempt to flood the client component with information resulting in induced false positives and/or induced false negatives. The client component can perform one or more checks to determine whether false login(s) have been attempted.

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: 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: The current application is-directed to flexible methods for motion-vector-based encoding of macroblocks within video: frames. The flexible methods for encoding video-frame macroblocks provide for segmentation of a video-frame block and encoding the segmented video-frame block by a segmented-block motion vector that includes a reference to a segmentation-defining region of a segmentation map and that also includes references to sources of intensity-and-color data for each segment. Segmented-block motion vectors provide for flexible segmentation-based encoding of video-frame blocks without sacrificing the coding efficiencies attendant with conventional motion-vector-based video encoding.

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: The subject disclosure is directed towards an immersive conference, in which participants in separate locations are brought together into a common virtual environment (scene), such that they appear to each other to be in a common space, with geometry, appearance, and real-time natural interaction (e.g., gestures) preserved. In one aspect, depth data and video data are processed to place remote participants in the common scene from the first person point of view of a local participant. Sound data may be spatially controlled, and parallax computed to provide a realistic experience. The scene may be augmented with various data, videos and other effects/animations.

Abstract: A person is provided with the ability to auditorily determine the spatial geometry of his current physical environment. A spatial map of the current physical environment of the person is generated. The spatial map is then used to generate a spatialized audio representation of the environment. The spatialized audio representation is then output to a stereo listening device which is being worn by the person.

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: An acoustic echo cancellation technique. The present adaptive acoustic echo cancellation technique employs a plurality of acoustic echo cancellation filters which use different adaptation techniques which may employ different parameters such as step size, to improve both the adaptation algorithm convergence time and misadjustment over previously known acoustic echo cancellation techniques.

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 multi-sensor sound source localization (SSL) technique is presented which provides a true maximum likelihood (ML) treatment for microphone arrays having more than one pair of audio sensors. Generally, this is accomplished by selecting a sound source location that results in a time of propagation from the sound source to the audio sensors of the array, which maximizes a likelihood of simultaneously producing audio sensor output signals inputted from all the sensors in the array. The likelihood includes a unique term that estimates an unknown audio sensor response to the source signal for each of the sensors in the array.

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 and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal.

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: Described are systems and methods performed by computer to reduce crosstalk produced by loudspeakers when rendering binaural sound that is emitted from the loudspeakers into a room. The room may have sound-reflecting surfaces that reflect some of the sound produced by the loudspeakers. To reduce crosstalk, a room model stored by the computer, is accessed. The room model models at least sound reflected by one or more of the physical surfaces. The room model is used to calculate a model of an audio channel from the loudspeakers to a listener. The model of the audio channel models sound transmission from the loudspeakers to the listener. The computer uses the model of the audio channel to cancel crosstalk from the loudspeakers when rendering the binaural sound.

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 phishing detection server component and method is provided. The component can be employed as part of a system to detect/phishing attacks. The phishing detection server component can receive password reuse event report(s), for example, from a protection component of client component(s). Due to the malicious nature of phishing in general, the phishing detection server component can be susceptible to attacks by phishers (e.g., by reverse engineering of the client component). For example, false report(s) of PREs can be received from phisher(s) in an attempt to overwhelm the server component, induce false positives and/or induce false negatives. Upon receipt of a PRE report, the phishing detection server component can first verify that the timestamp(s) are genuine (e.g., previously generated by the phishing detection server component). The report verification component can employ the timestamp(s) to verify veracity of the report (e.g., to minimize attacks by phishers).

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.