Abstract: Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

Abstract: A differential global positioning system (DGPS) processor can include an almost fixed integer ambiguity (AFIA) module for generating in real-time a multiple dimensional state vector of integer ambiguities and multiple dimensional corrections. The AFIA module can use double difference (DD) measurements for pseudo-range (PR) and carrier phase (CP) pairs generated from at least three global positioning system (GPS) receivers. A DGPS processor can be included in a high data rate real time attitude determination (RTAD) system to achieve high heading accuracy with high integrity.

Abstract: Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

Abstract: Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

Abstract: A method and system to handle single failure GPS faults in high integrity relative positioning systems is provided. The system is configured to receive a number of reference measurements in carrier-phase real time kinetic positioning systems and identify if a single fault exists. The system identifies the type of fault and mitigates the effects of the single fault to provide a positioning system with a high integrity that is suitable for applications involving risk to human lives.

Abstract: Systems and methods are provided for ultra-low latency decompression for a general-purpose audio input signal. In accordance with one implementation, a computer-implemented method is provided that includes decoding, by a processor, an input bit stream into quantization indices and residue quantization indices; applying an inverse quantization algorithm to the quantization indices to generate signal coefficients; applying an inverse transform to the signal coefficients to generate a time-domain reconstructed signal waveform; applying a stochastic noise synthesis algorithm to the residue quantization indices to generate a time-domain reconstructed residue waveform; combining, by the processor, the reconstructed signal waveform and the reconstructed residue waveform as a reconstructed signal waveform block; and generating an output signal by applying a boundary synthesis algorithm to the reconstructed signal waveform blocks.

Abstract: Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

Abstract: A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm.

Abstract: Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

Abstract: Systems and methods are provided for ultra-low latency compression. In accordance with one implementation, a method is provided that includes formatting, by a processor, an input audio signal into a plurality of overlapping time-domain blocks, transforming, by a processor, each time-domain block to a transform domain block comprising a plurality of coefficients, and partitioning the coefficients of each transform domain block into signal coefficients and residue coefficients. The method also includes quantizing the signal coefficients for each transform domain block and generating signal quantization indices indicative of such quantization, modeling the residue coefficients for each transform domain block as stochastic noise and generating residue quantization indices indicative of such quantization, and formatting the signal quantization indices and the residue quantization indices for each transform domain block as an output bit stream.

Abstract: Latency in a real-time electronic communication is dynamically managed. A communication delay arising from a receiving data buffer is measured and a latency adjustment necessary to adjust the size of the communication delay to within a predetermined range and an optimal range for a size of the communication delay are determined. Using these parameters, the number of samples for an audio playback data block passing through the receiving data buffer is modified.

Abstract: Methods and apparatus for providing high integrity probability of connect fix (PCF) in GPS navigation applications, such as precision approach and landing and airborne refueling. In an exemplary embodiment, an enlarged pull in region is used to compute protection levels. In an exemplary embodiment, geometric extra-redundancy is used to enhance PCF and PAF (probability of almost fixed). In an exemplary embodiment, geometric extra-redundancy almost fixed solutions provide superior accuracy and integrity for GPS navigation applications.

Abstract: Temporal drift correction may be provided in a real-time audio communication system by measuring a size of a receiving data buffer and comparing that size to a predetermined nominal data buffer size. An amount of temporal drift is characterized as a number of samples per audio playback data block based on the measured data buffer size and the nominal data buffer size. A number of samples to be inserted or removed for each audio playback data block to correct the temporal drift may be determined, and the number of samples for each audio playback data block may be modified. For example, an instantaneous size of the receiving data buffer may be measured, and if measured multiple times, may be averaged over a time period. Heuristic resampling of the audio playback data block also may be performed.

Abstract: A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm.

Abstract: Methods and apparatus for providing high integrity probability of connect fix (PCF) in GPS navigation applications, such as precision approach and landing and airborne refueling. In an exemplary embodiment, an enlarged pull in region is used to compute protection levels. In an exemplary embodiment, geometric extra-redundancy is used to enhance PCF and PAF (probability of almost fixed). In an exemplary embodiment, geometric extra-redundancy almost fixed solutions provide superior accuracy and integrity for GPS navigation applications.

Abstract: A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block-discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm.

Abstract: Latency in a real-time electronic communication is dynamically managed. A communication delay arising from a receiving data buffer is measured and a latency adjustment necessary to adjust the size of the communication delay to within a predetermined range and an optimal range for a size of the communication delay are determined. Using these parameters, the number of samples for an audio playback data block passing through the receiving data buffer is modified.

Abstract: Latency in a real-time electronic communication is dynamically managed. A communication delay arising from a receiving data buffer is measured and a latency adjustment necessary to adjust the size of the communication delay to within a predetermined range and an optimal range for a size of the communication delay are determined. Using these parameters, the number of samples for an audio playback data block passing through the receiving data buffer is modified.

Abstract: Temporal drift correction may be provided in a real-time audio communication system by measuring a size of a receiving data buffer and comparing that size to a predetermined nominal data buffer size. An amount of temporal drift is characterized as a number of samples per audio playback data block based on the measured data buffer size and the nominal data buffer size. A number of samples to be inserted or removed for each audio playback data block to correct the temporal drift may be determined, and the number of samples of each audio playback data block may be modified. For example, an instantaneous size of the receiving data buffer may be measured, and if measured multiple times, may be averaged over a time period. Heuristic resampling of the audio playback data block also may be performed.

Abstract: Temporal drift correction may be provided in a real-time audio communication system by measuring a size of a receiving data buffer and comparing that size to a predetermined nominal data buffer size. An amount of temporal drift is characterized as a number of samples per audio playback data block based on the measured data buffer size and the nominal data buffer size. A number of samples to be inserted or removed for each audio playback data block to correct the temporal drift may be determined, and the number of samples for each audio playback data block may be modified. For example, an instantaneous size of the receiving data buffer may be measured, and if measured multiple times, may be averaged over a time period. Heuristic resampling of the audio playback data block also may be performed.