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: Whether a microphone is connected to a real-time audio communication system of a computer may be detected by recording an audio sample through the real-time audio communication system, filtering a DC component is filtered out of the audio data, recognizing a pattern in the auto-correlation coefficients of the filtered audio sample, and determining whether a microphone is properly connected to the real-time audio communication system based on the values of the auto-correlation function coefficients and the predetermined values.

Abstract: Compressing the digitized time-domain continuous input signal typically includes formatting the input signal into a plurality of time-domain blocks having boundaries, forming an overlapping time-domain block by prepending a fraction of a previous time-domain block to a current time-domain block, transforming each overlapping time-domain block to a transform domain block including a plurality of coefficients, partitioning the coefficients of each transform domain block into signal coefficients and residue coefficients, quantizing the signal coefficients for each transformed 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. The continuous data may include audio data.

Abstract: Compressing the digitized time-domain continuous input signal typically includes formatting the input signal into a plurality of time-domain blocks having boundaries, forming an overlapping time-domain block by prepending a fraction of a previous time-domain block to a current time-domain block, transforming each overlapping time-domain block to a transform domain block including a plurality of coefficients, partitioning the coefficients of each transform domain block into signal coefficients and residue coefficients, quantizing the signal coefficients for each transformed 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. The continuous data may include audio data.

Abstract: Enabling low-latency reduction of quantization-induced block-discontinuities of continuous data formatted into a plurality of data blocks having boundaries typically includes forming an overlapping input data block by prepending a fraction of a previous input data block to a current input data block, identifying regions near the boundary of each overlapping input data block, and excluding regions near the boundary of each overlapping input data block and reconstructing an initial output data block from the remaining data of such overlapping input data block. The continuous data may include audio data and/or time-domain data. The low-latency reduction of quantization-induced block-discontinuities of continuous data may be applied to at least one of a coder and decoder.

Abstract: Compressing the digitized time-domain continuous input signal typically includes formatting the input signal into a plurality of time-domain blocks having boundaries, forming an overlapping time-domain block by prepending a fraction of a previous time-domain block to a current time-domain block, transforming each overlapping time-domain block to a transform domain block including a plurality of coefficients, partitioning the coefficients of each transform domain block into signal coefficients and residue coefficients, quantizing the signal coefficients for each transformed 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. The continuous data may include audio data.

Abstract: Enabling low-latency reduction of quantization-induced block-discontinuities of continuous data formatted into a plurality of data blocks having boundaries typically includes forming an overlapping input data block by prepending a fraction of a previous input data block to a current input data block, identifying regions near the boundary of each overlapping input data block, and excluding regions near the boundary of each overlapping input data block and reconstructing an initial output data block from the remaining data of such overlapping input data block. The continuous data may include audio data and/or time-domain data. The low-latency reduction of quantization-induced block-discontinuities of continuous data may be applied to at least one of a coder and decoder.

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: 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: Whether a microphone is connected to a real-time audio communication system of a computer may be detected by recording an audio sample through the real-time audio communication system, filtering a DC component is filtered out of the audio data, recognizing a pattern in the auto-correlation coefficients of the filtered audio sample, and determining whether a microphone is properly connected to the real-time audio communication system based on the values of the auto-correlation function coefficients and the predetermined values.

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: Both the energy and velocity contents of received Doppler and time shift signals are simultaneously displayed. A continuous range of imaging modes are provided ranging from velocity imaging to energy imaging. The user may select the particular imaging mode desired for a particular clinical application.

Abstract: An audio coder/decoder ("codec") that is suitable for real-time applications due to reduced computational complexity, and a novel adaptive sparse vector quantization (ASVQ) scheme and algorithms for general purpose data quantization. The codec provides low bit-rate compression for music and speech, while being applicable to higher bit-rate audio compression. The codec includes an in-path implementation of psychoacoustic spectral masking, and frequency domain quantization using the novel ASVQ scheme and algorithms specific to audio compression. More particularly, the inventive audio codec employs frequency domain quantization with critically sampled subband filter banks to maintain time domain continuity across frame boundaries. The input audio signal is transformed into the frequency domain in which in-path spectral masking can be directly applied. This in-path spectral masking usually results in sparse vectors.

Abstract: An audio coder/decoder ("codec") that is suitable for real-time applications due to reduced computational complexity, and a novel adaptive sparse vector quantization (ASVQ) scheme and algorithms for general purpose data quantization. The codec provides low bit-rate compression for music and speech, while being applicable to higher bit-rate audio compression. The codec includes an in-path implementation of psychoacoustic spectral masking, and frequency domain quantization using the novel ASVQ scheme and algorithms specific to audio compression. More particularly, the inventive audio codec employs frequency domain quantization with critically sampled subband filter banks to maintain time domain continuity across frame boundaries. The input audio signal is transformed into the frequency domain in which in-path spectral masking can be directly applied. This in-path spectral masking usually results in sparse vectors.