Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the M RI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the M RI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the MRI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

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: 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: 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 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: 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: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the MRI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

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: 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: 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.

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: 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.