Abstract: A system and method for compressing video is disclosed, in which video frames that are only between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method for compressing video is disclosed, in which video frames that are only between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method for compressing video is disclosed, in which video frames that are only between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method for compressing video is disclosed, in which video frames that are only between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method for compressing video is disclosed, in which video frames that are only between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method for compressing video is disclosed, in which video frames that between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal providing LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator also provides a pitch contour within the predetermined intervals.

Abstract: A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided.

Abstract: A method, system, and software product for transmitting TTY/TDD signals in a system employing low bit-rate voice compression are disclosed. The method includes receiving an input signal and generating a teletypewriter (TTY) indicator signal from the input signal. Whether or not the input signal is a TTY signal including a TTY character, is determined based on the TTY indicator signal. A TTY packet including the TTY character of the TTY signal is constructed and transmitted if the input signal is determined to be a TTY signal. A method, system, and software product for receiving and decoding TTY/TDD signal is also disclosed.

Abstract: A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided.

Abstract: A drift-free hybrid method of performing video stitching is provided. The method includes decoding a plurality of video bitstreams and storing prediction information. The decoded bitstreams form video images, spatially composed into a combined image. The image comprises frames of ideal stitched video sequence. The method uses prediction information in conjunction with previously generated frames to predict pixel blocks in the next frame. A stitched predicted block in the next frame is subtracted from a corresponding block in a corresponding frame to create a stitched raw residual block. The raw residual block is forward transformed, quantized, entropy encoded and added to the stitched video bitstream along with the prediction information. Also, the stitched raw residual block is inverse transformed and dequantized to create a stitched decoded residual block. The residual block is added to the predicted block to generate the stitched reconstructed block in the next frame of the sequence.

Abstract: A system determines a voicing measure as a measure of the degree of signal periodicity and uses the determined voicing measure to quantize the spectral magnitude of the slowly evolving waveform (SEW) and the modeling of the SEW and rapidly evolving waveform (REW) phase spectra.

Abstract: A low bit rate voice codec based on Frequency Domain Interpolation (FDI) technology is designed to operate at multiple rates of 4.0, 2.4, and 1.2 Kbps. At 4 Kbps, the codec uses a 20 ms frame size and a 20 ms lookahead for purposes of voice activity detection (VAD), noise reduction, linear prediction (LP) analysis, and open loop pitch analysis. The LP parameters are encoded using backward predictive hybrid scalar-vector quantizers in the line spectral frequency (LSF) domain after adaptive bandwidth broadening to minimize excessive peakiness in the LP spectrum. Prototype Waveforms (PW) are extracted every subframe or 2.5 ms from the LP residual and subsequently aligned and normalized. The PW gains are encoded separately using a backward predictive vector quantizer (VQ). The normalized and aligned PWs are separated into a magnitude component and a phase component. The phase component is encoded implicitly using PW correlations and a voicing measure which are jointly quantized using a VQ.

Abstract: A system and method for compressing video is disclosed, in which video frames that between consecutive I-frames are grouped into a video data set. The video data set is split into separate homogeneous files, and each of the homogeneous files are individually compressed. In one embodiment, the individually compressed files are multiplexed to form a bit stream.

Abstract: Encoding of prototype waveform components applicable to telecommunication systems provides improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates a codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions.

Abstract: An improved noise reduction algorithm is provided, as well as a voice activity detector, for use in a voice communication system. The voice activity detector allows for a reliable estimate of noise and enhancement of noise reduction. The noise reduction algorithm and voice activity detector can be implemented integrally in an encoder or applied independently to speech coding application. The voice activity detector employs line spectral frequencies and enhanced input speech which has undergone noise reduction to generate a voice activity flag. The noise reduction algorithm employs a smooth gain function determined from a smoothed noise spectral estimate and smoothed input noisy speech spectra. The gain function is smoothed both across frequency and time in an adaptive manner based on the estimate of the signal-to-noise ratio. The gain function is used for spectral amplitude enhancement to obtain a reduced noise speech signal.

Abstract: Encoding of prototype waveform components applicable to GeoMobile and Telephony Earth Station (TES) providing improved voice quality enabling a dual-channel mode of operation which permits more users to communicate over the same physical channel. A prototype word (PW) gain is vector quantized using a vector quantizer (VQ) that explicitly populates the codebook by representative steady state and transient vectors of PW gain for tracking the abrupt variations in speech levels during onsets and other non-stationary events, while maintaining the accuracy of the speech level during stationary conditions. The rapidly evolving waveform (REW) and slowly evolving waveform (SEW) component vectors are converted to magnitude-phase. The variable dimension SEW magnitude vector is quantized using a hierarchical approach, i.e., a fixed dimension SEW mean vector computed by a sub-band averaging of SEW magnitude spectrum, and only the REW magnitude is explicitly encoded.

Abstract: A method and apparatus for echo cancellation and noise reduction are provided that use synergy among system components. Double-talk detection is performed using either the voice activity detector of a codec or a secondary double-talk detector, depending on the signal-to-noise ratio (SNR) obtained from the encoder. The echo canceller is implemented via an adaptive filter and operates in a dual-mode. Under low SNR conditions, variable step-size methods, VAD-based double-talk detection and emergency coefficients are used. Under high SNR conditions, a secondary double-talk detector employing an echo loss return estimator and comparator for near-end and far-end levels is used, as well as a non-linear gain function and masking noise.

Abstract: An integrated circuit for processing a speech signal in accordance with a CELP standard includes a plurality of processing elements coupled to a data bus in parallel. Each processing element includes a multiplier and an accumulator. The integrated circuit further includes an auxiliary processing element, which is also coupled to the data bus and has a division unit and a comparator. The plurality of processing elements and the auxiliary processing element are also coupled in a pipeline formation.

Abstract: A speech mode based multi-stage vector quantizer is disclosed which quantizes and encodes line spectral frequency (LSF) vectors that were obtained by transforming the short-term predictor filter coefficients in a speech codec that utilizes linear predictive techniques. The quantizer includes a mode classifier that classifies each speech frame of a speech signal as being associated with one of a voiced, spectrally stationary (Mode A) speech frame, a voiced, spectrally non-stationary (Mode B) speech frame and an unvoiced (Mode C) speech frame. A converter converts each speech frame of the speech signal into an LSF vector and an LSF vector quantizer includes a 12-bit, two-stage, backward predictive vector encoder that encodes the Mode A speech frames and a 22 bit, four-stage backward predictive vector encoder that encodes the Mode 13 and the Mode C speech frames.