Hughes Electronics Corporation

The Directv Group, Inc.

Hughes Aircraft Company

Kumar Swaminathan

Srinivas Nandkumar

Udaya Bhaskar

Gaguk Zakaria

Prabhat K Gupta

Kalyan Ganesan

Ashish Banerji

Bangalore R Udaya Bhaskar

Brian Michael McCarthy, Sr

Murthy Vemuganti

David Alan Roos

Daniel Lin

Yue-Peng Zheng

Adrian J Morris

Shrirang Jangi

Seok Ho Kim

Robert Carroll Marquart

James Eryx Malcolm

Kirk H Berry

Filiz Basbug Ertem

Shvetal K Patel

P Vijay Kumar

Daniel Fraley

Debabrata Chowdhury

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.

System and methodology for video compression

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.

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. A voice activity detector adapted to process the LP parameters and the open loop pitch contour over the predetermined intervals is also provided as well as a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following functions: extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined invervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and provide a voicing measure where the voicing measure characterizes a degree of vocing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals. The voicing measure is provided for the purpose of regenerating a PW phase at a decoder; and providing improved quantization of the PW magnitude at an encoder. The voicing measure is encoded jointly with a PW nonstationarity measure vector using a spectrally weighted vector quantizer having a codebook partioned based on a voiced and unvoiced mode.

Voicing measure for a speech CODEC system

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. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and directly quantize the PW in a magnitude domain without further decomposition of the PW into complex components, where the direct quantization is performed by a hierarchical quantization method based on a voicing classification using fixed dimension vector quantizers (VQ's).

Prototype waveform magnitude quantization for a frequency domain interpolative speech codec system

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.

In-band transmission of TTY/TTD signals for systems employing low bit-rate voice compression

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. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and separate stationary and nonstationary components of the PW using a low complexity alignment process and a filtering process that introduce no delay. The ratio of the energy of the nonstationary component of the PW to that of the stationary component of the PW is averaged across 5 subbands to compute the nonstationarity measure as a frequency dependent vector entity. A measure of the degree of voicing of the residual is also computed using openloop pitchgain, pitch variance, relative signal power, PW correlation and PW nonstationarity in low frequency subbands. The nonstationarity measure and voicing measure are encoded using a 6-bit spectrally weighted vector quantization scheme using a codebook partitioned based on a voiced/unvoiced decision. At the decoder, a stationary component of PW is reconstructed as a weighted combination of the previous PW phase vector, a random phase perturbation and a fixed phase vector obtained from a voiced pitch pulse.

Prototype waveform phase modeling for a frequency domain interpolative speech codec system

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.

Voicing measure as an estimate of signal periodicity for a frequency domain interpolative speech codec system

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.

Spectral magnitude modeling and quantization in a frequency domain interpolative speech codec system

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. Smoothing employs critical frequency bands corresponding to the human auditory system. Swirl reduction is performed to improve overall human perception of decoded speech.

Method of noise reduction for speech codecs

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. The REW magnitude vector sequence is normalized to unity RMS value, resulting in a REW magnitude shape vector and a REW gain vector. The normalized REW magnitude vectors are modeled by a multi-band sub-band model which converts the variable dimension REW magnitude shape vectors, e.g., six dimensional REW sub-band vectors. The sub-band vectors are averaged over time, resulting in a single average REW sub-band vector for each frame. At the decoder, the full-dimension REW magnitude shape vector is obtained from the REW sub-band vector by a piecewise-constant construction. The REW phase vector is regenerated at the decoder based on the received REW gain vector and the voicing measure, which determines a weighted mixture of SEW component and a random noise that is passed through a high pass filter to generate the REW component. The high pass filter poles are adjusted based on the voicing measure to control the REW component characteristics. At the output the filter, the magnitude of the REW component is scaled to match the received REW magnitude vector.

Frequency domain interpolative speech codec system

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.

Parallel/pipeline VLSI architecture for a low-delay CELP coder/decoder

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.

Speech mode based multi-stage vector quantizer

The present invention relates to a device and a method for communicating in a mobile communication system. The method provides a carrier signal having a plurality of frames. Each frame has a plurality of time slots, and each time slot comprises a plurality of transmission bits. A group of time slots are assigned to a communication channel. A traffic burst signal having a plurality of traffic symbols is transmitted over the communication channel by transmitting a first preamble over one of the assigned time slots, and transmitting a second preamble and at least one of the traffic symbols over at least one of the other assigned time slots. The second preamble occupies fewer transmission bits than the first preamble. The apparatus for transmitting a telephony signal over an RF channel includes a modem receiving a digitized PCM telephony signal and producing a traffic burst signal, and a transmitting unit in communication with the modem for transmitting a FDMA/TDMA signal carrying a plurality of traffic burst signals. At least one of the traffic burst signals carries a limited preamble message including a header field and a unique word field and at least one digitized voice message associated with a telephone call. Another traffic burst signal carries at least one signal acquisition message including a unique word field.

Device and method for communicating in a mobile satellite system

A method of transporting speech information over a wireless cellular communications system is provided. By determining the existence and compatibility of the destination port with the origination port in a given telephone call, the present invention is capable of using only one compression step and one decompression step. Accordingly, voice signal degradation and delay associated with multiple compression/decompression steps may be reduced.

Growing community of inventors

North Potomac, MD, United States of America

Kumar Swaminathan