Texas Instruments Corporation

Bts Software Solutions, LLC

Telogy Networks, Inc.

Bts Software Software Solutions, LLC

Dunling Li

Gokhan Sisli

John Thomas Dowdal

Zoran Mladenovic

Daniel C Thomas

Sachin Adlakha

David Alston Lide

Bogdan R Kosanovic

Murray H Loew

Methods and systems for processing telemetry data that contains multiple data types is disclosed. Optimum multimodal encoding approaches can be used which can achieve data-specific compression performance for heterogeneous datasets by distinguishing data types and their characteristics at real-time and applying most effective compression method to a given data type. Using an optimum encoding diagram for heterogeneous data, a data classification algorithm classifies input data blocks into predefined categories, such as Unicode, telemetry, RCS and IR for telemetry datasets, and a class of unknown which includes non-studied data types, and then assigns them into corresponding compression models.

Realtime multimodel lossless data compression system and method

A method and system for transmitting data is disclosed. The method may include receiving data from two or more data sources. The method may also include selectively classifying the data into at least two input data streams, the at least two data streams including a one dimensional data stream, called an encoded data stream, and an imagery data stream, which can include still images or video or both. The method may also include separately compressing the one-dimensional data stream into a first compressed bit-stream and the imagery data stream into a second compressed bit-stream and combining the first compressed bit-stream and the second compressed bit-stream into a packetized bit-stream. The method may also include encrypting the packetized bit-stream to generate encrypted data packets prepared for transmission.

Realtime telemetry data compression system

A method for compressing data is disclosed. The method may include receiving data from one or more data sources. The method may also include selectively classifying the data into one or more data streams, the one or more data streams including at least PCM-encoded data and image data. The method may further include separately compressing the PCM-encoded data and the image data into first and second compressed bit-streams. The method may also include shuffling the first and second compressed bit-streams.

Low delay low complexity lossless compression system

A method for compressing data is disclosed. The method may include classifying the data from a plurality of data sources into classifications including at least PCM-encoded data and data structure information. The method may also include determining data states associated with the PCM-encoded data and the data structure information. The method may further include compressing the PCM-encoded data and the data structure information into compressed bit-streams based on the determined data states. The method may also include shuffling the compressed bit-streams to introduce randomness.

In a voice activity detection (VAD) device a method for defining tone signals comprises defining a threshold for zero amplitude change, calculating a zero crossing rate of a signal, extracting a set of parameters from a plurality of duration periods of the signal, defining a tolerance threshold between the plurality of duration periods when a zero amplitude change occurs, calculating a maximum difference between the plurality of duration periods, and comparing the maximum difference with the threshold. The method is implemented in the International Telecommunications Union (ITU) recommendation G.729 Annex B VAD.

Tone, modulated tone, and saturated tone detection in a voice activity detection device

In a method, apparatus and system for transmitting packet loss concealment (PLC) information, a subscriber device divides a voice sample into a plurality of packets, each including a plurality of successive frames having portions of the voice sample. The subscriber device determines if a predetermined look ahead time duration from the final frame of the plurality of successive frames in a current packet of the plurality of packets includes a noise to voice transition. When the predetermined look ahead time duration is determined to include the noise to voice transition, the subscriber device packs packing information regarding the predetermined look ahead time duration into the current packet. Finally, the subscriber device encodes the plurality of successive frames into the current packet for transmission.

Method, system and apparatus for providing signal based packet loss concealment for memoryless codecs

A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. A voice playout unit in the receiver shares its nominal delay buffer with a history buffer of a packet loss concealment algorithm up to a maximum limit described in a standard. This reduces or eliminates need to allocate memory for the history buffer. A history buffer can also be extended to retain an original portion of voice signal packets received prior to a packet loss as well as generated voice signals as they are generated. A scratch buffer is used as a working buffer and replaces the function of a pitch buffer.

Memory optimization packet loss concealment in a voice over packet network

Embodiments of the invention provide methods to calculate compression noise statistics of decompressed images in transform coding. They can be used in compressed image quality assessment, compression algorithm optimization, compression noise reduction, and other quantization and compression related applications.

Method and apparatus for processing analytical-form compression noise in images with known statistics

An overflow problem of LSF quantization in G.729 Annex B speech encoding which may lead to non-assignment of a codebook index. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Speech coder and method

A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. Packet losses are concealed by using the spectral analysis filter memory to smooth a signal gap and by using a technique for determining a maximum repeatable waveform range instead of using the pitch period to reproduce lost packets. The invention uses fewer processing resources and results in improved performance compared to a packet loss concealment algorithm under G.711 Appendix I standards.

Packet loss concealment for voice over packet networks

ITU Recommendation G.729 Annex E teaches in the implementation of a fixed codebook search to determine the selected sample combination providing the minimal difference between the original input speech and the reconstructed speech after implementation of the codec. A large number of sample sets are processed and the difference between the original input signal and the reconstructed signal for each set is determined and stored in a register. Under certain conditions, the register can overflow resulting in invalid difference values. When such a condition occurs, the fixed codebook search cannot determine the sample combination providing the minimal mean square error between the weighted input speech and the weighted reconstructed speech. An initialization vector for the codvec vector is used to provide valid data which conforms to the G.729 Annex E specifications and minimizes changes to the G.729 source code while providing robust quality signal processing in the event of register overflow condition.

Modification of fixed codebook search in G.729 Annex E audio coding

A distributed intelligence conferencing system having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing. The nodes DSP includes: A signal measuring device for measuring features of the signals from each of the participants. A single core speaker tracking algorithm identifies the active speaker on a local node based upon the relative signal features of each of the number of participant input signals. A communication device communicates the measured signal features for the active local participant input signals only to all other conferencing nodes. A multi core speaker tracking algorithm compares the features of the signals from the speakers on the local node with the features of the signals from the speakers on other nodes. A final determination is made to identify the active speakers for the entire conference so that only the signals from a limited number of speakers transmitted across the system bus.

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Rockville, MD, United States of America

Dunling Li