256 results about "Source encoding" patented technology

A method and apparatus for improving the speed and quality of end-to-end data or real-time media transmissions over an internet is disclosed. A media stream being transmitted to the internet is channel coded at the edge of the internet in order to free upstream bit rate for use in source coding the media. The channel coded media stream may then be decoded at a remote edge of the internet to recover lost packets.

A method for coding and decoding seismic data acquired, based on the concept of multishooting, is disclosed. In this concept, waves generated simultaneously from several locations at the surface of the earth, near the sea surface, at the sea floor, or inside a borehole propagate in the subsurface before being recorded at sensor locations as mixtures of various signals. The coding and decoding method for seismic data described here works with both instantaneous mixtures and convolutive mixtures. Furthermore, the mixtures can be underdetemined [i.e., the number of mixtures (K) is smaller than the number of seismic sources (I) associated with a multishot] or determined [i.e., the number of mixtures is equal to or greater than the number of sources). When mixtures are determined, we can reorganize our seismic data as zero-mean random variables and use the independent component analysis (ICA) or, alternatively, the principal component analysis (PCA) to decode. We can also alternatively take advantage of the sparsity of seismic data in our decoding process. When mixtures are underdetermined and the number of mixtures is at least two, we utilize higher-order statistics to overcome the underdeterminacy. Alternatively, we can use the constraint that seismic data are sparse to overcome the underdeterminacy. When mixtures are underdetermined and limited to single mixtures, we use a priori knowledge about seismic acquisition to computationally generate additional mixtures from the actual recorded mixtures. Then we organize our data as zero-mean random variables and use ICA or PCA to decode the data. The a priori knowledge includes source encoding, seismic acquisition geometries, and reference data collected for the purpose of aiding the decoding processing.The coding and decoding processes described can be used to acquire and process real seismic data in the field or in laboratories, and to model and process synthetic data.

A diatrophic, bio-physiological interface is self-contained with onboard intensification, filtering, and signal processing and is wirelessly enabled (idio-electrode), with multiple sensory system for bio-physiological measurements, described herein utilizes spatially resolved potential profiles from a cluster of mini electrodes to form constituent sets comprising mini sensorial electrodes. The sets of sub electrodes containing the clusters are jointly optimized to attain measurable gradient of some diagnostic value. The present invention provides a distinct lead-free single electrode that is rotationally invariant with onboard Digital Signal Processor for arrhythmia detection, source encoding, and passive and active wireless transmission. Additionally, in one aspect of the present invention the lead-free idio-electrode bio-physiological adapter allows for utmost clinical operational freedom and dramatically obviates the needs for leads of any length that invariably encumber the acquisition and performance of electrocardiogram recordings of any sort.

Method for simultaneous full-wavefield inversion of gathers of source (or receiver) encoded geophysical data to determine a physical properties model for a subsurface region, especially suitable for surveys where fixed receiver geometry conditions were not satisfied in the data acquisition. First, a shallow time window of the data (202) where the fixed receiver condition is satisfied is inverted by simultaneous encoded (203) source inversion (205). Then, the deeper time window of the data (208) is inverted by sparse sequential source inversion (209), using the physical properties model from the shallow time window (206) as a starting model (207). Alternatively, the shallow time window model is used to simulate missing far offset data (211) producing a data set satisfying the stationary receiver assumption, after which this data set is source encoded (212) and inverted by simultaneous source inversion (214).

An encoded information reading (EIR) terminal can comprise a microprocessor electrically coupled to a system bus, a memory communicatively coupled to the microprocessor, an encoded information reading (EIR) device, and a wireless communication interface. The EIR device can be selected from the group consisting of a bar code reading device, an RFID reading device, and a card reading device. The EIR device can be configured to perform outputting raw message data containing an encoded message and / or outputting decoded message data corresponding to an encoded message. The wireless communication interface can comprise a radio frequency front end configured to perform receiving a first radio signal and / or transmitting a second radio signal. The radio frequency front end can be electrically coupled to an analog-to-digital converter (ADC) which can be electrically coupled to the system bus and / or to a digital-to-analog converter (DAC) which can be electrically coupled to the system bus. The microprocessor can be configured to execute a base-band encoder software program and / or a base-band decoder software program. The base-band encoder software program can be configured to produce a first encoded bit stream by performing at least one of the following functions: source encoding of a first bit stream, encryption, channel encoding, multiplexing, modulation, frequency spreading, and media access control. The DAC can be configured to output to the radio frequency front end an analog signal corresponding to the first encoded bit stream. The ADC can be configured to output a second encoded bit stream corresponding to an analog signal produced by the radio frequency front end. The base-band decoder software program can be configured to produce a second bit stream corresponding to the second encoded bit stream by performing at least one of the following functions: media access control, frequency de-spreading, de-modulation, de-multiplexing, channel decoding, decryption, and source decoding.

This invention relates to a coordinate transmission based on joint network- channel coding utilizing the maximum system transmission rate to structure an odd-even check matrix, which includes coding odd-even check matrixes of a source channel, a relay channel and a network channel, and the source node codes the source information with the source channel odd-even check matrix and broadcasts it to the relay and the destination nodes, the relay node uses the relay channel matrix to code its own information and send it to the destination node and uses the network code matrix to code the source coded and relay coded information to get network coded information to pick up extra information and carry out low density odd-even check coding it to be sent to the destination node, which decodes the information sent by the source node and the relay node jointly.

The invention discloses an audio lossless compression encoding and decoding method and belongs to the field of information source encoding and decoding. In the method, a signal is framed adaptively according to related conditions of the former frame and the later frame of the signal, wherein a frame signal after framing is a combination of signals with similar signal characteristics, so that an encoder can achieve higher compression efficiency and is good for later integer wavelet transform and linear predictive encoding. For lossless compression encoding, a signal can be completely reconstructed, and the complete reconstructed characteristic of the signal is guaranteed by adopting integer lifting wavelet transform. Compared with the prior art, after a related adaptive framing-based module and an integer lifting wavelet-based de-correlation module are introduced, redundant information in an original signal can be well de-correlated, and the generated compressed data contains less redundant information. Therefore, the audio lossless compression encoding and decoding method can greatly improve the compression ratio by low computation complexity.

IP multicast enabled devices, systems and methods for use on an end-to-end IP multicast-enabled network are disclosed. An IP multicast system, device and method operable on the network includes an IP multicast-engine, and storage for storing instruction sets to instruct the engine to send messages according to a select multicast application. A plurality of devices become members of an IP multicast group such that sending a message to a single multicast address can provide for the concurrent control of, and the delivery of the multicast message to, the devices of the group. Error conditions in a multicast source may be handled by preserving the multicast session resources, and reassigning a multicast source address from a faulty source encoding device to an alternate device.

A communication system and method with orthogonal, block encoding is provided. Encoded signals are transmitted by repeating transmissions of symbol blocks with a phase or sign change selected for each block from a sequence of phase or sign changes. Different symbols are transmitted using orthogonal sequences. The decoding uses different orthogonal sequences for separating the received encoded signals into corresponding separate channels. The orthogonal encoding is removed from the encoded transmitted signals and corresponding ones of the repeated symbols are added in successively received repeated blocks after the orthogonal encoding is removed. A transmitter uses a digital source encoder to encode information into symbols, and each symbol is repeated a preselected number of times to successively produce groups of repeated bits. Each repeat bit is changed in phase or sing by application of a sign or phase change determined by a selected assigned orthogonal code associated with the transmitter. The sign changed bits are interleaved from a number of such groups to successively generate a number of blocks, each composed of the different sign or phase changed bits of the preselected number of repeated groups and having a collective sign or phase change corresponding to a common sign change or phase shared by all bits of the block. The interleaved blocks then modulate a radio signal for transmission.

The invention belongs to the field of mobile communication source coding and discloses a sparse matrix based compressed sensing processing method for hyperspectral remote sensing images. The method includes the steps of wavelet transform, data type transform, quantization, sparse matrix compressed encoding, OMP (orthogonal matching pursuit) decoding, data type inverse transform, inverse quantization and wavelet inverse transform, and the eight steps are executed sequentially. The method can be used for processing two types of remote sensing images, wherein the remote sensing images of the first type are not subjected to geometric correction, and the remote sensing images of the second type are subjected to geometric correction. Data type transform and data type inverse transform are added to the whole processing method aiming at the remote sensing images of the second type. The method for processing the hyperspectral remote sensing images has the advantages of high image compression proportion, saving of memory and computation space and high image restoration quality.

The present invention discloses a baseband chip and a method to implement multi-mode switching for a mobile terminal based on the baseband chip, the baseband chip of the mobile terminal in the present invention includes three subsystems of a source encoding / decoding subsystem 102, a wireless transmitting / receiving subsystem 101, a wireless transmitting / receiving and source encoding / decoding common control and process subsystem 100, the baseband chip may be implemented by using a single chip based on CMOS technology. The baseband chip of the present invention are suitably used to a TD-SCDMA mode mobile terminal or a multi-mode mobile terminal including TD-SCDMA, the new requirements proposed to the mobile terminal of the third generation mobile communication technology are met, and repeated design are decreased; furthermore, the method of switching among different communication modes for the mobile terminal based on the above mentioned baseband chip in the present invention implements the switching among different communication modes for the mobile terminal.

Provided is a Digital Multimedia Broadcasting (DMB) transmitting / receiving apparatus. The DMB apparatus for a high-quality video service includes: a source encoder for source-encoding audio and video signals into base layer streams and enhancement layer streams, individually; a base layer transmission processor for receiving the base layer streams from the source encoder, performing system encoding, multiplexing encoded base layer streams into transport streams, and outputting base layer transport streams; an enhancement layer transmission processor for receiving the enhancement layer streams from the source encoder, performing system encoding, multiplexing the encoded enhancement layer streams into enhancement layer transport streams, and outputting enhancement layer transport streams; and a layered digital broadcasting transmitter for receiving the base layer transport streams and the enhancement layer transport streams from the base layer transmission processor and the enhancement layer transmission processor, and modulating the base and enhancement layer transport streams into digital broadcasting signals.

The invention relates to a code conversion method and a code conversion device. The code conversion method comprises the following steps of receiving a request of performing format conversion on source codes in a source coding format input by a user, and analyzing the source codes according to the source coding format, obtaining data character strings contained in the source codes and syntax character strings in the source coding format; converting the syntax character strings in the source coding format into syntax character strings corresponding to a target coding format; combining the data character strings with the syntax character strings corresponding to the target coding format so as to form target codes in the target coding format. Thus, if a section of source codes, written according to the source coding format and possessed with a certain function, is needed to be applied to a language platform of which the supporting coding format is different from the source coding format, just converting the source coding format of the source codes into the coding format supported by the language platform, therefore the converted codes can be applied to the language platform, and the labor cost can be reduced in comparison with the prior art.

The invention discloses an audio coder switching method which belongs to the multimedia signal processing and source encoding field. The method comprises that: a plurality of audio sequences are selected as training data; according to a preset rule and the training data, a classifier is trained; the rule is achieved by adopting a subjective tone quality evaluating method after a plurality of audio sequences are tested; when an audio signal is input, according to the trained result, the audio signal is classified by the classifier and switched to a corresponding audio coder for coding. The invention uses a subjective tone quality test result to train the classifier in order that the classifier can approach to subjective hearing and select a proper coder for switching, thereby overcoming the disadvantages of large calculating amount of an AMR-WB+closed loop switching method; compared with an AMR-WB+open loop switching method, the invention greatly improves the switching accuracy and ensures the coding quality.

A method and apparatus for performing multiple descriptive source coding in which a plurality of homogeneous encoders are advantageously employed in combination with a corresponding plurality of advantageously substantially identical decoders. In particular, diversity is provided to the multiple encoders by modifying the quantization process in at least one of the encoders such that the modified quantization process is based at least on a quantization error resulting from the quantization process of another one of the encoders. In this manner, diversity among the multiple bit streams is obtained, and in particular, the quality of a reconstructed signal based on a combination of multiple decoded bit streams at the receiver is advantageously superior to that based on any one of the decoded bit streams. In accordance with a first illustrative embodiment of the present invention, two Pulse Code Modulation (PCM) coders are employed. In accordance with a second illustrative embodiment of the present invention, two Adaptive Differential Pulse Code Modulation (ADPCM) coders are employed. And in accordance with a third illustrative embodiment of the present invention, two Low-Delay Code Excited Linear Prediction (LD-CELP) coders are employed. In each case, diversity is ensured by an appropriate modification to the quantization process of at least one of the encoders, and the total error may be advantageously reduced when decoded bit streams from both coders are combined at the receiver.

The invention provides a water sound frequency hopping communication method for linear modulation signal modulation and demodulation, and relates to a water sound frequency hopping communication. The water sound frequency hopping communication method comprises the following steps: carrying out source encoding on original information so as to obtain compressed digital signals; carrying out channel coding; using data obtained by the channel coding as a linear modulation signal control code; controlling to generate a linear modulation signal according to a frequency hopping sequence as a carrier for modulation, so as to obtain modulation signals; carrying out D / A (Digital / Analogue) conversion and power amplification on the modulation signals; converting the modulated signals into sound waves to spread in an ocean underwater acoustic channel through a transmission energy converter; receiving and converting acoustic signals spreading in the ocean underwater acoustic channel into electric signals through the receiving energy converter; obtaining stimulation signals through front amplification and band-pass filtering; carrying out A / D (Analogue / Digital) conversion on the received stimulation signals; after detecting that the digital signals after the A / D conversion are synchronous, carrying out linear modulation signal demodulation center frequency and demodulation frequency to obtain demodulation data; and carrying out channel decoding on the demodulation data; and carrying out source decoding on the digital signals to obtain information.

The invention discloses a non-orthogonal multi-carrier digital modulation and demodulation method and device. The method includes the following steps that an available channel is divided into multiple sub carrier channels; source encoding, channel encoding, mapping, multi-carrier modulation and guard interval addition processing is performed on external input data to obtain a multi-carrier digital signal; the multi-carrier digital signal is converted into an analog signal which is then amplified and finally emitted into a wireless channel; the received signal is converted into an electric signal and amplified, and then the analog signal is converted into a digital received signal; through a parameter estimation algorithm, the digital received signal is estimated, so that the frequencies, amplitude values and phases of multiple carriers are obtained, and then demodulation is completed, and data are output. According to the method and device, the defect that carrier frequency deviation influences an OFDM orthogonal multi-carrier system and other orthogonal multi-carrier systems is overcome, the high utilization rate of a frequency spectrum can be achieved, and good resistance to Doppler is achieved.

The disclosed signaling message transmitting and receiving method for mobile digital multimedia broadcast physical layer comprises: adding message source coding mode information into the signaling message, modulating and sending the signaling message; demodulating the signaling message to obtain the coding mode information. Wherein, it can use single carrier wave, data sub-carrier and pilot frequency in OFDM system. This invention reduces decoding complexity.

The invention discloses an energy-saving sensor data collection method based on cross layer safe compression. High energy efficiency and safety are achieved by means of the CS cross layer safe compression method. The method comprises a CS-based safe compression method and a cross layer combined optimization method. According to the CS-based safe compression method, encryption and integrity checking is combined with compressed sensing, and safety and low redundancy rate of data are ensured; cross layer combined optimization relates to the source encoding rate, transmission power and TDMA time slot assignment of a physical layer and a MAC layer, and energy minimization is achieved by means of the nonlinearity convex optimization algorithm. The energy-saving sensor data collection method based on cross layer safe compression achieves higher data safety, optimized energy efficiency, smaller transmission information amount, and smaller system complexity.

The invention relates to a short-wave communication system based on fountain codes and single carrier frequency domain equalization and a transmission method thereof, and belongs to the technical field related to wireless communication systems, communication signal processing and channel encoding and decoding. Fountain codes are employed in the short-wave data transmission system, and source encoding-channel encoding-modulation combined optimized technology is used, thereby effectively shortening the file transmission time delay in short-wave unidirectional non-feedback channels, enhancing the reliability of a communication system under severe channel environment, improving file transmission efficiency, and effectively ensuring data reliability. Single carrier frequency domain equalization technology based on time-frequency mixed channel estimation is employed to greatly reduce the complexity of a receiver of a single carrier system, and greatly increase the capability of the system to resist frequency domain selectivity fading due caused by multipath transmission of short-wave channels.

A method of performing a rate-distortion optimization process comprising, for each of a plurality of target image portions to be encoded in each of a plurality of frames, selecting a preferred one of a set of encoding modes by optimizing a function comprising an estimate of distortion for the target image portion and a measure of bit rate required to encode the target image portion, wherein the estimate of distortion is based on source coding distortion and an estimate of a distortion that would be experienced due to possible loss over the channel; encoding the target image portion into the encoded video stream using the selected encoding mode; and transmitting the encoded video stream over the channel. The rate-distortion optimization process for a current one of the frames is performed in dependence on feedback received from the receiving terminal based on an earlier one of the frames.

A method of encoding input data (D1) in an encoder to generate corresponding encoded data (E2) is provided. The method includes: (a) splitting and / or transforming the input data (D1) into one or more data chunks, and analyzing symbols present in the input data (D1) and compressing the symbols as a function of occurrence of the symbols in the one or more data chunks; (b) generating one or more code tables, and / or one or more frequency tables, and / or one or more length of code word tables for the symbols present in the one or more data chunks; (c) computing one or more sets of indices relating the symbols in each data chunk and / or the compressed symbols to entries in the one or more code tables, and / or the one or more frequency tables, and / or the one or more length of code word tables; and (d) assembling the one or more sets of indices, together with the one or more frequency tables, and / or the one or more code tables, and / or information indicative of such one or more tables, for generating the encoded data (E2). There is also provided an encoder (50) which utilizes the method, together with a corresponding decoder (60), wherein the encoder (50) and the decoder (60) in combination form a codec (100).

A computer-implemented method for encoding using feedback. The method comprising includes encoding a base layer of a current frame, generating a residue of the current frame, and encoding the residue of said current frame. Also, in response to feedback from a receiver, wherein the feedback is based on the base layer of the current frame and previous correctly received enhancement frames, determining coding strategy for each block of the current frame and coding according to one or more of source coding and Wyner-Ziv coding.

A method of encoding a transmission signal to reduce multiple access interference at a receiver. A data source signal including a plurality of source symbols is received into a transmitter. The source symbols are encoded using low energy encoding forming encoded codewords. An identifying pattern is applied to the encoded codewords forming a transmission signal. In one embodiment, the identifying pattern is a unique pseudo-random pattern identifying the data source. Once the pattern has been added, multiple transmission signals can be summed together and then modulated onto a carrier wave. The transmission signals are then transmitted on the carrier wave which is at a pre-selected frequency. The low energy encoding format is such that a sequence of signals forming codewords is created. Each sequence of signals is characterized by an energy and normally a high bit consumes energy whereas a low bit requires substantially less energy.

The invention proposes a form of video source coding that is particularly suitable for transmission over a channel at a low bit rate and with a high error rate, such as a wireless transmission channel. It proposes to use block coding with motion prediction and employing a plurality of reference pictures. The reference pictures are subjected to block effect filtering. A reference picture is selected when coding a picture. By using a plurality of reference pictures, the invention limits transmission channel time propagation errors. Because the reference pictures are subjected to block effect filtering, transmission timing errors are corrected independently of intrinsic errors caused by block coding.

The invention relates to the acquisition of seismic data using many seismic sources simultaneously or where the sources are emitting in an overlapping time frame but where it is desired to separate the data traces into source separated data traces. The key is having each seismic source emit a distinctive pattern of seismic energy that may all be discerned in the shot records of all of the seismic receivers. Distinctive patterns are preferably based on time / frequency pattern that is distinctive like an easily recognized song, but may include other subtle, but recognizable features such a phase differences, ancillary noise emissions, and physical properties of the vibes such a the weight and shape of the pad and the reaction mass and the performance of the hydraulic system and prime energy source.

A sending device receives applications data blocks to be transmitted to a receiving device according to a clock rate defined by transport periods, the applications data blocks being encoded before transmission by application of a first source encoding up to the processing of a change request. The sending device determines a drift between a number of applications data blocks received by the sending device and a number of applications data blocks transmitted after encoding by the sending device. The sending device determines, taking account of the determined drift, a transition applications data block from which a second source encoding is applied in response to the change request. The transmission of the stream is then adapted in such a way that the transmission of the encoded transition data block is aligned with a start of a transport period.