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At least 469 records · Page 26
All sourcesNASA NTRSDOE OSTI
NASA NTRS1991

Compression of digital images over local area networks. Appendix 1: Item 3

Differential Pulse Code Modulation (DPCM) has been used with speech for many years. It has not been as successful for images because of poor edge performance. The only corruption in DPC is quantizer error but this corruption becomes quite large in the region of an edge because of the abrupt changes in the statistics of the signal. We introduce two improved DPCM schemes; Edge correcting DPCM and Edge Preservation Differential Coding. These two coding schemes will detect the edges and take action to correct them. In an Edge Correcting scheme, the quantizer error for an edge is encoded using a recursive quantizer with entropy coding and sent to the receiver as side information. In an Edge Preserving scheme, when the quantizer input falls in the overload region, the quantizer error is encoded and sent to the receiver repeatedly until the quantizer input falls in the inner levels. Therefore these coding schemes increase the bit rate in the region of an edge and require variable rate channels. We implement these two variable rate coding schemes on a token wing network. Timed token protocol supports two classes of messages; asynchronous and synchronous. The synchronous class provides a pre-allocated bandwidth and guaranteed response time. The remaining bandwidth is dynamically allocated to the asynchronous class. The Edge Correcting DPCM is simulated by considering the edge information under the asynchronous class. For the simulation of the Edge Preserving scheme, the amount of information sent each time is fixed, but the length of the packet or the bit rate for that packet is chosen depending on the availability capacity. The performance of the network, and the performance of the image coding algorithms, is studied.

Gorjala, Bhargavi↗
NASA NTRS1978

Effect of noisy phase reference on coherent detection of unbalanced QPSK signals

A phase error in the reference signal used in coherent detection of PSK signals causes a degradation in performance; this effect on bit error probability has been much discussed in the literature for BPSK and balanced QPSK systems. For unbalanced QPSK (UQPSK) systems, where the power and data rates in the two channels are unequal, and where the channel data symbols might also be different, the problem of computing the error probability is more complex, and only approximate solutions have been previously given. In this paper, exact expressions are derived for the conditional probability of error for the I and Q channels as a function of the channel power and data rate ratio. This is done for all combinations of Manchester and NRZ signalling schemes, and it is shown that in some cases the low data rate channel can be more seriously affected by a phase offset. Numerical results are also given for the average probability of error, assuming a biphase Costas loop is used for carrier synchronization.

Osborne, H. C.↗
NASA NTRS1988

Estimating Rates Of Single-Event Upsets

Mathematical model yields estimates of upper and lower bounds on rates of single-event upsets (SEU's) in logic circuits. SEU's are reversible changes in logic states, caused by single high-energy ions. Energetic ions passed through device along tracks perpendicular to surface, to determine parameters of model. Used to estimate rates caused by ions of various energies incident at arbitrary angles. Useful in design of integrated circuits as high-density memories for minimal susceptibility to bit errors.

Zoutendyk, John A.↗
NASA NTRS1988

Vehicle antenna for the mobile satellite experiment

A low profile, low cost, printed circuit, electronically steered, right hand circularly polarized phase array antenna system has been developed for the Mobile Satellite Experiment (MSAT-X) Program. The success of this antenna is based upon the development of a crossed-slot element array and detailed trade-off analyses for both the phased array and pointing system design. The optimized system provides higher gain at low elevation angles (20 degrees above the horizon) and broader frequency coverage (approximately 8 1/2 percent bandwidth) than is possible with a patch array. Detailed analysis showed that optimum performance could be achieved with a 19 element array of a triangular lattice geometry of 3.9 inch element spacing. This configuration has the effect of minimizing grating lobes at large scan angles plus it improves the intersatellite isolation. The array has an aperture 20 inches in diameter and is 0.75 inch thick overall, exclusive of the RF and power connector. The pointing system employs a hybrid approach that operates with both an external rate sensor and an internal error signal as a means of fine tuning the beam acquisition and track. Steering the beam is done electronically via 18, 3-bit diode phase shifters. A nineteenth phase shifter is not required as the center element serves as a reference only. Measured patterns and gain show that the array meets the stipulated performance specifications everywhere except at some low elevation angles.

Peng, Sheng Y.↗
NASA NTRS1989

High rate concatenated coding systems using bandwidth efficient trellis inner codes

High-rate concatenated coding systems with bandwidth-efficient trellis inner codes and Reed-Solomon (RS) outer codes are investigated for application in high-speed satellite communication systems. Two concatenated coding schemes are proposed. In one the inner code is decoded with soft-decision Viterbi decoding, and the outer RS code performs error-correction-only decoding (decoding without side information). In the other, the inner code is decoded with a modified Viterbi algorithm, which produces reliability information along with the decoded output. In this algorithm, path metrics are used to estimate the entire information sequence, whereas branch metrics are used to provide reliability information on the decoded sequence. This information is used to erase unreliable bits in the decoded output. An errors-and-erasures RS decoder is then used for the outer code. The two schemes have been proposed for high-speed data communication on NASA satellite channels. The rates considered are at least double those used in current NASA systems, and the results indicate that high system reliability can still be achieved.

Deng, Robert H.↗
NASA NTRS1994

An efficient system for reliably transmitting image and video data over low bit rate noisy channels

This research project is intended to develop an efficient system for reliably transmitting image and video data over low bit rate noisy channels. The basic ideas behind the proposed approach are the following: employ statistical-based image modeling to facilitate pre- and post-processing and error detection, use spare redundancy that the source compression did not remove to add robustness, and implement coded modulation to improve bandwidth efficiency and noise rejection. Over the last six months, progress has been made on various aspects of the project. Through our studies of the integrated system, a list-based iterative Trellis decoder has been developed. The decoder accepts feedback from a post-processor which can detect channel errors in the reconstructed image. The error detection is based on the Huber Markov random field image model for the compressed image. The compression scheme used here is that of JPEG (Joint Photographic Experts Group). Experiments were performed and the results are quite encouraging. The principal ideas here are extendable to other compression techniques. In addition, research was also performed on unequal error protection channel coding, subband vector quantization as a means of source coding, and post processing for reducing coding artifacts. Our studies on unequal error protection (UEP) coding for image transmission focused on examining the properties of the UEP capabilities of convolutional codes. The investigation of subband vector quantization employed a wavelet transform with special emphasis on exploiting interband redundancy. The outcome of this investigation included the development of three algorithms for subband vector quantization. The reduction of transform coding artifacts was studied with the aid of a non-Gaussian Markov random field model. This results in improved image decompression. These studies are summarized and the technical papers included in the appendices.

Costello, Daniel J., Jr.↗
NASA NTRS2006

Less-Complex Method of Classifying MPSK

An alternative to an optimal method of automated classification of signals modulated with M-ary phase-shift-keying (M-ary PSK or MPSK) has been derived. The alternative method is approximate, but it offers nearly optimal performance and entails much less complexity, which translates to much less computation time. Modulation classification is becoming increasingly important in radio-communication systems that utilize multiple data modulation schemes and include software-defined or software-controlled receivers. Such a receiver may "know" little a priori about an incoming signal but may be required to correctly classify its data rate, modulation type, and forward error-correction code before properly configuring itself to acquire and track the symbol timing, carrier frequency, and phase, and ultimately produce decoded bits. Modulation classification has long been an important component of military interception of initially unknown radio signals transmitted by adversaries. Modulation classification may also be useful for enabling cellular telephones to automatically recognize different signal types and configure themselves accordingly. The concept of modulation classification as outlined in the preceding paragraph is quite general. However, at the present early stage of development, and for the purpose of describing the present alternative method, the term "modulation classification" or simply "classification" signifies, more specifically, a distinction between M-ary and M'-ary PSK, where M and M' represent two different integer multiples of 2. Both the prior optimal method and the present alternative method require the acquisition of magnitude and phase values of a number (N) of consecutive baseband samples of the incoming signal + noise. The prior optimal method is based on a maximum- likelihood (ML) classification rule that requires a calculation of likelihood functions for the M and M' hypotheses: Each likelihood function is an integral, over a full cycle of carrier phase, of a complicated sum of functions of the baseband sample values, the carrier phase, the carrier-signal and noise magnitudes, and M or M'. Then the likelihood ratio, defined as the ratio between the likelihood functions, is computed, leading to the choice of whichever hypothesis - M or M'- is more likely. In the alternative method, the integral in each likelihood function is approximated by a sum over values of the integrand sampled at a number, 1, of equally spaced values of carrier phase. Used in this way, 1 is a parameter that can be adjusted to trade computational complexity against the probability of misclassification. In the limit as 1 approaches infinity, one obtains the integral form of the likelihood function and thus recovers the ML classification. The present approximate method has been tested in comparison with the ML method by means of computational simulations. The results of the simulations have shown that the performance (as quantified by probability of misclassification) of the approximate method is nearly indistinguishable from that of the ML method (see figure).

Hamkins, Jon↗
NASA NTRS2009

Hardware Implementation of Serially Concatenated PPM Decoder

A prototype decoder for a serially concatenated pulse position modulation (SCPPM) code has been implemented in a field-programmable gate array (FPGA). At the time of this reporting, this is the first known hardware SCPPM decoder. The SCPPM coding scheme, conceived for free-space optical communications with both deep-space and terrestrial applications in mind, is an improvement of several dB over the conventional Reed-Solomon PPM scheme. The design of the FPGA SCPPM decoder is based on a turbo decoding algorithm that requires relatively low computational complexity while delivering error-rate performance within approximately 1 dB of channel capacity. The SCPPM encoder consists of an outer convolutional encoder, an interleaver, an accumulator, and an inner modulation encoder (more precisely, a mapping of bits to PPM symbols). Each code is describable by a trellis (a finite directed graph). The SCPPM decoder consists of an inner soft-in-soft-out (SISO) module, a de-interleaver, an outer SISO module, and an interleaver connected in a loop (see figure). Each SISO module applies the Bahl-Cocke-Jelinek-Raviv (BCJR) algorithm to compute a-posteriori bit log-likelihood ratios (LLRs) from apriori LLRs by traversing the code trellis in forward and backward directions. The SISO modules iteratively refine the LLRs by passing the estimates between one another much like the working of a turbine engine. Extrinsic information (the difference between the a-posteriori and a-priori LLRs) is exchanged rather than the a-posteriori LLRs to minimize undesired feedback. All computations are performed in the logarithmic domain, wherein multiplications are translated into additions, thereby reducing complexity and sensitivity to fixed-point implementation roundoff errors. To lower the required memory for storing channel likelihood data and the amounts of data transfer between the decoder and the receiver, one can discard the majority of channel likelihoods, using only the remainder in operation of the decoder. This is accomplished in the receiver by transmitting only a subset consisting of the likelihoods that correspond to time slots containing the largest numbers of observed photons during each PPM symbol period. The assumed number of observed photons in the remaining time slots is set to the mean of a noise slot. In low background noise, the selection of a small subset in this manner results in only negligible loss. Other features of the decoder design to reduce complexity and increase speed include (1) quantization of metrics in an efficient procedure chosen to incur no more than a small performance loss and (2) the use of the max-star function that allows sum of exponentials to be computed by simple operations that involve only an addition, a subtraction, and a table lookup. Another prominent feature of the design is a provision for access to interleaver and de-interleaver memory in a single clock cycle, eliminating the multiple clock-cycle latency characteristic of prior interleaver and de-interleaver designs.

Moision, Bruce↗
NASA NTRS1973

Digital data detection and synchronization

The primary accomplishments have been in the analysis and simulation of receivers and bit synchronizers. It has been discovered that tracking rate effects play, a rather fundamental role in both receiver and synchronizer performance, but that data relating to recorder time-base-error, for the proper characterization of this phenomenon, is in rather short supply. It is possible to obtain operationally useful tape recorder time-base-error data from high signal-to-noise ratio tapes using synchronizers with relatively wideband tracking loops. Low signal-to-noise ratio tapes examined in the same way would not be synchronizable. Additional areas of interest covered are receiver false lock, cycle slipping, and other unusual phenomena, which have been described to some extent in this and earlier reports and simulated during the study.

Noack, T. L.↗
NASA NTRS1977

Communication system analysis for manned space flight

One- and two-dimensional adaptive delta modulator (ADM) algorithms are discussed and compared. Results are shown for bit rates of two bits/pixel, one bit/pixel and 0.5 bits/pixel. Pictures showing the difference between the encoded-decoded pictures and the original pictures are presented. The effect of channel errors on the reconstructed picture is illustrated. A two-dimensional ADM using interframe encoding is also presented. This system operates at the rate of two bits/pixel and produces excellent quality pictures when there is little motion. The effect of large amounts of motion on the reconstructed picture is described.

Schilling, D. L.↗
NASA NTRS1989

Mixture block coding with progressive transmission in packet video. Appendix 1: Item 2

Video transmission will become an important part of future multimedia communication because of dramatically increasing user demand for video, and rapid evolution of coding algorithm and VLSI technology. Video transmission will be part of the broadband-integrated services digital network (B-ISDN). Asynchronous transfer mode (ATM) is a viable candidate for implementation of B-ISDN due to its inherent flexibility, service independency, and high performance. According to the characteristics of ATM, the information has to be coded into discrete cells which travel independently in the packet switching network. A practical realization of an ATM video codec called Mixture Block Coding with Progressive Transmission (MBCPT) is presented. This variable bit rate coding algorithm shows how a constant quality performance can be obtained according to user demand. Interactions between codec and network are emphasized including packetization, service synchronization, flow control, and error recovery. Finally, some simulation results based on MBCPT coding with error recovery are presented.

Chen, Yun-Chung↗
NASA NTRS1993

Output data formatter for the Electronically Scanned Thinned Array Radiometer (ESTAR) instrument

A prototype Output Data Formatter (ODF) for the ESTAR (Electronically Scanned Thinned Array Radiometer) instrument has been designed and tested. It employs programmable logic devices to format and tag correlator data for transmission to Earth. After accepting 170 bits or correlator and error data in parallel, it appends an identification word and then serially passes the data to the Small Explorer Data System (SEDS) for transmission at a maximum rate of greater than 15 Mb/sec. Implemented with two reprogrammable field programmable gate arrays (FPGA's), each contained in a 132-pin plastic pin grid array (PGA) package, the design is cascadeable, fully testable, and low-power.

Chren, William A., Jr.↗
NASA NTRS1976

Convolutional coding at 50 Mbps for the Shuttle Ku-band return link

Error correcting coding is required for 50 Mbps data link from the Shuttle Orbiter through the Tracking and Data Relay Satellite System (TDRSS) to the ground because of severe power limitations. Convolutional coding has been chosen because the decoding algorithms (sequential and Viterbi) provide significant coding gains at the required bit error probability of one in 10 to the sixth power and can be implemented at 50 Mbps with moderate hardware. While a 50 Mbps sequential decoder has been built, the highest data rate achieved for a Viterbi decoder is 10 Mbps. Thus, five multiplexed 10 Mbps Viterbi decoders must be used to provide a 50 Mbps data rate. This paper discusses the tradeoffs which were considered when selecting the multiplexed Viterbi decoder approach for this application.

Batson, B. H.↗
NASA NTRS1985

Convolutionally-Coded Unbalanced QPSK Systems

Report discusses error-rate performance for three convolutionallycoded unbalanced quadriphase-shift-keying (UQPSK) communication systems with noisy carriers that introduce crosstalk. Systems analyzed unbalanced in sense that each transmits two data streams with different bit rates and (in some cases) different powers.

Divsalar, D.↗
NASA NTRS2000

A Real-Time High Performance Data Compression Technique For Space Applications

A high performance lossy data compression technique is currently being developed for space science applications under the requirement of high-speed push-broom scanning. The technique is also error-resilient in that error propagation is contained within a few scan lines. The algorithm is based on block-transform combined with bit-plane encoding; this combination results in an embedded bit string with exactly the desirable compression rate. The lossy coder is described. The compression scheme performs well on a suite of test images typical of images from spacecraft instruments. Hardware implementations are in development; a functional chip set is expected by the end of 2001.

Yeh, Pen-Shu↗
NASA NTRS1977

Adaptive delta modulation systems for video encoding

This paper describes several adaptive delta modulators designed to encode video signals. One- and two-dimensional ADM algorithms are discussed and compared. Results are shown for bit rates of 2 bits/pixel, 1 bit/pixel and 0.5 bits/pixel. Pictures showing the difference between the encoded-decoded pictures and the original pictures are presented. Results are also presented to illustrate the effect of channel errors on the reconstructed picture. A two-dimensional ADM using interframe encoding is also presented. This system operates at the rate of 2 bits/pixel and produces excellent quality pictures when there is little motion. We also describe and illustrate the effect of large amounts of motion on the reconstructed picture.

Lei, T.-L. R.↗
NASA NTRS2009

Error-Rate Bounds for Coded PPM on a Poisson Channel

Equations for computing tight bounds on error rates for coded pulse-position modulation (PPM) on a Poisson channel at high signal-to-noise ratio have been derived. These equations and elements of the underlying theory are expected to be especially useful in designing codes for PPM optical communication systems. The equations and the underlying theory apply, more specifically, to a case in which a) At the transmitter, a linear outer code is concatenated with an inner code that includes an accumulator and a bit-to-PPM-symbol mapping (see figure) [this concatenation is known in the art as "accumulate-PPM" (abbreviated "APPM")]; b) The transmitted signal propagates on a memoryless binary-input Poisson channel; and c) At the receiver, near-maximum-likelihood (ML) decoding is effected through an iterative process. Such a coding/modulation/decoding scheme is a variation on the concept of turbo codes, which have complex structures, such that an exact analytical expression for the performance of a particular code is intractable. However, techniques for accurately estimating the performances of turbo codes have been developed. The performance of a typical turbo code includes (1) a "waterfall" region consisting of a steep decrease of error rate with increasing signal-to-noise ratio (SNR) at low to moderate SNR, and (2) an "error floor" region with a less steep decrease of error rate with increasing SNR at moderate to high SNR. The techniques used heretofore for estimating performance in the waterfall region have differed from those used for estimating performance in the error-floor region. For coded PPM, prior to the present derivations, equations for accurate prediction of the performance of coded PPM at high SNR did not exist, so that it was necessary to resort to time-consuming simulations in order to make such predictions. The present derivation makes it unnecessary to perform such time-consuming simulations.

Moision, Bruce↗
NASA NTRS2000

Full Duplex, Spread Spectrum Radio System

The goal of this project was to support the development of a full duplex, spread spectrum voice communications system. The assembly and testing of a prototype system consisting of a Harris PRISM spread spectrum radio, a TMS320C54x signal processing development board and a Zilog Z80180 microprocessor was underway at the start of this project. The efforts under this project were the development of multiple access schemes, analysis of full duplex voice feedback delays, and the development and analysis of forward error correction (FEC) algorithms. The multiple access analysis involved the selection between code division multiple access (CDMA), frequency division multiple access (FDMA) and time division multiple access (TDMA). Full duplex voice feedback analysis involved the analysis of packet size and delays associated with full loop voice feedback for confirmation of radio system performance. FEC analysis included studies of the performance under the expected burst error scenario with the relatively short packet lengths, and analysis of implementation in the TMS320C54x digital signal processor. When the capabilities and the limitations of the components used were considered, the multiple access scheme chosen was a combination TDMA/FDMA scheme that will provide up to eight users on each of three separate frequencies. Packets to and from each user will consist of 16 samples at a rate of 8,000 samples per second for a total of 2 ms of voice information. The resulting voice feedback delay will therefore be 4 - 6 ms. The most practical FEC algorithm for implementation was a convolutional code with a Viterbi decoder. Interleaving of the bits of each packet will be required to offset the effects of burst errors.

Harvey, Bruce A.↗