Data compression for satellite images

An efficient data compression system is presented for satellite pictures and two grey level pictures derived from satellite pictures. The compression techniques take advantages of the correlation between adjacent picture elements. Several source coding methods are investigated. Double delta coding is presented and shown to be the most efficient. Both predictive differential quantizing technique and double delta coding can be significantly improved by applying a background skipping technique. An extension code is constructed. This code requires very little storage space and operates efficiently. Simulation results are presented for various coding schemes and source codes

Implementation issues in source coding

An edge preserving image coding scheme which can be operated in both a lossy and a lossless manner was developed. The technique is an extension of the lossless encoding algorithm developed for the Mars observer spectral data. It can also be viewed as a modification of the DPCM algorithm. A packet video simulator was also developed from an existing modified packet network simulator. The coding scheme for this system is a modification of the mixture block coding (MBC) scheme described in the last report. Coding algorithms for packet video were also investigated

Optimum quantization

Design of optimum quantizers for quantizer-input message signal and quantizer-input message signal contaminated by nois

Predictive Encoding of Digitized TV Pictures

In this report, we consider the problem of efficient intra-frame encoding of digitized TV pictures using Differential Pulse Code Modulation (DPCM) type encoders. Our goal is to reduce the average number of bits used to encode a pixel while subjective quality of the picture still remains acceptable. To this end, we propose some new predictors and compare them with the existing ones using the error criteria "essential maximum" and "variance" of the prediction errors. The two criteria have also been compared with respect to the subjective quality of the final encoded pictures. We propose to design the quantizers using "mean n-th power" distortion criterion. Effects of "n" on bit rate and subjective quality of the encoded picture have been studied as it varies from 1 to 6. In this way, we achieve reductions up to 6 to 1 in the bit rate with little degradation in the picture quality. We also consider the use of second and third order entropies of the encoded pictures to reduce the bit rate. Next, we adapt the encoding procedure to local structural variations in the picture by first segmenting it in three areas of varying detail and then using the concepts developed for non-adaptive encoding to encode the three areas differently. The resulting encoder yields better picture quality than before and can be used to encode a broad class of pictures.</p

Shuttle/TDRSS Ku-band downlink study

Assessing the adequacy of the baseline signal design approach, developing performance specifications for the return link hardware, and performing detailed design and parameter optimization tasks was accomplished by completing five specific study tasks. The results of these tasks show that the basic signal structure design is sound and that the goals can be met. Constraints placed on return link hardware by this structure allow reasonable specifications to be written so that no extreme technical risk areas in equipment design are foreseen. A third channel can be added to the PM mode without seriously degrading the other services. The feasibility of using only a PM mode was shown to exist, however, this will require use of some digital TV transmission techniques. Each task and its results are summarized

Differential encoding techniques applied to speech signals

The increasing use of digital communication systems has produced a continuous search for efficient methods of speech encoding. This thesis describes investigations of novel differential encoding systems. Initially Linear First Order DPCM systems employing a simple delayed encoding algorithm are examined. The systems detect an overload condition in the encoder, and through a simple algorithm reduce the overload noise at the expense of some increase in the quantization (granular) noise. The signal-to-noise ratio (snr) performance of such d codec has 1 to 2 dB's advantage compared to the First Order Linear DPCM system. In order to obtain a large improvement in snr the high correlation between successive pitch periods as well as the correlation between successive samples in the voiced speech waveform is exploited. A system called "Pitch Synchronous First Order DPCM" (PSFOD) has been developed. Here the difference Sequence formed between the samples of the input sequence in the current pitch period and the samples of the stored decoded sequence from the previous pitch period are encoded. This difference sequence has a smaller dynamic range than the original input speech sequence enabling a quantizer with better resolution to be used for the same transmission bit rate. The snr is increased by 6 dB compared with the peak snr of a First Order DPCM codea. A development of the PSFOD system called a Pitch Synchronous Differential Predictive Encoding system (PSDPE) is next investigated. The principle of its operation is to predict the next sample in the voiced-speech waveform, and form the prediction error which is then subtracted from the corresponding decoded prediction error in the previous pitch period. The difference is then encoded and transmitted. The improvement in snr is approximately 8 dB compared to an ADPCM codea, when the PSDPE system uses an adaptive PCM encoder. The snr of the system increases further when the efficiency of the predictors used improve. However, the performance of a predictor in any differential system is closely related to the quantizer used. The better the quantization the more information is available to the predictor and the better the prediction of the incoming speech samples. This leads automatically to the investigation in techniques of efficient quantization. A novel adaptive quantization technique called Dynamic Ratio quantizer (DRQ) is then considered and its theory presented. The quantizer uses an adaptive non-linear element which transforms the input samples of any amplitude to samples within a defined amplitude range. A fixed uniform quantizer quantizes the transformed signal. The snr for this quantizer is almost constant over a range of input power limited in practice by the dynamia range of the adaptive non-linear element, and it is 2 to 3 dB's better than the snr of a One Word Memory adaptive quantizer. Digital computer simulation techniques have been used widely in the above investigations and provide the necessary experimental flexibility. Their use is described in the text

Optimum quantization.

Bibliography: p.63-65

Transform coding techniques and their application in JPEG scheme.

by Chun-tat See.Thesis (M.Phil.)--Chinese University of Hong Kong, 1991.Includes bibliographical references.ACKNOWLEDGEMENTS --- p.iABSTRACT --- p.iiNOTATIONS --- p.ivTABLE OF CONTENTS --- p.viChapter 1. --- INTRODUCTION --- p.1-1Chapter 1.1 --- Introduction --- p.1-1Chapter 1.2 --- A Basic Transform Coding System --- p.1-2Chapter 1.3 --- Thesis Organization --- p.1-5Chapter 2. --- DYADIC MATRICES AND THEIR APPLICATION --- p.2-1Chapter 2.1 --- Introduction --- p.2-1Chapter 2.2 --- Theory of Dyadic Matrix --- p.2-2Chapter 2.2.1 --- Basic Definitions --- p.2-3Chapter 2.2.2 --- Maximum Size of Dyadic Matrix --- p.2-8Chapter 2.3 --- Application of Dyadic Matrix in Generating Orthogonal Transforms --- p.2-13Chapter 2.3.1 --- Transform Performance Criteria --- p.2-14Chapter 2.3.2 --- "[T1] = [P]Diag([DM2(4)],[A(4)])[Q]" --- p.2-19Chapter 2.3.3 --- "[T2] = [P]Diag([DM2(4)],[DM2(4)])[Q]" --- p.2-21Chapter 2.4 --- Discussion and Conclusion --- p.2-26Chapter 3. --- LOW SEQUENCY COEFFICIENT TRUNCATION (LSCT) CODING SCHEME --- p.3-1Chapter 3.1 --- Introduction --- p.3-1Chapter 3.2 --- DC Coefficient Estimation Schemes --- p.3-2Chapter 3.2.1 --- Element Estimation --- p.3-2Chapter 3.2.2 --- Row Estimation --- p.3-4Chapter 3.2.3 --- Plane Estimation --- p.3-7Chapter 3.3 --- LSCT Coding Scheme 1 and Results --- p.3-11Chapter 3.4 --- LSCT Coding Scheme 2 and Results --- p.3-17Chapter 3.5 --- Discussions and Conclusions --- p.3-21Chapter 4. --- VARIABLE BLOCK SIZE (VBS) CODING SCHEME --- p.4-1Chapter 4.1 --- Introduction --- p.4-1Chapter 4.2 --- Chen's VBS Coding Scheme and Its Limitation --- p.4-3Chapter 4.3 --- VBS Coding Scheme With Block Size Determined Using Edge Discriminator --- p.4-6Chapter 4.4 --- Simulation Results --- p.4-8Chapter 4.5 --- Discussions and Conclusions --- p.4-12Chapter 5. --- ENHANCEMENT OF JPEG INTERNATIONAL STANDARD --- p.5-1Chapter 5.1 --- Introduction --- p.5-1Chapter 5.2 --- The Basic JPEG International Standard --- p.5-2Chapter 5.2.1 --- Level Shift and Discrete Cosine Transform --- p.5-4Chapter 5.2.2 --- Uniform Quantization --- p.5-5Chapter 5.2.3 --- Coefficient Coding --- p.5-7Chapter 5.3 --- Efficient DC Coefficients Encoding --- p.5-8Chapter 5.3.1 --- The Minimum Edge Difference (MED) Predictor --- p.5-8Chapter 5.3.2 --- Simulation Results --- p.5-9Chapter 5.3.3 --- Pixel Domain Predictors --- p.5-13Chapter 5.3.4 --- Discussion and Conclusion --- p.5-15Chapter 5.4 --- JPEG Scheme Using Variable Block Size Technique --- p.5-15Chapter 5.4.1 --- Scheme 1 --- p.5-16Chapter 5.4.2 --- Scheme 2 --- p.5-25Chapter 5.4.3 --- Scheme 3 --- p.5-27Chapter 5.4.4 --- Scheme 4 --- p.5-29Chapter 5.4.5 --- Scheme 5 --- p.5-32Chapter 5.4.6 --- Discussions and Conclusions --- p.5-32Chapter 5.5 --- Conclusions --- p.5-33Chapter 6. --- CONCLUSIONS --- p.6-1Chapter 6.1 --- Summary of Research Work --- p.6-1Chapter 6.2 --- Contributions of Work --- p.6-2Chapter 6.3 --- Suggestions for Further Research --- p.6-3Chapter 7. --- REFERENCES --- p.7-1RESULT

Channel state testing in information decoding.

Massachusetts Institute of Technology. Dept. of Electrical Engineering. Thesis. 1965. Ph.D.Ph.D

Evaluation of glottal characteristics for speaker identification.

Based on the assumption that the physical characteristics of people's vocal apparatus cause their voices to have distinctive characteristics, this thesis reports on investigations into the use of the long-term average glottal response for speaker identification. The long-term average glottal response is a new feature that is obtained by overlaying successive vocal tract responses within an utterance. The way in which the long-term average glottal response varies with accent and gender is examined using a population of 352 American English speakers from eight different accent regions. Descriptors are defined that characterize the shape of the long-term average glottal response. Factor analysis of the descriptors of the long-term average glottal responses shows that the most important factor contains significant contributions from descriptors comprised of the coefficients of cubics fitted to the long-term average glottal response. Discriminant analysis demonstrates that the long-term average glottal response is potentially useful for classifying speakers according to their gender, but is not useful for distinguishing American accents. The identification accuracy of the long-term average glottal response is compared with that obtained from vocal tract features. Identification experiments are performed using a speaker database containing utterances from twenty speakers of the digits zero to nine. Vocal tract features, which consist of cepstral coefficients, partial correlation coefficients and linear prediction coefficients, are shown to be more accurate than the long-term average glottal response. Despite analysis of the training data indicating that the long-term average glottal response was uncorrelated with the vocal tract features, various feature combinations gave insignificant improvements in identification accuracy. The effect of noise and distortion on speaker identification is examined for each of the features. It is found that the identification performance of the long-term average glottal response is insensitive to noise compared with cepstral coefficients, partial correlation coefficients and the long-term average spectrum, but that it is highly sensitive to variations in the phase response of the speech transmission channel. Before reporting on the identification experiments, the thesis introduces speech production, speech models and background to the various features used in the experiments. Investigations into the long-term average glottal response demonstrate that it approximates the glottal pulse convolved with the long-term average impulse response, and this relationship is verified using synthetic speech. Furthermore, the spectrum of the long-term average glottal response extracted from pre-emphasized speech is shown to be similar to the long-term average spectrum of pre-emphasized speech, but computationally much simpler