76 research outputs found
Study and simulation of low rate video coding schemes
The semiannual report is included. Topics covered include communication, information science, data compression, remote sensing, color mapped images, robust coding scheme for packet video, recursively indexed differential pulse code modulation, image compression technique for use on token ring networks, and joint source/channel coder design
Traffic Management and Congestion Control in the ATM Network Model.
Asynchronous Transfer Mode (ATM) networking technology has been chosen by the International Telegraph and Telephony Consultative Committee (CCITT) for use on future local as well as wide area networks to handle traffic types of a wide range. It is a cell based network architecture that resembles circuit switched networks, providing Quality of Service (QoS) guarantees not normally found on data networks. Although the specifications for the architecture have been continuously evolving, traffic congestion management techniques for ATM networks have not been very well defined yet. This thesis studies the traffic management problem in detail, provides some theoretical understanding and presents a collection of techniques to handle the problem under various operating conditions. A detailed simulation of various ATM traffic types is carried out and the collected data is analyzed to gain an insight into congestion formation patterns. Problems that may arise during migration planning from legacy LANs to ATM technology are also considered. We present an algorithm to identify certain portions of the network that should be upgraded to ATM first. The concept of adaptive burn-in is introduced to help ease the computational costs involved in virtual circuit setup and tear down operations
A software based, 13 kbits/s real-time internet codec
Due to the character of the original source materials and the nature of batch digitization, quality control issues may be present in this document. Please report any quality issues you encounter to [email protected], referencing the URI of the item.Includes bibliographical references: p. 50-53.Issued also on microfiche from Lange Micrographics.Bandwidth usage is a prime concern to many on the Internet, especially for users on low bit rate channels. As video conferencing becomes more popular, the need for efficient software based compression of video and audio becomes more important. This work develops a scalable, real-time, software based speech codec for use on desktop computers. The system is based on subband coding, adaptive prediction, and Huffman coding, and is capable of bit rates below 13 kbits/s for communications quality audio. The quality may be 'scaled" up by allocating additional bits to the subbands. This coder has been successfully implemented in real-time on a Sun Sparc 10 platform
Performance of VBR packer video communications on an ethernet LAN: A trace-driven simulation study
Provision of multimedia communication services on today’s packet-switched network infrastructure is becoming increasingly feasible. However, there remains a lack of information regarding the performance of multimedia sources operating in bursty data traffic conditions. In this study, a videotelephony system deployed on the Ethernet LAN is simulated, employing high time-resolution LAN traces as the data traffic load. In comparison with Poisson traffic models, the trace-driven cases produce highly variable packet delays, and higher packet loss, thereby degrading video traffic performance. In order to compensate for these effects, a delay control scheme based on a timed packet dropping algorithm is examined. Simulations of the scheme indicate that improvements in real time loss rates of videotelphony sources can be achieved
Time and frequency domain algorithms for speech coding
The promise of digital hardware economies (due to recent advances in
VLSI technology), has focussed much attention on more complex and sophisticated
speech coding algorithms which offer improved quality at relatively
low bit rates.
This thesis describes the results (obtained from computer simulations)
of research into various efficient (time and frequency domain) speech
encoders operating at a transmission bit rate of 16 Kbps.
In the time domain, Adaptive Differential Pulse Code Modulation (ADPCM)
systems employing both forward and backward adaptive prediction were
examined. A number of algorithms were proposed and evaluated, including
several variants of the Stochastic Approximation Predictor (SAP). A
Backward Block Adaptive (BBA) predictor was also developed and found to
outperform the conventional stochastic methods, even though its complexity
in terms of signal processing requirements is lower. A simplified
Adaptive Predictive Coder (APC) employing a single tap pitch predictor
considered next provided a slight improvement in performance over ADPCM,
but with rather greater complexity.
The ultimate test of any speech coding system is the perceptual performance
of the received speech. Recent research has indicated that this
may be enhanced by suitable control of the noise spectrum according to
the theory of auditory masking. Various noise shaping ADPCM
configurations were examined, and it was demonstrated that a proposed
pre-/post-filtering arrangement which exploits advantageously the
predictor-quantizer interaction, leads to the best subjective
performance in both forward and backward prediction systems.
Adaptive quantization is instrumental to the performance of ADPCM systems.
Both the forward adaptive quantizer (AQF) and the backward oneword
memory adaptation (AQJ) were examined. In addition, a novel method
of decreasing quantization noise in ADPCM-AQJ coders, which involves the
application of correction to the decoded speech samples, provided
reduced output noise across the spectrum, with considerable high frequency
noise suppression.
More powerful (and inevitably more complex) frequency domain speech
coders such as the Adaptive Transform Coder (ATC) and the Sub-band Coder
(SBC) offer good quality speech at 16 Kbps. To reduce complexity and
coding delay, whilst retaining the advantage of sub-band coding, a novel
transform based split-band coder (TSBC) was developed and found to compare
closely in performance with the SBC.
To prevent the heavy side information requirement associated with a
large number of bands in split-band coding schemes from impairing coding
accuracy, without forgoing the efficiency provided by adaptive bit
allocation, a method employing AQJs to code the sub-band signals together
with vector quantization of the bit allocation patterns was also
proposed.
Finally, 'pipeline' methods of bit allocation and step size estimation
(using the Fast Fourier Transform (FFT) on the input signal) were examined.
Such methods, although less accurate, are nevertheless useful in
limiting coding delay associated with SRC schemes employing Quadrature
Mirror Filters (QMF)
Secure mobile radio communication over narrowband RF channel.
by Wong Chun Kau, Jolly.Thesis (M.Phil.)--Chinese University of Hong Kong, 1992.Includes bibliographical references (leaves 84-88).ABSTRACT --- p.1ACKNOWLEDGEMENT --- p.3Chapter 1. --- INTRODUCTION --- p.7Chapter 1.1 --- Land Mobile Radio (LMR) CommunicationsChapter 1.2 --- Paramilitary Communications SecurityChapter 1.3 --- Voice Scrambling MethodsChapter 1.4 --- Digital Voice EncryptionChapter 1.5 --- Digital Secure LMRChapter 2. --- DESIGN GOALS --- p.20Chapter 2.1 --- System Concept and ConfigurationChapter 2.2 --- Operational RequirementsChapter 2.2.1 --- Operating conditionsChapter 2.2.2 --- Intelligibility and speech qualityChapter 2.2.3 --- Field coverage and transmission delayChapter 2.2.4 --- Reliability and maintenanceChapter 2.3 --- Functional RequirementsChapter 2.3.1 --- Major system featuresChapter 2.3.2 --- Cryptographic featuresChapter 2.3.3 --- Phone patch facilityChapter 2.3.4 --- Mobile data capabilityChapter 2.4 --- Bandwidth RequirementsChapter 2.5 --- Bit Error Rate RequirementsChapter 3. --- VOICE CODERS --- p.38Chapter 3.1 --- Digital Speech Coding MethodsChapter 3.1.1 --- Waveform codingChapter 3.1.2 --- Linear predictive codingChapter 3.1.3 --- Sub-band codingChapter 3.1.4 --- VocodersChapter 3.2 --- Performance EvaluationChapter 4. --- CRYPTOGRAPHIC CONCERNS --- p.52Chapter 4.1 --- Basic Concepts and CryptoanalysisChapter 4.2 --- Digital Encryption TechniquesChapter 4.3 --- Crypto SynchronizationChapter 4.3.1 --- Auto synchronizationChapter 4.3.2 --- Initial synchronizationChapter 4.3.3 --- Continuous synchronizationChapter 4.3.4 --- Hybrid synchronizationChapter 5. --- DIGITAL MODULATION --- p.63Chapter 5.1 --- Narrowband Channel RequirementsChapter 5.2 --- Narrowband Digital FMChapter 5.3 --- Performance EvaluationChapter 6. --- SYSTEM IMPLEMENTATION --- p.71Chapter 6.1 --- Potential EMC ProblemsChapter 6.2 --- Frequency PlanningChapter 6.3 --- Key ManagementChapter 6.4 --- Potential Electromagnetic Compatibility (EMC) ProblemsChapter 7. --- CONCLUSION --- p.80LIST OF ILLUSTRATIONS --- p.81REFERENCES --- p.82APPENDICES --- p.89Chapter I. --- Path Propagation Loss(L) Vs Distance (d)Chapter II. --- Speech Quality Assessment Tests performedby Special Duties Unit (SDU
A comparison of adaptive predictors in sub-band coding
Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1987.Bibliography: leaves 82-85.by Paul Ning.M.S
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