Beyond Traditional Transform Coding
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The Discrete Cosine Transform (DCT): Theory and Application
The Discrete Cosine Transform (DCT): 1 Theory and Application Syed Ali Khayam Department of Electrical & Computer Engineering Michigan State University March 10th 2003 1 This document is intended to be tutorial in nature. No prior knowledge of image processing concepts is assumed. Interested readers should follow the references for advanced material on DCT. ECE 802 – 602: Information Theory and Coding Seminar 1 – The Discrete Cosine Transform: Theory and Application 1. Introduction Transform coding constitutes an integral component of contemporary image/video processing applications. Transform coding relies on the premise that pixels in an image exhibit a certain level of correlation with their neighboring pixels. Similarly in a video transmission system, adjacent pixels in consecutive frames2 show very high correlation. Consequently, these correlations can be exploited to predict the value of a pixel from its respective neighbors. A transformation is, therefore, defined to map this spatial (correlated) data into transformed (uncorrelated) coefficients. Clearly, the transformation should utilize the fact that the information content of an individual pixel is relatively small i.e., to a large extent visual contribution of a pixel can be predicted using its neighbors. A typical image/video transmission system is outlined in Figure 1. The objective of the source encoder is to exploit the redundancies in image data to provide compression. In other words, the source encoder reduces the entropy, which in our case means decrease in the average number of bits required to represent the image. On the contrary, the channel encoder adds redundancy to the output of the source encoder in order to enhance the reliability of the transmission. -
Block Transforms in Progressive Image Coding
This is page 1 Printer: Opaque this Blo ck Transforms in Progressive Image Co ding Trac D. Tran and Truong Q. Nguyen 1 Intro duction Blo ck transform co ding and subband co ding have b een two dominant techniques in existing image compression standards and implementations. Both metho ds actually exhibit many similarities: relying on a certain transform to convert the input image to a more decorrelated representation, then utilizing the same basic building blo cks such as bit allo cator, quantizer, and entropy co der to achieve compression. Blo ck transform co ders enjoyed success rst due to their low complexity in im- plementation and their reasonable p erformance. The most p opular blo ck transform co der leads to the current image compression standard JPEG [1] which utilizes the 8 8 Discrete Cosine Transform DCT at its transformation stage. At high bit rates 1 bpp and up, JPEG o ers almost visually lossless reconstruction image quality. However, when more compression is needed i.e., at lower bit rates, an- noying blo cking artifacts showup b ecause of two reasons: i the DCT bases are short, non-overlapp ed, and have discontinuities at the ends; ii JPEG pro cesses each image blo ck indep endently. So, inter-blo ck correlation has b een completely abandoned. The development of the lapp ed orthogonal transform [2] and its generalized ver- sion GenLOT [3, 4] helps solve the blo cking problem to a certain extent by b or- rowing pixels from the adjacent blo cks to pro duce the transform co ecients of the current blo ck. -
16.1 Digital “Modes”
Contents 16.1 Digital “Modes” 16.5 Networking Modes 16.1.1 Symbols, Baud, Bits and Bandwidth 16.5.1 OSI Networking Model 16.1.2 Error Detection and Correction 16.5.2 Connected and Connectionless 16.1.3 Data Representations Protocols 16.1.4 Compression Techniques 16.5.3 The Terminal Node Controller (TNC) 16.1.5 Compression vs. Encryption 16.5.4 PACTOR-I 16.2 Unstructured Digital Modes 16.5.5 PACTOR-II 16.2.1 Radioteletype (RTTY) 16.5.6 PACTOR-III 16.2.2 PSK31 16.5.7 G-TOR 16.2.3 MFSK16 16.5.8 CLOVER-II 16.2.4 DominoEX 16.5.9 CLOVER-2000 16.2.5 THROB 16.5.10 WINMOR 16.2.6 MT63 16.5.11 Packet Radio 16.2.7 Olivia 16.5.12 APRS 16.3 Fuzzy Modes 16.5.13 Winlink 2000 16.3.1 Facsimile (fax) 16.5.14 D-STAR 16.3.2 Slow-Scan TV (SSTV) 16.5.15 P25 16.3.3 Hellschreiber, Feld-Hell or Hell 16.6 Digital Mode Table 16.4 Structured Digital Modes 16.7 Glossary 16.4.1 FSK441 16.8 References and Bibliography 16.4.2 JT6M 16.4.3 JT65 16.4.4 WSPR 16.4.5 HF Digital Voice 16.4.6 ALE Chapter 16 — CD-ROM Content Supplemental Files • Table of digital mode characteristics (section 16.6) • ASCII and ITA2 code tables • Varicode tables for PSK31, MFSK16 and DominoEX • Tips for using FreeDV HF digital voice software by Mel Whitten, KØPFX Chapter 16 Digital Modes There is a broad array of digital modes to service various needs with more coming. -
Video/Image Compression Technologies an Overview
Video/Image Compression Technologies An Overview Ahmad Ansari, Ph.D., Principal Member of Technical Staff SBC Technology Resources, Inc. 9505 Arboretum Blvd. Austin, TX 78759 (512) 372 - 5653 [email protected] May 15, 2001- A. C. Ansari 1 Video Compression, An Overview ■ Introduction – Impact of Digitization, Sampling and Quantization on Compression ■ Lossless Compression – Bit Plane Coding – Predictive Coding ■ Lossy Compression – Transform Coding (MPEG-X) – Vector Quantization (VQ) – Subband Coding (Wavelets) – Fractals – Model-Based Coding May 15, 2001- A. C. Ansari 2 Introduction ■ Digitization Impact – Generating Large number of bits; impacts storage and transmission » Image/video is correlated » Human Visual System has limitations ■ Types of Redundancies – Spatial - Correlation between neighboring pixel values – Spectral - Correlation between different color planes or spectral bands – Temporal - Correlation between different frames in a video sequence ■ Know Facts – Sampling » Higher sampling rate results in higher pixel-to-pixel correlation – Quantization » Increasing the number of quantization levels reduces pixel-to-pixel correlation May 15, 2001- A. C. Ansari 3 Lossless Compression May 15, 2001- A. C. Ansari 4 Lossless Compression ■ Lossless – Numerically identical to the original content on a pixel-by-pixel basis – Motion Compensation is not used ■ Applications – Medical Imaging – Contribution video applications ■ Techniques – Bit Plane Coding – Lossless Predictive Coding » DPCM, Huffman Coding of Differential Frames, Arithmetic Coding of Differential Frames May 15, 2001- A. C. Ansari 5 Lossless Compression ■ Bit Plane Coding – A video frame with NxN pixels and each pixel is encoded by “K” bits – Converts this frame into K x (NxN) binary frames and encode each binary frame independently. » Runlength Encoding, Gray Coding, Arithmetic coding Binary Frame #1 . -
Predictive and Transform Coding
Lecture 14: Predictive and Transform Coding Thinh Nguyen Oregon State University 1 Outline PCM (Pulse Code Modulation) DPCM (Differential Pulse Code Modulation) Transform coding JPEG 2 Digital Signal Representation Loss from A/D conversion Aliasing (worse than quantization loss) due to sampling Loss due to quantization Digital signals Analog signals Analog A/D converter signals Digital Signal D/A sampling quantization processing converter 3 Signal representation For perfect re-construction, sampling rate (Nyquist’s frequency) needs to be twice the maximum frequency of the signal. However, in practice, loss still occurs due to quantization. Finer quantization leads to less error at the expense of increased number of bits to represent signals. 4 Audio sampling Human hearing frequency range: 20 Hz to 20 Khz. Voice:50Hz to 2 KHz What is the sampling rate to avoid aliasing? (worse than losing information) Audio CD : 44100Hz 5 Audio quantization Sample precision – resolution of signal Quantization depends on the number of bits used. Voice quality: 8 bit quantization, 8000 Hz sampling rate. (64 kbps) CD quality: 16 bit quantization, 44100Hz (705.6 kbps for mono, 1.411 Mbps for stereo). 6 Pulse Code Modulation (PCM) The 2 step process of sampling and quantization is known as Pulse Code Modulation. Used in speech and CD recording. audio signals bits sampling quantization No compression, unlike MP3 7 DPCM (Differential Pulse Code Modulation) 8 DPCM (Differential Pulse Code Modulation) Simple example: Code the following value sequence: 1.4 1.75 2.05 2.5 2.4 Quantization step: 0.2 Predictor: current value = previous quantized value + quantized error. -
Speech Compression
information Review Speech Compression Jerry D. Gibson Department of Electrical and Computer Engineering, University of California, Santa Barbara, CA 93118, USA; [email protected]; Tel.: +1-805-893-6187 Academic Editor: Khalid Sayood Received: 22 April 2016; Accepted: 30 May 2016; Published: 3 June 2016 Abstract: Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed. Keywords: speech coding; voice coding; speech coding standards; speech coding performance; linear prediction of speech 1. Introduction Speech coding is a critical technology for digital cellular communications, voice over Internet protocol (VoIP), voice response applications, and videoconferencing systems. In this paper, we present an abridged history of speech compression, a development of the dominant speech compression techniques, and a discussion of selected speech coding standards and their performance. We also discuss the future evolution of speech compression and speech compression research. We specifically develop the connection between rate distortion theory and speech compression, including rate distortion bounds for speech codecs. We use the terms speech compression, speech coding, and voice coding interchangeably in this paper. The voice signal contains not only what is said but also the vocal and aural characteristics of the speaker. As a consequence, it is usually desired to reproduce the voice signal, since we are interested in not only knowing what was said, but also in being able to identify the speaker. -
Shannon's Coding Theorems
Saint Mary's College of California Department of Mathematics Senior Thesis Shannon's Coding Theorems Faculty Advisors: Author: Professor Michael Nathanson Camille Santos Professor Andrew Conner Professor Kathy Porter May 16, 2016 1 Introduction A common problem in communications is how to send information reliably over a noisy communication channel. With his groundbreaking paper titled A Mathematical Theory of Communication, published in 1948, Claude Elwood Shannon asked this question and provided all the answers as well. Shannon realized that at the heart of all forms of communication, e.g. radio, television, etc., the one thing they all have in common is information. Rather than amplifying the information, as was being done in telephone lines at that time, information could be converted into sequences of 1s and 0s, and then sent through a communication channel with minimal error. Furthermore, Shannon established fundamental limits on what is possible or could be acheived by a communication system. Thus, this paper led to the creation of a new school of thought called Information Theory. 2 Communication Systems In general, a communication system is a collection of processes that sends information from one place to another. Similarly, a storage system is a system that is used for storage and later retrieval of information. Thus, in a sense, a storage system may also be thought of as a communication system that sends information from one place (now, or the present) to another (then, or the future) [3]. In a communication system, information always begins at a source (e.g. a book, music, or video) and is then sent and processed by an encoder to a format that is suitable for transmission through a physical communications medium, called a channel. -
Information Theory and Coding
Information Theory and Coding Introduction – What is it all about? References: C.E.Shannon, “A Mathematical Theory of Communication”, The Bell System Technical Journal, Vol. 27, pp. 379–423, 623–656, July, October, 1948. C.E.Shannon “Communication in the presence of Noise”, Proc. Inst of Radio Engineers, Vol 37 No 1, pp 10 – 21, January 1949. James L Massey, “Information Theory: The Copernican System of Communications”, IEEE Communications Magazine, Vol. 22 No 12, pp 26 – 28, December 1984 Mischa Schwartz, “Information Transmission, Modulation and Noise”, McGraw Hill 1990, Chapter 1 The purpose of telecommunications is the efficient and reliable transmission of real data – text, speech, image, taste, smell etc over a real transmission channel – cable, fibre, wireless, satellite. In achieving this there are two main issues. The first is the source – the information. How do we code the real life usually analogue information, into digital symbols in a way to convey and reproduce precisely without loss the transmitted text, picture etc.? This entails proper analysis of the source to work out the best reliable but minimum amount of information symbols to reduce the source data rate. The second is the channel, and how to pass through it using appropriate information symbols the source. In doing this there is the problem of noise in all the multifarious ways that it appears – random, shot, burst- and the problem of one symbol interfering with the previous or next symbol especially as the symbol rate increases. At the receiver the symbols have to be guessed, based on classical statistical communications theory. The guessing, unfortunately, is heavily influenced by the channel noise, especially at high rates. -
Source Coding: Part I of Fundamentals of Source and Video Coding Full Text Available At
Full text available at: http://dx.doi.org/10.1561/2000000010 Source Coding: Part I of Fundamentals of Source and Video Coding Full text available at: http://dx.doi.org/10.1561/2000000010 Source Coding: Part I of Fundamentals of Source and Video Coding Thomas Wiegand Berlin Institute of Technology and Fraunhofer Institute for Telecommunications | Heinrich Hertz Institute Germany [email protected] Heiko Schwarz Fraunhofer Institute for Telecommunications | Heinrich Hertz Institute Germany [email protected] Boston { Delft Full text available at: http://dx.doi.org/10.1561/2000000010 Foundations and Trends R in Signal Processing Published, sold and distributed by: now Publishers Inc. PO Box 1024 Hanover, MA 02339 USA Tel. +1-781-985-4510 www.nowpublishers.com [email protected] Outside North America: now Publishers Inc. PO Box 179 2600 AD Delft The Netherlands Tel. +31-6-51115274 The preferred citation for this publication is T. Wiegand and H. Schwarz, Source Coding: Part I of Fundamentals of Source and Video Coding, Foundations and Trends R in Signal Processing, vol 4, nos 1{2, pp 1{222, 2010 ISBN: 978-1-60198-408-1 c 2011 T. Wiegand and H. Schwarz All rights reserved. No part of this publication may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, mechanical, photocopying, recording or otherwise, without prior written permission of the publishers. Photocopying. In the USA: This journal is registered at the Copyright Clearance Cen- ter, Inc., 222 Rosewood Drive, Danvers, MA 01923. Authorization to photocopy items for internal or personal use, or the internal or personal use of specific clients, is granted by now Publishers Inc for users registered with the Copyright Clearance Center (CCC). -
Explainable Machine Learning Based Transform Coding for High
1 Explainable Machine Learning based Transform Coding for High Efficiency Intra Prediction Na Li,Yun Zhang, Senior Member, IEEE, C.-C. Jay Kuo, Fellow, IEEE Abstract—Machine learning techniques provide a chance to (VVC)[1], the video compression ratio is doubled almost every explore the coding performance potential of transform. In this ten years. Although researchers keep on developing video work, we propose an explainable transform based intra video cod- coding techniques in the past decades, there is still a big ing to improve the coding efficiency. Firstly, we model machine learning based transform design as an optimization problem of gap between the improvement on compression ratio and the maximizing the energy compaction or decorrelation capability. volume increase on global video data. Higher coding efficiency The explainable machine learning based transform, i.e., Subspace techniques are highly desired. Approximation with Adjusted Bias (Saab) transform, is analyzed In the latest three generations of video coding standards, and compared with the mainstream Discrete Cosine Transform hybrid video coding framework has been adopted, which is (DCT) on their energy compaction and decorrelation capabilities. Secondly, we propose a Saab transform based intra video coding composed of predictive coding, transform coding and entropy framework with off-line Saab transform learning. Meanwhile, coding. Firstly, predictive coding is to remove the spatial intra mode dependent Saab transform is developed. Then, Rate- and temporal redundancies of video content on the basis of Distortion (RD) gain of Saab transform based intra video coding exploiting correlation among spatial neighboring blocks and is theoretically and experimentally analyzed in detail. Finally, temporal successive frames. -
Signal Processing, IEEE Transactions On
IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 50, NO. 11, NOVEMBER 2002 2843 An Improvement to Multiple Description Transform Coding Yao Wang, Senior Member, IEEE, Amy R. Reibman, Senior Member, IEEE, Michael T. Orchard, Fellow, IEEE, and Hamid Jafarkhani, Senior Member, IEEE Abstract—A multiple description transform coding (MDTC) comprehensive review of the literature in both theoretical and method has been reported previously. The redundancy rate distor- algorithmic development, see the comprehensive review paper tion (RRD) performance of this coding scheme for the independent by Goyal [1]. and identically distributed (i.i.d.) two-dimensional (2-D) Gaussian source has been analyzed using mean squared error (MSE) as The performance of an MD coder can be evaluated by the re- the distortion measure. At the small redundancy region, the dundancy rate distortion (RRD) function, which measures how MDTC scheme can achieve excellent RRD performance because fast the side distortion ( ) decreases with increasing redun- a small increase in redundancy can reduce the single description dancy ( ) when the central distortion ( ) is fixed. As back- distortion at a rate faster than exponential, but the performance ground material, we first present a bound on the RRD curve for of MDTC becomes increasingly poor at larger redundancies. This paper describes a generalization of the MDTC (GMDTC) scheme, an i.i.d Gaussian source with the MSE as the distortion mea- which introduces redundancy both by transform and through sure. This bound was derived by Goyal and Kovacevic [2] and correcting the error resulting from a single description. Its RRD was translated from the achievable region for multiple descrip- performance is closer to the theoretical bound in the entire range tions, which was previously derived by Ozarow [3]. -
Fundamental Limits of Video Coding: a Closed-Form Characterization of Rate Distortion Region from First Principles
1 Fundamental Limits of Video Coding: A Closed-form Characterization of Rate Distortion Region from First Principles Kamesh Namuduri and Gayatri Mehta Department of Electrical Engineering University of North Texas Abstract—Classical motion-compensated video coding minimum amount of rate required to encode video. On methods have been standardized by MPEG over the the other hand, a communication technology may put years and video codecs have become integral parts of a limit on the maximum data transfer rate that the media entertainment applications. Despite the ubiquitous system can handle. This limit, in turn, places a bound use of video coding techniques, it is interesting to note on the resulting video quality. Therefore, there is a that a closed form rate-distortion characterization for video coding is not available in the literature. In this need to investigate the tradeoffs between the bit-rate (R) paper, we develop a simple, yet, fundamental character- required to encode video and the resulting distortion ization of rate-distortion region in video coding based (D) in the reconstructed video. Rate-distortion (R-D) on information-theoretic first principles. The concept of analysis deals with lossy video coding and it establishes conditional motion estimation is used to derive the closed- a relationship between the two parameters by means of form expression for rate-distortion region without losing a Rate-distortion R(D) function [1], [6], [10]. Since R- its generality. Conditional motion estimation offers an D analysis is based on information-theoretic concepts, elegant means to analyze the rate-distortion trade-offs it places absolute bounds on achievable rates and thus and demonstrates the viability of achieving the bounds derives its significance.