Coding and Compression - University of Missouri
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CODING AND COMPRESSION
PRESENTED BY: PING CHEN
CECS401 UMC
DATE: April, 29 2000
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Coding and Compression
Introduction
Lossless Data Compression
Runlength, Huffman, Dictionary compression
Audio
ADPCM, LPC, CELP
Image
hierarchical coding, subband coding
MPEG
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Introduction
A key problem with multimedia is the huge
quantities of data that result from raw digitized
data of audio, image or video source.
The main goal for coding and compression is to
alleviate the storage, processing and
transmission costs for these data.
There are a variety of compression techniques
commonly used in the Internet and other
system.
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Introduction
The components of a system are
capturing, transforming, coding and
transmitting.
Sample
Transform
Coding
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Introduction
Sampling --- Analog to Digital Conversion.
An input signal is converted from some continuously
varying physical value(e.g. pressure in air, or
frequency or wavelength of light) into a continuously
electrical signal by some electro-mechanical device.
This continuously varying electrical signal can then
be converted to a sequence of digital values, called
samples, by some analog to digital conversion circuit.
Two factors determine the accuracy of the
sample with the original continuous signal:
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Introduction
The maximum rate at which we sample.
Based on Nyquist’s theorem, the digital sampling rate must
be twice of the highest frequency in continuous signal.
The number of bits used in each sample. (known as
the quantization level.)
however, it is often not necessary to capture all
frequencies in the original signal.
For example, voice is comprehensible with a much smaller
range of frequencies that we can actually hear.
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Introduction
The goal of transform is to decorrelate the
original signal, and this decorrelation results in
the signal energy being redistributed among
only a small set of transform coefficients.
The original data can be transformed in a
number of ways to make it easier to apply
certain compression techniques.
The most common transform in current
techniques are the Discrete Cosine Transform
and wavelet transform.
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Lossless Data Compression
Lossless means the reconstructed image doesn’t
lose any information according to the original
one.
There is a huge range of lossless data
compression techniques.
The common techniques used are:
runlength encoding
Huffman coding
dictionary techniques
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Lossless Data Compression
Runlength compression
Removing repetitions of values and replacing them
with a counter and single value.
Fairly simple to implement.
Its performance depends heavily on the input data
statistics. The more successive value it has, the more
space we can compress.
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Lossless Data Compression
Huffman compression
Use more less bits to represent the most frequently
occurring characters/codeword values, and more bits
for the less commonly occurring once.
It is the most widespread way of replacing a set of
fixed size code words with an optimal set of different
sized code words, based on the statistics of the input
data.
Sender and receiver must share the same codebook
which lists the codes and their compressed
representation.
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Lossless Data Compression
Dictionary compression
Look at the data as it arrives and form a dictionary.
when new input comes, it look up the dictionary. If the
new input existed, the dictionary position can be
transmitted; if not found, it is added to the dictionary
in a new position, and the new position and string is
sent out.
Meanwhile, the dictionary is constructed at the
receiver dynamically, so that there is no need to carry
out statistics or share a table separately.
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Audio
The input audio signal from a microphone is
passed through several stages:
firstly, a band pass filter is applied eliminating
frequencies in the signal that we are not interested
in.
then the signal is sampled, converting the analog
signal into a sequence of values.
This is then quantised, or mapped into one of a set
of fixed value.
These values are then coded for storage or
transmission.
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Audio
Some techniques for audio compression:
ADPCM
LPC
CELP
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Audio
ADPCM -- Adaptive Differential Pulse Code
Modulation
ADPCM allows for the compression of PCM encoded
input whose power varies with time.
Feedback of a reconstructed version of the input
signal is subtracted from the actual input signal,
which is quantised to give a 4 bits output value.
This compression gives a 32 kbit/s output rate.
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Audio
Transmitter
Original
Em
Xm
+
Em*
Qunatizer
Coder
Channel
+
Xm'
Xm*
Predictor
+
Receiver
Channel
Reconstructed
Em*
Decoder
+
Xm*
+
Xm'
Predictor
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Audio
LPC -- Linear Predictive Coding
The encoder fits speech to a simple, analytic model
of the vocal tract. Only the parameters describing the
best-fit model is transmitted to the decoder.
An LPC decoder uses those parameters to generate
synthetic speech that is usually very similar to the
original.
LPC is used to compress audio at 16 Kbit/s and
below.
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Audio -- CELP
CELP -- Code Excited Linear Predictor
CELP does the same LPC modeling but then
computers the errors between the original speech
and the synthetic model and transmits both model
parameters and a very compressed representation of
the errors.
The result of CELP is a much higher quality speech at
low data rate.
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Image
Hierarchical Coding
based on the idea that coding will be in the form of
quality hierarchy where the lowest layer of hierarchy
contains the minimum information for intelligibility.
It is ideal for transmission over packet switched
network, low level packets can be filtered out
wherever a low bandwidth link is encountered and
still delivering a better quality to sites.
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Image
Subband Coding
an example of an encoding algorithm that can map
onto hierarchical coding.
based on the fact that the low spatial frequencies
components of a picture do carry most of the
information within the picture.
The picture can thus be divided into its spatial
frequencies components.
Allocate each subband to one of the hierarchy layers.
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