Arithmetic coding is a common algorithm used in both lossless and lossy data compression algorithms. It is an entropy encoding technique, in which the frequently seen symbols are encoded with fewer bits than rarely seen symbols. It has some advantages over wellknown techniques like Huffman coding. This article will describe the CACM87 implementation of arithmetic coding in detail, giving you a good understanding of all the details needed to implement it.
On a historical note, this post is an update of an article I wrote over twenty years ago. That article was published in the print edition of Dr. Dobb’s Journal, which meant that a lot of editing was done in order to avoid excessive page count. In particular, that Dr. Dobb’s piece combined two topics: a description of arithmetic coding along with a discussion of compression using PPM (Prediction by Partial Matching).
Because this new piece will be published on the web, space considerations are no longer a big factor, which I hope will allow me to do justice to the details of arithmetic coding. PPM, a worthy topic of its own, will be discussed in a later article. I hope that this new effort will be, while annoyingly long, the thorough explanation of the subject I wanted to do in 1991.
I think the best way to understand arithmetic coding is to break it into two parts, and I’ll use that idea in this article. First I will give a description of how arithmetic coding works, using regular floating point arithmetic implemented using standard C++ data types. This allows for a completely understandable, but slightly impractical implementation. In other words, it works, but it can only be used to encode very short messages.
The second section of the article will describe an implementation in which we switch to doing a special type of math on unbounded binary numbers. This is a somewhat mindboggling topic in itself, so it helps if you already understand arithmetic coding – you don’t have get hung up trying to learn two things at once.
To wrap up I will present working sample code written in modern C++. It won’t necessarily be the most optimized code in the world, but it is portable and easy to add to your existing projects. It should be perfect for learning and experimenting with this coding technique.
Fundamentals
The first thing to understand about arithmetic coding is what it produces. Arithmetic coding takes a message (often a file) composed of symbols (nearly always eightbit characters), and converts it to a floating point number greater than or equal to zero and less than one. This floating point number can be quite long – effectively your entire output file is one long number – which means it is not a normal data type that you are used to using in conventional programming languages. My implementation of the algorithm will have to create this floating point number from scratch, bit by bit, and likewise read it in and decode it bit by bit.
This encoding process is done incrementally. As each character in a file is encoded, a few bits will be added to the encoded message, so it is built up over time as the algorithm proceeds.
The second thing to understand about arithmetic coding is that it relies on a model to characterize the symbols it is processing. The job of the model is to tell the encoder what the probability of a character is in a given message. If the model gives an accurate probability of the characters in the message, they will be encoded very close to optimally. If the model misrepresents the probabilities of symbols, your encoder may actually expand a message instead of compressing it!
Encoding With Floating Point Math
The term arithmetic coding has to cover two separate processes: encoding messages and decoding them. I’ll start by looking at the encoding process with sample C++ code that implements the algorithm in a very limited form using C++ double
data. The code in this first section is only useful for exposition – don’t try to do any real compression with it.
To perform arithmetic encoding, we first need to define a proper model. Remember that the function of the model is to provide probabilities of a given character in a message. The conceptual idea of an arithmetic coding model is that each symbol will own its own unique segment of the number line of real numbers between 0 and 1. It’s important to note that there are many different ways to model character probabilities. Some models are static, never changing. Others are updated after every character is processed. The only two things that matter to us are that 1) the model attempts to accurately predict the probability a character will appear, and 2) the encoder and decoder have identical models at all times.
As an example, we can start with an encoder that can only encode an alphabet of 100 different characters. In a simple static model we will start with capital letters, then the lower case letters. This means that the first symbol, ‘A’, will own the number line from 0 to .01, ‘B’ will own .01 to .02, and so on. (In all cases, this is strictly a halfclosed interval, so the probability range for ‘A’ is actually >= 0 and < .01.)
With this model my encoder can represent the single letter ‘B’ by outputting a floating point number that is less than .02 and greater than or equal to .01. So for example, an arithmetic encoder that wanted to create that single letter could output .15 and be done.
An encoder that just outputs single characters is not much use though. To encode a string of symbols involves a slightly more complicated process. In this process, the first character defines a range of the number line that corresponds to the section assigned to it by the model. For the character ‘B’, that means the message is between .01 and .02.
The next character in the message then further divides that existing range proportionate to its current ownership of the number line. So some other letter that owns the very end of the number line, from .99 to 1.0 would change the range from [.01,.02) to [.0199, .020). This progressive subdividing of the range is just simple multiplication and addition, and is best understand with a simple code sample. My first pass in C++, which is far from a working encoder, might look like this:
double high = 1.0; double low = 0.0; char c; while ( input >> c ) { std::pair<double,double> p = model.getProbability(c); double range = high  low; high = low + range * p.second; low = low + range * p.first; } output << low + (highlow)/2;
After the entire message has been processed, we have a final range, [low,high). The encoder outputs a floating point number right in the center of that range.
Examining the Floating Point Prototype
The first pass encoder is demonstrated in the attached project as fp_proto.cpp
. To get it working I also needed to define a simple model. In this case I've created a model that can encode 100 characters, with each having a fixed probability of .01, starting with 'A' in the first position. To keep things simple I've only fleshed the class out enough to encode the capital letters from the ASCII character set:
struct { static std::pair<double,double> getProbability( char c ) { if (c >= 'A' && c <= 'Z') return std::make_pair( (c  'A') * .01, (c  'A') * .01 + .01); else throw "character out of range"; } } model;
So in this probability model, 'A' owns the range from 0.0 to 0.01, 'B' from .01 to .02, 'C' from .02 to .03, and so on. (Note that this is not an accurate or effective model, but its simplicity is useful at this point.) For a representative example, I called this encoder with the string "WXYZ". Let's walk through what happens in the encoder.
We start with high
and low
set to 1.0 and 0.0. The encoder calls the model to get the probabilities for letter 'W', which returns the interval [0.22, 0.23)  the range along the probability line that 'W' owns in this model. If you step over the next two lines, you'll see that low
is now set to 0.22, and high
is set to 0.23.
If you examine how this works, you'll see that as each character is encoded, the range between high
and low
becomes narrower and narrower, but high
will always be greater than low
. Additionally, the value of low
is always increasing, and value of high
is always decreasing. These invariants are important in getting the algorithm to work properly.
So after the first character is encoded, we know that the no matter what other values are encoded, the final number in the message will be less than .23 and greater than or equal to .22. Both low
and high
will be greater than equal to 0.22 and less than .23, and low
will be strictly less than high
. This means that when decoding, we are going to be able to determine that the first character is 'W' no matter what happens after this, because the final encoded number will fall into the range owned by 'W'. The narrowing process is roughly shown in the figure below:
Let's see how this narrowing works when we process the second character, 'X'. The model returns a range of [.23, .24) for this character, and the subsequent recalculation of high
and low
results in and interval of [.2223, .2224). So high
and low
are still inside the original range of [.22, .23), but the interval has narrowed.
After the final two characters are included, the output looks like:
Encoded message: 0.2223242550
I'll talk more about how the exact value we want to output needs to be chosen, but in theory at least, for this particular message, any floating point number in the interval [0.22232425,0.22232426) should properly decode to the desired values.
Decoding With Floating Point Math
I find the encoding algorithm to be very intuitive. The decoder reverses the process, and is no more complicated, but the steps might not seem quite as obvious. A first pass algorithm at decoding this message would look something like this:
void decode(double message) { double high = 1.0; double low = 0.0; for ( ; ; ) { double range = high  low; char c = model.getSymbol((message  low)/range); std::cout << c; if ( c == 'Z' ) return; std::pair<double,double> p = model.getProbability(c); high = low + range * p.second; low = low + range * p.first; } }
The math in the decoder basically reverses the math from the encode side. To decode a character, the probability model just has to find the character whose range covers the current value of the message. When the decoder first starts up with the sample value of 0.22232425, the model sees that the value falls between the interval owned by 'W': [0.22,0.23), so the model returns W. In fp_proto.cpp
, the decoder portion of the simple model looks like this:
static char getSymbol( double d) { if ( d >= 0.0 && d < 0.26) return 'A' + static_cast<int>(d*100); else throw "message out of range"; }
In the encoder, we continually narrow the range of the output value as each character is processed. In the decoder we do the same narrowing of the portion of the message we are inspecting for the next character. After the 'W' is decoded, high
and low
will now define an interval of [0.22,0.23), with a range of .01. So the formula that calculates the next probability value to be decoded, (message  low)/range
, will be .2324255, which lands right in the middle of of the range covered by 'X'.
This narrowing continues as the characters are decoded, until the hardcoded end of message, letter 'Z' is reached. Success!
Notes and Summary So Far
The algorithm demonstrated to this point can be tested with fp_proto.cpp
in the attached project. You need to be very aware that we have seen so far is not really a valid or useful encoder. It does, however, demonstrate with some accuracy the general flow of the algorithm. The key observations so far are:
 Characters are encoded according to their position in the probability range [0, 1).
 We keep track of the current state using variables
high
andlow
. A valid output result will be any number in the range [low
,high
).  As each character is encoded, it compresses the range [
low
,high
) in a way that corresponds exactly to its position in the probability range.  We have a few invariants that result from the math used in the algorithm. The value of
low
never decreases, the value ofhigh
never increases, andlow
is always less thanhigh
.
The big problem with this demonstration algorithm is that it depends on C++ double
values to hold the message state. Floating point variables have limited precision, which means that eventually, as the range between high
and low
continues to narrow, the distance between them becomes too small to represent with floating point variables. With the model used here, you can encode 10 characters or so, but after that the algorithm won't work.
The rest of this article will present a fully functional algorithm. Conceptually it will look very much like the code already presented. The big difference is that the variables used in the encoder and decoder, high
, low
, and message
, are no longer going to be of the C++ type double
. Instead they will be arbitrarily long binary variables.
In order for our program to handle binary numbers of arbitrary length, they will be processed a bit at a time, with bits being read in, calculations being performed, and then bits being output and then discarded as they are no longer needed. The details of how this is done are where all of the work comes into play in this algorithm.
In addition to those modifications, the rest of this article will cover a few other points that have not been dealt with yet, including:
 How to deal with the end of the stream.
 How to choose the actual output number from the range [low, high).
 Why arithmetic coding is superior to Huffman coding as an entropy coder.
Unbounded Precision Math With Integers
In this, the second part of the exposition, you are going to have to wrap your head around some unconventional mathematics. The algorithm that was described in the first part of this article is still going to be faithfully executed, but it will no longer be implemented using variables of type double
. It will instead be implemented with integer variables and integer math, albeit in interesting ways.
The basic concept that we implement is this: the values of high
, low
, and the actual encoded message, are going to be binary numbers of unbounded length. In other words, by the time we finish encoding the Project Gutenberg copy of Moby Dick, high
and low
will be millions of bits long, and the output value itself will be millions of bits long. All three will still represent a number greater than or equal to 0, and less than 1.
An even more interesting facet of this is that even though the three numbers in play are millions of bits long, each time we process a character we will only do a few operations of simple integer math  32 bits, or perhaps 64 if you like.
Number Representation
Recall that in the reference version of the album, low and high were initialized like this:
double high = 1.0; double low = 0.0;
In the integer version of the algorithm, we switch to a representation like this:
unsigned int high = 0xFFFFFFFFU; unsigned int low = 0;
Both numbers have an implied decimal point leading their values, which would mean that high
is actually (in hex) 0.FFFFFFFF, or in binary, 0.1111...1111, and low
is 0.0. The number that we output will likewise have an implied decimal point before the first bit.
But this is not quite right  in the first implementation high
was 1.0. The value 0.FFFFFFF is close, but it is just a bit less than 1.0. How is this dealt with?
This is where a bit of mindbending math comes into play. Although high
has 32 bits in memory, we consider it to have an infiinitely long trail of binary 1's trailing off the right end. So it isn't just 0.FFFFFFFF, that string of F's (or 1's) continues off into infinity  they are out there, but haven't been shifted into memory yet.
And while it may not be obvious, Google can help convince you that an infinite string of of 1's starting with 0.1 is actually equal to 1. (The short story: 2x  x = 1.0, therefore x = 1.0.) Likewise, low
is considered to be an infinitely long binary string, with 0's hanging off the end out to the last binary place.
Resulting Changes to the Model
Remember that the final implementation is going to be entirely implemented using integer math. Previously, the model returned probabilities as a pair of floating point numbers representing the range that a particular symbol owns.
In the updated version of the algorithm, a symbol still owns a specific range of on the number line greater than equal to 0 and less than 1. But we are now going to represent these using a pair of fractions: upper
/denom
and lower
/denom
. This doesn't actually affect our model code too much. The sample model we used in the previous section returned, for example, .22 and .23 for character 'W'. Now, instead, it will return {22, 23, 100} in a structure called prob
.
The Real Encoder  First Pass
Before getting into final, working code, I'm presenting some code that implements the basic algorithm, but takes a couple of shortcuts around real problems. This notquitedone code looks like this:
unsigned int high = 0xFFFFFFFFU; unsigned int low = 0; char c; while ( input >> c ) { int range = high  low + 1; prob p = model.getProbability(c); high = low + (range * p.upper)/p.denominator; low = low + (range * p.lower)/p.denominator; for ( ; ; ) { if ( high < 0x80000000U ) output_bit( 0 ); else if ( low >= 0x80000000U ) output_bit( 1 ); else break; low <<= 1; high << = 1; high = 1; } }
The first part of this loop is conceptually and functionally identical to the floating point code given in part 1. We take the range  that is, the difference between and low
and high
, and we allocate some subset of that range to the character being encoded, depending on the probability returned from the model. The result is that low
gets a little larger, and high
gets a little smaller.
The second part of the code is a little more interesting. The while
loop is new, and what it does is new as well  the simplified floating point algorithm didn't have anything like this. It performs range checks on low
and high
, looking for situations in which the values have the same most significant bit.
The first check looks to see if high
has dropped below 0x80000000, in which case its MSB is 0. Because we know that low
is always less than high, its MSB will also be 0. And because the two values only get closer to one another, the MSB of both values will forever be 0.
The other range check looks to see if low
has increased above 0x7FFFFFFF, in which case both it and high
will have MSB values of 1, and will always have an MSB of 1.
In either of these cases, remember that we have three invariants to work with: high
only decreases, low
only increases, and high
is always greater than low
. So once high
has a leading bit of 0, it will never change. Once low
has a leading bit of 1, it will never change. If this is the case, we can output that bit to the output stream  we know what it is with 100% certainty, so let's shift it out to the output stream and get rid of it.
And after we have output the bit, we discard it. Shifting low
and high
one bit to the left discards that MSB. We shift in a 1 into the least significant bit of high
, and a 0 into the least significant of low
. Thus, we keep working on 32 bits of precision by expelling the bits that no longer contribute anything to the precision of our calculations. In this particular implementation, we just keep 32 bits in working registers, with some additional number already sent to the output, and some other number pending for input.
The figure below shows how the math system now works while in the middle of some arbitrary compression run. Even though we are using 32bit math, the algorithm is now dealing with arbitrarily long versions of high
and low
:
As
low
and high
converge, their matching digits are shifted out on the left side, presumably to a file. These digits are never going to change, and are no longer needed as part of the calculation. Likewise, both numbers have an infinite number of binary digits that are being shifted in from the right  1's for high
and 0's for low
.
A Fatal Flaw and the Workaround
In the code just presented we have a pretty reasonable way of managing the calculation that creates an arbitrarily long number. This implementation is good, but it isn't perfect.
We run into problems with this algorithm when the values of low
and high
start converging on a value of 0.5, but don't quite cross over the line that would cause bits to be shifted out. A sequence of calculations that runs into this problem might produce values like these:
low=7C99418B high=81A60145 low=7FF8F3E1 high=8003DFFA low=7FFFFC6F high=80000DF4 low=7FFFFFF6 high=80000001 low=7FFFFFFF high=80000000
Our algorithm isn't quite doing what we want here. The numbers are getting closer and closer together, but because neither has crossed over the 0.5 divider, no bits are getting shifted out. This process eventually leaves the values of the two numbers in a disastrous position.
Why is it disastrous? The initial calculation of range
is done by subtracting low
from high
. In this algorithm, range
stands in as a proxy for the number line. The subsequent calculation is intended to find the proper subsection of that value for a given character. For example, in the earlier example, we might have wanted to encode a character that has a range of [0.22,0.23). If the value of range
is 100, then we can clearly see that the character will subdivide that with values of 22 and 23.
But what happens if low
and high
are so close that range
just has value of 1? Our algorithm breaks down  it doesn't matter what the probability of the next character is if we are going to try to subdivide the range [0,1)  we get the same result. And if we do that identically for every character, it means the decoder is not going to have any way to distinguish one character from another.
Ultimately this can decay to the point where low == high
, which breaks one of our key invariants. It's easy to work out what kind of disaster results at that point.
Later on in this article I'll go over some finer points in the algorithm that show why you run into trouble long before the two values are only 1 apart. We'll come up with specific requirements for the accuracy needed by the value of range
. Even without those detailed requirements, clearly something needs to be done here.
The fix to the algorithm is shown in the code below. This version of the algorithm still does the normal output of a single bit when low
goes above 0.5 or high
drops below it. But when the two values haven't converged, it adds a check of the next most significant bits to see if we are headed towards the problem of nearconvergence. This will be the case when the two most significant bits of high
are 10 and the two most significant bits of low
are 01. When this is the case, we know that the two values are converging, but we don't yet know what the eventual output bit is going to be.
unsigned int high = 0xFFFFFFFFU; unsigned int low = 0; int pending_bits = 0; char c; while ( input >> c ) { int range = high  low + 1; prob p = model.getProbability(c); high = low + (range * p.upper)/p.denominator; low = low + (range * p.lower)/p.denominator; for ( ; ; ) { if ( high < 0x80000000U ) { output_bit_plus_pending( 0 ); low <<= 1; high << = 1; high = 1; } else if ( low >= 0x80000000U ) { output_bit_plus_pending( 1 ); low <<= 1; high << = 1; high = 1; } else if ( low >= 0x40000000 && high < 0xC0000000U ) pending_bits++; low << = 1; low &= 0x7FFFFFFF; high << = 1; high = 0x80000001; } else break; } } void output_bit_plus_pending(bool bit, int &pending_bits) { output_bit( bit ); while ( pending_bits ) output_bit( !bit ); }
So what do we do when we are in this nearconverged state? We know that sooner or later, either high
is going to go below 0.5 or low
will go above it. In the first case, both values will have leading bits of 01, and in the second, they will both have leading bits of 10. As this convergence increases, the leading bits will extend to either 01111... or 10000..., with some finite number of digits extended.
Given this, we know that once we figure out what the first binary digit in the string is going to be, the subsequent bits will all be the opposite. So in this new version of the algorithm, when we are in the nearconvergence state, we simply discard the second most significant bit of high
and low
, shifting the remaining bits left, while retaining the MSB. Doing that means also incrementing the pending_bits
counter to acknowledge that we need to deal with it when convergence finally happens. An example of how this looks when squeezing out the converging bit in low
is shown here below. Removing the bit from high
is basically identical, but of course a 1 is shifted into the LSB during the process.
The bit twiddling that makes this happen can be a bit difficult to follow, but the important thing is that the process has to adhere to the following rules:
 The low 30 bits of both
low
andhigh
are shifted left one position.  The least significant bit of
low
gets a 0 shifted in.  The least significant bit of
high
gets a 1 shifted in.  The MSB of both words is unchanged  after the operation it will still be set to 1 for
high
and 0 forlow
.
The final change that ties all this together is the introduction of the new function output_bit_plus_pending()
. Each time that we manage this nearconvergence process, we are know that another bit has been stored away  and we won't know whether it is a one or a zero. We keep a count of all these consecutive bits in pending_bits
. When we finally reach a situation where an actual MSB can be output, we doit, plus all the pending bits that have been stored up. And of course, the pending bits will be the opposite of the bit being output.
This fixed up version of the code does everything we need to properly encode. The final working C++ code will have a few differences, but they are mostly just tweaks to help with flexibility. The code shown above is more or less the final product.
The Real Decoder
I've talked about some invariants that exist in this algorithm, but one I have skipped over is this: the values of high
and low
that are produced as each character is processed in the encoder will be duplicated in the decoder. These values operate in lockstep, right down to the least significant bit.
A consequence of this is that the code in the decoder ends up looking a lot like the code in the encoder. The manipulation of high
and low
is effectively duplicated. And in both the encoder and the decoder, the values of these two variables are manipulated using a calculated range
, along with the probability for the given character.
The difference between the two comes from how we get the probability. In the encoder, the character is known because we are reading it directly from the file being processed. In the decoder, have to determine the character by looking at value of the message we are decoding  where it falls on the [0,1) number line. It is the job of the model to figure this out in function getChar()
.
The compressed input is read into a variable named value
. This variable is another one of our pseudoinfinite variables, like high
and low
, with the primary difference being what is shifted into it in the LSB position. Recall that high
gets an infinite string of 1's shifted in, and low
gets an infinite string of 0's. value
gets something completely different  it has the bits from the encoded message shifted into. So at any given time, value
contains 32 of the long string of bits that represent the number that the encoder created. On the MSB side, bits that are no longer are used in the calculation are shifted out of value
, and on the LSB side, as those bits are shifted out, new bits of the message are shifted in.
The resulting code looks like this:
unsigned int high = 0xFFFFFFFFU; unsigned int low = 0; unsigned int value = 0; for ( int i = 0 ; i < 32 ; i++ ) { value <<= 1; value += m_input.get_bit() ? 1 : 0; } for ( ; ; ) { unsigned int range = high  low + 1; unsigned int count = ((value  low + 1) * m_model.getCount()  1 ) / range; int c; prob p = m_model.getChar( count, c ); if ( c == 256 ) break; m_output.putByte(c); high = low + (range*p.high)/p.count 1; low = low + (range*p.low)/p.count; for( ; ; ) { if ( low >= 0x80000000U  high < 0x80000000U ) { low <<= 1; high <<= 1; high = 1; value <<= 1; value += m_input.get_bit() ? 1 : 0; } else if ( low >= 0x40000000 && high < 0xC0000000U ) { low << = 1; low &= 0x7FFFFFFF; high <<= 1; high = 0x80000001; value <<= 1; value += m_input.get_bit() ? 1 : 0; } else break; } }
So this code looks very similar to the final encoder. The updating of the values is nearly identical  it adds the update of value
that updates in tandem with high
and low
. The nature of the algorithm introduces another invariant: value
will always be greater than or equal to low
and less than high
.
A Sample Implementation
All of this is put together in the production code included in ari.zip. (Links are at the end of this article.) The use of templates makes this code very flexible, and should make it easy to plug it into your own applications. All the needed code is in header files, so project inclusion is simple.
In this section I'll discuss the various components that need to be put together in order to actually do some compression. The code package has four programs:
fp_proto.cpp
, the floating point prototype program. Useful for experimentation, but not for real work.compress.cpp
, which compresses a file using command line arguments.decompress.cpp
, which decompresses a file using command line arguments.tester.cpp
, which puts a file through a compress/decompress cycle, tests for validity, and outputs the compression ratio.
The compression code is implemented entirely as a set of template classes in header files. These are:
compressor.h
, which completely implements the arithmetic compressor. The compressor class is parameterized on input stream type, output stream type, and model type.decompressor.h
, the corresponding arithmetic decompressor with the same type parameters.modelA.h
, a simple order0 model that does an acceptable job of demonstrating compression.model_metrics.h
, a utility class that helps a model class set up some types used by the compressor and decompressor.bitio.h
andbyteio.h
, the streaming classes that implement bitoriented I/O.
This article is going to skip over the details pertaining to the bitoriented I/O classes implemented in bitio.h
and byteio.h
. The classes provided will allow you to read or write from std::iostream
and FILE *
sources and destinations, and can be pretty easily modified for other types. Details on the implementation of these classes are in my article
C++ Generic Programming Meets OOP, which includes a bitoriented I/O class as an illustrative example.
All of the code I use requires a C++11 compiler, but could be modified to work with earlier versions without much trouble. The makefile will build the code with g++ 4.6.3 or clang 3.4. The solution file included will build the projects with Visual C++ 2012.
model_metrics.h
In the code I've shown you so far, I blithely assume that your architecture uses 32bit math and efficiently supports unsigned integer math. This is a bit of an unwanted and unneeded restriction, so my production code will define the math parameters using templates. The way it works is that the compressor and decompressor classes both take a MODEL
type parameter, and they rely on that type to provide a few typedefs and constants:
CODE_VALUE 
The integer type used to perform math. In the sample code shown so far we assumed this was unsigned int , but signed and longer types are perfectly valid. 
MAX_CODE 
The maximum value of a code  in other words the highest value that high can reach. If we are using all 32 bits of an unsigned in, this would be 0xFFFFFFF, but as we will see shortly, this will normally be quite a bit smaller than the max that can fit into an int or unsigned int. 
ONE_FOURTH 
The three values used when testing for convergence. In the sample code these were set to full 32bit values of 0x40000000, 0x80000000, and 0xC0000000, but they will generally be smaller than this. The values could be calculated from MAX_CODE , but we let the model define them for convenience. 
struct prob 
This is the structure used to return probability values from the model. It is defined in the model because the model will know what types it wants the three values in this structure to be. The three values are low , high , and count , which define the range of a given character being encoded or decoded. 
The header file model_metrics.h
contains a utility class that helps the model class figure out what these types are. In general, the choice of CODE_VALUE
is going to be defined using the natural int or unsigned int type for your architecture. Calculating the value of MAX_CODE
requires a little bit more thinking though.


modelA.h
I haven't talked too much about modeling in this article. The focus is strictly on the mechanics of arithmetic coding. We know that this depends on having an accurate model of character probabilities, but getting deep into that topic requires another article.
For this article, the sample code uses a simple adaptive model, which I call modelA
. This model has character probabilities for 257 symbols  all possible symbols in an eightbit alphabet, plus one additional EOF symbol. All of the characters in the model start out with a count of 1, meaning each has an equal chance of being seen, and that each will take the same number of bits to encode.
After each character is encoded or decoded, its count is updated in the model, making it more likely to be seen, and thus reducing the number of bits that will be required to encode it in the future.
The model is implemented by using a CODE_VALUE
array of size 258. The range for a given character i is defined by the count at i and the count at i+1, with the total count being found at location 257. Thus, the routine to get the probability looks like this:
prob getProbability(int c) { prob p = { cumulative_frequency[c], cumulative_frequency[c+1], cumulative_frequency[257] }; if ( !m_frozen ) update(c); pacify(); return p; }
Because of the way arithmetic coding works, updating the count for character i means the counts for all characters greater than i have to be incremented as well, because adjusting the range for one position on the number line is going to squeeze everyone to the right  the counts in cumulative_frequency
are in fact cumulative, representing the sum of all characters less than i. This means the update function has to work its way through the whole array. This is a lot of work, and there are various things we could do to try to make the model more efficient. But ModelA
is just for demonstration, and the fact that the array probably fits neatly in cache means this update is adequate for now.
One additional factor we have to watch out for is that the we can't exceed the maximum number of frequency bits in our count. This is managed by setting a flag to freeze the model once the maximum frequency is reached. Again, there are a number of things we could do to make this more efficient, but for demonstration purposes this is adequate:
void inline update(int c) { for ( int i = c + 1 ; i < 258 ; i++ ) cumulative_frequency[i]++; if ( cumulative_frequency[257] >= MAX_FREQ ) m_frozen = true; }
The decoder has to deal with the same issues when looking up a character based on the scaled_value
. The code does a linear search of the array, whose only saving grace is that it ought to be sitting in the cache:
prob getChar(CODE_VALUE scaled_value, int &c) { for ( int i = 0 ; i < 257 ; i++ ) if ( scaled_value < cumulative_frequency[i+1] ) { c = i; prob p = {cumulative_frequency[i], cumulative_frequency[i+1], cumulative_frequency[257]}; if ( !m_frozen) update(c); return p; } throw std::logic_error("error"); }
In a future article, the notion of how to develop efficient models will be a good topic to cover. You won't want to use modelA
as part of a production compressor, but it does a great job of letting you dig into the basics of arithmetic compression.
When you instantiate modelA
you have three optional template parameters, which you can use to tinker with the math used by the compressor and decompressor. (Of course, the compressor and decompressor have to use the same parameters).
template<typename CODE_VALUE_ = unsigned int, int CODE_VALUE_BITS_ = (std::numeric_limits<CODE_VALUE_>::digits + 3) / 2, int FREQUENCY_BITS_ = std::numeric_limits<CODE_VALUE_>::digits  CODE_VALUE_BITS_> struct modelA : public model_metrics<CODE_VALUE_, CODE_VALUE_BITS_, FREQUENCY_BITS_> {
On 32 bit compiler, opting for all defaults will result in you using 17 bits for CODE_VALUE
calculations, and 15 bits for frequency counts. Static assertions in the model_metrics
class check to insure that your selections will result in correct compression.
compressor.h
This header file contains all the code that implements the arithmetic compressor. The class that does the compression is a template class, parameterized on the input and output classes, as well as the model. This allows you to use the same compression code with different types of I/O in as efficient a manner as possible.
The compressor engine is a simple C++ object that has an overloaded operator()()
, so the normal usage is to instantiate the engine then call it with input, output, and model parameters. Because of the way that C++ manages template instantiation, the easiest way to actually use the engine is via a convenience function, compress()
, that takes care of those details, and can be called without template parameters, as in:
std::ifstream input(argv[1], std::ifstream::binary); std::ofstream output(argv[2], std::ofstream::binary); modelA<int, 16, 14> cmodel; compress(input, output, cmodel);
You can see the (simple) implementation of the convenience function in the attached source.
The operator()()
is where all the work happens, and it should look very similar to the trial code shown earlier in this article. The difference is that what is shown here is fully fleshed out, with all the details taken care of:
int operator()() { int pending_bits = 0; CODE_VALUE low = 0; CODE_VALUE high = MODEL::MAX_CODE; for ( ; ; ) { int c = m_input.getByte(); if ( c == 1 ) c = 256; prob p = m_model.getProbability( c ); CODE_VALUE range = high  low + 1; high = low + (range * p.high / p.count)  1; low = low + (range * p.low / p.count); // // On each pass there are six possible configurations of high/low, // each of which has its own set of actions. When high or low // is converging, we output their MSB and upshift high and low. // When they are in a nearconvergent state, we upshift over the // nexttoMSB, increment the pending count, leave the MSB intact, // and don't output anything. If we are not converging, we do // no shifting and no output. // high: 0xxx, low anything : converging (output 0) // low: 1xxx, high anything : converging (output 1) // high: 10xxx, low: 01xxx : near converging // high: 11xxx, low: 01xxx : not converging // high: 11xxx, low: 00xxx : not converging // high: 10xxx, low: 00xxx : not converging // for ( ; ; ) { if ( high < MODEL::ONE_HALF ) put_bit_plus_pending(0, pending_bits); else if ( low >= MODEL::ONE_HALF ) put_bit_plus_pending(1, pending_bits); else if ( low >= MODEL::ONE_FOURTH && high < MODEL::THREE_FOURTHS ) { pending_bits++; low = MODEL::ONE_FOURTH; high = MODEL::ONE_FOURTH; } else break; high <<= 1; high++; low <<= 1; high &= MODEL::MAX_CODE; low &= MODEL::MAX_CODE; } if ( c == 256 ) //256 is the special EOF code break; } pending_bits++; if ( low < MODEL::ONE_FOURTH ) put_bit_plus_pending(0, pending_bits); else put_bit_plus_pending(1, pending_bits); return 0; } inline void put_bit_plus_pending(bool bit, int &pending_bits) { m_output.put_bit(bit); for ( int i = 0 ; i < pending_bits ; i++ ) m_output.put_bit(!bit); pending_bits = 0; }


decompressor.h
The decompressor code mirrors the compressor code. The engine is a template class that is parameterized on the input and output stream types, and the model type. A convenience function called decompress()
makes it easy to instantiate the engine and then decompress without needing to declare template parameters:
std::ifstream input(argv[1], std::ifstream::binary); std::ofstream output(argv[2], std::ofstream::binary); modelA<int, 16, 14> cmodel; decompress(input, output, cmodel);
The convenience function is in the attached source package.
The actual operator code is shown here. This is very much like the sample code shown earlier, with all the additional loose ends tied up so that this is productionready:
int operator()() { CODE_VALUE high = MODEL::MAX_CODE; CODE_VALUE low = 0; CODE_VALUE value = 0; for ( int i = 0 ; i < MODEL::CODE_VALUE_BITS ; i++ ) { value <<= 1; value += m_input.get_bit() ? 1 : 0; } for ( ; ; ) { CODE_VALUE range = high  low + 1; CODE_VALUE scaled_value = ((value  low + 1) * m_model.getCount()  1 ) / range; int c; prob p = m_model.getChar( scaled_value, c ); if ( c == 256 ) break; m_output.putByte(c); high = low + (range*p.high)/p.count 1; low = low + (range*p.low)/p.count; for( ; ; ) { if ( high < MODEL::ONE_HALF ) { //do nothing, bit is a zero } else if ( low >= MODEL::ONE_HALF ) { value = MODEL::ONE_HALF; //subtract one half from all three code values low = MODEL::ONE_HALF; high = MODEL::ONE_HALF; } else if ( low >= MODEL::ONE_FOURTH && high < MODEL::THREE_FOURTHS ) { value = MODEL::ONE_FOURTH; low = MODEL::ONE_FOURTH; high = MODEL::ONE_FOURTH; } else break; low <<= 1; high <<= 1; high++; value <<= 1; value += m_input.get_bit() ? 1 : 0; } } return 0; }

Implementation Notes  Using Exceptions for Fatal Errors
Using C++ exceptions properly can be a really difficult task. Ensuring that all possible paths through stack unwinding leave your program in a correct state is not for the faint of heart. This is less of a problem when you are using exceptions strictly as a fatal error mechanism, as I discuss here.
The demo code I use here throws exceptions in response to the following error conditions:
 EOF on the input stream.
 A request for a character from the model with a
scaled_value
larger than the maximum value currently defined  this is a logic error that should never happen.
Tightening up this code would give some other places that this would be useful  for example failures when writing output.
Propagating errors this way is efficient, and helps keep the code clean  no checking for failures from the model or I/O objects, and it provides a very consistent way to implement error handling when you write new classes for modeling or I/O.
To catch these errors, the normal template for compressing or decompressing is going to look like this:
try { // // set up I/O and model // then compress or decompress return 0; //the success path } catch (std::exception &ex) { std::cerr << "Failed with exception: " << ex.what() << "\n"; } return 255; //failure path
Implementation Notes  Building the Demo Programs
There are four executables that can be built from the attached source package:
fp_proto:  The floating point prototype code. This code is useful to demonstrate the basics of how arithmetic coding works, but isn't much good for real work, with reasons outlined in the article. 
compress:  A compressor that accepts an input and output file name on the command line, then compresses using a simple order0 model. You should see modestly good compression with this file. To get superior compression requires some work on more advanced modeling. 
decompress:  A decompressor that accepts an input and output file name on the command line. This will work properly with files created by the correspond compress program. 
tester:  A test program that accepts a single file name on the command line. It executes a compress to a temporary file, then a decompress, then compares the result with the original file. The app exits with 0 if the comparison succeeds, 255 in the event of a failure 
If you are working on a Windows box, you can build all four applications from the enclosed ari.sln
solution file. This should work with Visual C++ 12 or later.
If you are working on Linux, you can build all four files with the attached makefile, using:
make all
By default this will use g++ to compile, but if you change one option in the file you can easily switch to clang. The code has been tested with g++ version 4.6.3, and clang++ version 3.4.
The code uses a few C++11 features, so if you try to build the projects with earlier versions of any of these compilers you may run into some problems  mostly with the I/O code. It should be reasonably easy to strip out some of the functionality and create classes that will support either iostreams
or FILE *
objects.
Implementation Notes  Testing
The tester
program provides a good way to exercise the code. If you are on a Linux system, you create a directory called corpus
, populate it with a tree of as many files as you like, then execute make test
, which will run the tester program against all files. If a failure occurs, the process should abort with message, allowing you to see which file failed:
mrn@mrnubuntu12:$ make test find corpus type f print0  xargs 0r n 1 ./tester compressing corpus/about/services/trackback/index.html... 5.41378 compressing corpus/about/services/feed/index.html... 5.71502 compressing corpus/about/services/index.html... 5.41378 compressing corpus/about/trackback/index.html... 5.39995 compressing corpus/about/feed/index.html... 5.25144 compressing corpus/about/serial1/trackback/index.html... 5.43231 compressing corpus/about/serial1/feed/index.html... 5.32922 compressing corpus/about/serial1/index.html... 5.43231 compressing corpus/about/tdcb/trackback/index.html... 5.37245
The tester
app also prints out the number of bits/byte that was achieved by the compressor, that output can be collected to provide stats.
It's a little harder to do this with a oneliner under Windows, so I am afraid I don't have an easy bulk test option.
Implementation Notes  Logging
When an arithmetic compressor goes bad, it can be exceedingly difficult to determine what went wrong.
There are two common sources of error. The encoder itself can go awry, some error with bit twiddling, or failure to maintain the invariants that have been discussed ad nauseum here.
The second possible source of error is in the model itself. There are a few things that can go wrong here, but the most common error is a loss of synch between the encoder and decoder models. For things to work properly, both models have to operate in perfect lockstep.
Some low level logging can be turned on in the encoder by building with the LOG
constant defined. You can do this by editing the makefile and adding DLOG
to the compiler command line, or defining it in the C++PreprocessorPreprocessor Definitions area of the project properties for Windows builds.
Once you have the projects reconfigured, you can do a clean and rebuild of the projects. From that point on, the compressor will create a log file called compressor.log
, and the decompressor will create a file called decompressor.log
. These log files will output the state of the compressor or decompressor as each character is encoded or decoded, with output like this:
0x2f(/) 0x0 0x1ffff => 0x5da2 0x5f9f 0x2f(/) 0xd100 0x1cfff => 0xff74 0x1016d 0x0d 0xba00 0x1b6ff => 0xc6b2 0xc7ab 0x0a 0xb200 0x1abff => 0xbb9d 0xbc92 0x2f(/) 0x9d00 0x192ff => 0xcb2f 0xce01 0x2f(/) 0xcbc0 0x1807f => 0xed8d 0xf04f 0x20 0x6340 0x113ff => 0x7a19 0x7ac4 0x20 0x3200 0x189ff => 0x5e4d 0x60e7 0x42(B) 0x2680 0x173ff => 0x83a0 0x84e2 0x57(W) 0xa000 0x1e2ff => 0x11492 0x115c8 0x54(T) 0x9200 0x1c8ff => 0xfe53 0xff7c 0x2e(.) 0x5300 0x17cff => 0x8a98 0x8bb4 0x43(C) 0x9800 0x1b4ff => 0xe994 0xeaa2 0x50(P) 0x9400 0x1a2ff => 0xef56 0xf056 0x50(P) 0x5600 0x156ff => 0xac4c 0xae31
The data you see there is the character being encoded, both in hex and text representation (if printable), followed by the values of low
and high
before and after the character is processed. If you diff the two log files, the only difference between the two should be in the last character position, because the decompressor doesn't update its state when it sees an EOF:
mrn@mrnubuntu12:$ diff compressor.log decompressor.log 10587c10587 < 0x100 0x27a0 0x10cff => 0x10cfa 0x10cff  > 0x100 0x27a0 0x10cff mrn@mrnubuntu12:$
Any other differences indicate an error. You can use the line number of the position to determine exactly where things went awry.
EOF
By itself, an entropy encoder is not a magic bullet. A good compressor needs sophisticated modeling to back it up. With that good model, an arithmetic encoder will give you excellent results, and should outperform Huffman or other older and more revered coders. There is an entire class of coders referred to as Range encoders that use the ideas here, tailored for greater efficiency.
The computational requirements of an arithmetic encoder are definitely going to be higher than with something like Huffman, but modern CPU architectures should be able to minimize the impact this has on your program. Arithmetic encoders are particularly well suited for adaptive models when compared to Huffman coding.
Source is attached here.
Your comments and corrections are welcome, and will help improve this post as time goes on.
Thanks for reading.
 Mark
23 users commented in " Data Compression With Arithmetic Coding "
Followup comment rss or Leave a Trackback[...] have a substantially improved article that covers Arithmetic Coding only which you can find here: Data Compression With Arithmetic Coding. I strongly encourage you to read this article first for an improved take on the [...]
Found this through comp.compression list. This is a really nice article. I am just about to try wrapping my head around the jpeg 2000 MQcoder, so this will help me a lot.
Thanks!
Aaron
@Aaron, thanks for reading!
Glad to see you posting I miss your code. I hope you expand a little more on bijective endings. There is a way to think in terms of bits or bytes. So that there are one to one bijective transforms between the two. Its both easier and harder than most people think. But to me it sort of polishes either Huffman or Arithmetic.
Take Care
Dear Mr Nelson,
I am a language student hoping to get my license as an EnglisgDutch translater in the Netherlands. I selected your article as one of the works that I need to translate in order to get my license.
I bought a copy of the first edition of your book on datacompression 25 years ago and indeed find the article an improvement on the chapter dedicated to arithmetic compression there.
Nevertheless, going through the text, translating into Dutch, I think that the following is incorrect. You write:
“A typical number would be 17, which would require that we read three bits into value before we start encoding”
Shouldnt this be 3 bytes and before we start decoding?
Thanks for the article, anyhow
John Janssen
Yes John, I believe you are correct. Thanks for checking, and good luck with the project.
 Mark
Mark, I just wanted to leave a thank you note. I’ve always found your explanations of compression arcana the most lucid. I needed to implement arithmetic encoding this week, and after reading a few articles on the web, I remembered to reach for my 1992 edition of the Data Compression Book, which I used back in the day as background material for a custom compressor I built for seismic data. I found this page with your latest take on arithmetic coding, and I’m grateful you are keeping it fresh. DCB is a great piece of work. I recommend it to everyone.
@Hugh:
Thanks for the kind words, I appreciate it and I’m glad to have been some help!
 Mark
Hi Mark
Thank you for this great article on arithmetic coding. Without your article, it would have taken me much longer to grasp the transition from theory to implementation.
Best regards
Roman
Hi,
Thanks for the article, Mark. It's very helpful. I have a question, though. Code in the article says:
But in that case range will be 0 (because of the overflow). Am I missing something?
@wonder:
No, that underflow would be a problem. However, that code in the article is just trying to demonstrate a bit of the concept.
In the actual implementation, as you read further on, you will see that the algorithm has to be close attention to the number of bits used in the math.
In the final implementation, the range variable will typically be a long, and the number of bits used by high and low will be low enough that underflow and overflow cannot occur.
If you download and compile the source you will see that the model metrics header figures out reasonable values for these things.
Keep working with the source to improve your understanding  this algorithm is kind of challenging, getting the deep understanding in your brain takes some practice.
 Mark
Hi Mark, thanks for your previous reply.
I'm writing implementation of arithmetic coder (mostly for educational purposes) and hit some problem that apparently solved in your code. I even found how, but the trick is that I don't completely understand math mechanics behind that solution. The problem is in interval scaling and integer arithmetic.
Lets say current interval is [lower=640, upper=65920) (I'm using intervals that are open on the right side, but I'm pretty sure it's irrelevant). Its length is 65280. Current symbol to encode has cumulative frequency interval [7/258, 8/258). Mapping this (frequency) interval on code interval gives: [lower=65280*7/258=1771, upper=65280*8/258=2024). Now imagine that we want to decode that symbol. Current decoder interval is the same [lower=640, upper=65920) and current value is 1771. Lets map 1771 to frequency interval 1771*258/65280=6. As you can see we will choose incorrect interval [6/258, 7/258) instead of [7/258, 8/258). That happens because the naive mapping I'm using is not reversible i.e. 1772 maps to 7 and 7 maps to 1771:
In your code you use following formula to map current value to frequency interval:
((code_value + 1)*freq_range  1)/code_range
But I can't understand why exactly how that works. Can you give any hint where I can read more about that?
@wonder.mice:
You've got one basic problem that is breaking you down, you aren't calculating the position of a given code in the interval properly.
In your example, the difference between upper and lower is 65280. So when you are trying to calculate the width of the range, you come up with 1771 and and 2024 as your interval.
That's correct, the width of the interval is bounded by 1771 and 2024. But it doesn't start at 0, it starts at 640. So your final interval is 1771+640, 2024+640.
The math involved in dividing up the line is pretty simple, but you need to get it right. You can see how your initial calculation is not working by simply adjusting the intervals. Imagine that low is 1,000,000 and high is 1,065,280. With your calculation, the value of the character is still at position 1771  but that isn't even in the current range! So the method you are using is clearly broken.
Figuring out how these calculations work will be more interesting when you look at the bottom and top ranges, 0/n to x/n, and x/n to n/n. For those cases you know that the new low and high values after update have to incorporate the old low, or the old high.
 Mark
Sorry, I cut some corners to emphasize the problem I'm facing, but that made my post harder to understand. Of course my final interval is [1771+640, 2024+640). However, in decoder, I subtract 640 again before scaling ("value  low" part). And when calculating range those 640 will eliminate each other: (2024+640)  (1771+640) == 2024  1771. So I just throw them away in my previous post as irrelevant details. My bad. The point is that we have two intervals. Interval A=[0, Model::getCount()) and interval B=[0, upperlower). When encoding, we have two values x and x' from interval A (cumulative frequency range) and we want to map them to y and y' from interval B.
y=B * x / A
y'=B * x' / A
When decoding, we have y and y' from B and we want to map them _back_ to x and x' from A. Following formulas will NOT work:
x=A * y / B
x'=A * y' / B
Example: A=[0, 2), B=[0, 5). Let x = 1, then (y=B * x / A) y = 2. Then (x=A * y / B) x=0 (but we expect it to be 1 again). So, this formula doesn't correctly reverse the encoder mapping. The correct formulas are:
x=(A * (y + 1)  1) / B
x'=(A * (y' + 1)  1)/ B
So my question was: how exactly this (correct) formula was obtained and why it works.
But after some more thinking (after writing the first post), I think I understand it. Lets say A = 3 and B = 5. Since we want to map one to another, lets make it of equal length by "multiplying" A on B and B on A:
We see, that mapping from A to B is just y=floor(B*x/A). To do the reverse, we need to take next y (hence, y + 1) and find how many B we can fit into it: A*(y + 1)/B. But we need to avoid case where A*(y + 1) % B == 0, because in that case we will get x + 1, not x we are looking for. For that purpose we subtract 1: ( A*(y + 1)  1)/B.
As you can see my explanation is clumsy and not formal, so I would like to hear a better on, if somebody has it.
Thanks again for this post, I am still referring to it as I learn about arithmetic coding. Would you consider a follow up post about the MQ coder used in JPEG 2000 and JBIG2 ?
Cheers,
Aaron
Hi Aaron,
Wish I knew enough about it to write an article!
 Mark
This is a great article.
Can we limit the maximum number of bits per code word to x bits?
@rachana:
It's a little difficult to determine how many bits are being used per code word. It's not a code like, say Huffman coding, in which a symbol has a specific integral number of bits.
In arithmetic coding, codes tend to have close to ln(p) bits, where p is the probability of the appearance of a symbol in your model using base 2. The actual number of bits used will be as close to that as the precision your model allows.
So in order to restrict a symbol to a certain number of bits, you have to adjust your model to not generate a probability that will be less than 1/2&bits or something like that. Since you control the model this is totally under your control
 Mark
Hey mark Nelson.
your work is profound.i just appreciate,
your explanations are easy to read and understand.
with regards from Kenya.
Hi, Mark
An excellent article, thank you. After reading it [I hope that] I understand a practical implementation of arithmetic coding much better than after reading and rereading of respective Wikipedia page.
It seems you made a small mistake in the part called "The Real Decoder".
In article:
if ( low >= 0x80000000U  high < 0x80000000U ) {
low <<= 1;
high <= 0x40000000 && high < 0xC0000000U ) {
low << = 1;
low &= 0x7FFFFFFF;
high <= 0x80000000U  high < 0x80000000U ) {
low <<= 1;
high << = 1;
high = 1;
value <= 0x40000000 && high < 0xC0000000U ) {
low << = 1;
low &= 0x7FFFFFFF;
high << = 1;
high = 0x80000001;
value <<= 1;
value += m_input.get_bit() ? 1 : 0;
} else
Now let's look if after reading your article I will be able to understand an entropy encoding part of JPEGXR codec. That is, assuming that it is actually adaptive arithmetic coder. So far, I couldn't figure out even that. I guess, I am not good at understanding minimally commented source codes.
Hi, Mark
An excellent article, thank you. After reading it [I hope that] I understand a practical implementation of arithmetic coding much better than after reading and rereading of respective Wikipedia page.
It seems you made a small mistake in the part called “The Real Decoder”.
In article:
Now let's look if after reading your article I will be able to understand an entropy encoding part of JPEGXR codec. That is, assuming that it is actually adaptive arithmetic coder. So far, I couldn't figure out even that. I guess, I am not good at understanding minimally commented source codes.
Oh, sorry. It's still a mess. Hard to live without preview.
Hopefully, my point was clear. Feel free to remove or edit my messy posts.
Regards, Michael.
@Michael,
I don't understand what you are trying to say.
The code you added is missing some keywords  for example you removed the "if" from a couple of statements.
You don't say what you think the problem is.
 Mark
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