2/10/2009

02-10-09 - Image Compression Blues

Good lord the world of image compression is so screwed up. There's no standard image test set (the old Kodak images are archaic, and even then people use different variants of Lena, this new test images set is pretty good but it's not standard so fuck), there's no standard error measure (people even compute RMSE and PSNR differently), and really we should be using a perceptual measure, but again there's no good standard perceptual measure (it is nice to see that things like SSIM are catching on - however SSIM is a bit vague in its specification of blocking, and there are various implementations that do it differently, so we're back in the fucked up comparing apples-to-oranges methodology).

Making it all worse is that people keep making new shitty standards and claiming they "look good to their eyes". I mentioned before that the HD Photo guys were saying some sort of silly things about error. Well guess what, it sucks. I just found this nice benchmark with graphs that shows HD Photo doing much worse than even old baseline JPEG (!!) under the SSIM metric. WTF, how can you do worse than JPEG !? Bush league IMO.

I also wasted some time this morning looking at PGF (libPGF) . PGF is a semi-open wavelet library. It does have some good properties. The code is actually very simple and semi-readable. Compression performance is a bit better than baseline JPEG. On the minus side, compression performance is not anywhere close to state of the art. Even my very simple/fast "cbwave" beats it handily.

BTW looking at the PGF source code this is what it seems to do :

It uses a very simple small integer lifting wavelet transform. It's 5/3 tap transform, perhaps it's the Le Gall transform which is also used in JPEG2000 ? It does not do anything smart about memory flow for the subbands, it's basically a bad brute-force implementation, which means "cbwave" can beat it easily for speed.

The coder is kind of interesting. It's a bitplane based coder with no entropy coder. It just uses a certain kind of bit-packing that makes small streams when there are lots of zeros, it's similar to the old EZW or SPIHT type of zero-tree coding just with bit sequences. Obviously these codecs have implicit modeling and "entropy coding" built in to the bit sequence spec, they just avoid the arithmetic coder. The method in PGF works by breaking the subbands into blocks, choosing a linear walk order on the block, and then doing linear RLE on the bitplanes. The significance bitstreams are basically a few scattered ones with big blocks of zeros, and the RLE is just coding that out.

Another thing I found that I wasn't aware of is the new T.851 variant of JPEG1 ; basically it's a new arithmetic coder called Q15 stuck on the back end of JPEG instead of Huffman or the old QM coder. The IJG is pushing for this, I don't really know what the status is. The performance should be fine. At low bit rates a deblocking filter helps a lot and could make this a decent choice.

02-10-09 - Fixed Block Size Embedded DCT Coder

A while ago when I wrote about DXTC I also included numbers for something new that I called "CodeTree" or "CodeLinear". I figured I should take a second to write down what they do before I forget.

CodeTree and CodeLinear are two variants of a Fixed Block Size Embedded DCT Coder. In the post above I wrote :

Fixed bitrate blocks inherently gives up even more. It kills your ability to do any rate-distortion type of optimization. You can't allocate bits where they're needed. You might have images with big flat sections where you are actually wasting bits (you don't need all 64 bits for a 4x4 block), and then you have other areas that desperately need a few more bits, but you can't gived them to them.

So, what if we keep ourselves constrained to the idea of a fixed size block and try to use a better coder? What is the limit on how well you can do with those constraints? I thought I'd see if I could answer that reasonably quickly.

What I made is an 8x8 pixel fixed rate coder. It has zero side information, eg. no per-image tables. (it does have about 16 constants that are used for all images). Each block is coded to a fixed bit rate. Here I'm coding to 4 bits per pixel (the same as DXT1) so that I can compare RMSE directly, which is a 32 byte block for 8x8 pixels. It also works pretty well at 24 byte blocks (which is 1 bit per byte), or 64 for high quality, etc.

This 8x8 coder does a lossless YCoCg transform and a lossy DCT. Unlike JPEG, there is no quantization, no subsampling of chroma, no huffman table, etc. Coding is via an embedded bitplane coder with zerotree-style context prediction. I haven't spent much time on this, so the coding schemes are very rough. CodeTree and CodeLinear are two different coding techniques, and neither one is ideal.

Now lets go into more details about how this is done. All the basics are quite simple, but we have to be careful about the details. To achieve the fixed block size, I write an "embedded" style stream, and then just truncate it down to the fixed rate for each block. In order for this to be okay, we have to be very careful that we are writing the most important bits first. For very simple blocks (such as single flat colors) we will be able to write the whole block exactly within the rate limit, and no truncation will occur. More typically the lossless size of the block will be very large (maybe 100 bytes or so) and we're chopping it down to 24, 32, or 64 bytes. (obviously any fixed rate size is possible, but those are the main ones that we consider).

We're going to be sending bit planes. We need to order the bit planes from most important to least important, and also order the values within each plane. We have 8x8 pixels, 3 colors for each pixel, and 12 bits for sample. That means we are transmitting 8*8*3*12 = 2304 bits in each block, and we want each bit to be in order of importance. It's crucial to be precise about what we mean by "importance". What we mean is that the signal reconstruction from each bit in order has an expected decreasing contribution to the L2 norm. (L2 norm because we are measuring quality with RMSE - if you have a different quality metric you should order the bits for that metric). (Note that higher bits are always more important than lower bits, so we really only need to worry about ordering the 8*8*3 = 192 samples and we know to descend the bitplanes in order - though there is the possibility of sending the top bitplanes of some coefficients before the top bitplanes of other coefficients).

We're applying two transforms - the DCT and the color conversion. We need both of them to be "Unitary" transforms so that we know how our transformed coefficients relate to RGB pixel values. That is, every value in post-transform space should affect the L2 norm in the same way. Many many people mess this up.

First we have to be careful with the color conversion. I chose YCoCg because it's easy and exact and it works fine. But the standard YCoCg lossless code that people use scaled up the CoCg by *2 compared to Y. That shifts the way the bit planes relate to RGB errors. We could remember this and step through the planes differently, but it's easier just to shift Y left by one bit. That means the bottom bit plane of Y is always zero. In our coder we could just not send these bits when we get down to that bit plane, but I don't bother, because if we get down to the very bottom bit plane it means we're sending the data lossless. Specifically I do this :


void RGB_to_YCoCg(int r,int g,int b, int & y,int & Co,int & Cg)
{
   Co = r - b;
   int t = b + (Co/2);
   Cg = g - t;
   y = t + (Cg/2) - 128;
// scale up Y so its at the same magnitude as Co and Cg - the bottom empty bit never gets sent
   y <<= 1; 
}

The next thing is to be careful with the DCT normalization. A lot of the DCT code out there does funny scalings on the various values. You can test this by taking an 8x8 block, setting to all zero, put a 1 in one of the bins, and then run your IDCT. The output will have little values all over, but the squared sum (L2 norm) should be a constant value no matter where you put that 1. If your DCT is not Unitary, you can use this to apply scalings to fix it.

So now we have a DCT + color conversion that's correct about value scaling. The DCT I used increases the dynamic range up to 12 bits (from 8) but the large values are very rare; the typical maximum value is around 512, but 4095 does occur (for example if your block is all 255, the DC value you be very large).

Now, for sample ordering by importance, the right thing obviously is the KLT. We want to send the samples of each block in an order where the first has the most energy, the next has the most of the remaining energy, etc.. This is exactly the KLT (aka the PCA). Now, we can't actually use the KLT because we get to send zero side information. (that is, a decoder must be able to grab one block at random and decode it, there's zero header). Fortunately, for most images, the DCT is very close to the KLT. That is, the first coefficient has the largest expected value, the second coefficient has the next largest, in order. Note that this is without any perceptual scaling or quantization - it's simply a property of the transform that it is rotating energy towards the lower ceofficients for most cases. If the input were random, the output would also be random with every sample having equal expected value.

Now we have to code the data; I tried two ways, they performed similarly, I'm not sure which is better. Both coders use arithmetic coding. That's largely just because it's easy for me, it's not inherent to this method necessarily. Arithmetic coding just makes it easy for me to make sure I'm not writing redundancy in the stream. Both methods use non-adaptive arithmetic coders. That means probabilities are stored as constants, not transmitted. I write the whole block lossless, then just truncate it. To decode, I take the truncated block and implicitly stuff zeros after the code stream, so if the arithmetic decoder reads past the end, it gets zero bits. The code stream must be designed correctly so that zero bits in the code stream always select the most likely value on decode.

CodeTree

This is a tree-based bitplane coder similar to EZW or the "EZDCT" that I wrote long ago. It uses zigzag scan and a "parent" relationship. Bits are sent as 2x2 "blocks". A block has a single parent bit. In the 8x8 DCT there are 16 blocks, each block has a parent in the previous "subband" formed by pretending the DCT is a wavelet transform (see the old EZDCT papers for more details). (there are many pages on the standard wavelet tree structure ; see : for example )

Bits are sent in a fixed order : first loop on bit planes, then loop on blocks, then loop on color planes. That is :


top bit plane :
   block 0 Y
   block 0 Co
   block 0 Cg
   block 1 Y
   block 1 Co
   ...
next bit plane :
   block 0 Y
   block 0 Co
   ...

The blocks are traversed in zigzag order, starting with block 0 = the DC (the upper-left most 2x2 LL in a wavelet interpretation).

Each block that we send consists of 4 bits in the current bitplane level. We pack those 4 bits together to make a value in [0,15]. When we send that block, we have some context information. We have already sent all of those samples previous bitplanes, so we can use the values of the bits in those samples previous bitplanes (in fact, we just use a boolean for each sample - were any of the previous bits on or not). We have also sent the parent sample's bits up to and including the current bitplane. Again I just use a single boolean - was it on. This is just a kind of "significance" mask that is familiar if you know about wavelet coding at all.

The bits within a block are obviously correlated with each other and this is modeled *implicitly* by coding then as a combined 4-bit symbol. Using the combined symbol also means I get 4 bits in one arithmetic decode. When a sample is previously significant, then the remaining bits are close to random (though not quite, zero is slightly more likely). When a sample is not previously significant, but the parent is, it's quite likely that the sample will become significant. If the sample is not previously signficant, and neither is the parent, then it's very likely the bit will still be zero. This is what the arithmetic coding gets us. In particular the first few bit planes that we send are almost always all zero and we need to send that very compactly. So previous zeros strongly predict more zeros. These probabilities were measured on a large test set and stored as constants.

When a sample first becomes signifcant (its first nonzero bit is sent) its sign is then immediately encoded as a raw bit. Note that there are also some special case contexts. The 0th Y sample has no parent - its parent is treated as always on. The 0th Co and Cg use the 0th Y as their parent. All other samples have a parent available and code normally.

The decoder works in the obvious way, but there is one detail : missing bit reconstruction.

When the decoder runs out of bits to read (it hits the end of the fixed size block), it has probably only read some of the samples and some of the bit planes. Because of the way we ordered things, usually what has been dropped is primarily the high frequency information in the Co and Cg. Note that we didn't do any CoCg subsampling, but implicitly that tends to happen automatically, because that's what gets truncated out. Often we have lost quite a few of the bottom bits of the samples. If we just stop decoding, those bottom bits are currently zero, like :

3 bottom bits not received
sample value = 40
sample bits : 0 0 0 0 1 0 1 [ 0 0 0 ]
[] bits not received

Now, just leaving them zero is not too bad, and it's what many people do, but we can do better. This mainly becomes important at very low bit rates (4 bits per pixel or less) because we are discarding a lot of bottom bits. What we want to do is reconstruct those bottom bits to their expected value in the range.

That is, we've gotten the top bits, so we know the true value is in the range [40,47]. What we should do is restore the value to the average (weighted by probability) in that range. The values are not equally likely, so it's not just the middle. Generally DCT coefficients are pretty close to laplacian, centered at zero. The exact distribution depends on the image, or even the block. One thing we could do to be very precise is measure the energy of what we did successfuly receive and try to guess the laplacian distribution for this block. I just measured the average in practice, and found on most images it is around 40% of the gap. That is, in this case it would be :

3 bottom bits not received
add 8 * 0.4 = 3.2 -> 3
sample value = 43
sample bits : 0 0 0 0 1 0 1 0 1 1

BTW if the previous bits received are all zero, you must not do this, restore the unreceived bits to zero. Obviously more generally what you should be doing is modeling the unreceived bits based on the received bits and the average energy of the block.

Now, I have a suspicion that you could do something neater here for visual quality. Rather than always restore to 40% of the gap, randomly restore to somewhere in 0% - 50% of the gap (or so). Note that if you restore to zero, the error looks like smoothing. That is, when you restore the unreceived bits to zero, you are letting the shape of the transform basis functions show through. Restoring the unknown to random will reproduce detail - it just might not be the right detail. At low bit rate, you can either get an image which becomes very smooth looking, or an image that become grainy in a weird way. In order for this to be ideal you need some per-block sensitivity - if you can tell that the block is pretty noisy, then he grainy random restoration is probably better; if the block looks pretty smooth, then restoring to zero is probably better.

CodeLinear

CodeTree is pretty straightforward (it's very similar to standard EZW type coders) but it does require a lot of constants to be tweaked. There are 16 symbols (4 bit block) and 8 contexts, so there are 16*8 = 128 int constants used by the arithmetic coder. This is undesirable for various reasons, one of which is the risk of overtraining to the training set.

So I wanted a coding scheme that used fewer tweak constants and came up with CodeLinear. CodeLinear throws away the tree structures and codes bits in linear order, as you might guess from the names. The bits are ordered by expected importance, and coding is done carefully to try to send each bitplane in order of most valuable to least valuable bits.

Again I always send the bitplanes in order from top bit to bottom bit. Within each bitplane, the samples and colors are no longer scanned in a simple order.

Instead we take all 8*8*3 (=192) samples, and reorder them based on expected importance. For simplicity, I use a static reorder that was computed on the training set. Note that unlike the arithmetic coding coefficients, this reorder is not succeptible to overtraining because it's almost always similar on all images and even if you get it slightly wrong it doesn't hurt too badly.

Because of this ordering we expect the samples bits to turn on roughly in order. That is, expect sample at index 0 to turn on first, then sample 1, 2, 3, etc. That is, we expect them to look like :


1 0 0 0 0 0 0 0
- 1 1 0 1 0 0 0
- - - 1 - 1 1 0
- - - - - - - 1
- - - - - - - -

(the vertical axis is the bitplane #, the horizontal axis is the sample #)
dashes are bits in samples that were previously significant in higher bit-planes
the 0s and 1s are part of the "significance front"

Or something like that - the index of the highest bit is generally decreasing.

We then send the bits in significance passes. We keep track of the highest-indexed sample that was significant in a previous bitplane, we'll call it "max_significant". (obviously this starts at zero). That is, for all samples indexed > max_significant , all of their bits sent so far are zero. For samples <= max_significant some may be significant and some may not be (of course the one *at* max_significant is always significant).

To send a bit plane we send three passes over the bits (note that using max_significant we don't actually have to walk over all the bits, in fact we generally rarely touch a bit more than once).

Pass 1 : previously signficant. For all indeces <= max_significant , if the value had previous bits on, then send the current bit. The current bit is likely to have a lot of information. It's arithmetic coded with a fixed probability.

Pass 2 : in signficant range. For all indeces <= max_significant , if the value did not have previous bits on (e.g. not coded in pass 1), then send the bit with another arithmetic probability. If the bit was on, that value is now significant, then send its sign bit raw.

Pass 3 : indeces > max_significant. These values all had no previous bits on. Send whether the current bit is now on with an arithmetic coded bit, if it comes on send the sign and also move max_significant up to this index as it is now on. We usually are sending lots of zero bits here, and the few on bits that we send are close to the start. For efficiency reasons, we thus code this differently. At each step first send a flag whether all remaining bits are zero or not. If they are not, we know at least one on bit follows. We then send how many indeces away that one bit is. We send this distance with unary. So we are doing two binary arithmetic codes, but we never actually send a "bit on/off" code. The decoder pseudocode looks like :


while( arithDecodeBit() ) // any remaining on
{
   max_significant ++;
   while( arithDecodeBit() ) // decode unary
      max_significant++;   
   // bit at max_significant now turns on :
   sample[ max_significant ] |= current_bit;
   // decode sign too
}

note that you have to arrange your code so that a zero bit terminates the loops, thus when the stream is truncated by zeros you don't get infinite loops.

That's it, pretty simple. We do reconstruction of missing bits like CodeTree does. You can see it uses 4 binary arithmetic codes, which means 4 constants which must be tweaked. Note that binary arithmetic coding with constant probabilities can be made very very fast. You could in fact use table-based coders that do no multiplies or divides (see the Howard-Vitter papers for example which are not patented). Ideally you would tweak a few different sets of 4 constants for different image types and be able to select a constant set for each image. Depending on your situation this may or may not be possible.

Note that there's an obvious possibility for improvement in CodeLinear if you did some R/D optimization in the encoder. In particular, if a value with a high index comes on prematurely all by its lonesome, it can cost a lot of bits to send and not contribute much to distortion. This value should really just get quashed. For example if the bits look something like :


1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 1 0 
- 1 1 0 0 0 0 0 0 0 0 0 0 0 0 - 0 
- - - 1 1 1 1 0 0 1 1 0 0 0 0 - 0 
- - - - - - - 1 1 - - 1 1 1 1 - 0
- - - - - - - - - - - - - - - - 1

that one lone bit way out there really boofoos things, it makes the whole area preceding it come on too soon. Note that there are also obviously more sophisticated models of "significance" that you could use other than the simple "max_significant" index thing. I haven't explored that.

02-10-09 - How to fight patents

Some rich folks like Gates and Buffet could create a consortium that just patents everything it can. It would also accept "donations" of patents from people like us who want to make sure their work is free ("defensive patents") - eg if you invent something, you submit the paper to them and they patent it.

Obviously this institution could just make all its patents free for anyone to use. But it could also use them as a hammer to force open other patents.

For example, you could make all your patents absolutely free for any other institution who does not own any patents. If you do own any patents, then you cannot use our patents unless you also make your patents completely free under the same license. (sort of like GPL).

This might be too restrictive, but on the plus side if you just got some good patents into this institute, that would force lots of other people to open up theirs under the same license, and in the ideal world it would quickly snowball.

(of course this was the idea of GPL open source code as well, and I would say that has generally been a complete failure; the real result of the GPL is that nobody serious can use GPL code, so all the GPL libraries out there are just wasted).

2/09/2009

02-09-09 - void pointer

Urg WTF why doesn't C++ let void * be implicitly cast to any other pointer type !? It's fucking void * obviously I'm using that because it's untyped or type-unknown so I'm going to be casting it. Forcing me to cast in situations like that where it's obvious just makes me more inclined to cast all the time without thinking about it, which is much worse.

Being too strict about enforcing stupid rules is very negative. It makes people ignore the rules. You need to complain only when it really matters.

2/04/2009

02-04-09 - Exceptions

I'm not really high on the whole Herb Sutter super-exception safe everything, but they are really nice for error handling some times. They have two big wins in my opinion (vs. just having error return values and checking with if statements) :

1. Consolidating error handling code and putting the error-handling code in the place that actually can respond to the error in a reasonable way.

2. Correct default behavior if you are lazy or screw up and don't handle the error - it automatically propagates out rather than just being silently ignored. Standard if-checking badly violates one of my guiding principles in coding : if you don't write the right code, it just silently breaks; I always want my programs to noisily scream when I don't write the right code.

Anyway, I'm thinking about this because handling Async IO errors is a huge pain. The big standard problem comes from file opens. The thing is, my "fopen" doesn't actually open the file, it just queues an open request that will get done at some time in the future. If the open fails (say because the file doesn't exist), you don't actually see that until you try to get a char off the file and it's no good.

With if-checking it looks like :


File = fopen();
// no need to check error here really because this did nothing

... do other work to give it some time

char c = fgetc(File);
// ! boom this can fail because the file didn't actually open

// so fgets is no good, you have to do :
char c;
EStat status = fgetc(File,&c);
if ( status == error )
   ... clean up file, back out work already done, christ

With exceptions it's a million times cleaner. You just say the File ops can throw various IOExceptions. One of those IOExceptions is "FileNotFound". That exception might be thrown in the fopen() or in the next fgetc() or whatever.

In fact with exceptions and class back-out you can also have it automatically undo the temp work you did :

try {

File = fopen();

class X;
X.DoStuff();

char c = fgetc(File); // this might throw if the open actually failed

X.Read(File);

... okay, all good. now :

World.Apply(X);

} catch

Sadly I can't use exceptions for various reasons (aside from all the practical problems with exceptions, lots of clients have them disabled, so there you go), but my god the error stuff for this async IO is so nasty. One way I'm handling it is by sort of simulating the exception code style without exceptions.

What I do is still use an fgetc() that just returns a char. If the file fails to open, fgetc() just returns zeros, and an error flag is set in the file. That way you can go ahead and write straight line code without checking errors, and then once in a while you can check the error flag. So the code looks like :

{
File = fopen();

class X;
X.DoStuff();

char c = fgetc(File); // this just sets File.error

X.Read(File);

if ( File.ok )
   World.Apply(X);
// else X will just be destructed and not be loaded into the world

}

Sadly it's not actually so neat in practice. In particular with many layers of wrapping, exceptions just do What You Want magically. With an error flag I have to be pushing it through manually all the time. A lot of my files are actually a raw disk file, wrapped in a double-buffer file layer, wrapped in an LZ decompress layer. Each layer has to push out the status to the parent, and errors can occur at any point in the code because of the async nature.

2/02/2009

02-01-09 - Swap and Templates Part 2

So I'm back into the nightmare of std::swap and template overloading. I've been working on my cb::hash_table a bit, which I built on cb::vector so I'm seeing a lot of stuff with vector. One issue is when the vector changes capacity, you have to copy all the data over and delete the old ones. The normal way to implement vector is to do this with a bunch of uninitialized copy constructor calls, then call the destructor on all the old data. But that copy can be very expensive, and worse if your objects are large in memory it means you can temporarily double your memory use.

So I had the idea to use swap. When the vector capacity changes, first go ahead and default construct everything in the new space so it's valid. Then ::swap the new to the old, and then destruct the old vector. This is of course actually not ideal, what you really want is a single call that would "swap_construct" and object so that you don't have to default construct anything. But anyhoo, for many types of data it's easy to make a very cheap default constructor and a very fast swap.

Apparently this idea is old, and in fact newer versions of the Dinkum STL do this, they call it Swaptimization which you can find on that page if you search for swap. BTW it's also obvious that sometimes swapping is worse, eg. for vector< int > - but it's never much worse. If you wanted maximimum speed on silly cases like that, you could use a type-traits template to choose whether the class prefers swapping or copying. Actually in my old versions of my own vector I had a lot of type-traits to select the most efficient operation, but I've ripped that all out now. I find it too fragile and just also not important. (note that most STL implementations now use type traits for acceleration; I think this is really loathsome, because it's not exposed in a standard way, so you can't get to it for your own types - it makes them fact on basic types and their own types, and slow on your types, which blows - the whole idea of generic programming is that you can adapt it to work on your types, but the standards people have recently started giving special treatment to things in the STL, for example the STL gets different name lookup rules now (wtf)). (for type traits in the STL see for example stuff like "_IsOKToMemCpy" and "has_trivial_destructor" in STLport.)

Anyhoo, the problem now is that you need to make sure the right swap() is called. Ruh roh. You can't really correct overload std::swap in the std namespace ; see for example : STL defects #225-227 or PDF post about swap problems by Alan Griffiths . or C++ Standards - The "swap" Problem

So, what might you do instead? One option is to define your own swap() in your own namespace, and rely on the Koenig lookup to find your swap (cuz your objects are in your namespace). That doesn't work, partly because the STL calls std::swap explicitly. I'm not sure why they do that instead of just calling swap(). Apparently new MSVC uses ADL_Swap or some funny mumbo jumbo to try to fix this.

In any case, I have a hacky solution that I'm using for now.

I go into the STL headers to the definition of std::swap. It should look something like this :

   template < class _Tp >
   inline void swap(_Tp& __a, _Tp& __b) {
     _Tp __tmp = __a;
     __a = __b;
     __b = __tmp;
   }   

I change that to :

   template < class _Tp >
   inline void swap(_Tp& __a, _Tp& __b) {
     MySwap(a,b);
   }   

Now in my own code I have MySwap :

template < typename Type >
struct swap_functor
{
   void operator () (Type &a,Type &b)
   {
      Type c = a; a = b; b = c;
   }
};

template< typename Type > inline void MySwap(Type &a,Type &b)
{
   swap_functor< Type >()(a,b);
}

Which redirects MySwap to a functor, and the default implementation is the usual swap.

The reason to go to a functor is that you can take advantage of partial specialization due to the stronger support for class templates as opposed to function templates. Thus you can easily partial-specialize to all types of vectors (for example) :

template < class t_entry > 
struct swap_functor< cb::vector< t_entry > >
{
   void operator () ( cb::vector< t_entry > & _Left, cb::vector< t_entry > & _Right)
   {
      _Left.swap(_Right);
   }
};

The reason we do the redirects like this is that when you call anything in std::algorithms, they will call std::swap. First of all they will call the STL-provided overrides for std::swap for various STL types (like std::vector and std::string). Then they will call to the default, which calls MySwap. MySwap will go to the specializations that I wanted for my types. Finally it will fall back to the default swap.

This is the only way I know of to get specialization for both my types and the STL types. Note that to get the specializations on the std types, you have to call std::swap() - it's the "outermost" swap, and you always need to call to the leaves.

Now, for example, if I wanted to also get somebody else's library integrated, call it "somelib" that implements "somelib::swap" , I would do all the above, but the cb::swap_functor default implementation would be changed to call somelib::swap. Again everyone should call std::swap to get all the specializations.

Obviously this is not ideal because it involves digging into other people's libraries.

BTW using Swap in vectors is obviously semantically much better, even if it isn't an optimization. If you consider default constructed objects to be "nulls" and objects that have been set up and pushed into a vector to be "initialized", then the standard copy way of growing a vector temporarily makes a bunch of new "initialized" objects. The swap way just moves them and doesn't temporary change the number. A common case is stuffing something like a shared_ptr ref-counted pointer into a vector - the swap way keeps the ref counts all invariant, while the copy way temporarily bumps the refs up then back down again.

BTW one thing I noticed while testing this is that std::sort doesn't seem to use swap (!?). I always thought it did. Other algorithms, like std::reverse and std::random_permutation do call std::swap. sort seems to invoke the copy constructor and assignment a lot. The RAD version rr::sort does work entirely through swap, which means it should beat std::sort very trivially in bad cases, like vector< vector< int > >

ADDENDUM : thanks to the commenters; I think what I like best is to make my swap_functor default to doing a *byte* swap. Then any object which wants to be swapped and can't stand to be byte swapped (very rare) would override swap_functor. Doing it that way means almost nobody ever has to override swap_functor, it just uses the default (byte swap) almost always.

BTW the cool way to byte swap is something like this :


#pragma pack(push)
#pragma pack(1)
template < int n_count >
  struct Bytes { char bytes[n_count]; };
#pragma pack(pop)

template < typename t_type >
void ByteCopy(t_type * pTo,const t_type & from)
{
   typedef Bytes< sizeof(t_type) > t_bytes;

   *(reinterpret_cast< t_bytes * >(pTo)) = reinterpret_cast< const t_bytes & >(from);
}

template < typename t_type >
void ByteSwap(t_type & a,t_type & b)
{
   typedef Bytes< sizeof(t_type) > t_bytes;

   t_bytes c; // don't use t_type cuz we don't want a constructor or destructor
   ByteCopy(&c,reinterpret_cast< const t_bytes & >(a));
   ByteCopy(&a,b);
   ByteCopy(&b,reinterpret_cast< const t_type & >(c));
}

The nice thing about this is for small types the compiler does the right thing and just generates mov instructions. For large types you would want to call a memswap function, but you would never do this on large types so punt.

1/30/2009

01-30-09 - SetFileValidData and async writing

Extending and async writing files on Windows is a bad situation. See for example Asynchronous Disk I/O Appears as Synchronous on Windows . Basically you should just not even try, but if you really want to give it a go :

1. Open your file for OVERLAPPED and also for NO_BUFFERING. Buffering helps reads, but it severely hurts writes. It takes my write speeds from 80 MB/sec down to like 30 MB/sec. I suspect that the reason is that buffering causes the pages to be read in before they are written out. (it's sort of like using cached memory - it will fetch in the pages even though you're doing nothing but stomping all over them).

2. Use the undocumented NtSetInformationFile to resize the file to its full size before you write anything. SetEndOfFile will only work for page size granularity, NtSetInformationFile can do arbitrary sizes. BTW this is also better for fragmentation than just writing lots of data onto the end of the file, which can cause NTFS to give you lots of allocation chains.

3. Use SetFileValidData to tell Windows that whatever is in the sectors on disk is okay. If you don't use SetFileValidData, Windows will first zero out each sector before you get to touch it. This is like a security thing, but obviously it's pretty bad for perf to basically write the whole file twice. SetFileValidData will fail unless you first ask for your process to get the right permissions, which of course will only work for processes running as administrator. Okay, I did all that. this post is okay but dear god don't read the thread.

If you do all those things right - your WriteFile() calls will actually be asynchronous. The actual speed win is not huge. Part of the problem is the next issue :

When I do all that, I start hitting some kind of weird OS buffer filling issue. I haven't tried to track down exactly what's happening because I don't really care that much, but what I see is that the writes are totally async and very fast (> 100 MB/sec) for some random amount of time (usually up to about 10 MB of writing or so) and then suddenly randomly start having huge delays. The write speed then goes down to 40 MB/sec or so.

ADDENDUM : when I say "you should not even try" I mean you should just live with WriteFile() being synchronous. It's plenty fast. Just run it from a thread and it still looks async to your thread (you need a thread anyway because OpenFile and CloseFile are very slow and synchronous; in fact the only thing you can rely on actually being fast and async is ReadFile). Also just live with the fact that Windows is zeroing the disk before you write it, everyone else does.

01-30-09 - Stack Tracing on Windows

There are basically 3 ways to capture stack traces on Windows.

1. Manually walking ebp/esp ; this steps back through the frame pointers, it relies on the callers stack being pushed. This is basically what RtlCaptureStackBackTrace or DmCaptureStackBackTrace does, but you can also just write it yourself very easily. The advantage of this is it's reasonably fast. The disadvantage is it doesn't work on all CPU architectures, and it doesn't work with the frame pointer omission optimization.

For info on RtlCaptureStackBackTrace, see Undocumented NT Internals or MSDN

2. StackWalk64. This is the new API you're supposed to use. The advantage is it works on all CPUs and it even works with frame pointer omission (!). But you can see from that latter fact that it must be very slow. In order to work with FPO it loads the PDB and uses the instruction pointer map to figure out how to trace back. It also can trace through lots of system calls that normal ebp-walking fails on.

See gamedev.net or ExtendedTrace for examples. But it's really too slow.

3. Manual push/pop in prolog/epilog. Uses the C compiler to stick a custom enter/leave on every function that does a push & pop to your own stack tracker. Google Perftools has an option to work this way. The "MemTracer" project works this way (more on MemTracer some day). The nice thing about this is it works on any architecture as long as the prolog/epilog is supported. The disadvantage is it adds a big overhead even on functions that you never trace. That rather sucks. Stacktraces are very rare in my world, so I want to pay the cost of them only when I actually do them, I don't want to be pushing & popping stack info all the time.

1/28/2009

01-28-09 - Graph Memory

So we have a thing to track memory allocs with a stack trace, la di da, no big whoop. I log it all out to a file. So I wrote a thing to parse them into a hierarchy and spit them out with tabs for tabview . That's awesome.

Then I thought, hey, Atman makes these awesome graphs with "graphviz" so maybe I'll try that. One disadvantage of the pure hierarchy view in tabview is that you can't really see the flow when lines merge back up. That is, call graphs are not *trees* they're *DAGs*. Sometimes the stack hierarchy forks apart but then comes back together again. Graphviz should be able to show this neatly. Graphviz makes "SVG" files that you can just load with Firefox (Firefox 3's support is much better than 2's).

Anyway I made some graphs with various options and it's kinda cool. Here's an example : Allocs1.svg (you'll need to use ctrl-wheel to zoom out to see anything). Apparently if you click this link Firefox does nothing good. You have to download it - then open it with Firefox. WTF. Apparently it's my Verio servers doing the wrong thing . Yegads I hate the web.

Not bad, but I'm a little disappointed with graphviz's layout abilities. In particular the actual cell and edge layout seems very good, but they seem to have literally zero code to try to put the labels in good places.

In a bit of odd deja vu, this was one of the very first things I ever worked on as a professional programmer in 1991; I worked for a GIS company that had an old COBOL GIS database engine that worked with Census data and such; I wrote them a new C front end with graphics for PC's and such, and one of the things you have to do is take this raw street data with lat/long coordinates and street names and do some nice munging to make it look okay; a big part of that is a lot of heuristics about putting the labels for streets in good places (and when to repeat the label, etc.).


ADDENDUM : talking to Sean I realized you really want the graph to have different sizing/color options, be hierarchical, interactive, and stable.

That sounds hard, but I don't think it actually is. The key thing that makes it easy is that there is very good hierarchy in this information, and you can create the graph incrementally. I think that means you can just use simple iterative penalty methods to make the graph stable.

Here's my proposal :

Start with the graph at very coarse granularity ; maybe directory granularity of you have a few directories and that makes sense, else file granularity. Whatever coarse level so you have < 32 nodes or so. Just use a solver like graphviz to make this initial graph.

Now, interactively the user can click any group to expand its hierarchy. When that happens the big cell splits into various pieces. You just create the new pieces inside the parent and make the new edges - and then you just let them time evolve with a semi-physical iterative evoluton.

You apply a penalty force for intersection with neighbors to drive the nodes apart so there's no overlap. You similarly apply forces with the edges to make them never intersect edges. And the edges also act kind of like springs, applying forces to try to be short and straight. Stable 2d physics is a pretty solved problem so you just let them run until they settle down. Note that as they spread apart they can force the other nodes in the graph to move around, but it's all nice and smooth and stable.

I think it's much easier to treat the canvas as just infinitely large and let your nodes move apart all they need to. Graphviz does everything oriented towards being printed on a page which is not necessary for the interactive view.


ADDENDUM 2 : after figuring out some evil things I've got graph_viz making better graphs. Here's an example : allocs2 SVG (direct clicking should work now too).

See comments on this post for details.

I'm still not happy with the edge labeling, and the FF SVG navigation is not ideal, but it's useable now.

1/27/2009

01-27-09 - Oddworld Memory Memories

I've been working on some memory allocator related junk recently (just for laughs I made my SmallAllocator in cblib work lock free and timed it running on a bunch of threads; an alloc takes around 200 clocks; I think there may be some cache contention issues). Anyway it reminded me of some funny stuff :

Munch's Oddysee was a wild painful birth; we were shipping for Xbox launch and crunching like mad at the end. We had lots of problems to deal with, like trying to rip as much of the fucking awful NetImmerse Engine out as possible to get our frame rate up from 1 fps (literally 6 months before ship we had areas of the game running at 1 fps) (scene graphs FTL). And then there were the DMusic bugs. Generally I found that dealing with XBox tech support guys, the graphics and general functionality guys have been really awesome, really helpful, etc. Eventually we trapped the exception in the kernel debugger and it got fixed.

Anyhoo... in the rush to ship and compromise, one of the things we left out was the ability to clean up our memory use. We just leaked and fragmented and couldn't release stuff right, so we could load up a level, but we couldn't do a level transition. Horrible problem you have to fix, right? Nah, we just rebooted the Xbox. When you do a level transition in Munch, you might notice the loading screen pops up and is totally still for a few seconds, and then it starts moving, and the screen flashes briefly during that time. That's your Xbox rebooting, giving us a fresh memory slate to play with. (shortly after launch the Xbox guys forbid developers from using this trick, but quite a few games in the early days shipped with this embarassing delay in their level load).

For Stranger's Wrath we used my "Fixed Restoring" Small Allocator, which is a standard kind of page-based allocator. We were somewhat crazy and used a hefty amount of allocations; our levels were totally data driven and objects could be different types and sizes depending on designer prefs, so we didn't want to lock into a memory layout. The different levels in the game generally all get near 64 MB, but they use that memory very differently. We wanted to be able to use things like linked lists of little nodes and STL hashes and not worry about our allocator. So the Small Allocator provides a small block allocator with (near) zero size overhead (in the limit of a large # of allocations). That is, there are pages of some fixed size, say 16 KB, and each page is assigned to one size of allocation. Allocations of that size come out of that page just by incrementing a pointer, so they're very fast and there's zero header size (there is size overhead due to not using a whole page all the time).

That was all good, except that near the end when all the content started coming together I realized that some of our levels didn't quite fit no matter how hard we squeezed - after hard crunching they were around 65 MB and we needed to get down to 63 or so, and also the Fixed Restoring Allocator was wasting some space due to the page granularity. I was assigning pages to each size of allocation, rounding up to the next alignment of 8 or 16 or 32. Pages would be given out as needed for each size bucket. So if a size bucket was never used, pages for that size would never be allocated.

This kind of scheme is pretty standard, but it can have a lot of waste. Say you allocate a lot of 17 byte items - that gets rounded up to 24 or 32 and you're wasting 7 bytes per item. Another case is that you allocate a 201 byte item - but only once in the entire game ! You don't need to give it a whole page, just let it allocate from the 256-byte item page.

Now in a normal scenario you would just try to use a better general purpose allocator, but being a game dev near shipping you can do funny things. I ran our levels and looked at the # of allocations of each size and generated a big table. Then I just hard-coded the Fixed Restoring allocator to make pages for exactly the sizes of allocations that we do a lot of. So "Stranger's Wrath" shipped with allocations of these sizes :


static const int c_smallBlockSizes[] = { 8,16,24,32,48,64,80,96,116,128,136,156,192,224,256,272 };

(The 116, 136, 156, and 272 are our primary GameObject and renderable types). (yes, we could've also just switched those objects to a custom pool allocator for the object, but that would've been a much bigger code change, which is not something I would want to do very close to shipping when we're trying to be in code lockdown and get to zero bugs).

old rants