As I mentioned in the previous post, I’m polishing NihAV and improving some bits here and there. In this post I’d like to describe what has been done and what should be done (but not necessarily will be done).
For about a month I’ve been working on adding various functionality and functionality to support that functionality and resolving recursive dependency on that functionality as well. I wanted to have a better means to test decoder output so I tried to write a simple player. Which required implementing missing bits in nihav_core::scale and implementing sound conversion. And then implementing support for floating point numbers in nihav_core::io::byteio (at least Rust offers functionality to convert float into its integer representation and back). And then optimising scaler since it took two thirds of playback CPU. And also implementing seeking in demuxers (so that nihav-tool can start decoding from some position and skip uninteresting first frames).
Let’s start with the lest significant feature—seeking. I’ve introduced SeekIndex structure that can be filled by demuxer on opening format and later used to calculate seek position. I wonder if it’s worth to add automatic index building or not. At least I expect to have significantly less demuxers than decoders so it can be changed later with relative ease. Currently AVI and RM demuxers support seeking.
Now to testing. There are two different things: internal crate testing and manual testing with nihav-player. First, let’s talk about internal testing.
Rust allows modules to have separate test functionality so normally built crate won’t contain it but cargo test will build special version with tests enabled, run it and report which tests failed with the output they produce. Usually I abuse this functionality to debug decoder (since I don’t have to build full nihav-tool but rather just two-three crates required for decoder and its family). So previously test function just decoded payload and optionally output sequence of images or wave file. It’s good enough for manual checking or to ensure that it works but it’s not good enough for regression testing. So I’ve finally added MD5 hash support in testing function so now I can also compare decoded output to MD5 for the whole output or per-frame (I still need a better solution for floating point audio but it can wait). Also because I always operate either packed buffers or in native endianness and calculate MD5 directly on those buffers in the way defined by me it should produce the same hash on big-endian systems as well (it’s just I still remember the times when I forgot to add output colourspace conversion for high-bitdepth or RGBA output for FATE test in order to fix them on PowerPC).
And finally the hairiest part—nihav-player. It’s very primitive player that can only play audio and video hopefully in sync and quit when ‘q’ is pressed. And yet it both required a lot of additional code to be written and reduced the complexity of testing—for instance it demonstrates chroma artefacts (or simply swapped chroma planes) and playing a file takes less time than waiting for decoding to finish and reviewing frames one by one.
Here’s a list of features I had to add or improve just for player:
Another thing I had to improve for my player is sdl crate. Yes, my main development system is too old for SDL2 (with or without sdl2-build crate) so I stuck to SDL1. The Rust wrapper lacks support for YUV overlays (it was trivial to add) and sane audio handling (i.e. audio callback is just a function pointer which is hardly usable; I’ve changed it to accept some audio callback track in style of SDL2 wrapper).
The player itself took a lot of fiddling because I barely remember how I played with SDL more than ten years ago and I’ve not played with multi-threading much. In either case spawning threads with moving variables and channels for passing data between threads was a bit bumpy but it works. In result I have main thread that demuxes data, passes it to separate audio and video decoding threads, and then receives decoded video surfaces or overlays from video thread and displays them when the time comes (or discards if it’s too late—I should add a control for decoder to skip e.g. B-frames during decoding). Audio thread decodes audio and puts it into FIFO which gets emptied by audio callback when SDL calls it. Video thread decodes video, converts and/or scales into RGB32 or YUV420 and sends results back to main thread in either SDL RGB surface or YUV overlay (to my surprise they are not mutually exclusive).
In either case that’s just a first attempt to build proof of a concept player just for looking how decoders work and eventually I should write a proper one with better syncing, interactivity and other bells and whistles. The current one does its task good so I could watch a lot of samples for various decoders and see what should be fixed. I’ve fixed bugs in Indeo 3-5 decoders already plus swapped chroma planes in other decoders but H.263-based decoders still suck a lot and VP3-7 decoders are even worse. So there’s still a lot of fixing for me left.
At least after all those fixes and rewriting player I should have something substantial to show the world.
In the previous post I’ve mentioned that one of the levels contain an unfinished witch puzzle probably based on Match Two that also contains various photos of the (presumably team and their families (about a dozen of them actually). I also wanted to tell a bit more about it and since I’m working on boring parts of NihAV and it will take a while, why not tell the story now?
Here’s the actual mini-game background:
There are two areas of interest there—actual playing field where tiles are placed and smaller area where probably the clip from the movie would play on winning (the dimensions seem to fit). There are about twenty various symbols that can be placed there (plus some that form the word “witch”). Here are some of them:
The other pictures include scales, crumbs or stones, ladle, lead ingot, fire, newt (…it will get better!), log, bridge, apples and more.
And the playing area is suspiciously the same size as the pictures. I can only make theories that the pictures would be revealed there if you win the game normally or that it was an Easter egg in an unfinished mini-game. Here’s one for the reference:
I still have a faint hope that ScummVM developers will support the game one day and then we find out how it’s supposed to work (or that my guesses were all wrong as usual). Until then I can just occupy myself with something else.
First of all, unlike Rheinland Palatine railways I’ve not travelled on all local railways here because of certain geographic peculiarities it takes less time to get to the farthest corners of Rheinland-Pfalz from here than to places around Bodensee. And I can’t visit all historic railways either because one of them is located in the middle of nowhere with the only connection being a bus that comes there maybe twice a day.
Nevertheless, I’ve seen most of them and here I’d like to describe my impressions about them.
Let’s start with the main line. Due to its strategic position at the border of Germany, Baden has connections with various neighbouring countries. So the main line would be the backbone of Europe connecting North and South, from Rotterdam to Genoa along the Rhine. One of the important parts of it is the track from Mannheim to Basel with branches that connect it to France as well. Unfortunately its importance was demonstrated when because of botched tunnelling they had to close the part between Rastatt and Offenburg and it had been causing chaos for many weeks because there are no sane alternatives to it: all other lines are single-track and French ones are not electrified—while Rheintalbahn is four-track for most of its length (which was mandated by an international agreement between Switzerland, Germany and Italy—yes, it’s that important).
I mentioned branches to France before. There are two of those: around Strasbourg and Mulhouse (there had been more but France decided to cut ties like they were afraid of German army or something) and those are essentially the main connections between Germany and France. The only other double-track line goes via Saarland and it’s rather a joke. It takes 1 hour 22 minutes from Mannheim to Saarbrücken with express train like ICE or TGV and 1 hour 36 minutes with regional train. That’s all you need to know about how fast that line is. The other lines are unelectrified single-track railways connecting rural towns: Strasbourg-Lauterbourg-Wörth am Rhein, Strasbourg-Wissembourg-Winden(-Neustadt an der Weinstraße), Metz-Apach-Perl(-Trier).
Now let’s talk about less important lines that are mostly connected to the main one. In Mannheim there’s a line going to Eberbach, Lauda and Würzburg (very scenic, I’d definitely visit it again) with branches to Miltenberg and Aschaffenburg. Near Bruchsal there’s also a high-speed branch to Stuttgart. From Karlsruhe you can take Residenzbahn to Stuttgart (fun fact: previously there were both Regio-Express and Inter-Regio-Express trains from Karlsruhe to Stuttgart with no other difference beyond that Inter- part, now there are only IREs left). From Offenburg the main line branches into Schwarzwaldbahn to Konstanz and further to Switzerland (very scenic but I’ve seen almost everything there). From Freiburg there’s a line to Donaueschingen (i.e. to Schwarzwaldbahn) known mostly for running through Höllental with a station Himmelreich there (for non-German speaking people like me: it’s Heaven Kingdom town in Hell Valley).
And in Basel the main line transforms into Hochrheinbahn that runs along the Rhine to Konstanz. The line has several connections to Switzerland, like Waldshut and Koblenz are connected with the oldest rail bridge over the Rhine. It is also remarkable that some stations on Swiss territory still belong to Germany, the most known is Basel Badischer Bahnhof but there are more in e.g. Beringen and Neuhausen.
Of course it’s worth mentioning that some rail lines are serviced by trams (it’s called Karlsruhe model for a reason), sometimes only by trams. For example you can ride S5 from Karlsruhe to Bietigheim-Bissingen which is also the end station for S5 coming from Stuttgart. And S4 for a long time hold the record for the longest tram line in the world (more than 150 kilometres from Öhringen to Karlsruhe and then to Freudenstadt), now it has been split into S4 and S8.
And now historical railways.
There are only three historical railways in Baden plus some smaller neglected branch lines (like Neckarbischofsheim-Hüffenhardt) that have only occasional traffic. And some routes are so scenic that they have historic trains running on them like part of Schwarzwaldbahn from Triberg or the ring around Kaiserstuhl. The lines I’m talking about are Kandertalbahn (Kandern-Haltingen, close to Swiss border), Sauschwänzlebahn and international historical railway Etzwilen-Singen.
Kandertalbahn is nice but without any distinct features, Sauschwänzlebahn is so remarkable that I wrote a separate post about it long time ago and Etzwilen-Singen line is worth talking about here. It runs across the Rhine from Switzerland to Baden (unfortunately not the full way to Singen, the track between Rielasingen and Singen requires some re-constructing. So while it’s Etzwilen-Singen line, the train I took ran from Stein am Rhein (hauled by electric locomotive to Etzwilen and changed to proper steam engine there) to Rielasingen. And the second class part of carriages looked exactly like third class part but that’s some Swiss peculiarities I don’t care about.
It might be a surprise but despite being part of high-speed corridor Paris-Stuttgart-München, Württemberg has just one bit of high speed line connecting Stuttgart to Mannheim and Stuttgart-Ulm part is very slow speed (essentially it takes an hour no matter what train kind it is). Why? Mountains and too steep curve. That’s why they’re building a new line and burying main station in Stuttgart (the infamous Stuttgart 21 project). And since it costs a lot of money they decided this is more important than building missing tracks on Rheintalbahn as the agreement demands. At least the railway from Stuttgart to Friedrichshafen via Ulm plays central role in the song Auf de’ schwäb’sche Eisebahne, the only railway-related German song I know.
The other notable railways are Stuttgart-Aalen-Crailsheim-Nürnberg, Miltenberg-Lauda-Crailsheim (it’s known as Westfrankenbahn), Aalen-Ulm and so-called Ring Railway (Villingen-Rottweil-Aldingen-Tuttlingen-Geisingen-Donaueschingen-Villingen). There are many other railways but they are not that remarkable.
And I should mention the quarter of German cogwheel railways (or half if you exclude Bavaria as Bavarians do) is located in Stuttgart—Zacke (the line Marienplatz-Degerloch). Speaking of German cogwheel railways, there was one connecting Reutlingen and Kleinengstingen but it got dismantled back in 1969 leaving confusing names like Südbahnhof Nord (South Station North) station in Reutlingen and Zahnradbahn Honau-Lichtenstein society that sometimes run steam trains near Reutlingen on tracks that have no relation to that old cogwheel railway at all.
Speaking about still existing historical railways, there is a lot of them of varying forms and gauges.
And now for something about the same. Here’s a picture of train on Öchsle:
While talking about historical railways in Württemberg it’s worth mentioning Ulmer Eisenbahn-Freunde socienty (Railway Friends of Ulm if translated directly). They run steam trains on some of those lines and on other railways too (like Karlsruhe-Ettlingen-Bad Herrenalb which is serviced by trams).
Here’s a train operated by them in Gerstetten:
It was built in 1921 in Karlsruhe and it’s worth saying some words about the manufacturer. Maschinenbau-Gesellschaft Karlsruhe went bankrupt in 1929 but before that it was not merely a place where locomotives were built and whose founder later founded Maschinenfabrik Esslingen as well (this one still exists) but it was also a place where such people as Niklaus Riggenbach (one of the pioneers of cogwheel railways; he moved to Karlsruhe when he decided he wants to know how to build locomotives), Carl Benz (son of locomotive driver and inventor of car), Gottlieb Daimler (the guy who put internal combustion engine wherever possible) and Wilhelm Maybach (guess why he got a car and couple of streets named after him). It’s always nice to find out about such development of seemingly unremarkable things.
Finally NihAV got full-feature VP7 decoding support (well, except one very exotic case for a very exotic mode) so now I can move to other things like actually making various decoders bit-exact, fixing other bugs in them, adding missing pieces of code for player and even documenting stuff. I hope to give a presentation of my work on VDD 2020 or FOSDEM 2021 (whichever accepts it) and I want to have something decent to present by then.
Anyway, here’s a review of VP7.
First of all, I’d like to note complete lack of cooperation from Baidu. Back in the day when they just bought On2 and presented VP8 people asked them to release VP7 as open source. To which they replied that their developers are working on adding new features to the codec and they don’t have time to locate the old sources. Even better, we have managed to find VP6 and VP7 specification on some Chinese site and Baidu people could not even bother to release that. As it was said in certain series of games, press <F>+<U> to pay respect.
Well, maybe they were too ashamed of it because it looks like half-finished VP8 specification without source code in the appendix. It has the same passages written by Captain Obvious like this passage:
A reasonable “divide and conquer” approach to implementation of a decoder is to begin by decoding streams composed exclusively of key frames. After that works reliably, interframe handling can be added more easily than if complete functionality were attempted immediately.
And very informative things like:
const Prob kfUVmodeProb [numUVmodes - 1] = { ??, ??, ??};
Nevertheless, it was my primary source along with the binary specification which obviously has all those missing tables, pieces of code and more.
So let’s move to overall VP7 description. VP7 is Duck rip-off of H.264: the same coding method as used in VP5 and VP6 (to the point where I could reuse bits for decoding coefficients and motion vectors from my VP5/6 decoder) with overall structure of H.264 (4×4 blocks clustered into 16×16 macroblocks, optional separate 4×4 block for luma DCs, spatial prediction and such). It’s funny that if you consider how it does certain things then VP8 looks like a very minor improvement on VP7 (fixed probabilities are internally calculated from the fixed tree weights, golden frame can be partially updated after decoding a frame and other details). Another fun fact is that VP7 decoder recognizes three FOURCCs—VP70, VP71 and ONYX (and you can still see onyxc_int.h in libaom). VP71 is claimed to be error-resilient profile and it has some small changes in frame header and it can signal to preserve scan order and some frequencies (that cannot be changed regardless) and restore them after the frame has been decoded.
The nastiest part there is so-called macroblock features: if they are enabled then each macroblock can have something in its decoding process altered. First feature forces decoder to use different quantiser for current macroblock, second one tells decoder to use different loop filter strength, third one tells decoder to update golden frame with this macroblock and the last feature tells decoder to reconstruct macroblock in weird way. There are two parts there: residue reconstruction (i.e. how to apply reconstructed 4×4 blocks to the macroblock) and motion prediction. Residue reconstruction has four modes: normal, interlaced, very interlaced (i.e. using every fourth line instead every second one) and koda extremely interlaced (each 4×4 subblock is treated as 16×1 line instead). Motion prediction has all these modes and also warped motion (when you copy not from a rectangle but rather from a parallelogram with 45-degree angle). I can understand many things but “copy from a line like it’s a 4×4 block” is too much for me and thus I left it unimplemented (not that I have any samples for it either).
But other than that I have all features implemented (even though I don’t have samples to test them) and decoder produces recognizable picture so I’m happy with that and can move to other things as I stated in the beginning.
And in the unlikely case somebody reads this post and asks that question—no, I’m not going to implement VP8/VP9/AV1 decoder. Those are not Duck codecs and I’d rather reverse engineer some obscure codec instead.
I’ve spent two days on something that I wanted to do long time ago but postponed because of various reasons including complexity. But now thanks to Ghidra I’ve finally managed to pry into resources of Monty Python and the Quest for the Holy Grail and decode videos (and more!) stored there.
Probably it was VP7 (again) that made me look into it, but since Ghidra has decompiler for 16-bit code as well, I’ve tried it on libraries of surprisingly small game engine (pity that ScummVM does not even plan to support it but I’m in a similar situation with codecs). And what do you know, they have a special library dealing with resource files that included unpacking images (but not audio—there’s audio library for that).
First, let’s talk about resource file organisation and why it baffled me for too long. The files are organised into several chunks: header that contains offsets of the other chunks at the end of it (around bytes 0xE2..0x141), then you have actual table of contents (more about it later), then some small binary data chunk, then list of strings related to file names, then some chunk that looks like game script (in binary form), then palette and finally another list of strings that looks like variables list. It is no wonder I could not do anything useful with it since actual table of contents is hidden somewhere near the end of file but not exactly at the end. Also each game scene corresponds to its own archive which makes it clearer what to expect there (previous version used in Monty Python’s Complete Waste of Time even has a description right in the header and table of contents stored right after it, I might look at it later but no promises).
Now, files in the resource archive. The catalogue has only file type, its size and offset in the archive and nothing else. There are more than a dozen file types, 1 being images (both background and sprites), 2 being movies (the thing I was hunting), 4 is for MIDI tracks, 9 is for digitised audio, the rest I don’t care about.
Let’s move to simple things like music and audio. Music is stored in its custom format with four bytes per command except when high bit is set (then it’s delta time for next command). Why so? Probably because it’s being played by sending single MIDI commands and at least on Windows it requires packing them into 32-bit word anyway. Audio is stored in files with some header and either raw form or with IMA ADPCM compression. The notable thing is that it does not use any multiplications in any form—instead it has precomputed table for all eight possible values for each step.
Images are another interesting topic. First of all, palette is stored as 944-byte file with 32-bit entries (so it’s just 236 colours) but somehow decoded files use it with bias of ten, i.e. decoded index 13 corresponds to palette entry 3, index 27 is for entry 17 etc etc. I have no idea what it does with first colours in the palette (though sometimes images are not colorised properly so maybe they need different palette bias or a different palette at all).
Second, images employ two different compression schemes: RLE and LZW. RLE is rather trivial except that it uses opcode zero to signal end of line (and double zero for end of image):
LZW-compressed image splits image into chunks that may be uncompressed or compressed with LZW using 10, 11 or 12 bits per index. And after all these years I was still able to correctly guess it was LZW and write a decoder that worked fine without resorting to any documentation or even studying the decompiled code (I just needed to realize that it reads sequences of n-bits indexes and does something with them to output bytes).
And finally the movie format. Those are actually stored in two files—movie data and companion frame index (i.e. frame type, size and offset). After extracting frames I could not realize what to do with them until I noticed that frame type 2 all have the constant size of 11025 except for first few. Obviously they turned out to be IMA ADPCM compressed audio frames (and since they were the first frames in every movie it was hard to guess the format looking at semi-random data without header). Frame type 0 turned out to be the same image format as normal pictures.
Frame type 1 turned out to be inter-frames that use index 0 for pixels that should not be updated.
Unfortunately even if I now have a way to decode the files I cannot add direct support for them in NihAV since it requires at least three different files to play it (palette, frame index and frame data). At least now I know I can transcode them into something when the need arises.
Now let’s talk about Ghidra a bit. Without it this probably would have not happened since I’m not that young to spend time on translating tons of disassembly (and REC failed to do a good job). Ghidra support for 16-bit code is far from perfect with all that mess with far pointers consisting of two 16-bit words, external DLL functions linked by ordinals (so I had to match them by hoof and rename where possible) and general confusion with int treated as 4-byte in some contexts but 2-byte in another. And yet it did the job and helped me to understand how the game libraries work and that’s what really counts.
And as a bonus here’s a team photo extracted from concentration mini-game related to the witch scene (not the Simon Says clone with fires but probably Match Two clone that I’ve never seen in-game before):
VP7 is such a nice codec that I decided to distract myself a little with something else. And that something else turned out to be MidiVid codec family. It turned out to be quite peculiar and somehow reminiscent of Duck codecs.
The family consists of three codecs:
I’ve actually added MidiVid decoder to NihAV because it’s simple (two hundred lines including boilerplate and tests) and way more fun than working on VP7 decoder. Now I’ll describe them and hopefully you’ll understand why it reminds me of Duck codecs despite not being similar in design.
This is a simple hold-and-modify video codec that had been used in some games back in PS2/Xbox era. The frame data can be stored either unpacked or packed with LZSS and it contains the following kinds of data: change mask for 8×8 blocks (in case of interframe—if it’s zero then leave block as is, otherwise decode new data for it), 4×4 block codebook data (up to 512 entries), high bits for 9-bit indices (if we have 257-512 various blocks) and 8-bit indexes for codebook.
The interesting part is that LZSS scheme looked very familiar and indeed it looks almost exactly like lzss.c from LZARI author (remember that? I still do), the only differences is that it does not use pre-filled window and flags are grouped into 16-bit word instead of single byte.
This one is a special best as it combines two completely different compression methods: the same LZSS as before and something used by BWT-based compressor (to the point that frame header contains FTWB or ZTWB IDs).
I’m positively convinced it was copied from some BTW-based compressor not just because of those IDs but also because it seems to employ the same methods as some old BTW-based compressor except for the Burrows–Wheeler transform itself (that would be too much for the old codecs): various data preprocessing methods (signalled by flags in the frame header), move-to-front coding (in its classical 1-2 coding form that does not update first two positions that much) plus coding coefficients in two groups: first just zero/one/large using order-3 adaptive model and then values larger than one using single order-1 adaptive model. What made it suspicious? Preprocessing methods.
MVLZ has different kinds of preprocessing methods: something looking like distance coding, static n-gram replacement, table prediction (i.e. when data is treated as series of n-bit numbers and the actual numbers are replaced with the difference between previous ones) and x86 call preprocessing (i.e. that trick when you change function call address from relative into absolute for better compression ratio and then undo it during decompression; known also as E8-preprocessing because x86 call opcode is E8 <32-bit offset> and it’s easy to just replace them instead of adding full disassembler to the archiver). I had my suspicions as n-gram replacement (that one is quite stupid for video codecs and it only replaces some values with some binary values that look more related to machine code than video) but the last item was a dead give-away. I’m pretty sure that somebody who knows open-source BWT compressors of late 1990s will probably recognize it even from this description but sadly I’ve not been following it that closely being more attracted to multimedia.
This codec is based on some static codebook for packing all values: block types, motion vectors and actual coefficients. Each block in macroblock can be coded with one of four modes: empty (fill with 0x80 in case of intra), DC only, just few coefficients DCT, and full DCT. As usual various kinds of data are grouped and coded as single array.
Motion compensation is full-pixel and unlike its predecessor it operates in YUV420 format.
This was an interesting detour but I have to return back to failing to start writing VP7 decoder.
P.S. I’ll try to document them with more details in the wiki soon.
P.P.S. This should’ve been a post about railways instead but I guess it will have to wait.
I’ve wasted enough time on AVC decoder for On2 family so while it’s not working properly for those special cases I’m moving to VP7 regardless.
For those who don’t know (or forgot; or never had a reason to care) On2 AAC is AAC-LC rip-off with some creative reconstruction modes added to the usual long/short windows. I’ve failed to understand how it works before and I fail to understand how it works still. But at least some details are a bit clearer now that I’ve analysed the whole codec from scratch with less guesswork.
The codec has three IDs that it recognizes: 0x500, 0x501 and 0x1234. First two are different only in the aspect that one handles singular packets and another one handles several packets glued together prefixed with size. The last ID is simply recognized but it does not have any special handling.
The tricky part is some special modes that do some heavy processing of data. For most modes you invoke IMDCT and that’s all, here you do some QMF-like filtering (probably for transients extraction), then you perform RDFT (previously I thought it was plain FFT but after long investigation it turned out to be RDFT after all) on quarters, merge those quarters using filters that look like convolution filters for four sub-bands, perform RDFT again on the whole block and add some transients. And after that you still may need to reverse the data before using permuted window for overlap-add operation. In other words it’s not fun and I lack education for recognizing all those algorithms used, why they’re used and where it goes wrong.
So hopefully I’ll return to it some day to fix it for good but now VP7 awaits (so I can at least formally declare Duck codecs family done and move to implementing missing bits in the framework itself).
When I think I’ve learned enough about Dingo Pictures there’s something new appears. From Toys review I’ve learned that Roswitha Haas also published a book (or maybe several) that are directly related to the subject at hoof. And of course being curious I’ve decided to buy it.
Amazon mentions two other books—an adaptation of Land Before Time and Jimmy Button but I dared not to buy them (one has definitely nothing to do with Dingo Pictures and I fear to find out that the first one is not about Tio).
Anyway, let’s look at the alternative version of King of the Animals Part 2 (spoiler: it’s still much better than Disney remake).
First let’s mention some technical technical details: this book was released by Unipart in 1999, the same year as King of the Animals Part 2 but it had completely different artists (from Bayreuth even, while the publisher is located in Hessen not far from Dingo studio) with a bit different style as well. The same publisher printed children book adaptations of Disney stories too. But who cares about that, let’s look at the pictures!
If you wonder who is that magnificent warthog crocodile on the cover, that’s Dundee. If you wonder why he does not look like himself in the movie it’s because for some reason they’ve decided to cast warthog Bobo as Kemo’s friend (no idea why, artistic license?).
The story is still largely the same: there’s this king of the animals (aka lion), he has a son Kemo, there’s that black panther with his own son Pino and minions.
And there are diamonds. My God!
Pity they could not have a climactic fight here but here’s the best shot at it:
The ending is significantly different from the movie though and I’m not going to spoil it.
Overall, this was an interesting experience and while I’m not going to research it further I still hope more books might re-appear in the future.
While it’s summer and I’d rather travel around (or suffer from heat when I can’t), there has been some progress on NihAV. Now I can decode VP5 and VP6 files. Reconstruction still sucks because it takes a lot of effort to make perfect reconstruction and I’m too lazy to do that when simple demonstration that the decoder works would suffice.
Anyway, now I can decode both VP5 and VP6 files including interlaced ones. Interlacing in VP5/6 is done in very simple way like many other codecs: there’s a bit for each macroblock telling whether macroblock should be output in interlaced form or not.
Of course this being VPx family, they had to do it with some creativity. First you decode base interlaced bit probability, which is stored as 8-bit value while all other bit probabilities are stored in 7 bits. Then you derive actual probability for interlaced bit and decode it before any other macroblock information (including macroblock type—it’s that important). Probability is derived by companding base probability depending on whether last macroblock was interlaced (then probability is halved) or not (then it’s remapped to fit 128-255 range)—except for the first macroblock in a row which would use the base probability without modifications. And for VP6 you also have to use different starting scan order (band assignment for each coefficient, now it’s shuffled). This is so trivial that one would wonder why this has not been done in libavcodec decoder yet.
There are three possible things to do next: polish current implementation, move to AVC (On2 AVC that is) or move to AVC (Duck VP7 which is AVC ripoff). But probably I’ll simply keep doing nothing instead.
… gibt’s gar viele Haltstatione,
Kowelenz, Bingen, Bad Kreuznach, Lautre und Waldfischbach.
Ehm, sorry, wrong federal land for that song.
I think I can call my hobby project of travelling on all rail lines in Rheinland-Pfalz that are still in service complete.
Well, actually there are two tracks that I have not visited: couple of kilometres (about three) north from Unkel to the land border (when I went there the trains were going just to Unkel because of road works and I see no point of revisiting it again) and Zellertalbahn (Monsheim-Münchweiler, that one has been closed for at least this and last year and I’m not sure it will open next year—if it does I’ll go there); same for draisine-only tracks—I’m not a cyclist. Anyway, let’s start with the overview of Palatinate railway network.
Rheinland-Pfalz is divided into two parts by the river so the rail network looks like two nets connected in several points (essentially near Koblenz and somewhat Mainz). There is one and a half high-speed train lines (one going from Frankfurt Airport to Airport Köln/Bonn and something going from Mannheim via Kaiserslautern to Saarbrücken and Paris but it’s definitely not a high speed line). The right bank network consists essentially of a railway along the Rhine and several lines that connect neighbouring federal lands (Hessen and Nordrhein-Westfallen) including the high-speed one. The left bank network has several lines that form a very good connectivity: railways on the Rhine side go all way from Lauterbourg to Ludwigshafen to Mainz to Koblenz to Bonn, there’s an East-West line connecting Mannheim to Saarbrücken via Neustadt and Kaiserslautern, there’s another East-West line connecting Koblenz to Trier and Luxembourg, there’s yet another line connecting Mainz to Idar-Oberstein and Saarbrücken, there’s West border line going from Saarbrücken to Trier and Bonn, there’s North-South line going from Bingen to Kaiserslautern and Pirmasens (consisting of several railways actually), there’s a line Worms-Bingen, there are some lines connecting Karlsruhe and Wörth to Neustadt, Pirmasens and Saarbrücken, and there are still more. There are some local lines that branch from those and have a terminus in some small place you’ve never heard of. And there are some museum lines as well. Here is the official map from Deutsche Bahn for regional lines operated on regular basis (so no high-speed or museum lines there). As for special lines I can name semi-museum line from Linz (not on Danube) to Kalenborn, museum lines Kuckucksbähnel (from Neustadt to Elmstein), Brohltalbahn (Brohl-Engeln, it has the slowest express I’ve ever seen) and narrow-gauge museum Stumpfwaldbahn (which kinda duplicates Ramsen-Eiswoog part of regular line).
My tickets from museum lines (maybe I should start collecting them).
It would take too long to describe all of the lines so I’ll just talk about some peculiarities. First of all, it’s scenic and you have a lot of scenery kinds to choose from: there is the largest forest in Germany, there are nice mountains with different geology, there’s Middle Rhine with castles at every mountain, there are fields and even former volcanic area.
Second, Rheinland-Pfalz probably has more abandoned and dismantled railways than those in service. Just look at the list in German Wickedpedia—it looks like most current branches were part of longer railways that were closed later and many branches are remains of much longer railways.
Third, Rheinland-Pfalz is weird. Southern part of it is quite French to the point that you can see both TER and TGV trains there and stops are sometimes announced in French too. But in the very Eastern part of Rheinland-Pfalz you have Girod and in Northern part there’s Monreal. In Eifel. Why not have normal German names like Vogelweh, Geilhausen or Waldfischbach? And there’s weird RB90 route (Limburg an der Lahn—Siegen) that usually does not go all the way so you have to transfer in Westerburg from RB90 to RB90. But the real something is track numbering on the stations. I’m not talking about train arriving to “Gleis Ost” which is the only track here, I’m talking about numbers. Normally you number your tracks from one and sequentially above, maybe reserving ten-something or hundred-something for some auxiliary tracks but that’s not enough for Rheinland-Pfalz and Saarland. Few examples: on route Saarbrücken-Lebach Merchweiler has tracks 20 and 30, Gennweiler has tracks 40 and 50 while Illingen has tracks 41 and 51; Betzdorf has tracks 101-106 and 112 and neighbouring stations have tracks 40x (on two different stations), 50x and 60x.
And finally some words about special railways there:
There is a lot that can be told about Rheinland-Pfalz and its railways but I hope that this short review gave you at least an idea of what to expect there.
Recently I’ve been contacted by some guy working on a mod campaign for Heroes of Might and Magic III. The question was about the encoder for videos there. And since the original one is not likely to exist, I just wrote a simple one that would take PGMYUV image sequence and encode it. Here’s the gzipped source.
It took a couple of evenings to do that mostly because I still have weak symptoms of creeping perfectionism (thankfully it’s treated with my laziness). BIKb does not have Huffman-coded bundles, so the simplest straightforward encoding would be: write block type bundle (13-bit size and 4-bit elements), write empty other bundles, write several bundles containing pixels and you’re done. There’s a proper approach: write a full-featured encoder that takes input in several formats and that encodes using all possible features selecting the best quality for the target bitrate. There’s a hacky approach—translate later versions of Bink into BIKb (and then you remember that it has different motion compensation scheme so this approach won’t work). I’ve chosen something simple yet with some effectiveness: write an encoder that employs only vector quantisation and motion compensation for non-overlapped blocks plus add a quality setting so users can play with output size/quality if they really need it.
So how does the block encoding work? Block truncation coding, the fast and good way to quantise block into two colours (many video codecs back in the day used it and only some dared to use vector quantisation for more than two different values per block). Essentially you just calculate average pixel value and select two values depending on how many pixels in the block are larger than average and by how much they deviate. And here’s where quality parameter comes into play: depending on it encoder sets the threshold above which block is coded as is (aka full mode) instead as two colours and pattern in which they occur (of course if it’s a solid-colour fill it’s always coded as such). As I said, it’s simple but quite effective. Motion compensation is currently lossless i.e. encoder will try to find only the block that matches exactly (again, it can be improved but that would only lead to longer implementation times and even longer debugging times). This makes me appreciate the work on Smacker and Bink 1 video codecs and encoders for them even more.
Overall, it was a nice diversion from implementing Duck decoders for NihAV but I should probably return to it. The sooner it’s done the sooner I can move to something more exciting like finally experimenting with vector quantisation, or trying to write a player, or something else entirely. I avoid making plans but there are many possibilities at hoof so I just need to pick one.
Dedicated to Peter Ross, who wrote an opensource VP4 decoder (that is not committed to CEmpeg yet at the time of the writing).
The codecs from VP3 to VP6 form a single codec family that is united not merely by the design but even by the header—every frame in this codec (sub)family has the same header format. And the leaked VP6 format specification still calls the version field there Vp3VersionNo (versions 0-2 used by VP3, 3 is used by VP4, 5 is for VP5 and 6-8 is for VP6). VP7 changed the both the coding principles to mimic H.264 and the header format too. And you can call it the golden age for Duck because it’s when it gained popularity with VP3 donated to open-source community (and xiphed to Theora which was the only patent-free(ish) opensource video codec with decent performance back then) to its greatest success found in VP6, employed both in games and in Flash video (remember when BaidUTube still used .flv with VP6 and N*llyMos*r ADPCM or Speex?). And today, having gathered enough material, I want to give an overview of these codecs. Oh, and NihAV can decode VP30 and VP31 now.
It is hard to say how this codec was designed but looks like it was mostly good for network transmission with the risk of packet loss (but then why it does not have CRCs?) and for doing stuff not like other codecs do.
Overall format can be described as being very weird variation on progressive JPEG. Image (all three planes) is represented as a sequence of 32×32 pixel super-blocks (part of super-block outside plane boundaries is not coded), each of them contains 4×4 macroblocks, each of those contains usual 8×8 blocks. The block metadata (coding type and motion vectors) are coded as single chunks of data for all blocks before coefficients. Coefficients are also coded in sequence: DCs for all blocks first, then 1st AC for all blocks (where present), then 2nd AC … up to the 63rd AC. There are two kinds of frames—intra and inter. Inter-frames can use either previous frame or last intra frame (called golden frame) as the reference (VP6 finally allowed any inter-frame to become new golden frame but let’s talk about it later).
As you can see, it’s robust against truncated frames but I cannot imagine such scenario being important for the codec not used in video calls (and that was VP7—way later than VP3). Also that new block ordering required a new way to walk them so there are no jumps between last block in current super-block and first block in next super-block. Hence the Hilbert curve suitable for macroblocks and blocks alike.
IMO codec was designed not for efficiency but rather for being sufficiently different from the rest. And while I don’t like the result, I must admit that at least it’s quite original in some parts and I appreciate the variety.
Anyway, beside golden frames, peculiar block walk pattern and data partitioning the other notable thing is block coefficients coding. While the method remains the same (you need to code sparse array essentially), instead of using the standard run-length codebook VP3 uses semi-fixed scheme with entries grouped in categories (fixed number of end-of-block flags, longer amount of EOB flags, fixed coefficient, small coefficient in fixed range, zero run of one plus small coefficient and such). In result you need to read one of 32 possible tokens using one of several codebooks and then unpack token by following the fixed scheme. For the reference, H.263 coefficients codebook contains over hundred entries (and still need escape values handling).
Now let’s review VP3 flavours.
VP30 is obviously the zeroth version (it does not even have a version in the header and stores dimensions in macroblocks there instead). Block coded flags are coded hierarchically: first there are a bit-run coded flags for super-blocks whether they contain coded macroblocks, then for all non-empty super-blocks there are bit-run coded flags for macroblock that contain coded blocks, and finally there are bit-run coded flags for blocks in coded macroblocks (obviously in intra frames all block are coded and are of intra type so such metadata is not present).
After that you have macroblock types (just for coded macroblocks of course) which is read with a fixed codebook. There are seven macroblock types: intra, skip, copy with MV, copy using motion vector from last inter block, four MVs per macroblock, copy from golden frame from the same position, copy from golden frame using MV. The WTFiest moment for me is that uncoded intra block in VP3 (which may happen only in inter frame) actually means skipped block.
After that you have motion vector information for the macroblocks that require it using fixed coding scheme for motion vector components. As you can see above, the only kind of MV prediction is re-using motion vector from last inter block with coded MV (VP5 will improve this).
Coefficients are coded using three codebooks (one for DCs, one for intra block ACs and one for inter block ACs) with codebook set selected from one of five possible depending on quantiser. DC prediction is simple: just add last DC from the block of the same type (intra for intra, any inter for inter block). Also it has been using bit-exact DCT that remained until VP7 decided to switch to H.264-like transforms.
Loop filter seems to be the same and survives even in VP8 (under the name of simple loop filter) but it’s not used on output picture since VP4…
Fun fact: VP30 decoder was written in C++ and binary VP31 (and maybe even VP4) decoder still contains it along with C code for decoding VP31.
And as I’ve mentioned in the beginning, NihAV can decode it now while libavcodec/vp3.c still can’t.
VP31 is a small improvement on the above and it’s the version people know the most (if you include Theora based on it). Block coding information is now coded in following layers: first it codes information for partially-coded super-blocks, then it classifies the rest into fully coded/uncoded super-blocks and finally it signals which blocks in partially coded super-blocks are actually coded.
Macroblock types are coded using either raw indices or unary coding plus one of the predefined sets of values (or a custom one).
Motion vectors now can be decoded using fixed coding scheme (slightly different one) or raw 5-bit value plus sign (even for zeroes). Also now there’s an inter block type that uses second last MV.
Coefficients are now coded using a set of five codebooks (DC, ACs 1-5, ACs 6-14, ACs 15-27 and ACs 28-63) for luma and for chroma. Codebooks are selected from a set of sixteen using two indices read from bitstream for DCs and (after DCs are decoded) two indices read from bitstream for ACs. DC prediction is now complex and uses weighted DCs from neighbours depending which of them of the same kind are available (intra, last frame reference or golden frame reference).
And there is an elusive VP33 (aka VP3 version 2) which has some leftover code in the VP31 opensource dump including the tables and block coded flags.
While most of the code is the same there are two main changes: block coded flags is completely different now and there’s now a completely new set of tables. Block coded flags are decoded on macroblock basis: first you decode flags denoting fully coded macroblocks, then you decode flags for partially coded macroblocks and finally you decode coded block pattern using previously decoded block pattern and 3-5 bits of data. The rest goes exactly as VP31.
Either this was an experimental codec or simply a leftover they forgot to remove but while binary VP3 decoder supports it, there’s nothing known about any samples or means to encode them. Call it Bink-d of VP codecs.
Now enters VP4. Essentially it’s VP33 with some tweaks and some changes in frame reconstruction. The main differences are motion vector coding scheme (it’s more complex now) and the fact that loop filter is not applied to the decoded frame but used on motion block source instead (i.e. it copies block from source frame to the temporary buffer and if it contains edge pixels then it applies loop filter on it). The same scheme was employed in VP5 and VP6 as well. Also now it stores table indices before all coefficients (instead of DC luma table index, DC chroma table index, all DCs, AC luma tables index, AC chroma tables index, AC1, AC2, …) and has some more information in the frame header but that’s all I can think of. Also DC prediction has been changed once again (seems to be simply an average of neighbouring DCs from the same kind blocks).
The only fun facts about it is that some of its code is already present in opensource VP31 dump (being mostly VP33) and that binary VP4 decoder contains code and tables for bilinear and bicubic motion interpolation employed in VP5/6. I’m yet to look at VP5 and VP6 binary specifications so I don’t know whether they contain hints on VP7 but that seems unlikely.
This is a codec that radically changed two things: data is now coded using their bool coder (a variant of arithmetic coder for coding bits that uses fixed probabilities instead of an adaptive model) that is used to decode bits for values and codes in fixed Huffman trees (also that means that now there are twelve DCT token categories instead of 32 like before). This scheme will survive until VP9 and has some chances to reappear in AV2 (just a guess—we know who develops it and nothing else beside that fact).
In result the codec could return to having ordinary 16×16 macroblocks without super-blocks (and code blocks instead of coefficient slice). Also now motion vector prediction relies on using so-called nearest and near MVs that come from neighbour blocks visited in certain order.
The only major drawback I can name is that now inter frames may rely on previous ones since they can transmit probability updates and thus dropped frame will result in complete garbage instead of damaged frame converging back to normal image.
And now VP5 can be tweaked for performance. So the notable changes include: signalling that this inter frame is a new golden frame and new data coding and partitioning modes. Now if you want to have lower latency you can split your data into two parts and start decoding coefficients as soon as you decode corresponding macroblock information. Or if you really care about speed you can switch to old way of coding by constructing Huffman trees from model probabilities and having special handling for zero/EOB runs. Also now there’s an interlaced mode and alpha support. No wonder the codec was popular on the Web until H.264 came to power.
From VP30 to VP6.2 it was a steady evolution of the codec (beside brief return from bool coding for Huffman trees in VP6) with most concepts remaining the same and the biggest changes being between VP4 and VP5—new bool coder, qpel MC and switch from super-blocks to ordinary macroblocks (and new MV prediction scheme to some extent). But other things like data partitioning, DCT tokens, DCT itself and loop filter remained the same. The same way VP6 into VP7 change will involve changing of coding blocks into 4×4 ones, new transforms and spatial prediction, but many other things like bool coder or MC interpolation still remain there.
Maybe I’ll revisit it later if I ever get to VP7 decoder but for now this should give you an idea how Duck codecs changed over the time and why it ended like that (maybe one day somebody will manage to answer a question “Why platypus? Just why?”). Meanwhile I’m looking sideways for AV2 aka VP11 and its possible collision course with H.266 aka VVC.
As I’ve mentioned in the previous post, I’ve finally tried rust-clippy to see what issues and suggestions it will have on my code. The results are not disappointing if you take the tool name seriously.
The output (at least in my case) can be divided into four categories: actual bugs spotted, useful suggestions, (mostly) useless suggestions and counterproductive suggestions.
Let’s start with actual bugs spotted. It spotted two cases when condition has identical code in both branches. So I had to remove such useless condition in one case and fix the data source taken in another. Also an error on casting pointer to several-byte buffer as pointer to integer. Depending on stack alignment and CPU architecture this may result in a buggy load so better fix it (but then this issue will be mentioned again when we get to the third category).
Useful suggestions were much more plentiful. I started developing NihAV in Rust a bit more than two years ago and the language has changed quite a bit since then so some parts of code were a bit outdated (e.g. nowadays you can simply write Struct { field1, field2 } if you initialise it from field1 and field2 variables and have the same fields; previously you had to write Struct { field1: field1, field2: field2 }). Plus of course there are features I did not know about either because they were introduced later or I simply did not think about it being possible. Here are some examples I encountered:
vec![elem; size] to create vector of fixed size filled with element (previously I invoked Vec::with_capacity() and then vec.resize() which is more error-prone);start..=end instead of start..end+1 (IIRC this was still an unstable feature when I learned about it so no wonder I forgot about it);println!() is equivalent to println!("") (and guess who uses it for debugging);ptr.offset(size as isize) you can simply write ptr.add(size);slice.swap(idx1, idx2) for the cases when you need to swap two values in an array;u32 to u32);Vec when passing the reference to slice would work as good (and even save some MiNicycles on access); same for borrowing Box.There were some stylistic suggestions that I found good (replacing some indexed loops with loops on iterators and copying slice with a call instead of loop) but mostly they belong to the next category, useless suggestions.
Yes, that’s where rust-clippy starts to live up to its name. Most of its suggestions were in the format “hey, you’re trying to use clear imperative programming to do the work that can be expressed with iterators in 10x text”. I understand that it’s more to the Rust style and it helps to prevent errors in some cases, but when it suggests to replace simple loop like for i in 3..7 { dst[i] = src[3 - i]; } with something that won’t fit onto single like I say it’s too much. Or how it suggested to change simple mem::transmute::<&mut $inttype, &mut [u8; $size]>(&mut buf) into &mut *(&mut buf as *mut $inttype as *mut [u8; $size]) (sure, it does the same but I find the former easier to comprehend).
Similar story with X as u32 versus u32::from(X). I understand it helps catching the cases where X changes type and cannot be converted to destination type losslessly any more. But it breaks formatting and makes it unnatural (and close to C++) in the sense that previously the code was read like “calculate X and cast it to type” and now it’s “convert to type from calculated X”. And by breaking formatting I mean that I have a lot of code in form
let a = br.read(4)? as usize;
let b = br.read(6)?;
let c = br.read_bool()?;
let d = br.read(3)? as u64;
and by using u64::from(br.read(3)?) I’d break nice tabular form for no obvious merit.
di
Another cases is when rust-clippy suggests refactoring conditions. When I write
if is_intra {
if block.coded { do a }
else { do b }
} else {
if block.coded { do c }
else { do d }
}
I format it this way because it reflects the logic of the code and I do not want to make it into } else if block.coded { do c } else { do d }. Similarly if I don’t join let a; if foo { do stuff; a = val; } else { a = val2; } into single let a = if foo { ... val } else { val2 }; it’s because I have enough code in that condition that I don’t want to indent it even further (sure, make fun of me because I worked in text mode and still don’t like lines longer than 80-100 characters).
There’s more like “you should implement Default trait for each structure with new() method” and “you should write constants like 0x00_0001” and “enums should not have different prefix in the name, you should consider removing Mode from Mode16k, Mode5_5k and Mode8k” but let’s move to counterproductive suggestions already.
First issue is that rust-clippy interacts with macros quite poorly. For example I have a macros that expands 4-byte array into 32-bit tag and it suggests using u32::from(arr[0]) instead of (arr[0] as u32) but if you do that it will have an error instead of warning because it can’t check types in macro. Also here’s one of the WTFiest examples: I have a macro for filter implementation that takes coefficients as arguments and calculates the value. It has a code -$c1 * $src[$off + 1] and when $c1=-1 you have a warning “about –x is double negation”. Actually I remember that example from Feuer’s C. Puzzle Book. where #define NEG(x) -x NEG(-x) expanded to –x (and thus I put all macro arguments into parentheses) but I expected Rust to handle that correctly (otherwise this WTF goes to linter). In either case -(-1) is what I meant in that case and that’s how it works.
And now the most infuriating example that the tool treats like a warning or an error depending on situation.
As I write video decoders, I have a need to apply a filter or transform on some data. And IMO this form:
dst[off - 2 * step] = a;
dst[off - 1 * step] = b;
dst[off + 0 * step] = c;
dst[off + 1 * step] = d;
I think this conveys the meaning and looks much better than
dst[off - 2 * step] = a;
dst[off - step] = b
dst[off] = c;
dst[off + step] = d;
with whatever spaces you insert there (and it’s not possible to use iterators here quite often too). But rust-clippy treats that as a warning if you’re multiplying by one, adding zero or constant multiplied by zero and gives outright error on variable multiplied by zero or an expression like N - N (which happens when you have constant base and one of the offsets equal to it).
So my experience with this linter was mostly negative—all those neat things were out-weighted by the sheer amount of useless warnings (and I don’t like to add dozens of #[allow(clippy::useless_warning)] to lib.rs for many modules) and some stuff that would simply break code clarity for no obvious benefit. Overall, it’s a nice tool and though I got something useful from it, it may be a beginner in the language aka ordinary M$ Office user who would benefit the most from it and it will mostly hinder people writing advanced code unless you turn off a lot of warnings and even some errors.
Since the clean-up work on NihAV is done and I progress with Truemotion VP3 decoder, it’s a good time to talk about what I’ve actually done—there’s even more material to write waiting in the queue.
The intent was to make all frame-related stuff thread-safe and improve efficiency a bit. In order to do the former I had to replace most of the references from Rc<RefCell<T>> to Arc<T> and while doing it I introduced aliases like type NAFrameRef = Arc<NAFrame> and .into_ref() methods to convert object into ref-counted version. This helped when I tried switching from one implementation of reference counter to another and will make it easy to switch again if I ever need that (hopefully not). Now about improved efficiency and how it’s related to the ref-counting.
There’s a straightforward way of dealing with frames: you allocate the picture, fill it, dispose, allocate a new one, etc etc. And there’s a more effective way: you allocate several pictures at once, select an unused one, fill, return to the pool when it’s not needed any more. That is where reference counting comes into play and where Rust default structures don’t help. Frame pool owns the reference and decoder gets a second copy. And Rust Arc is intended for single ownership: when you try to access the shared object it will simply clone it so you end up working with a copy (which defies the purpose). So I had to NIH my own NABufferRef<T> which keeps reference counts and still allows shared access even for writing (currently it does that in all cases but if I need to add some guards the API won’t have to be changed for that). The implementation is very simple: the structure contains a raw pointer to a structure that contains actual object and AtomicUsize counter. The whole implementation is ~2.2kB relying just on std crate.
And finally I’ve made a picture pool. The difference between picture and frame is all additional metadata picture should not care about (like timestamps, stream information and such). Because of the design decisions I have three different picture formats (implemented for 8-, 16- and 32-bit element sizes, Rust does not like aliasing after all), which means I need to provide decoder with all three picture pools because we can’t say in advance which one codec will use (if at all—the option to allocate new non-pooled pictures is still there). Also I want to keep those pools external in case the code around it wants to do keep more pictures in it (e.g. 2-3 pictures required by decoder and 25 pictures pre-buffered for the display). This resulted in a structure called NADecoderSupport that contains picture pools and may have something else added late. Of course people might argue that it’s much better to have AVCodecContext with a myriad of fields you can set directly or via utility functions but I’d rather not have one single structure. Though it might be a good place to put various decoder options there (so that decoder can ignore them at its leisure).
Since I said I did it to increase efficiency I should probably give some numbers too: RealVideo 3/4/6 decoders now use buffer pool (for three frames obviously) and reallocate it on format change. Decoding time got reduced by 4-5% from using the pool. Currently I don’t care about speed much but I may convert more decoders to it if the need arises.
In conclusion I want to say that even I did not enjoy doing that work much, it was needed and gave me some experience plus some improvements in code and design. So it was not a wasted effort.
P.S. I also installed rust-clippy since it’s in stable now and tried to fix errors and warnings it reported. But that is a story for another post.
Today I want to talk about local dynasty that was rather short-lived but left quite an impressive legacy.
So, about a thousand years ago in the South-Westernmost corner of Germany there was a noble lord by the name of Berthold who started a dynasty of Zähringen and his eldest son founded the line of margraves of Baden. And while the dynasty of Zähringen has only six rulers (Berthold, Berthold, Berthold, Konrad, Berthold and Berthold) it’s known for building and rebuilding a dozen of towns (yes, exactly twelve), some of which are important even to this day. Since I have nothing better to do at the weekends I decided to visit them all.
Berthold I founded Weilheim an der Teck, the only Zähringer town founded in Württemberg.
Berthold II is known for founding the Abbey of St. Peter (and a town forming near it), Freiburg (in Breisgau of course) and reforming a Swiss company Rheinfelden AG (though Swiss claim that AG stands for canton Aargau and not joint-stock company).
Berthold III and Konrad I somehow evaded the traditional duty and built nothing. Though Freiburg still honours Berthold III for developing Freiburg into proper city and not a mere settlement by the castle.
Berthold IV, son of Konrad I, decided to follow the suit of his grandfather and founded Freiburg (this time in Üechtland). He also rebuilt Murten, founded Burgdorf and Neuenburg (in Baden, don’t confuse it with Neuchâtel).
Berthold V probably remembered that his ancestors were margraves of Dietrichsbern (or Verona as locals call it) and decided to found his own Bern (in Üechtland). He also rebuilt and developed Thun (the castle he built there is still one the best places there), Bräunlingen and Villingen.
Since Berthold V had no offspring, the lands of duchy of Zähringen went to different other noble houses (including dukes of Fürstenberg who are famous for building a lot of castles in Czechia) and ended in hands of Swiss house of Habsburg (who, obviously, are famous for ruling Austria). Thus one can see former Austrian towns even on Rhine.
Fun fact: some of those places have suspiciously similar coats of arms (all images are from Wickedpedia).
Bern: 
Neuenburg: 
Karlsruhe: 
Would you believe it has nothing to do with heraldic colours of Zähringen being red and yellow (also adopted by margraves of Baden, one of which founded Karlsruhe)?
Now, about the towns and my impressions on them.
Another interesting thing is how much the founders are honoured in all those places. Most of them have Zähringerstraße or Zähringerplatz (and Zürich has both despite Berthold IV being just a vogt there). There are exceptions though: Freiburg in Breisgau has Zähringerstraße in a suburb called Zähringen and actual city has Bertoldstraße instead; Weilheim simply mentions that it’s one of Zähringerstädte and has their coat of arms near the church; Freiburg im Üechtland has its main bridge named after Zähringen; Thun and Murten do not seem to have any mentions. While many other places in Baden have Zähringerstraße (like Karlsruhe or Heidelberg) probably because of the relation to the house of Baden, I think they deserve recognition for building new towns. It’s a pity there were not so many nobles like them.
Since obviously I have nothing better to do, here’s a description of loop filter in Bink2 as much as I understand it (i.e. not much really).
First, the loop filter makes decision on two factors: motion vector difference between adjacent blocks is greater than two or it selects filter strength depending on number of coefficients coded in the block (that one I don’t remember seeing before). The filter is the same in all cases (inter/intra, luma/chroma, edge/inside macroblock), only the number of pixels filtered varies between zero and two on each side (more on that later). This is nice and elegant design IMO.
Second, filtering is done after each macroblock, horizontal edges first, vertical edges after that—but not necessarily for all macroblocks. Since BIKi or BIKj encoder can signal “do not deblock macroblocks in these rows and columns” by transmitting set of flags for columns and rows.
Third, in addition to normal filtering decoder can do something that I still don’t understand but it looks like whole-block overlapping in both directions (and it is performed in actual decoding but I don’t know what happens with the result of it).
And the filter itself is not that interesting (assuming we filter buf[0] buf[1] | buf[2] buf[3]:
diff0 = buf[2] - buf[1] + 8 >> 4;
diff1 = diff0 * 4 + 8 >> 4;
if (left_strength >= 2)
buf[0] = clip8(buf[0] + diff0);
if (left_strength >= 1)
buf[1] = clip8(buf[1] + diff1);
if (right_strength >= 1)
buf[2] = clip8(buf[2] - diff1);
if (right_strength >= 2)
buf[3] = clip8(buf[3] - diff0);
Strength is determined like this: 0 — more than 8 coefficients coded, 1 – MV difference or 4-7 coefficients coded, 2 — 1-3 coefficients coded, 3 — no coefficients coded.
Overall, the loop filter is nice and simple if you ignore the existence of some additional filter functions and very optimised implementation that is not that much fun to untangle.
Update: the alternative function seems to be some kind of block reconstruction based on DCs. In case it’s intra block with less than four coefficients coded it will take all neighbouring DCs, select those not differing by more than a frame-defined threshold and smooth the differences. I still don’t understand its purpose in full though.
I’m proud to say that NihAV got TrueMotion 2X support. For now only intra frames are supported but 75% of the samples I have (i.e. three samples) have just intra frames. At least I could check that it works as supposed.
First, here’s codec description after I managed to write a working decoder for it. TrueMotion 2X is another of those codecs that’s closer to TrueMotion 1 in design. It still uses the same variable-length codebook instead of Huffman coding (actually only version 5 of this codec uses bit reading for anything). It also uses “apply variable amount of deltas per block” approach but instead of old fixed scheme it now defines twenty-something coding approaches and tells decoder which ones to use in current frame. That is done because block size now can be variable too (but it’s always 8 in all files I’ve seen). And blocks are grouped in tiles (usually equivalent to one row of blocks but again, it may vary). The frame data obfuscation that XORs chunks inside the frame with a 32-bit key derived in a special way is not worth mentioning.
Second, the reference is quite peculiar too. It decodes frame data by filling an array of pointers to the functions that decode each line segment with proper mode, move to the next line and repeat. And those functions are in handwritten assembly—they use stack pointer register for decoder context pointer (that has original ESP saved somewhere inside), which also means they do not use stack space for anything and instead of returning they simply jump to the next routine until the final one restores the stack and returns properly. Thankfully Ghidra allows to assign context argument to ESP and while decompile still looks useless, assembly has proper references in the form mov EDX, dword ptr [ctx->luma_pred + ESP].
And finally, I could not check what binary specification really does because MPlayer could not run it. At first I tried running working combination of WMP+Win98 under OllyDbg in QEMU but it was painfully slow and even more painful to look at the memory state. In result I’ve managed to run TM2X decoder in MPlayer which then served as a good reference. The trick is that you should not try to run tm2X.dll (it’s really hopeless) but rather to take tm2Xdec.ax (or deceptively named tm20dec.ax from the same distribution that can handle TM2X unlike its earlier versions), patch one byte for check in DLL init and it works surprisingly well after that.
So what’s next? Probably I’ll just add missing features for the second TM2X sample (the other two samples are TM2A), maybe add Bink2 deblocking feature—since I’d rather have that decoder complete—and move to improving overall NihAV design. Frame management needs proper rework before I add more codecs—I want to change into a thread-safe version before I add more decoders. Plus I’ll need to add some missing bits for a player. There’s a lot of work still to do but I’m pleased that I still managed to do something.
So NihAV finally got Discworld Noir BMV support and I’ve tested it on all samples from the game to see if it works correctly. Here’s a sample frame:
(I still remember the song Samael plays there and have it somewhere ripped in its full MPEG audio layer II glory).
Now I want to talk about the format since it’s quite different from anything else. BMV used in Discworld II was simple but with two quirks: it employed integer coding using variable amount of nibbles (that were read as bytes but a nibble could be saved and used later) and it could decode frame either from the beginning to end or from end to beginning (reading frame data from the end too!). DW3 BMV is even stranger and let’s start with audio part. Audio codec is very simple: you have 41-byte block with one byte signalling which quantised values tables should be used for both channels and 32 indices for each channel packed into 16-bit words. The main peculiarity is that data is aligned to 16-bit and mode byte can be either in the beginning or at the end of the block. That’s a bit unusual but not strange. Well, it turns out it aligns for the absolute position in a file so my demuxer has to signal whether audio data was at even or odd position. And video is even stranger.
As I wrote previously, video codec is 16-bit now and still employs nibble variable integer coding and copy/repeat/new pixels mode. Luckily there is no backwards decoding mode yet the codec is tricky without it. First of all, where previously there were just three plain modes now we have combinations of those with bytes or nibbles signalling what should be done (i.e. copy/repeat/put new pixels fixed amount of times and then do the other operation another fixed amount of times). And they have different meaning depending on what was the last operation (copy/repeat/put for fixed amount or with arbitrary large one). And if there’s a nibble left unread after last operation or not. But that’s not all! While previously reading new pixels meant just reading a byte, 16-bit pixels can be compressed a bit more. In result we read 1-3 bytes per pixel: first read index byte, remap it, if it’s in range 00..F7 then return pixel in an array, if it’s in range F8..FE then read another byte and use it as an index in one of seven secondary “palettes”; if it’s FF then simply read explicit 2-byte value from the stream. The reference simply used an array pointing to the various functions performing this. And of course palettes can be updated in the beginning of each frame.
Surprisingly, decoder implementation takes about 28kB with a quarter of it being tables. That’s for both audio and video decoder. This is on par with other game decoders (GDV, Smacker and VMD) and feels significantly smaller than the reference (which is about 30kB in a stripped assembled .o file and over 200kB as assembly).
Overall it was hard to comprehend and tricky to debug too. Nevertheless now it’s over and I can probably move to TrueMotion 2X. Or whatever I decide to do when I’m bored enough.
I’ve made some significant progress on REing Discworld Noir BMV.
First, I put opcodes meaning into table (it’s probably the only case when I had to use spreadsheet for REing) to figure out the meaning.
Normal mode of operation have these opcodes:
00**00*0 — perform extra-long copy;00**00*1 — perform extra-long invoking of pixel functions;00**xxx0 — copy xxx-1 pixels;00**xxx1 — invoke pixel function xxx-1 times;xxxx000y — copy xxxx+3+y pixels;xxxx001y — invoke pixel function xxxx+3+y times;< xxx0yyy0 — copy yyy-1 pixels and then invoke pixel function xxx-3 times;xxx0yyy1 — invoke pixel function yyy-1 times and then repeat last value xxx-3 times;xxx1yyy0 — copy yyy-1 pixels and then repeat last value xxx-3 times;xxx1yyy1 — invoke pixel function yyy-1 times and then copy xxx-3 pixels.Then depending on last operation performed mode is changed to: something special for 00****** opcodes, no change for repeat, “after copy mode” and “after pixel func mode” for obvious cases.
After copy mode opcodes:
xxxxxxx0 — invoke normal mode opcode xxxxxxx1;00**00** — extended repeat;00**xxx1 — repeat last value xxx-1 times;xxxx00y1 — repeat last value xxxx+3+y times;xxx0yyy0 — repeat last value yyy-1 times and copy xxxx-3 pixels;xxx0yyy1 — repeat last value yyy-1 times and copy xxxx-3 pixels;xxx1yyy1 — repeat last value yyy-1 times and invoke pixels function xxxx-3 times.After pixel function mode is simple: xxxxxxx0 opcode is after copy mode xxxxxxx1 opcode and xxxxxxx1 opcode is normal mode xxxxxxx0 opcode.
The special modes may have secondary opcodes that usually boil down to the same thing: either do some more of the same and proceed normally or calculate next opcode instead of reading it from the stream.
And now the second thing: I’ve managed to rip the relevant code from the disassembly, fix it for NASM to handle plus added some fixed to make it possible to invoke externally and linked it against my own small program that parses BMV file, decodes the first video frame and dumps it into file. That approach works so I have something to test my new implementation against. Also since NASM makes all labels visible it’s easy to make debugger report each opcode as it gets called.
To summarise, I have good understanding of the algorithm and I have a working binary specification. This should be enough to finish it soon (unless I get distracted by something else of course).
As I’ve mentioned before, NihAV now can decode Bink2 files more or less decently. I don’t have many samples but I can decode all samples I could find from KB2f to KB2j quite well (the only exception is KB2a — there’s only one partial sample known with no indication which game uses it and no version of RAD tools understands it either).
I’ve omitted support for full-resolution Bink2 files (no samples) and reconstruction is not perfect because there’s an in-loop deblocking filter with some additional crazy functions invoked in some cases. It’s messy and does not affect actual bitstream decoding so I’m not going to work on it now. Maybe if I get some inspiration later…
Anyway, I moved to REing Discworld Noir BMV format instead. While the game is nice, it’s hard to run on any modern OS and there’s almost no hope for its engine being reimplemented. So maybe I’ll be able to re-watch cutscenes from it… I’ve figured out container and audio long time ago, video is not that easy. It seems to be an upgrade of Discworld II BMV that outputs 16-bit video but the way it’s implemented is baffling.
While locating the functions responsible for the decoding was easy, understanding them was hard. For the disassembler. No, the code was recognized fine but its flow makes even disassembler freak out. Here’s how it works:
00-3F, then you usually have to construct length word, perform some pixel output loop and then jump to another opcode handler which performs some operation and then jump to the address calculated by the previous operation;Anyway, I’ve made some progress and I reckon it will be possible to support this format in NihAV though maybe not soon enough.
I am working on PowerPC SIMD optimizations for x264. I was playing with SAD functions and was thinking it would be nice to have something similar to x86 PSADBW for computing the sum of absolute differences. Luca suggested me to try using the #power9 vec_absd. Single vec_absd( fenc, pix0v ) replaces vec_sub( vec_max( fencv, pix0v ), vec_min( fencv, pix0v) ). My patch can be found here. To make it work -mcpu=power9 must be set. The patch contains the macro that makes its code backward compatible with POWER8:#ifndef __POWER9_VECTOR__
#define vec_absd(a, b) vec_sub(vec_max(a, b), vec_min(a, b))
#endif
vec_absd (the numbers are ratios of AltiVec/C checkasm timings):
vec_absd for SAD and SSD functions makes 8% improvement. Which is amazing for such a small change.I decided to write SIMD optimizations for HEVC decoder inverse transform (which is IDCT approximation) for ARMv7. (Here is an interesting post about DCT.) The inverse transform for HEVC operates on 4x4, 8x8, 16x16 and 32x32 blocks and I have finished them recently. For each block there are 2 functions, one for 8 bitdepth and the other for 10 bitdepth:
vpush/vpop {q4-q7} ) when one wants to use them. VPUSH/VPOP pushes and pops to/from stack. pop lr and then bx lr to return but it's better to return with simply pop {pc} . mov rx, sp and rx, sp, #15 add rx, rx, #buffer_size sub sp, sp, rx sp now points to the buffer. After using the buffer, the stack pointer has to be restored with add sp, sp, rx . I tried my skills at optimising HEVC. My SIMD IDCT (Inverse Discrete Cosine Transform) for HEVC decoder was merged lately. What I did was 4x4, 8x8, 16x16 and 32x32 IDCTs for 8 and 10 bitdepths. Both 4x4 and 8x8 are supported on 32-bit CPUs but 16x16 and 32x32 are 64-bit only.
The larger transforms calls the smaller ones, 32 calls 16, 16 calles 8 and so on, so 4x4 is used by all the other transforms. Here is how the actual assembly looks:
; void ff_hevc_idct_4x4__{8,10}_(int16_t *coeffs, int col_limit) *coeffs is a pointer to coefficients I want to transform. They are loaded to XMM registers and then
; %1 = bitdepth
%macro IDCT_4x4 1
cglobal hevc_idct_4x4_%1, 1, 1, 5, coeffs
mova m0, [coeffsq]
mova m1, [coeffsq + 16]
TR_4x4 7, 1, 1
TR_4x4 20 - %1, 1, 1
mova [coeffsq], m0
mova [coeffsq + 16], m1
RET
%endmacroTR_4x4 macro is called. This macro transforms the coeffs according to the following equations: res00 = 64 * src00 + 64 * src20 + 83 * src10 + 36 * src30
res10 = 64 * src01 + 64 * src21 + 83 * src11 + 36 * src31
res20 = 64 * src02 + 64 * src23 + 83 * src12 + 36 * src32
res30 = 64 * src03 + 64 * src23 + 83 * src13 + 36 * src33
Because the transformed coefficients are written back to the same place, "res" (as residual) is used for the results and "src" for the initial coefficients. The results from the calculations are then scaled res = (res + add_const) >> shift and the (4x4) block of the results is transposed. The macro is called again to perform the same transform but this time to rows.
; %1 - shift
; %2 - 1/0 - SCALE and Transpose or not
; %3 - 1/0 add constant or not
%macro TR_4x4 3
; interleaves src0 with src2 to m0
; and src1 with scr3 to m2
; src0: 00 01 02 03 m0: 00 20 01 21 02 22 03 23
; src1: 10 11 12 13 -->
; src2: 20 21 22 23 m1: 10 30 11 31 12 32 13 33
; src3: 30 31 32 33
SBUTTERFLY wd, 0, 1, 2
pmaddwd m2, m0, [pw_64] ; e0
pmaddwd m3, m1, [pw_83_36] ; o0
pmaddwd m0, [pw_64_m64] ; e1
pmaddwd m1, [pw_36_m83] ; o1
%if %3 == 1
%assign %%add 1 << (%1 - 1)
mova m4, [pd_ %+ %%add]
paddd m2, m4
paddd m0, m4
%endif
SUMSUB_BADC d, 3, 2, 1, 0, 4
%if %2 == 1
psrad m3, %1 ; e0 + o0
t psrad m1, %1 ; e1 + o1
psrad m2, %1 ; e0 - o0
psrad m0, %1 ; e1 - o1
;clip16
packssdw m3, m1
packssdw m0, m2
; Transpose
SBUTTERFLY wd, 3, 0, 1
SBUTTERFLY wd, 3, 0, 1
SWAP 3, 1, 0
%else
SWAP 3, 2, 0
%endif
%endmacro
The larger transforms are a bit more complicated but they works in a similar way.
There are the results benchmarked by checkasm bench_new() function for the bitdepth 8 (the results are similar for bitdepth 10). Checkasm can benchmark SIMD functions with --bench option, in my case the full command was:
I asked Kostya Shishkov, an experienced ARM developer, to check my basic NEON knowledge. So here are his questions and my answers to them:
vmov.i32 q0, #42 - move immediate constant 42 to q0 SIMD register, suffix i32 specifies the data type, 32-bit integer in this case, as q0 is 128-bit register, there will be 4 32-bit 42 constants mov r1, #16 - move number 16 to r1 GPR register add r0, r1 - add 16 bytes to the address stored in r0 vst1.s16 {q0}, [r0] - store the content of q0 to the address stored in r1r0 and store the result in r1mov r2, \step - move the constant step to r2 vld1.s16 {d0}, [r1], r2 - store d0 content to the address in r1, then update r1 = r1 + r2 vmov d0[0], r1I decided to organise the Libav sprint again, this time in a small village near Pelhřimov. The participants:
I rented a cosy cottage for the sprint. It was surprisingly warm for the end of September and we enjoyed a nice garden sitting not only with the table and chairs but even with a couch. The sitting was under the roof and it was possible to work outside which was really pleasant for me. There was also a grill so we grilled some sausages for one of our dinners. It started to rain the last day and making a fire in the fireplace made a nice feeling.
Some time ago Niels Möller proposed a new method of bitreading that should be faster then the current one (here). It is an interesting idea and I decided to try it. Luca Barbato considered it to be a good idea and had his company sponsored this work. The new bitstream reader (bitstream.h) is faster in many cases and is never slower than the existing one (get_bits.h).
static inline unsigned int get_bits(GetBitContext *s, int n){ register int tmp; OPEN_READER(re, s);
UPDATE_CACHE(re, s); tmp = SHOW_UBITS(re, s, n); LAST_SKIP_BITS(re, s, n); CLOSE_READER(re, s); return tmp;} The new bitstream reader is written to be easier to use, more consistent and to be easier to follow. It is better documented, the functions are named according to the current naming convetions and to be consistent with the bytestream reader naming. bitstream_read_32() reads bits from the 0-32 range and replaces get_bits() get_bits_long() get_bitsz() bitstream_peek_32() replaces show_bits() show_bits_long() show_bits1() bitstream_skip() replaces skip_bits1() skip_bits() skip_bits_long() Sometimes it's very useful to print out how some parameters changes during the program execution.
When writing the new version of some piece of code one usually needs to compare it with the old one to be sure it behaves the same in every case. Especially the corner cases might be tricky and I spent a lot of time with them while my code worked fine in general.
For example when I was working on my ASF demuxer, I was happy there's an old demuxer and I can compare their behaviour. When debugging the ASF, I wanted to know the state of I/O context. In that time lu_zero (who was mentoring me) created a set of macros which printed logs for every I/O function (here). For example there's the macro for avio_seek() function (which is equivalent to fseek()).
#define avio_seek(s, o, w) ({ \
int64_t _ret = avio_seek(s, o, w); \
int64_t _pos = avio_tell(s); \
av_log(NULL, AV_LOG_VERBOSE|AV_LOG_C(154), "0x%08"PRIx64" - %s:%d seek %p %"PRId64" %d -> %"PRId64"\n", \
_pos, __FUNCTION__, __LINE__, s, o, w, _ret); \
_ret; \
}) When such a macro was present in my demuxer, for all the calls of avio_seek the following information was printed _pos = avio_tell(s): the offset in the demuxed file __FUNCTION__ : preprocessor define that contains the name of the function being compiled to know which function called avio_seek __LINE__ : preprocessor define that contains the line number of the original source file that is being compiled to know from what line avio_seek was called from s, o, w : the values of the parameters avio_seek was called with _ret: the avio_seek return value __FILE__: preprocessor define contains the name of the file being compiled (this one was not used in the example but might be useful when one needs more complex log). _ret; as the last statement in this macro because its value serves as the value of the entire construct. If the last _ret; would be omitted in my example, the return value of this macro expression would be printf return value. The underscores in _ret or _pos variables are used to be sure it does not shadow some other variables with the same names.lu_zero for teaching me about it. The support from the more experienced developers is the thing I really love about Libav.
I made a split my complex dcadec bit-exact patch (https://patches.libav.org/patch/59013/) to the several parts. The first part which contains changing the dcadec core to work with integer coefficients instead of converting the coefficients to floats just after reading them was sent to the mailing list (https://patches.libav.org/patch/59141/). Such a change was expected to slow down the decoding process. Therefore I made some measurements to examine how much slower decoding is after my patch.
I decoded this sample: samples.libav.org/A-codecs/DTS/dts/dtswavsample16.wav 10 times and measured the user time between invocation and termination with the "time" command:
time ./avconv -f dts -i dtswavsample16.wav -f null -c pcm_f32le null,counted the average real time of avconv run and repeated everything for the master branch. The duration of the dtswavsample16.wav is ~4 mins and I wanted to look at the slow down for the longer files. Hence I used relatively new loop option for the avconv (http://sasshkas.blogspot.cz/2015/08/the-loop-option-for-avconv.html) to create ~24 mins long file from the initial file by looping it 6x with
./avconv -loop 6 -i dtswavsample16.wav -c copy dts_long.wav.I decoded this longer dts file 10x again for both new integer and old float coefficients core and counted the averages.
When playing a multimedia file, one usually wants to seek to reach different parts of a file. Mostly, containers allows this feature but it might be problem for streamed files.
Withing the libavformat, seeking is performed with function (inside the demuxer) called read_seek. This function tries to find matching timestamp for the requested position (offset) in the played file.
There are 2 ways to seek through the file. One of them is when file contains some kind of index, which matches positions with appropriate timestamps. In this case index entries are created by calling av_add_index_entry. If index entries are present, av_index_search_timestamp, which is called inside the read_seek, looks for the closest timestamp for the requested position. When the file does not provide such entries, one can look for the requested position with ff_seek_frame_binary. For doing so, read_timestamp function has to be created inside the demuxer.
Read_timestamp takes required position and stream index and then tries to find offset of the beginning of the closest packet wich is key frame with matching stream index. While doing this, read_timestamp reads timestamps for all the packets after given position and creates index entries. When the key frame with matching stream index is found, read_timestamp upgrades required position and returns timestamp matching to it.
I was told to test my ASF demuxer with the zzuf utility. Zuff is a fuzzer, it changes random bits in the program's input which simulates damaged file or unexpected data.
For testing ASF's behaviour I want to feed avconv with some corrupted wmv files and see what will happen. Because I want to fuzz in several different ways I want to vary seed (the initial value of zzuf’s random number generator). I'll do this with command:
while true; SEED=$RANDOM; for file *wmv; do zzuf -M -l -r 0.00001 -q -U 60 -s $SEED ./avconv -i "file" -f null -c copy - || echo $SEED $file >> fuzz; done; done;.
I got the file fuzz which is the list of seed paired with filename. Now I need to use zzuf for creating damaged files to check the problem with valgrind. I'll use the list to determine the seed which caused some crash for creating my damaged file:
zzuf -M -l -r 0.00001 -q -U 60 -s myseed < somefile.wmv | cat out.asf.
Now I'll just use valgrind to find out what happened:
valgrind ./avconv -i out.asf -f null -c copy -.
I tried to test the ASF demuxer with different tricky samples and with FATE
and the demuxer behaved well but testing with zzuf detected several new crashes. Mainly it was insane values sizes and it was easy to fix them by adding some more checks. Zzuf is a great thing for testing.
Pelhřimov is small but very nice town in Czech Republic approximately 120 km from the capital Prague and I decided to organize a small but nice Libav sprint in it.
The participants and the topics were:
dca_decode_frame to handle all the extensions and working with the new options for them more systematicly. I decided to improve the Libav DTS decoder - dcadec. Here I want to explain what are its problems now and what I would like to do about them.
DTS encoded audio stream consists of core audio and may contain extended audio. Dcadec supports XCH and XLL extensions but X96, XXCH and XBR extensions are waiting to be implemented - I'd like to implement them later.
For the DTS lossless extension - XLL, the decoded output audio should be a bit for bit accurate reproduction of the encoded input. However there are some problems:
dequantization (with int -> float conversion)I'm working now on modifying the core to work with integer coefficients and then convert them to floats before QMF filtering for lossy output but use bitexact QMF (intermediate LFE coefficients should be always integers and I think it's not correct in the current version) for lossless output. Also I added an option called
↓
inverse ADPCM (when needed)
↓
VQ decoding (when needed)
↓
filtering: QMF, LFE, downmixing (when needed)
↓
float output.
-force_fixed to force fixed-point reconstruction for any kind of input.XLL extension is not detected sometimes and the core audio only is decoded in this case. I want to fix this issue as well.I'd like to add the loop option to avconv. This option allows to repeat an input file given number of times, so the output contains specified number of inputs. The command is ./avconv -loop n -i infile outfile, n specifies how many times the input file should be looped in the output.
How does this work?
After processing the input file for the first time, avconv calls new seek_to_start function to seek back to the beginning of the file. av_seek_frame is called to perform seeking itself but there are other things needed for loop option to work.
1) flush
Flush decoder buffers to take out delayed frames. In avconv this is done by calling process_input_file with NULL as frame, process_input_packet had to be modified a little to not to signal EOF on the filters when seeking.
2) timestamps (ts)
To have correct timestamps in the "after seeking" part of the output stream they have to be corrected with ts = ts_{from the demuxer} + n * (duration of the input stream), n is number of times the input stream was processed so far . This duration is the duration of the longest stream in a file because all the streams have to be processed (or played) before starting the next loop. The duration of the stream is the last timestamp - the first timestamp + duration of the last frame. For the audio streams one "frame" is usually a constant number of samples and its duration is number of samples/sample rate. Video frames on the other side are displayed unevenly so their average framerate can be used for the last frame duration if available or if the average framerate is not known the last frame duration is just 1 (in the current time base).
https://github.com/sasshka/libav/commit/90f2071420b6fd50eea34982475819248e5f6c8f
I am hearing a lot of persons interested in open-source and giving back to the community. I think it can be an exciting experience and it can be positive in many different ways: first of all more contributors mean better open-source software being produced and that is great, but it also means that the persons involved can improve their skills and they can learn more about how successful projects get created.
So I wondered why many developers do not do the first step: what is stopping them to send the first patch or the first pull-request? I think that often they do not know where to start or they think that contributing to the big projects out there is intimidating, something to be left to an alien form of life, some breed of extra-good programmers totally separated by the common fellows writing code in the world we experience daily.
I think that hearing the stories of a few developers that have given major contributions to top level project could help to go over these misconceptions. So I asked a few questions to this dear friend of mine, Luca Barbato, who contributed among the others to Gentoo and VLC.
Let’s start from the beginning: when did you start programming?
I started dabbling stuff during high school, but I started doing something more consistent at the time I started university.
What was your first contribution to an open-source project?
I think either patching the ati-drivers to work with the 2.6 series or hacking cloop (a early kernel module for compressed loops) to use lzo instead of gzip.
What are the main projects you have been involved into?
Gentoo, MPlayer, Libav, VLC, cairo/pixman
How did you started being involved in Gentoo? Can you explain the roles you have covered?
Daniel Robbins invited me to join, I thought “why not?
During the early times I took care of PowerPC and [Altivec](http://en.wikipedia.org/wiki/AltiVec), then I focused on the toolchain due the fact it gcc and binutils tended to break software in funny ways, then multimedia since altivec was mainly used there. I had been part of the Council a few times used to be a recruiter (if you want to join Gentoo feel free to contact me anyway, we love to have more people involved) and I’m involved with community relationship lately.
Note: Daniel Robbins is the creator of Gentoo, a Linux distribution.
Are there other less famous projects you have contributed to?
I have minor contributions in quite a bit of software due. The fact is that in Gentoo we try our best to upstream our changes and I like to get back fixes to what I like to use.
What are your motivations to contribute to open-source?
Mainly because I can =)
Who helped you to start contributing? From who you have learnt the most?
Daniel Robbins surely had been one of the first asking me directly to help.
You learn from everybody so I can’t name a single person among all the great people I met.
How did you get to know Daniel Robbins? How did he helped you?
I was a gentoo user, I happened to do stuff he deemed interesting and asked me to join.
He involved me in quite a number of interesting projects, some worked (e.g. Gentoo PowerPC), some (e.g. Gentoo Games) not so much.
Do your contributions to open-source help your professional life?
In some way it does, contrary to the assumption I’m just seldom paid to improve the projects I care about the most, but at the same time having them working helps me when I need them during the professional work.
How do you face disagreement on technical solutions?
I’m a fan of informed consensus, otherwise prototypes (as in “do, test and then tell me back”) work the best.
To contribute to OSS are more important the technical skills or the diplomatic/relation skills?
Both are needed at different time, opensource is not just software, you MUST get along with people.
Have you found different way to organize projects? What works best in your opinion? What works worst?
Usually the main problem is dealing with poisonous people, doesn’t matter if it is a 10-people project or a 300+-people project. You can have a dictator, you can have a council, you can have global consensus, poisonous people are what makes your community suffer a lot. Bonus point if the poisonous people get clueless fan giving him additional voices.
Did you ever sent a patch for the Linux kernel?
Not really, I’m not fond of that coding style so usually other people correct the small bugs I stumble upon before I decide to polish my fix so it is acceptable =)
Do you have any suggestions for people looking to get started contributing to open-source?
Pick something you use, scratch your own itch first, do not assume other people are infallible or heroes.
ME: I certainly agree with that, it is one of the best advices. However if you cannot find anything suitable at the end of this post I wrote a short list of projects that could use some help.
Can you tell us about your best and your worst moments with contribution to OSS?
The best moment is recurring and it is when some user thanks you since you improved his or her life.
The worst moment for me is when some rabid fan claims I’m evil because I’m contributing to Libav and even praises FFmpeg for something originally written in Libav in the same statement, happened more than once.
What are you working on right now and what plans do you have for the future?
Libav, plaid, bmdtools, commonmark. In the future I might play a little more with [rust](http://www.rust-lang.org/).
Thanks Luca! I would be extremely happy if this short post could give to someone the last push they need to contribute to an existing open-source project or start their own: I think we could all use more, better, open-source software. So let’s write it.
One thing I admire in Luca is that he is always curious and ready to jump on the next challenge. I think this is the perfect attitude to become an OSS contributor: just start play around with the things you like and talk to people, you could find more possibilities to contribute that you could imagine.
…and one final thing: Luca is also the author of open-source recipes: he created the recipes of two types of chocolate bars dedicated to Libav and VLC. You can find them on the borgodoro website.
I suggest to take a look at his blog.
Well, just in case you are eager to start writing some code and you are looking for some projects to contribute to here there are a few, written with different technologies. If you want to start contributing to any of those and you need directions just drop me a line (federico at tomassetti dot me) and I would be glad to help!
If you are interested in contributing to Libav, you can take a look at this post: there I explained how I submitted my first patch (approved in the meantime!). It is written in C.
You could be also interested in plaid: it is a Python web application to manage git patches sent by e-mail (there are a few projects using this model like libav or the linux kernel)
WorldEngine, it is a world generator written in Python
Plate-tectonics, it is a library for plate tectonics simulation. It is written in C++
JavaParser a Java parser, written in Java
Incremental Java parser, an incremental Java parser, written in Scala
The post How people get started contributing to open-source? A few questions to Luca Barbato, contributor to Gentoo, MPlayer, Libav, VLC, cairo/pixman appeared first on Federico Tomassetti - Consultant Software Engineer.
I happened to have a few hours free and I was looking for some coding to do. I thought about VLC, the media player which I have enjoyed so much using over the years and I decided that I wanted to contribute in some way.
To start helping in such a complex process there are a few steps involved. Here I describe how I got my first patched accepted. In particular I wrote a patch for libav, the library behind VLC.
I started by reading the wiki. It is a very helpful starting point but the process to setup the environment and send a first patch was not yet 100% clear to me so I got in touch with some of the developers of libav to understand how they work and how I could start lending an hand with something simple. They explained me that the easier way to start is by solving issues reported by static analysis tools and style checkers. They use uncrustify to verify that the code is adhering to their style guidelines and they run coverity to check for potential issues like memory leaks or null deferences. So I:
After a few minutes the patch was approved by a committer, ready to be merged. The day after it made its way to the master branch. Yeah!
First of all, let’s clone the git repository:
git clone git://git.libav.org/libav.git
Alternatively you could use the GitHub mirror, if you want to.
At this point you may want to install all the dependencies. The instructions are platform specific, you can find them here. If you have Mac Os-X be sure to have installed yasm, because nasm does not work. If you have installed both configure will pick up yasm (correctly). Just be sure to run configure after installing yasm.
If everything goes well you can now build libav by running:
./configure make
Note that it is fine to build in-tree (no need to build in a separate directory).
Now it is time to run the tests. You will have to specify one directory where to download some samples, later used by tests. Let’s assume you wanted to put your samples under ~/libav-samples:
mkdir ~/libav-samples # This download the samples make fate-rsync SAMPLES=~/libav-samples # This run the tests make fate
Did everything run fine? Good! Let’s start to patch then!
First of all we need to find an open issue. Visit Coverity page for libav at https://scan.coverity.com/projects/106. You will have to ask for access and wait that someone grants it to you. When you will be able to login you will encounter a screen like this:
Here, this seems an easy one! The variable oggstream has been allocated by av_mallocz (basically a wrapper for malloc) but the result values has not been checked. If the allocation fails a NULL pointer is returned and when we will try to access it at the next line things are going end up unpleasantly. What we need to do is to check the return value of av_mallocz and if it is NULL we should return an error. The appropriate error to return in this case is AVERROR(ENOMEM). To get this information… you have to start reading code, getting familiar with the way of doing business of this codebase.
Libav follows strict rules about the comments in git commits: use git log to look at previous commits and try to use the same style.
I think many of you are familiar with GitHub and the whole process of submitting a patch for revision. GitHub is great because it made that process so easy. However there are some projects (notably including the Linux kernel) which adopts another approach: they receive patches by e-mail.
Git has a functionality that permits to submit a patch by e-mail with a simple command. The patch will be sent to the mailing list, discussed, and if approved the e-mail will be downloaded, processed through git and committed in the official repository. Does it sound cumbersome? Well, it sounds to me, spoiled as I am by GitHub and similar tools but, you know, if you go in Rome you should behave as the Romans do, so…
# This install the git extension for sending patches through e-mail sudo apt install git-email # This submit a patch built using the last commit git send-email -1 --to libav-devel@libav.org
Now, many of you are using gmail and many of you have enable 2-factor authentication (right? If not, you should). If this is you case you will get an error along this lines:
Password for 'smtp://f.tomassetti@gmail.com@smtp.gmail.com:587': 5.7.9 Application-specific password required. Learn more at 5.7.9 http://support.google.com/accounts/bin/answer.py?answer=185833 cj12sm14743233wjb.35 - gsmtp
Here you can find how to create a password for this goal: https://support.google.com/accounts/answer/185833 The name of the application that I had to create was smtp://f.tomassetti@gmail.com@smtp.gmail.com:587. Note that I used the same name specified in the previous error message.
If things go well an e-mail with your patch will be sent to the mailing-list, someone will look at it and accept it. Most of the times you will receive suggestions about possible adjustments to be done to improve your password. When it happens you want to submit a new version of your patch in the same thread which contains your first version of the patch and the e-mails commenting it.
To do that you want to update your patch (typically using git commit –amend) and then run something like:
git send-email -1 --to libav-devel@libav.org --in-reply-to Message-ID: 54E0F459.3090707@gentoo.org
Of course you need to find out the message-id of the e-mail to which you want to reply. To do that in gmail select the “Show original” item from the contextual menu for the message and in the screen opened look for the Message-Id header.
There are also web applications which are used to manage the patches sent by e-mail. Libav is currently using Patchwork for managing patches. You can see it deployed at: https://patches.libav.org/project/libav-devel/list/. Currently another tool has been developed to replace patchwork. It is named Plaid and I tried to help a little bit with that also 
Mine has been a very small contribution, and in the future I hope to be able to do more. But being a maintainer of other open-source projects I learned that also small help is useful and appreciated, so for today I feel good.
Please, if I am missing something help me correct this post
The post How to contribute to Libav (VLC): just got my first patch approved appeared first on Federico Tomassetti - Consultant Software Engineer.
I've participated to last Libav sprint in Torino. I made new ASF demuxer for Libav, but during the testing problems with rtsp a mms protocols has appeared. Therefore, my main task during the sprint was to fix these issues.
It was second time I was at such sprint and also my second Torino visit and the sprint was even better than I expected. It's really nice to see people I'm communicating throught the irc channel in person, the thing I like about Libav a lot is its friendly community. But the most important thing for me as the most unexperienced person among skilled developers was naturally their help. My mentors from OPW participated the sprint and as a result all the issues was fixed and patch was sent to the ML (https://patches.libav.org/patch/55682/). Also, these personal consultations can be very productive in learning new things and because I'm not native English speaker I realized few days I have to speak or even think in English are really helpful for getting better in it.
The last day of the sprint we had a trip to a really magical place called Sacra di San Michele (http://www.sacradisanmichele.com/).
After my challenge with the fused multiply-add instructions I managed to find some time to write a new test utility. It’s written ad hoc for unpaper but it can probably be used for other things too. It’s trivial and stupid but it got the job done.
What it does is simple: it loads both a golden and a result image files, compares the size and format, and then goes through all the bytes to identify how many differences are there between them. If less than 0.1% of the image surface changed, it consider the test a pass.
It’s not a particularly nice system, especially as it requires me to bundle some 180MB of golden files (they compress to just about 10 MB so it’s not a big deal), but it’s a strict improvement compared to what I had before, which is good.
This change actually allowed me to explore one change that I abandoned before because it resulted in non-pixel-perfect results. In particular, unpaper now uses single-precision floating points all over, rather than doubles. This is because the slight imperfection caused by this change are not relevant enough to warrant the ever-so-slight loss in performance due to the bigger variables.
But even up to here, there is very little gain in performance. Sure some calculation can be faster this way, but we’re still using the same set of AVX/FMA instructions. This is unfortunate, unless you start rewriting the algorithms used for searching for edges or rotations, there is no gain to be made by changing the size of the code. When I converted unpaper to use libavcodec, I decided to make the code simple and as stupid as I could make it, as that meant I could have a baseline to improve from, but I’m not sure what the best way to improve it is, now.
I still have a branch that uses OpenMP for the processing, but since most of the filters applied are dependent on each other it does not work very well. Per-row processing gets slightly better results but they are really minimal as well. I think the most interesting parallel processing low-hanging fruit would be to execute processing in parallel on the two pages after splitting them from a single sheet of paper. Unfortunately, the loops used to do that processing right now are so complicated that I’m not looking forward to touch them for a long while.
I tried some basic profile-guided optimization execution, just to figure out what needs to be improved, and compared with codiff a proper release and a PGO version trained after the tests. Unfortunately the results are a bit vague and it means I’ll probably have to profile it properly if I want to get data out of it. If you’re curious here is the output when using rbelf-size -D on the unpaper binary when built normally, with profile-guided optimisation, with link-time optimisation, and with both profile-guided and link-time optimisation:
% rbelf-size -D ../release/unpaper ../release-pgo/unpaper ../release-lto/unpaper ../release-lto-pgo/unpaper
exec data rodata relro bss overhead allocated filename
34951 1396 22284 0 11072 3196 72899 ../release/unpaper
+5648 +312 -192 +0 +160 -6 +5922 ../release-pgo/unpaper
-272 +0 -1364 +0 +144 -55 -1547 ../release-lto/unpaper
+7424 +448 -1596 +0 +304 -61 +6519 ../release-lto-pgo/unpaper
It’s unfortunate that GCC does not give you any diagnostic on what it’s trying to do achieve when doing LTO, it would be interesting to see if you could steer the compiler to produce better code without it as well.
Anyway, enough with the microptimisations for now. If you want to make unpaper faster, feel free to send me pull requests for it, I’ll be glad to take a look at them!
RealAudio files have several possible interleavers. The simplest is “Int0”, which means that the packets are in order. Today, I was contrasting “Int4” and “genr”. They both require rearranging data, in highly similar but not identical ways. “genr” is slightly more complex than “Int4”.
A typical Int4 pattern, writing to subpacket 0, 1, 2, 3, etc, would read data from subpacket 0, 6, 12, 18, 24, 30, 36, 42, 48, 54, 60, 66, 1, 7, 13, etc, in that order – assuming subpkt_h is 12, as it was in one sample file. It is effectively subpacket_h rows of subpacket_h / 2 columns, counting up by subpacket_h / 2 and wrapping every two rows.
A typical genr pattern is a little trickier. For subpacket_h = 14, and the same 6 columns per row as above, the pattern to read from looks like 0, 12, 24, 36, 48, 60, 72, 6, 18, 30, 42, 54, 66, 78, 1, etc.
I spent most of today implementing genr, carefully working with a paper notebook, pencil, Python, and a terse formula from the old implementation:
case DEINT_ID_GENR:
for (x = 0; x < w/sps; x++) avio_read(pb, ast->pkt.data+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
After various debug printfs, a lot of quality time in GDB running commands like x /94x (pkt->data + 14 * 94), a few interestingly garbled bits of audio playback, and a mentor pointing out I have some improvements to make on header parsing, I can play (some) genr files.
I have also recently implemented SIPR support, and it works in both RA and RM files. RV10 video also largely works.
I've solved lost packets problems and finally my ASF demuxer started to work right at "ideal samples in vacuum". So the time for fixing memory leaks had come and valgrind helped me a lot with this issue. After memory leaks was solved I had to start testing my demuxer on various samples of ASF format multimedia files. As expected, I've found many samples my demuxer failed for. The reasons was different - mostly it was my mistakes, misunderstood or overlooked parts of specs, but I think I found a case that needed unusual handling specs didn't mention about.
some of problems was caused for example by
* improper subpayloads handling - one should be really careful while reading specs to avoid problems for less common cases like one subpayload is inside single payload and there's is padding inside payload itself (while padding after payload is 0), but there was other problems too
* I had to revise padding handling for all possible cases
* ASF file has 3 places where ASF packet size is told - twice in the header objects and once in a packet itself, and specs are not specifying what should one do when they differs or at least I didn't found it
* some stupid mistakes like when I just forgot to do something after adding new block to my code was really annoying
Funny thing was when I fixed my demuxer for one group of samples and another one that worked before started to fail, I fixed this new group and third group failed. I was so much annoyed by this, but many mistakes I did was caused by my inexperience and I think one (at least me) just have to do all of these mistakes to get better.
Finally, all basic parts of ASF demuxer seems to work somehow.
At last two weeks I fixed various bugs in my code and I hope packets handling is correct now. Only problem is that few packets at the end of the Data Object are still lost. Because I wanted a small break from this problem, my mentors allowed me to implement basic seeking first. ASF demuxer can now read index entries from Simple Index Object and adds them with av_add_index_entry to AVStream. So when Simple Index Object is present in an ASF file, my demuxer can seek to the requested time.
Skeleton of the new ASF demuxer was written, but only audio was demuxed properly now. Problem is complicated video frames handling in ASF format. I hope I finally found out how to process packets properly. ASF packet can contain single payload, single payload with subpayloads, multiple payloads or multiple payloads with subpayloads inside some of them. Every subpayload is always one frame, but single payload can be whole frame or just part of it. When ASF packet contains multiple payloads inside it, each of them can be one frame but it can be just fragment of it as well. When one of mulptiple payloads contains subpayloads, each of subpayload is one frame and it can be processed as AVPacket.
For the case of fragmented frame in ASF packet I have to store several unfinished frames in ASFPacket structures that I've created for this purpose. There should not be more than one unfinished frames per stream, so I have one ASFPacket in each ASFStream (ASFStream is structure for storing ASF stream properties). ASFPacket contains pointer to AVBufferRef where unfinished frame is stored. When frame is finished I can forward pointer to buffer with data to AVPacket, set its properties like size, timestamps and others and finally return AVPacket.
I introduced many bugs to my code that was working (at least ASF packets was parsed right and audio worked) and now I'm working on fixing all of them.
I was accepted for OPW, May - August 2014 round with project "Rewrite the ASF demuxer". First task from my mentors was to create wiki page about ASF (Advanced Streaming Format), it was created at https://wiki.libav.org/ASF.
Interesting notes about other containers: http://codecs.multimedia.cx/?p=676.
Next task from my mentors was to write simple program which reads asf file and prints its structure, i.e. list of asf objects, metadata and codec information. ASF file consists of so called ASF Objects. There're 3 top-level objects - Header Object, Data Object and Index Object. Especially Header Object can contain many other objects to provide different asf features, for example Codec List Object for codec information or Metadata object for metadata. One can recognise object with GUID, which is 16 byte array (each byte is number) that identifies object type. I was confused about the fact the GUID number you read from the file is not matching the GUID from specs. For some historical reasons one have to modify GUIDs from specs (reorder the numbers) for match GUID read from the file.
My program is working now and can list objects, codecs and metadata info, but it ignores Index Objects by then. I hope I'll add support for them soon. Also I want to print offsets for each object and read Data Object deeper.
Today, I learned how to use framecrc as a debug tool. Many Libav tests use framecrc to compare expected and actual decoding. While rewriting existing code, the output from the old and new versions of the code on the same sample can be checked; this makes a lot of mistakes clear quickly, including ones that can be quite difficult to debug otherwise.
Checking framecrcs interactively is straightforward: ./avconv -i somefile -c:a copy -f framecrc -. The -c:a copy specifies that the original, rather than decoded, packet should be used. The - at the end makes the output go to stdout, rather than a named file.
The output has several columns, for the stream index, dts, pts, duration, packet size, and crc:
0, 0, 0, 192, 2304, 0xbf0a6b45
0, 192, 192, 192, 2304, 0xdd016b78
0, 384, 384, 192, 2304, 0x18da71d6
0, 576, 576, 192, 2304, 0xcf5a6a07
0, 768, 768, 192, 2304, 0x3a84620a
It is also unusually simple to find out what the fields are, as libavformat/framecrcenc.c spells it out quite clearly:
static int framecrc_write_packet(struct AVFormatContext *s, AVPacket *pkt)
{
uint32_t crc = av_adler32_update(0, pkt->data, pkt->size);
char buf[256];
snprintf(buf, sizeof(buf), “%d, %10″PRId64″, %10″PRId64″, %8d, %8d, 0x%08″PRIx32″\n”,
pkt->stream_index, pkt->dts, pkt->pts, pkt->duration, pkt->size, crc);
avio_write(s->pb, buf, strlen(buf));
return 0;
}
Keiler, one of my Libav mentors, patiently explained the above; I hope documenting it helps other people who are starting with Libav development.
Most recently, I have been adding documentation to Libav. Today, my work included writing a demuxer howto. In the last couple of weeks, I have also reimplemented RealAudio 1.0 support (2.0 is in progress), and learned more about Coccinelle and undefined behavior in C. Blog posts on these topics are pending.
My first patch for undefined behavior eliminates left shifts of negative numbers, replacing a << b (where a can be negative) with a * (1 << b). This change fixes bug686, at least for fate-idct8x8 and libavcodec/dct-test -i (compiled with ubsan and fno-sanitize-recover). Due to Libav policy, the next step is to benchmark the change. I was also asked to write a simple benchmarking HowTo for the Libav wiki.
First, I installed perf: sudo aptitude install linux-tools-generic
I made two build directories, and built the code with defined behavior in one, and the code with undefined behavior in the other (with ../configure && make -j8 && make fate). Then, in each directory, I ran:
perf stat --repeat 150 ./libavcodec/dct-test -i > /dev/null
The results were somewhat more stable than with –repeat 30, but it still looks much more like noise than a meaningful result. I ran the command with –repeat 30 for both before the recorded 150 run, so both would start on equal footing. With defined behavior, the results were “0.121670022 seconds time elapsed ( +- 0.11% )”; with undefined behavior, “0.123038640 seconds time elapsed ( +- 0.15% )”. The best of a further three runs had the opposite result, shown below:
% cat undef.150.best
perf stat –repeat 150 ./libavcodec/dct-test -i > /dev/null
Performance counter stats for ‘./libavcodec/dct-test -i’ (150 runs):
120.427535 task-clock (msec) # 0.997 CPUs utilized ( +- 0.11% )
21 context-switches # 0.178 K/sec ( +- 1.88% )
0 cpu-migrations # 0.000 K/sec ( +-100.00% )
226 page-faults # 0.002 M/sec ( +- 0.01% )
455’393’772 cycles # 3.781 GHz ( +- 0.05% )
<not supported> stalled-cycles-frontend
<not supported> stalled-cycles-backend
1’306’169’698 instructions # 2.87 insns per cycle ( +- 0.00% )
89’674’090 branches # 744.631 M/sec ( +- 0.00% )
1’144’351 branch-misses # 1.28% of all branches ( +- 0.18% )
0.120741498 seconds time elapse
% cat def.150.best
Performance counter stats for ‘./libavcodec/dct-test -i’ (150 runs):
120.838976 task-clock (msec) # 0.997 CPUs utilized ( +- 0.11% )
21 context-switches # 0.172 K/sec ( +- 1.98% )
0 cpu-migrations # 0.000 K/sec
226 page-faults # 0.002 M/sec ( +- 0.01% )
457’077’626 cycles # 3.783 GHz ( +- 0.08% )
<not supported> stalled-cycles-frontend
<not supported> stalled-cycles-backend
1’306’321’521 instructions # 2.86 insns per cycle ( +- 0.00% )
89’673’780 branches # 742.093 M/sec ( +- 0.00% )
1’148’393 branch-misses # 1.28% of all branches ( +- 0.11% )
0.121162660 seconds time elapsed ( +- 0.11% )
I also compared the disassembled code from jrevdct.o, before and after the changes to have defined behavior (using gcc (Ubuntu 4.8.2-19ubuntu1) 4.8.2 on x86_64).
In the build directory for the code with defined behavior:
objdump -d libavcodec/jrevdct.o > def.dis
sed -e 's/^.*://' def.dis > noline.def.dis
In the build directory for the code with undefined behavior:
objdump -d libavcodec/jrevdct.o > undef.dis
sed -e 's/^.*://' undef.dis > noline.undef.dis
Leaving aside difference in jump locations (despite the fact that they can impact performance), there are two differences:
diff -u build_benchmark_undef/noline.undef.dis build_benchmark_def/noline.def.dis
– 0f bf 50 f0 movswl -0x10(%rax),%edx
+ 0f b7 58 f0 movzwl -0x10(%rax),%ebxi
It’s switched to using a zero-extension rather than sign-extension in one place.
– 74 1c je 40 <ff_j_rev_dct+0x40>
– c1 e2 02 shl $0x2,%edx
– 0f bf d2 movswl %dx,%edx
– 89 d1 mov %edx,%ecx
– 0f b7 d2 movzwl %dx,%edx
– c1 e1 10 shl $0x10,%ecx
– 09 d1 or %edx,%ecx
– 89 48 f0 mov %ecx,-0x10(%rax)
– 89 48 f4 mov %ecx,-0xc(%rax)
– 89 48 f8 mov %ecx,-0x8(%rax)
– 89 48 fc mov %ecx,-0x4(%rax)
+ 74 19 je 3d <ff_j_rev_dct+0x3d>
+ c1 e3 02 shl $0x2,%ebx
+ 89 da mov %ebx,%edx
+ 0f b7 db movzwl %bx,%ebx
+ c1 e2 10 shl $0x10,%edx
+ 09 da or %ebx,%edx
+ 89 50 f0 mov %edx,-0x10(%rax)
+ 89 50 f4 mov %edx,-0xc(%rax)
+ 89 50 f8 mov %edx,-0x8(%rax)
+ 89 50 fc mov %edx,-0x4(%rax)
Leaving aside differences in register use:
– 0f bf d2 movswl %dx,%edx
There is one extra movswl instruction in the version with undefined behavior, at least with the particular version of the particular compiler for the particular architecture checked.
This is an example of a null result while benchmarking; neither version performs better, although any given benchmark has one or the other come out ahead, generally by less than the variance within the run. If this were a suggested performance change, it would not make sense to apply it. However, the point of this change was correctness; a performance increase is not expected, and the lack of a performance penalty is a bonus.
One of my fantastic OPW mentors prepared a “Welcome task package”, of self-contained, approachable, useful tasks that can be done while getting used to the code, and with a much smaller scope than the core objective. This is awesome. To any mentors reading this: consider making a welcome package!
Step one of it is to use ubsan with gdb. This turned out to be somewhat intricate, so I have decided to supplement the wiki’s documentation with a step-by-step guide for Ubuntu 14.04.
1) Install clang-3.5 (sudo aptitude install clang-3.5), as Ubuntu 14.04 comes with gcc 4.8, which does not support -fsanitize=undefined.
2) Under libav, mkdir build_ubsan && cd build_ubsan && ../configure --toolchain=clang-usan --extra-cflags=-fno-sanitize-recover (alternatively, –cc=clang –extra-cflags=-fsanitize=undefined –extra-ldflags=-fsanitize=undefined can be used instead of –toolchain=clang-usan).
3) make -j8 && make fate
4) Watch where the tests die (they only die if –extra-cflags=-fno-sanitize-recover is used). For me, they died on TEST idct8x8. Running make V=1 fate and asking my mentors pointed me towards libavcodec/dct-test -i, which is dying on jrevdct.c:310:47: with “runtime error: left shift of negative value -14”. If you really want to err on the side of caution, make a second build dir, and ./configure --cc=clang && make -j8 && make fate in it, making sure it does not fail… this confirms that the problem is related to configuring with –toolchain=clang-usan (and, it turns out, with -fsanitize=undefined).
5) It’s time to use the information my mentor pointed out on the wiki about ubsan at https://wiki.libav.org/Security/Tools – specifically, the information about useful gdb breakpoints. I put a modified version of the b_u definitions into ~/.gdbinit. The wiki has been updated now, but was originally missing a few functions, including one that turns out to be relevant: __ubsan_handle_shift_out_of_bounds
6 Run gdb ./libavcodec/dct-test, then at the gdb prompt, set args -i to set the arguments dct-test was being run with, and then b_u to load the ubsan breakpoints defined above. Then start the program: type run at the gdb prompt.
7) It turns out that a problem can be found, and the program stops running. Get a backtrace with bt.
680 in __ubsan_handle_shift_out_of_bounds ()
#1 0x000000000048ac96 in __ubsan_handle_shift_out_of_bounds_abort ()
#2 0x000000000042c074 in row_fdct_8 (data=<optimized out>) at /home/me/opw/libav/libavcodec/jfdctint_template.c:219
#3 ff_jpeg_fdct_islow_8 (data=<optimized out>) at /home/me/opw/libav/libavcodec/jfdctint_template.c:273
#4 0x0000000000425c46 in dct_error (dct=<optimized out>, test=<optimized out>, is_idct=<optimized out>, speed=<optimized out>) at /home/me/opw/libav/libavcodec/dct-test.c:246
#5 main (argc=<optimized out>, argv=<optimized out>) at /home/me/opw/libav/libavcodec/dct-test.c:522
It would be nice to see a bit more detail, so I wanted to compile the project so that less would be optimized out, and eventually settled on -O1 because compiling with ubsan and without optimizations failed (which I reported as bug 683). This led to a slightly better backtrace:
#0 0x0000000000491a70 in __ubsan_handle_shift_out_of_bounds ()
#1 0x0000000000492086 in __ubsan_handle_shift_out_of_bounds_abort ()
#2 0x0000000000434dfb in ff_j_rev_dct (data=<optimized out>) at /home/me/opw/libav/libavcodec/jrevdct.c:275
#3 0x00000000004258eb in dct_error (dct=0x4962b0 <idct_tab+64>, test=1, is_idct=1, speed=0) at /home/me/opw/libav/libavcodec/dct-test.c:246
#4 0x00000000004251cc in main (argc=<optimized out>, argv=<optimized out>) at /home/me/opw/libav/libavcodec/dct-test.c:522
It is possible to work around the problem by modifying the source code rather than the compiler flags: FFmpeg did so within hours of the bug report – the commit is at http://git.videolan.org/?p=ffmpeg.git;a=commit;h=bebce653e5601ceafa004db0eb6b2c7d4d16f0c0 ! Both FFmpeg and Libav have also merged my patch to work around the problem (FFmpeg patch, Libav patch). The workaround of using -O1 was suggested by one of my mentors, lu_zero; –disable-optimizations does not actually disable all optimizations (in practice, it leaves in ones necessary for compilation), and it does not touch the -O1 that –toolchain=clang-usan now sets.
Wanting a better backtrace leads to the next post: a detailed guide to narrowing down a bug in a the C compiler, Clang. Yes, I know, the problem is never a bug in the C compiler – but this time, it was.
What’s the fun of only running code on platforms you physically have? Portability is important, and Libav actively targets several platforms. It can be useful to be able to try out the code, even if the hardware is totally unavailable.
Here is how to run Libav’s tests under aarch64, on x86_64 hardware and Ubuntu 14.04. This guide is provided in the hopes that it saves someone else 20 hours or more: there is a lot of once-excellent information which has become misleading, because a lot of progress has been made in aarch64 support. I have tried three approachs – building with Linaro’s cross-compiler, building under QEMU user emulation, and building under QEMU system emulation, and cross-compiling. Building with a cross-compiler is the fastest option. Building under user emulation is about ten times slower. Building under system emulation is about a hundred times slower. There is actually a fourth option, using ARM Foundation Model, but I have not tried it. Running under QEMU user emulation is the only approach I managed to make entirely work.
For all three approaches, you will want a rootfs; I used Ubuntu Core. You can download Ubuntu Core for aarch64 (a minimal rootfs; see https://wiki.ubuntu.com/Core to learn more), and untar it (as root) into a new directory. Then, set an environment variable that the rest of this guide/set of notes uses frequently, changing the path to match your system:
export a64root=/path/to/your/aarch64/rootdir
Approach 1 – build under QEMU’s user emulation.
Step 1) Set up QEMU. The days when using SUSE branches were necessary are over, but it still needs to be statically linked, and not all QEMU packages are. Ubuntu has a static QEMU:
sudo aptitude install qemu-user-static
This package also sets up binfmt for you. You can delete broken or stale binfmt information by running:
echo -1 > /proc/sys/fs/binfmt_misc/archnamehere – this can be useful, especially if you have previously installed QEMU by hand.
Step 2) Copy your QEMU binary into the chroot, as root, with:
cp `which qemu-aarch64-static` $a64root/usr/bin/
Step 3) As root, set up the aarch64 image so it can do DNS resolution, so you can freely use apt-get:
echo 'nameserver 8.8.8.8' > $a64root/etc/resolv.conf
Step 4) Chroot into your new system. Run chroot $a64root /bin/bash as root.
At this point, you should be able to run an aarch64 version of ls, and confirm with file /bin/ls that it is an aarch64 binary.
Now you have a working, emulated, minimal aarch64 system.
On x86, you would run aptitude build-dep libav, but there is no such package for aarch64 yet, so outside of the chroot, on the normal system, I installed apt-rdepends and ran:
apt-rdepends --build-depends --follow=DEPENDS libav
With version information stripped out, the following packages are considered dependencies:
debhelper frei0r-plugins-dev libasound2-dev libbz2-dev libcdio-cdda-dev libcdio-dev libcdio-paranoia-dev libdc1394-22-dev libfreetype6-dev libgnutls-dev libgsm1-dev libjack-dev libmp3lame-dev libopencore-amrnb-dev libopencore-amrwb-dev libopenjpeg-dev libopus-dev libpulse-dev libraw1394-dev librtmp-dev libschroedinger-dev libsdl1.2-dev libspeex-dev libtheora-dev libtiff-dev libtiff5-dev libva-dev libvdpau-dev libvo-aacenc-dev libvo-amrwbenc-dev libvorbis-dev libvpx-dev libx11-dev libx264-dev libxext-dev libxfixes-dev libxvidcore-dev libxvmc-dev texi2html yasm zlib1g-dev doxygen
Many of the libraries do not have current aarch64 Ubuntu packages, and neither does frei0r-plugins-dev, but running aptitude install on the above list installs a lot of useful things – including build-essential. The full list is in the command below; the missing packages are non-essential.
Step 5) Set it up: apt-get install aptitude
aptitude install git debhelper frei0r-plugins-dev libasound2-dev libbz2-dev libcdio-cdda-dev libcdio-dev libcdio-paranoia-dev libdc1394-22-dev libfreetype6-dev libgnutls-dev libgsm1-dev libjack-dev libmp3lame-dev libopencore-amrnb-dev libopencore-amrwb-dev libopenjpeg-dev libopus-dev libpulse-dev libraw1394-dev librtmp-dev libschroedinger-dev libsdl1.2-dev libspeex-dev libtheora-dev libtiff-dev libtiff5-dev libva-dev libvdpau-dev libvo-aacenc-dev libvo-amrwbenc-dev libvorbis-dev libvpx-dev libx11-dev libx264-dev libxext-dev libxfixes-dev libxvidcore-dev libxvmc-dev texi2html yasm zlib1g-dev doxygen
Now it is time to actually build libav.
Step 6) Create a user within your chroot: useradd -m auser, and switch to running as that user: sudo -u auser bash, and type cd to go to the home directory.
Step 7) Run git clone git://git.libav.org/libav.git, then ./configure --disable-pthreads && make -j8 (change the 8 to approximately the number of CPU cores you have).
On my hardware, this takes 10-11 minutes, and ‘make fate’ takes about 16. Disabling pthreads is essential, as qemu-user does not handle threads well, and running the tests hangs randomly without it.
Approach 2: cross-compile (warning: I do not have the tests working with this approach).
1) Start by getting an aarch64 compiler. A good place to get one is http://releases.linaro.org/latest/components/toolchain/binaries/; I am using http://releases.linaro.org/latest/components/toolchain/binaries/gcc-linaro-aarch64-linux-gnu-4.8-2014.04_linux.tar.xz . Untar it, and add it to your path:
export PATH=$PATH:/path/to/your/linaro/tools/bin
2) Make the cross-compiler work. Run aptitude install lsb lib32stdc++6. Without this, invoking the compiler will say “No such file or directory”. See http://lists.linaro.org/pipermail/linaro-toolchain/2012-January/002016.html.
3) Under the libav directory (run git clone git://git.libav.org/libav.git if you do not have one), type mkdir a64crossbuild; cd a64crossbuild. Make sure the libav directory is somewhere under $a64root (it should simplify running the tests, later).
4)./configure --arch=aarch64 --cpu=generic --cross-prefix=aarch64-linux-gnu- --cc=aarch64-linux-gnu-gcc --target-os=linux --sysroot=$a64root --target-exec="qemu-aarch64-static -L $a64root" --disable-pthreads
This is a minimal variant of Jannau’s configuration – a developer who has recently done a lot of libav aarch64 work.
5) Run make -j8. On my hardware, it takes just under a minute.
6) Run make fate. Unfortunately, both versions of QEMU I tried hung on wait4 at this point (in fft-test, fate-fft-4), and used an extra couple of hundred megabytes of RAM per second until I stopped QEMU, even if I asked it to wait for a remote GDB. For anyone else trying this, https://lists.libav.org/pipermail/libav-devel/2014-May/059584.html has several useful tips for getting the tests to run after cross-compilation.
Approach 3: Use QEMU’s system emulation. In theory, this should allow you to use pthreads; in practice, the tests hung for me. The following May 9th post describes what to do: http://www.bennee.com/~alex/blog/2014/05/09/running-linux-in-qemus-aarch64-system-emulation-mode/. In short: git clone git://git.qemu.org/qemu.git qemu.git && cd qemu.git && ./configure --target-list=aarch64-softmmu && make, then
./aarch64-softmmu/qemu-system-aarch64 -machine virt -cpu cortex-a57 -machine type=virt -nographic -smp 1 -m 2048 -kernel aarch64-linux-3.15rc2-buildroot.img --append "console=ttyAMA0" -fsdev local,id=r,path=$a64root,security_model=none -device virtio-9p-device,fsdev=r,mount_tag=r
Then, under the buildroot system, log in as root (no password), and type mkdir /mnt/core && mount -t 9p -o trans=virtio r /mnt/core. At this point, you can run chroot /mnt/core /bin/bash, and follow the approach 1 instructions from useradd onwards, except that ./configure without –disable-pthreads should theoretically work. On my system, ./configure takes a bit over 5 minutes with this approach. Running make is quite slow; time make took 113 minutes. Do not use -j – you are limited to a single core, so -j would slow compilation down slightly. However, make fate consistently hung on acodec-pcm-alaw, and I have not yet figured out why.
Things not to do:
Applying to OPW requires an initial contribution. The Libav IRC channel suggested porting the asettb filter from FFmpeg, so I did (version 5 of the patch was merged upstream, in two parts: a rename patch and a content patch; the FFmpeg author was credited as author for the latter, while I did a signed-off-by). I also contributed a 3000+ line documentation patch, standardizing the libavfilter documentation and removing numerous English errors, and triaged a few bugs, git bisecting the one that was reproducible.
And how it nearly ruined another video coding standard.
Everyone knows that interlacing was a trick in the '80s for pseudo motion compensation with analogue video. This more or less worked because it mimicked how television worked back then. This technique was preserved when flat panels for pc and tv were introduced, for a mix of backward compatibility and technical limitations, and video coding features interlacing in MPEG2 and H264 and similar.
However as with black and white, TACS and Gopher, old technology has to be replaced with modern and efficient technology, as a trade off of users' interests and technology providers' market prospects. In case you are not familiar, interlacing is a mess to support, makes decoding slower and heavily degrades quality. People saying that interlacing saves bandwidth do not know much about video coding and bad marketing claiming that higher resolution is better than higher framerate has an effect too.
So, when ITU and then MPEG set out to establish the mandates for a new video standard capable of superseding H264, it was decided that interlacing was old enough, did more harm than good and it was time for retirement: HEVC was going to be the first video codec to officially deprecate interlacing.
Things went pretty swell during its development, until a few months before the completion of the standard. A group of US companies complained that the proposed tools were not sufficient (a set of SEI messages and treating fields like progressive frames) and heavily protested with both standardisation bodies. ITU firmly rejected the idea (with the video group chair threatening to step down) while MPEG set out to understand the needs of the industry and see if there was anything that could be done.
An ad-hoc group was established to see if there was any evidence that interlaced coding tool would have improved the situation. Things looked really shady, the Requirements group even mentioned that it was the first time that an AhG was established to look for evidence, instead of establishing an AhG because there was evidence. Several liasons from EBU and other DVB members tried to point out this absurdity while the threat of adding interlacing back in HEVC became real. Luckily the first version of the specifications got published in the meantime, so this decision didn't slow down the standardisation process.
Why so much love towards interlacing? Well in the "rebellious" group defence, it is true that interlaced content in HEVC is less performant than in H264; however it is also true that such deinterlaced content in HEVC outperforms H264 in any configuration. Truth is that mass marketed deinterlacers (commonly found in televisions for example) bring a lot of royalty income, so it is normal that companies with vested interests would prefer to have interlacing in a soon-popular video standard like HEVC. Also in markets like US where the network operator (which has control on the encoding but not on the video source) might differ from the content provider, it could be politically difficult to act as a carrier only if you have to deinterlace a video.
However these problems are actually not enough for forcing every encoder, decoder, analyser to support a deprecated technology like interlacing. Technical problems can be solved with good deinterlacers at the top of the distribution chain, while political ones can be solved amending contracts. Plus having progressive only video will definitely improve quality and let the industry concentrate on other delicate subjects, like bit depth, both properties going in favour of users' interests.
At the last MPEG meeting, the "rebellious" group which had been working on reintroducing interlacing for a year provided no real evidence that interlaced coding tools would improve HEVC at all. The only sensible solution was to disband the group over this wasted effort and support progressive video only, which is what happened luckily. So now both ITU and MPEG support progressive video only and this has finally nailed it.
Interlacing is dead, long live progressive.
Written by Vittorio Giovara (projectsymphony@gmail.com)
Published under a CC-BY-SA 3.0 license.
I am very glad to announce that Libav 10 has been released!
This has a bunch of features that I contributed to, in particular regarding stereoscopic video and interlaced filtering, but more importantly this release has the work of an awesome group of people which has been carried out for a whole year. This is the magic of open source!
I joined the group more or less one year ago, with some patches regarding an obscure H.264 specification which I then later reimplemented in HEVC and then I wrote a few filters I needed and then designed an API and then, wow! A whole year passed without me noticing, and I am still around, sending patches to the same group of people who welcomed someone who had problems with shifting values (sad but true story)!
I met the team both at VDD and FOSDEM and they've been the most exciting conferences I ever went to (and I went to a lot of them). I couldn't believe I was with the devteam of my favourite multimeida opensource projects I've been following since I was a kid! Until a year ago, I saw the names from the commits and the blogposts from both VideoLAN and Libav projects and I had been thinking "Oh wouldn't it be so cool to be like one of them".
The answer is yes, it definitely would, and it's something that can happen if one is really committed in it! The Libav Info page states "Being a committer is a duty, not a privilege", but it sure does feel like one.
Thanks for this exciting year guys, I look forward to the next ones.
...using latest modern tools!
X264 and VLC are two of the most awesomest opensource software you can find on-line and of course the pose no problem when you compile them on a Unix environment. Too bad that sometimes you need to think of Windowze as well, so we need a way to crosscompile that software: in this blogpost, I'll describe how to achieve that, using modern tools on a Ubuntu 12.04 installation.
[0] Sources
It goes without saying that without the following guides, I'd have had a much harder time!
http://alex.jurkiewi.cz/blog/2010/cross-compiling-x264-for-win32-on-ubuntu-linux
https://bbs.archlinux.org/viewtopic.php?id=138128
http://wiki.videolan.org/Win32Compile
http://forum.videolan.org/viewtopic.php?f=32&t=101489
So a big thanks to all the original authors!
[1] Introduction
When you crosscompile you just use the same tools and toolchains that you are used to, gcc, ld and so on, but configured (and compiled) so that they produce executable code for a different platform. This platform can vary both in software and in hardware and it is usually identified by a triplet: the processor architecture, the ABI and the operating system.
What we are going to use here is i686-w64-mingw32, which identifies any x86 cpu since the Pentium III, the w64 ABI used on modern Windows NT systems (if I'm not wrong), and the mingw32 architecture, that is the Windows gcc variant.
[2] Prerequisites
Note that the name of the packages might be slightly different according to your distribution. We are going to need a quite recent mingw-runtime for VLC (>=3.00) which has not yet landed on Ubuntu, so we'll take it from our Debian cousins.
Execute this command$ wget http://ftp.jp.debian.org/debian/pool/main/m/mingw-w64/mingw-w64-dev_3.0~svn4933-1_all.deb
$ sudo dpkg -i mingw-w64-dev_3.0~svn4933-1_all.deb
and then install stock dependencies
$ sudo dpkg -i gcc-mingw-w64 g++-mingw-w64$ sudo dpkg -i pkg-config yasm subversion cvs git-core $ mkdir -p ~/win32-cross/{src,lib,include,share,bin}#!/bin/shPlease not the use of the CFLAGS variables: without all the static parameters, the executable will dynamically link gcc, so you'll need to bundle the equivalent dll. I prefer to have one single exe, so everything goes static, but I'm not really sure which flag is actually needed. If you have any idea, please drop me a line.
TRIPLET=i686-w64-mingw32
export CC=$TRIPLET-gcc
export CXX=$TRIPLET-g++
export CPP=$TRIPLET-cpp
export AR=$TRIPLET-ar
export RANLIB=$TRIPLET-ranlib
export ADD2LINE=$TRIPLET-addr2line
export AS=$TRIPLET-as
export LD=$TRIPLET-ld
export NM=$TRIPLET-nm
export STRIP=$TRIPLET-strip
export PATH="/usr/i586-mingw32msvc/bin:$PATH"
export PKG_CONFIG_PATH="$HOME/win32-cross/lib/pkgconfig/"
export CFLAGS="-static -static-libgcc -static-libstdc++ -I$HOME/win32-cross/include -L$HOME/win32-cross/lib -I/usr/$TRIPLET/include -L/usr/$TRIPLET/lib"
export CXXFLAGS="$CFLAGS"
exec "$@"
$ cd ~/win32-cross/src
$ wget -qO - http://zlib.net/zlib-1.2.7.tar.gz | tar xzvf -
$ cd zlib-1.2.7
$ ../../mingw ./configure
$ sed -i"" -e 's/-lc//' Makefile
$ make
$ DESTDIR=../.. make install prefix=$ cd ~/win32-cross/src
$ git clone git://git.libav.org/libav.git
$ cd libav
$ ./configure \
--target-os=mingw32 --cross-prefix=i686-w64-mingw32- --arch=x86 --prefix=../.. \
--enable-memalign-hack --enable-gpl --enable-avisynth --enable-runtime-cpudetect \
--disable-encoders --disable-muxers --disable-network --disable-devices
$ make
$ make install$ cd ~/win32-cross/src
$ svn checkout http://ffmpegsource.googlecode.com/svn/trunk/ ffms
$ cd ffms
$ ../../mingw ./configure --host=mingw32 --with-zlib=../.. --prefix=$HOME/win32-cross
$ ../../mingw make
$ make install$ cd $HOME/win32-x264/src
# Create a CVS auth file on your machine
$ cvs -d:pserver:anonymous@gpac.cvs.sourceforge.net:/cvsroot/gpac login
$ cvs -z3 -d:pserver:anonymous@gpac.cvs.sourceforge.net:/cvsroot/gpac co -P gpac
$ cd gpac
$ chmod +rwx configure src/Makefile
# Hardcode cross-prefix
$ sed -i'' -e 's/cross_prefix=""/cross_prefix="i686-w64-mingw32-"/' configure
$ ../../mingw ./configure --static --use-js=no --use-ft=no --use-jpeg=no \ --use-png=no --use-faad=no --use-mad=no --use-xvid=no --use-ffmpeg=no \ --use-ogg=no --use-vorbis=no --use-theora=no --use-openjpeg=no \ --disable-ssl --disable-opengl --disable-wx --disable-oss-audio \ --disable-x11-shm --disable-x11-xv --disable-fragments--use-a52=no \ --disable-xmlrpc --disable-dvb --disable-alsa --static-mp4box \ --extra-cflags="-I$HOME/win32-cross/include -I/usr/i686-w64-mingw32/include" \ --extra-ldflags="-L$HOME/win32-cross/lib -L/usr/i686-w64-mingw32/lib"
# Fix pthread lib name
$ sed -i"" -e 's/pthread/pthreadGC2/' config.mak
# Add extra libs that are required but not included
$ sed -i"" -e 's/-lpthreadGC2/-lpthreadGC2 -lwinmm -lwsock32 -lopengl32 -lglu32/' config.mak
$ make
# Make will fail a few commands after building libgpac_static.a# (i586-mingw32msvc-ar cr ../bin/gcc/libgpac_static.a ...).# That's fine, we just need libgpac_static.a i686-w64-mingw32-ranlib bin/gcc/libgpac_static.a $ cp bin/gcc/libgpac_static.a ../../lib/
$ cp -r include/gpac ../../include/
Finally we can compile x264 at full power! The configure script will provide a list of what features have been activated, make sure everything you need is there!$ cd ~/win32-cross/src
$ git clone git://git.videolan.org/x264.git
$ cd x264
$ ./configure --cross-prefix=i686-w64-mingw32- --host=i686-w64-mingw32 \ --extra-cflags="-static -static-libgcc -static-libstdc++ -I$HOME/win32-cross/include" \ --extra-ldflags="-static -static-libgcc -static-libstdc++ -L$HOME/win32-cross/lib" \ --enable-win32thread
$ make$ git clone git://git.videolan.org/vlc.git vlc$ cd vlc And let's get the dependencies through the contrib scripts, qt4 needs to be compiled by hand as the version in Ubuntu repositories doesn't cope well with the rest of the process. I also had to remove some of the files because they were of the wrong architecture (mileage might vary here) .$ mkdir -p contrib/win32
$ cd contrib/win32
$ ../bootstrap --host=i686-w64-mingw32
$ make prebuilt
$ make .qt4$ rm ../i686-w64-mingw32/bin/{moc,uic,rcc}$ cd -$ ./bootstrap$ mkdir win32 && cd win32$ ../extras/package/win32/configure.sh --host=i686-w64-mingw32$ ./compile$ make package-win-commonFor future reference, all (or most of) functions and structs in libav have a prefix that indicates the exposure of that functions. Those are
Well, I've finished the new audio decoding API, which has been merged into Libav master. The new audio encoding API is basically done, pending a (hopefully final) round of review before committing.
Next up is audio timestamp fixes/clean-up. This is a fairly undefined task. I've been collecting a list of various things that need to be fixed and ideas to try. Plus, the audio encoding API revealed quite a few bugs in some of the demuxers. Today I started a sort of TODO list for this stage of the project. I'll be editing it as the project continues to progress.
For the past few weeks I've been working on a new project sponsored by FFMTech. The entire project involves reworking much of the existing audio framework in libavcodec.
Part 1 is changing the audio decoding API to match the video decoding API. Currently the audio decoders take packet data from an AVPacket and decode it directly to a sample buffer supplied by the user. The video decoders take packet data from an AVPacket and decode it to an AVFrame structure with a buffer allocated by AVCodecContext.get_buffer(). My project will include modifying the audio decoding API to decode audio from an AVPacket to an AVFrame, as is done with video.
AVCODEC_MAX_AUDIO_FRAME_SIZE puts an arbitrary limit on the amount of audio data returned by the decoder. For example, each FLAC frame can hold up to 65536 samples for 8 channels at 32-bit sample depth, which is 2097152 bytes of raw audio, but AVCODEC_MAX_AUDIO_FRAME_SIZE is only 192000. Using get/release_buffer() for audio decoding will solve this problem. It will, however, require changes to every audio decoder. Most of those changes are trivial since the frame size is known prior to decoding the frame or is easily parsed. Some of the changes are more intrusive due to having to determine the frame size prior to allocating and writing to the output buffer.
As part of the preparation for the new API, I have been cleaning up all the audio decoder, which has been quite tedious. I've found some pretty surprising bugs along the way. I'm getting close to finishing that part so I'll be able to move on to implementing the new API in each decoder.
So, I've moved on from AHT now, and it's on to Spectral Extension (SPX). I got the full syntax working yesterday, now I just need to figure out how to calculate all the parameters. I have a feeling this will help quality quite a bit, especially when used in conjunction with variable bandwidth/coupling. My vision for automatic bandwidth adjustment is starting to come together.
SPX encoding/decoding is fairly straightforward, so I expect this won't take too long to implement. Similar to channel coupling, the encoder writes coarsely banded scale factors for frequencies above the fully-encoded bandwidth, along with noise blending factors. The decoder copies lower frequency coefficients to the upper bands, multiplies them by the scale factors, and blends them with noise (which has been scaled according to the band energy and the blending factors in the bitstream). For the encoder, I just need to make the reconstructed coefficients match the original coefficients as closely as possible by calculating appropriate spectral extension coordinates and blending factors. Also, like coupling coordinates, the encoder can choose how often to resend the parameters to balance accuracy vs. bitrate.
Once SPX encoding is working properly, I'll revisit variable bandwidth. However, instead of adjusting the upper cutoff frequency (which is somewhat complex to avoid very audible attack/decay), it will adjust the channel coupling and/or spectral extension ranges to keep the cutoff frequency constant while still adjusting to changes in signal complexity to keep a more stable quality level at a constant bitrate. This could also be used in a VBR mode with constrained bitrate limits.
If you want to follow the development, I have a separate branch at my Libav github repository.
http://github.com/justinruggles/Libav/commits/eac3_spx
I finally got the complete AHT syntax working properly. Unfortunately, the quality seems to be lower at all bitrates than with the normal AC-3 quantization. I'm hoping that I just need to pick better gain values, but I have a suspicion that some of the difference is related to vector quantization, which the encoder has no control over (a basic 6-dimensional VQ minimum distance search is the best it can do).
My first step is to find out for sure if choosing better gain values will help. One problem is that the bit allocation model is saying we need X number of bits for each mantissa. Using mode=0 (all zero gains) gives exactly X number of bits per mantissa (with no overhead for encoding the gain values), but the overall quality is lower than with normal AC-3 quantization or even GAQ with simplistic mode/gain decisions. So I think that means there is some bias built-in to the AHT bit allocation table that assumes GAQ will appropriately fine-tune the final allocations. Additionally, it could be that AHT should not always be turned on when the exponents are reused in blocks 1 through 5 (the condition required to use AHT). This is probably the point where I need a more accurate bit allocation model...
edit: After analyzing the bit allocation tables for AC-3 vs. E-AC-3, it seems there is no built-in bias in the GAQ range. They are nearly identical. So the difference is clearly in VQ. Next step, try a direct comparison of quantized mantissas using VQ vs. linear quantization and consider that in the AHT mode decision.
edit2: dct+VQ is nearly always worse than linear quantization... I also tried turning AHT off for a channel if the quantization difference was over a certain threshold, but as the threshold approached zero, the quality approached that with AHT turned off. I don't know what to do at this point... *sigh*
note: analyzation of a commercial E-AC-3 sample using AHT shows that AHT is always turned on when the exponent strategy allows it.
edit3: It turns out that the majority of the quality difference was in the 6-point DCT. If I turn it off in both the encoder and decoder (but leave the quantization the same) the quality is much better. I hope it's a bug or too much inaccuracy (it's 25-bit fixed-point) in my implementation... If not then I'm at another dead-end.
edit4: I'm giving up on AHT for now. The DCT is definitely correct and is very certainly causing the quality decrease. If I can get my hands on a source + encoded E-AC-3 file from a commercial encoder that uses AHT then I will revisit this. Until then, I have nothing to analyze to tell me how using AHT can possibly produce better quality.
Well, I finally got a working E-AC-3 encoder committed to Libav. The bitstream format does save a few bits here and there, but the overall quality difference is minimal. However, it will be the starting point for adding more E-AC-3 features that will improve quality.
The first feature I completed was support for higher bit rates. This is done in E-AC-3 by using fewer blocks per frame. A normal AC-3 frame has 6 blocks of 256 samples each, but E-AC-3 can reduce that to 1, 2, or 3 blocks. This way a small range can be used for the per-frame bit rate, but it still allow for increasing the per-second bit rate. For example, 5.1-channel E-AC-3 content on HD-DVDs was typically encoded at 1536 kbps using 1 block per frame.
Currently I am working on implementing AHT (adaptive hybrid transform). The AHT process uses a 6-point DCT on each coefficient across the 6 blocks in the frame. It basically uses the normal AC-3 bit allocation process to determine quantization of each DCT-processed "pre-mantissa" but it uses a finer resolution for quantization and different quantization methods. I have the 6-point DCT working and one of the two quantization methods. Now I just need to finish the other quantization method and implement mantissa bit counting and bitstream output.
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