Narabu is a new intraframe video codec. You probably want to read part 1, part 2, part 3, part 4 and part 5 first. Like I wrote in part 5, there basically isn't a big splashy ending where everything is resolved here; you're basically getting some graphs with some open questions and some interesting observations. First of all, though, I'll need to make a correction: In the last part, I wrote that encoding takes 1.2 ms for 720p luma-only on my GTX 950, which isn't correct I remembered the wrong number. The right number is 2.3 ms, which I guess explains even more why I don't think it's acceptable at the current stage. (I'm also pretty sure it's possible to rearchitect the encoder so that it's much better, but I am moving on to other video-related things for the time being.) I encoded a picture straight off my DSLR (luma-only) at various resolutions, keeping the aspect. Then I decoded it a bunch of times on my GTX 950 (low-end last-generation NVIDIA) and on my HD 4400 (ultraportable Haswell laptop) and measured the times. They're normalized for megapixels per second decoded; remember that doubling width (x axis) means quadruple the pixels. Here it is: I'm not going to comment much beyond two observations:

• Caches matter, even on GPU. This is the same data over and over again (so small images get an unrealistic boost), so up to a certain point, it's basically all in L1.
• The GTX 950 doesn't really run away from the Intel card before it's getting enough data to chew on. Bigger GPUs don't have faster cores they're just more parallel.

Encoding only contains the GTX 950 because I didn't finish the work to get that single int64 divide off: This is interesting. I have few explanations. Probably more benchmarking and profiling would be needed to make sense of any of it. In fact, it's so strange that I would suspect a bug, but it does indeed seem to create a valid bitstream that is decoded by the decoder. Do note, however, that seemingly even on the smallest resolutions, there's a 1.7 ms base cost (you can't see it on the picture, but you'd see it in an unnormalized graph). I don't have a very good explanation for this either (even though there are some costs that are dependent on the alphabet size instead of the number of pixels), but figuring it out would probably be a great start for getting the performance up. So that concludes the series, on a cliffhanger. :-) Even though it's not in a situation where you can just take it and put it into something useful, I hope it was an interesting introduction to the GPU! And in the meantime, I've released version 1.6.3 of Nageru, my live video mixer (also heavily GPU-based) with various small adjustments and bug fixes found before and during Tr ndisk. And Movit is getting compute shaders for that extra speed boost, although parts of it is bending my head. Exciting times in GPU land :-)

We're streaming live from Tr ndisk 2017, first round of the Norwegian ultimate frisbee series, today from 0945 CET and throughout the day/weekend. It's an interesting first for Nageru in that it's sports, where everything happens much faster and there are more demands for overlay graphics (I've made a bunch of CasparCG templates). I had hoped to get to use Narabu in this, but as the (unfinished) post series indicates, I simply had to prioritize other things. There's plenty of new things for us anyway, not the least that I'll be playing and not operating. :-) Feel free to tune in to the live stream, although we don't have international stream reflectors. It's a fun sport with many nice properties. :-) There will be YouTube not too long after the day is over, too. Edit: All games from Saturday and Sunday are now online; see this YouTube list. Most commentary is in Norwegian, although some games are in English. I was happy everything worked well, and the production crew did a great job (I was busy playing), but of course there's tons of small things we want to improve for next time.

Narabu is a new intraframe video codec. You probably want to read part 1, part 2, part 3 and part 4 first. At this point, we've basically caught up with where I am, so thing are less set in stone. However, let's look at what qualitatively differentiates encoding from decoding; unlike in interframe video codecs (where you need to do motion vector search and stuff), encoding and decoding are very much mirror images of each other, so we can intuitively expect them to be relatively similar in performance. The procedure is DCT, quantization, entropy coding, and that's it. One important difference is in the entropy coding. Since our rANS encoding is non-adaptive (a choice made largely for simplicity, but also because our streams are so short) works by first signaling a distribution and then encoding each coefficient using that distribution. However, we don't know that distribution until we've DCT-ed all blocks in the picture, so we can't just DCT each block and entropy code the coefficients on-the-fly. There are a few ways to deal with this:

• DCT and quantize first, then store to a temporary array, count up all the values, make rANS distributions, and then entropy code from that array. This is conceptually the simplest, but see below.
• Use some sort of approximate distribution, which also saves us from doing the actual counting. This has the problem that different images look different, so it loses efficiency (and could do so quite badly). Furthermore, it means this approximate distribution could never have a 0 for any frequency, which also is a net-negative when we're talking about only 12-bit resolution.
• Use the values from the previous frame, since video tends to have correlation between frames (and we can thus expect the distribution to be a better fit than some generic distribution decided ahead of time). Has many of the same problems as previous point, though, plus we get back the overhead of counting. Still, an interesting possibility that I haven't tried in practice yet.
• Do DCT and quantization just to get the values for counting, then do it all over again later when we have the distribution. Saves the store to the temporary array (the loads will be the same, since we need to load the image twice), at the cost of doing DCT twice. I haven't tried this either.
• A variant on the previous one: Do DCT on a few random blocks (10%?) to get an approximate distribution. I did something similar to this at some point, and it was surprisingly good, but I haven't gone further with it.

As you can see, tons of possible strategies here. For simplicity, I've ended up with the former, although this could very well be changed at some point. There are some interesting subproblems, though: First of all, we need to decide the data type of this temporary array. The DCT tends to concentrate energy into fewer coefficients (which is a great property for compression!), so even after quantization, some of them will get quite large. This means we cannot store them in an 8-bit texture; however, even the bigger ones are very rarely bigger than 10 bits (including a sign bit), so using 16-bit textures wastes precious memory bandwidth. I ended up slicing the coefficients by horizontal index and then pairing them up (so that we generate pairs 0+7, 1+6, 2+5 and 3+4 for the first line of the 8x8 DCT block, 8+15, 9+14, 10+13 and 11+12 for the next line, etc.). This allows us to pack two coefficients into a 16-bit texture, for an average of 8 bits per coefficient, which is what we want. It makes for some slightly fiddly clamping and bit-packing since we are packing signed values, but nothing really bad. Second, and perhaps surprisingly enough, counting efficiently is nontrivial. We want a histogram over what coefficients are used the more often, ie., for each coefficient, we want something like ++counts[dist][coeff] (recall we have four distinct distributions). However, since we're in a massively parallel algorithm, this add needs to be atomic, and since values like e.g. 0 are super-common, all of our GPU cores will end up fighting over the cache line containing counts[dist][0]. This is not fast. Think 10 ms/frame not fast. Local memory to the rescue again; all modern GPUs have fast atomic adds to local memory (basically integrating adders into the cache, as I understand it, although I might have misunderstood here). This means we just make a suitably large local group, build up our sub-histogram in local memory and then add all nonzero buckets (atomically) to the global histogram. This improved performance dramatically, to the point where it was well below 0.1 ms/frame. However, our histogram is still a bit raw; it sums to 1280x720 = 921,600 values, but we want an approximation that sums to exactly 4096 (12 bits), with some additional constraints (like no nonzero coefficients). Charles Bloom has an exposition of a nearly optimal algorithm, although it took me a while to understand it. The basic idea is: Make a good approximation by multiplying each frequency by 4096/921600 (rounding intelligently). This will give you something that very nearly rounds to 4096 either above or below, e.g. 4101. For each step you're above or below the total (five in this case), find the best single coefficient to adjust (most entropy gain, or least loss); Bloom is using a heap, but on the GPU, each core is slow but we have many of them, so it's better just to try all 256 possibilities in parallel and have a simple voting scheme through local memory to find the best one. And then finally, we want a cumulative distribution function, but that is simple through a parallel prefix sum on the 256 elements. And then finally, we can take our DCT coefficients and the finished rANS distribution, and write the data! We'll have to leave some headroom for the streams (I've allowed 1 kB for each, which should be ample except for adversarial data and we'll probably solve that just by truncating the writes and accepting the corruption), but we'll compact them when we write to disk. Of course, the Achilles heel here is performance. Where decoding 720p (luma only) on my GTX 950 took 0,4 ms or so, encoding is 1,2 ms or so, which is just too slow. Remember that 4:2:2 is twice that, and we want multiple streams, so 2,4 ms per frame is eating too much. I don't really know why it's so slow; the DCT isn't bad, the histogram counting is fast, it's just the rANS shader that's slow for some reason I don't understand, and also haven't had the time to really dive deeply into. Of course, a faster GPU would be faster, but I don't think I can reasonably demand that people get a 1080 just to encode a few video streams. Due to this, I haven't really worked out the last few kinks. In particular, I haven't implemented DC coefficient prediction (it needs to be done before tallying up the histograms, so it can be a bit tricky to do efficiently, although perhaps local memory will help again to send data between neighboring DCT blocks). And I also haven't properly done bounds checking in the encoder or decoder, but it should hopefully be simple as long as we're willing to accept that evil input decodes into garbage instead of flagging errors explicitly. It also depends on a GLSL extension that my Haswell laptop doesn't have to get 64-bit divides when precalculating the rANS tables; I've got some code to simulate 64-bit divides using 32-bit, but it doesn't work yet. The code as it currently stands is in this Git repository; you can consider it licensed under GPLv3. It's really not very user-friendly at this point, though, and rather rough around the edges. Next time, we'll wrap up with some performance numbers. Unless I magically get more spare time in the meantime and/or some epiphany about how to make the encoder faster. :-)

Narabu is a new intraframe video codec. You probably want to read part 1, part 2 and part 3 first. So we're at the stage where the structure is in place. How do we decode? Once we have the structure, it's actually fairly straightforward: First of all, we need to figure out where each slice starts and ends. This is done on the CPU, but it's mostly just setting up pointers, so it's super-cheap. It doesn't see any pixels at all, just lengths and some probability distributions (those are decoded on the CPU, but they're only a few hundred values and no FP math is involved). Then, we set up local groups of size 64 (8x8) with 512 floats of shared memory. Each group will be used for decoding one slice (320 blocks), all coefficients. Each thread starts off doing rANS decoding (this involves a lookup into a small LUT, and of course some arithmetic) and dequantization of 8 blocks (for its coefficient); this means we now have 512 coefficients, so we add a barrier, do 64 horizontal IDCTs (one 8-point IDCT per thread), add a new barrier, do 64 vertical IDCTs, and then finally write out the data. We can then do the next 8 coefficients the same way (we kept the rANS decoding state), and so on. Note how the parallelism changes during the decoding; it's a bit counterintuitive at first, and barriers are not for free, but it avoids saving the coefficients to global memory (which is costly). First, the parallelism is over coefficients, then over horizontal DCT blocks, then over vertical DCT blocks, and then we start all over again. In CPU multithreading, this would be very tricky, and probably not worth it at all, but on the GPU, it gives us tons of parallelism. One problem is that the rANS work is going to be unbalanced within each warp. There's a lot more data (and thus more calculation and loading of compressed data from memory) in the lower-frequency coefficients, which means the other threads in the warp are wasting time doing nothing when they do more work. I tried various schemes to balance it better (like making even larger thread groups to get e.g. more coefficient 0s that could work together, or reordering the threads' coefficients in a zizag), but seemingly it didn't help any. Again, the lack of a profiler here is hampering effective performance investigations. In any case, the performance is reasonably good (my GTX 950 does 1280x720 luma-only in a little under 0.4 ms, which equates to ~1400 fps for full 4:2:2). As a side note, GoPro open-sourced their CineForm HD implementation the other day, and I guess it only goes to show that these kind of things really belong on the GPU; they claim similar performance numbers to what I get on my low-end NVIDIA GPU (923.6 fps for 1080p 4:2:2, which would be roughtly 1800 fps on 720p60), but that's on a 4GHz 8-core Broadwell-E, which is basically taking the most expensive enthusiast desktop CPU you can get your hands on and then overclocking it. Kudos to GoPro for freeing it, though (even under a very useful license), even though FFmpeg already had a working reverse-engineered implementation. :-) Next time, we'll look at the encoder, which is a bit more complicated. After that, we'll do some performance tests and probably wrap up the series. Stay tuned.

Review: The Black Gryphon, by Mercedes Lackey & Larry Dixon

Series:	Mage Wars #1
Publisher:	DAW
Copyright:	1994
Printing:	January 1995
ISBN:	0-88677-643-0
Format:	Mass market
Pages:	460

The Mage Wars series (or the Gryphon series, which isn't its official title but which is in all of the titles) is part of the sprawling Valdemar mega-series, but it's a prequel to all of the other stories. It's also slightly challenging if you're reading in publication order, since it was published simultaneously with the Mage Storms series. If you're following publication order, in theory you should interleave the two series, but I hate doing that. I'm therefore reading it after Mage Winds and before Mage Storms. We'll see whether that was a good idea when I get to the next series. You could, if you really wanted to, read this series before any other Valdemar book. As a prequel from the deep past of Valdemar's world, it doesn't depend on the other series, and you'll get a rediscovery of lost knowledge feel from later books. The downside is that it's a rather boring introduction, and that order would spoil a lot of the revelatory flow of the other series (particularly Elspeth's adventures in the Mage Winds books). I'm now getting into the Valdemar books that I've only read once. I've been putting off continuing my Valdemar re-read because this series was next and I remember being rather bored with it the first time I read it. But I'm re-reading for the world-building and background as much as for the characters, and this is a huge chunk of world background that fills in the bones underneath Winds of Fate and its sequels. Here's why Dhorisha is a crater, here's why the Pelagiris forests are such a mess, here's where Ma'ar starts, here's the origin of both the gryphons and K'Leshya, and here, finally, we get to see the legendary Urtho on the page. The problem with writing novels set in the epic backstory of your universe is that it's hard to live up to the drama that readers have invented for themselves. A lot of The Black Gryphon is background to events Valdemar readers already know will happen, creating a corresponding lack of surprise. I reached the end of the book and said "yup, that's pretty much what everyone said had happened." Lackey and Dixon do try to do some interesting things here, one of which being the backgrounding of the war. The Black Gryphon starts in the middle of a long-running conflict between Urtho and Ma'ar and doesn't follow the generals or the battles. The protagonists, instead, are a kestra'chern (a type of psychiatrist and spiritual healer who also uses sex, with the expected conflicts of people who incorrectly think of them as prostitutes) and the eventual leader of the gryphons (Skandranon, who is referenced in later books and who provides the title). We get some combat scenes from Skandranon and later another gryphon, but a lot of the book is Amberdrake fighting the effects of the war instead of the details of the war itself. There's a deep and moving story in that idea, and in some of the attached love stories that play out in the army camps. There's also a great story somewhere around Urtho: a brilliant but detached mage who is way out of his depth trying to run an army but smart enough to gather good people around him. He's also a creator of new life, including the gryphons. The Black Gryphon tries to talk about Urtho's paternalism, the weird emotional currents of his relationship with his creations, and the places Urtho keeps things from others for, supposedly, their own good. If this book had looked a bit deeper at the support structure for an army that's trying to be humane, or at the ways in which Urtho strays far too close to being an abusive tyrant through inaction despite having the best of intentions at every step, I think it could have said something significant. Unfortunately, that's not this book. This book is full of relentlessly black and white morality (the flaw of much of the Valdemar series) that bleaches away interesting shades of grey. Urtho is good and wise by authorial fiat, and Ma'ar is the same utterly irredeemable force of evil that he is in other books. The story skitters over Urtho's odd tyrannies, making them all better with the pure power of friendship and good intentions. There just isn't much emotional depth, and while I don't expect that of Lackey in general, this story really needed that depth to work. What we get instead is repetition, as Lackey and Dixon hit the same emotional notes with Amberdrake repeatedly. This is one of those books that makes me wonder if Lackey was trying to write too many novels in a short time than was good for their individual quality. (Collaborations often mean that the lesser-known name is doing all the work, but Dixon is Lackey's husband and the tone of the book is sufficiently Lackey that I don't think that happened here.) It felt padded by Amberdrake turning over the same emotional rocks repeatedly, to largely the same effect. This is, in short, not Lackey's finest effort, although it does have its moments. As always, Lackey is at her best when writing psychological healing narratives. Zhaneel's story is a bit too easy, but the dynamic between Amberdrake and Winterhart is the best part of the book. And The Black Gryphon does tell the reader exactly what led up to the Cataclysm and why. There are no major surprises, but there are some small ones, and it's a nice payoff for the lore-obsessed (like me). This is missable unless you want the full world-building behind Valdemar's past, and it's not the best writing. But if you're heavily invested in the Valdemar universe, it's at least readable and provides an important bit of the history. Followed by The White Gryphon. Rating: 5 out of 10

Narabu is a new intraframe video codec. You probably want to read part 1 and part 2 first. Now having a rough idea of how the GPU works, it's obvious that we need our codec to split the image into a lot of independent chunks to be parallel; we're aiming for about a thousand chunks. We'll also aim for a format that's as simple as possible; other people can try to push the frontiers on research, I just want something that's fast, reasonably good, and not too hard to implement. First of all, let's look at JPEG. It works by converting the image to Y'CbCr (usually with subsampled chroma), split each channel into 8x8 blocks, DCT them, quantize and then use Huffman coding to encode those coefficients. This is a very standard structure, and JPEG, despite being really old by now, is actually hard to beat, so let's use that as a template. But we probably can't use JPEG wholesale. Why? The answer is that the Huffman coding is just too serial. It's all just one stream, and without knowing where the previous symbol ends, we don't know where to start decoding the next. You can partially solve this by having frequent restart markers, but these reduce coding efficiency, and there's no index of them; you'd have to make a parallel scan for them, which is annoying. So we'll need to at least do something about the entropy coding. Do we need to change anything else to reach our ~1000 parallel target? 720p is the standard video target these days; the ~1M raw pixels would be great if we could do all of them independently, but we obviously can't, since DCT is not independent per pixel (that would sort of defeat the purpose). However, there are 14,400 DCT blocks (8x8), so we can certainly do all the DCTs in parallel. Quantization after that is trivially parallelizable, at least as long as we don't aim for trellis or the likes. So indeed, it's only entropy coding that we need to worry about. Since we're changing entropy coding anyway, I picked rANS, which is confusing but has a number of neat properties; it's roughly as expensive as Huffman coding, but has an efficiency closer to that of arithmetic encoding, and you can switch distributions for each symbol. It requires a bit more calculation, but GPUs have plenty of ALUs (a typical recommendation is 10:1 calculation to memory access), so that should be fine. I picked a pretty common variation with 32-bit state and 8-bit I/O, since 64-bit arithmetic is not universally available on GPUs (I checked a variation with 64-bit state and 32-bit I/O, but it didn't really matter much for speed). Fabian Giesen has described how you can actually do rANS encoding and decoding in parallel over a warp, but I've treated it as a purely serial operation. I don't do anything like adaptation, though; each coefficient is assigned to one out of four rANS static distributions that are signaled at the start of each stream. (Four is a nice tradeoff between coding efficiency, L1 cache use and the cost of transmitting the distributions themselves. I originally used eight, but a bit of clustering showed it was possible to get it down to four at basically no extra cost. JPEG does a similar thing, with separate Huffman codes for AC and DC coefficients. And I've got separate distributions for luma and chroma, which also makes a lot of sense.) The restart cost of rANS is basically writing the state to disk; some of that we would need to write even without a restart, and there are ways to combine many such end states for less waste (I'm not using them), but let's be conservative and assume we waste all of it for each restart. At 150 Mbit/second for 720p60, the luma plane of one frame is about 150 kB. 10% (15 kB) sounds reasonable to sacrifice to restarts, which means we can have about 3750 of them, or one for each 250th pixel or so. (3750 restarts also means 3750 independent entropy coding streams, so we're still well above our 1000 target.) So the structure is more or less clear; we'll DCT the blocks in parallel (since an 8x8 block can be expressed as 8 vertical DCTs and then 8 horizontal DCTs, we can even get 8-way parallelism there), and then encode at least 250 coefficients into an rANS stream. I've chosen to let a stream encompass a single coefficient from each of 320 DCT blocks instead of all coefficients from 5 DCT blocks; it's sort of arbitrary and might be a bit unintuitive, but it feels more natural, especially since you don't need to switch rANS distributions for each coefficient. So that's a lot of rambling for a sort-of abstract format. Next time we'll go into how decoding works. Or encoding. I haven't really made up my mind yet.

Narabu is a new intraframe video codec. You may or may not want to read part 1 first. The GPU, despite being extremely more flexible than it was fifteen years ago, is still a very different beast from your CPU, and not all problems map well to it performance-wise. Thus, before designing a codec, it's useful to know what our platform looks like. A GPU has lots of special functionality for graphics (well, duh), but we'll be concentrating on the compute shader subset in this context, ie., we won't be drawing any polygons. Roughly, a GPU (as I understand it!) is built up about as follows: A GPU contains 1 20 cores; NVIDIA calls them SMs (shader multiprocessors), Intel calls them subslices. (Trivia: A typical mid-range Intel GPU contains two cores, and thus is designated GT2.) One such core usually runs the same program, although on different data; there are exceptions, but typically, if your program can't fill an entire core with parallelism, you're wasting energy. Each core, in addition to tons (thousands!) of registers, also has some shared memory (also called local memory sometimes, although that term is overloaded), typically 32 64 kB, which you can think of in two ways: Either as a sort-of explicit L1 cache, or as a way to communicate internally on a core. Shared memory is a limited, precious resource in many algorithms. Each core/SM/subslice contains about 8 execution units (Intel calls them EUs, NVIDIA/AMD calls them something else) and some memory access logic. These multiplex a bunch of threads (say, 32) and run in a round-robin-ish fashion. This means that a GPU can handle memory stalls much better than a typical CPU, since it has so many streams to pick from; even though each thread runs in-order, it can just kick off an operation and then go to the next thread while the previous one is working. Each execution unit has a bunch of ALUs (typically 16) and executes code in a SIMD fashion. NVIDIA calls these ALUs CUDA cores , AMD calls them stream processors . Unlike on CPU, this SIMD has full scatter/gather support (although sequential access, especially in certain patterns, is much more efficient than random access), lane enable/disable so it can work with conditional code, etc.. The typically fastest operation is a 32-bit float muladd; usually that's single-cycle. GPUs love 32-bit FP code. (In fact, in some GPU languages, you won't even have 8-, 16-bit or 64-bit types. This is annoying, but not the end of the world.) The vectorization is not exposed to the user in typical code (GLSL has some vector types, but they're usually just broken up into scalars, so that's a red herring), although in some programming languages you can get to swizzle the SIMD stuff internally to gain advantage of that (there's also schemes for broadcasting bits by voting etc.). However, it is crucially important to performance; if you have divergence within a warp, this means the GPU needs to execute both sides of the if. So less divergent code is good. Such a SIMD group is called a warp by NVIDIA (I don't know if the others have names for it). NVIDIA has SIMD/warp width always 32; AMD used to be 64 but is now 16. Intel supports 4 32 (the compiler will autoselect based on a bunch of factors), although 16 is the most common. The upshot of all of this is that you need massive amounts of parallelism to be able to get useful performance out of a CPU. A rule of thumb is that if you could have launched about a thousand threads for your problem on CPU, it's a good fit for a GPU, although this is of course just a guideline. There's a ton of APIs available to write compute shaders. There's CUDA (NVIDIA-only, but the dominant player), D3D compute (Windows-only, but multi-vendor), OpenCL (multi-vendor, but highly variable implementation quality), OpenGL compute shaders (all platforms except macOS, which has too old drivers), Metal (Apple-only) and probably some that I forgot. I've chosen to go for OpenGL compute shaders since I already use OpenGL shaders a lot, and this saves on interop issues. CUDA probably is more mature, but my laptop is Intel. :-) No matter which one you choose, the programming model looks very roughly like this pseudocode:

for (size_t workgroup_idx = 0; workgroup_idx < NUM_WORKGROUPS; ++workgroup_idx)     // in parallel over cores
        char shared_mem[REQUESTED_SHARED_MEM];  // private for each workgroup
        for (size_t local_idx = 0; local_idx < WORKGROUP_SIZE; ++local_idx)    // in parallel on each core
                main(workgroup_idx, local_idx, shared_mem);
         
 

except in reality, the indices will be split in x/y/z for your convenience (you control all six dimensions, of course), and if you haven't asked for too much shared memory, the driver can silently make larger workgroups if it helps increase parallelity (this is totally transparent to you). main() doesn't return anything, but you can do reads and writes as you wish; GPUs have large amounts of memory these days, and staggering amounts of memory bandwidth. Now for the bad part: Generally, you will have no debuggers, no way of logging and no real profilers (if you're lucky, you can get to know how long each compute shader invocation takes, but not what takes time within the shader itself). Especially the latter is maddening; the only real recourse you have is some timers, and then placing timer probes or trying to comment out sections of your code to see if something goes faster. If you don't get the answers you're looking for, forget printf you need to set up a separate buffer, write some numbers into it and pull that buffer down to the GPU. Profilers are an essential part of optimization, and I had really hoped the world would be more mature here by now. Even CUDA doesn't give you all that much insight sometimes I wonder if all of this is because GPU drivers and architectures are meant to be shrouded in mystery for competitiveness reasons, but I'm honestly not sure. So that's it for a crash course in GPU architecture. Next time, we'll start looking at the Narabu codec itself.

Narabu is a new intraframe video codec, from the Japanese verb narabu ( ), which means to line up or be parallel. Let me first state straight up that Narabu isn't where I hoped it would be at this stage; the encoder isn't fast enough, and I have to turn my attention to other projects for a while. Nevertheless, I think it is interesting as a research project in its own right, and I don't think it should stop me from trying to write up a small series. :-) In the spirit of Leslie Lamport, I'll be starting off with describing what problem I was trying to solve, which will hopefully make the design decisions a lot clearer. Subsequent posts will dive into background information and then finally Narabu itself. I want a codec to send signals between different instances of Nageru, my free software video mixer, and also longer-term between other software, such as recording or playout. The reason is pretty obvious for any sort of complex configuration; if you are doing e.g. both a stream mix and a bigscreen mix, they will naturally want to use many of the same sources, and sharing them over a single GigE connection might be easier than getting SDI repeaters/splitters, especially when you have a lot of them. (Also, in some cases, you might want to share synthetic signals, such as graphics, that never existed on SDI in the first place.) This naturally leads to the following demands:

• Intraframe-only; every frame must be compressed independently. (This isn't strictly needed for all use cases, but is much more flexible, and common in any kind of broadcast.)
• Need to handle 4:2:2 color, since that's what most capture sources give out, and we want to transmit the raw signals as much as possible. Fairly flexible in input resolution (divisible by 16 is okay, limited to only a given set of resolutions is not).
• 720p60 video in less than one CPU core (ideally much less); the CPU can already pretty be busy with other work, like x264 encoding of the finished stream, and sharing four more inputs at the same time is pretty common. What matters is mostly a single encode+decode cycle, so fast decode doesn't help if the encoder is too slow.
• Target bitrates around 100-150 Mbit/sec, at similar quality to MJPEG (ie. 45 dB PSNR for most content). Multiple signals should fit into a normal GigE link at the same time, although getting it to work over 802.11 isn't a big priority.
• Both encoder and decoder robust to corrupted or malicious data; a dropped frame is fine, a crash is not.
• Does not depend on uncommon or expensive hardware, or GPUs from a specific manufacturer.
• GPLv3-compatible implementation. I already link to GPLv3 software, so I don't have a choice here; I cannot link to something non-free (and no antics with dlopen(), please).

There's a bunch of intraframe formats around. The most obvious thing to do would be to use Intel Quick Sync to produce H.264 (intraframe H.264 blows basically everything else out of the sky in terms of PSNR, and QSV hardly uses any power at all), but sadly, that's limited to 4:2:0. I thought about encoding the three color planes as three different monochrome streams, but monochrome is not supported either. Then there's a host of software solutions. x264 can do 4:2:2, but even on ultrafast, it gobbles up an entire core or more at 720p60 at the target bitrates (mostly in entropy coding). FFmpeg has implementations of all kinds of other codecs, like DNxHD, CineForm, MJPEG and so on, but they all use much more CPU for encoding than the target. NDI would seem to fit the bill exactly, but fails the licensing check, and also isn't robust to corrupted or malicious data. (That, and their claims about video quality are dramatically overblown for any kinds of real video data I've tried.) So, sadly, this leaves only really one choice, namely rolling my own. I quickly figured I couldn't beat the world on CPU video codec speed, and didn't really want to spend my life optimizing AVX2 DCTs anyway, so again, the GPU will come to our rescue in the form of compute shaders. (There are some other GPU codecs out there, but all that I've found depend on CUDA, so they are NVIDIA-only, which I'm not prepared to commit to.) Of course, the GPU is quite busy in Nageru, but if one can make an efficient enough codec that one stream can work at only 5% or so of the GPU (meaning 1200 fps or so), it wouldn't really make a dent. (As a spoiler, the current Narabu encoder isn't there for 720p60 on my GTX 950, but the decoder is.) In the next post, we'll look a bit at the GPU programming model, and what it means for how our codec needs to look like on the design level.

I'm currently trying to make sense of some still image coding (more details to come at a much later stage!), and for a variety of reasons, I've chosen to use rANS as the entropy coder. However, there's an interesting little detail that I haven't actually seen covered anywhere; maybe it's just because I've missed something, or maybe because it's too blindingly obvious, but I thought I would document what I ended up with anyway. (I had hoped for something even more elegant, but I guess the obvious would have to do.) For those that don't know rANS coding, let me try to handwave it as much as possible. Your state is typically a single word (in my case, a 32-bit word), which is refilled from the input stream as needed. The encoder and decoder works in reverse order; let's just talk about the decoder. Basically it works by looking at the lowest 12 (or whatever) bits of the decoder state, mapping each of those 2^12 slots to a decoded symbol. More common symbols are given more slots, proportionally to the frequency. Let me just write a tiny, tiny example with 2 bits and three symbols instead, giving four slots:

Lowest bits	Symbol
00	0
01	0
10	1
11	2

Note that the zero coefficient here maps to one out of two slots (ie., a range); you don't choose which one yourself, the encoder stashes some information in there (which is used to recover the next control word once you know which symbol there is). Now for the actual problem: When storing DCT coefficients, we typically want to also store a sign (ie., not just 1 or 2, but also -1/+1 and -2/+2). The statistical distribution is symmetrical, so the sign bit is incompressible (except that of course there's no sign bit needed for 0). We could have done this by introducing new symbols -1 and -2 in addition to our three other ones, but this means we'll need more bits of precision, and accordingly larger look-up tables (which is negative for performance). So let's find something better. We could also simply store it separately somehow; if the coefficient is non-zero, store the bits in some separate repository. Perhaps more elegantly, you can encode a second symbol in the rANS stream with probability 1/2, but this is more expensive computationally. But both of these have the problem that they're divergent in terms of control flow; nonzero coefficients potentially need to do a lot of extra computation and even loads. This isn't nice for SIMD, and it's not nice for GPU. It's generally not really nice. The solution I ended up with was simulating a larger table with a smaller one. Simply rotate the table so that the zero symbol has the top slots instead of the bottom slots, and then replicate the rest of the table. For instance, take this new table:

Lowest bits	Symbol
000	1
001	2
010	0
011	0
100	0
101	0
110	-1
111	-2

(The observant reader will note that this doesn't describe the exact same distribution as last time zero has twice the relative frequency as in the other table but ignore that for the time being.) In this case, the bottom half of the table doesn't actually need to be stored! We know that if the three bottom bits are >= 110 (6 in decimal), we have a negative value, can subtract 6, and then continue decoding. If we are go past the end of our 2-bit table despite that, we know we are decoding a zero coefficient (which doesn't have a sign), so we can just clamp the read; or for a GPU, reads out-of-bounds on a texture will typically return 0 anyway. So it all works nicely, and the divergent I/O is gone. If this pickled your interest, you probably want to read up on rANS in general; Fabian Giesen (aka ryg) has some notes that work as a good starting point, but beware; some of this is pretty confusing. :-)

On releasing modified versions of GPLv3 software in binary form only (quote anonymized):

And in my opinion it's perfectly ok to give out a binary release of a project, that is a work in progress, so that people can try it out and coment on it. It's easier for them to have it as binary and not need to compile it themselfs. If then after a (long) while the code is still only released in binary form, then it's ok to start a discussion. But only for a quick test, that is unneccessary. So people, calm down and enjoy life!

I wonder at what point we got here.

Post-Solskogen, there hasn't been all that many commits in the main Nageru repository, but that doesn't mean the project is standing still. In particular, I've been working with NVIDIA to shake out a crash bug in their drivers (which in itself uncovered some interesting debugging techniques, although in the end, the bug turned out just to be uncovered by the boring standard technique of analyzing crash dumps and writing a minimal program to reproduce). But I've also been looking at intraframe codecs; my sort-of plan was to go to VideoLAN Dev Days to present my findings, but unfortunately, there seems to be a schedule conflict, so instead, you can have some scattered random notes:

• JPEG is really impressive technology. It's made in 1992, and it's still frighteningly close to state of the art! Beating it without getting a lot slower isn't trivial at all; witness that even with WebP etc., we still don't have a JPEG-killer because it's just so incredibly good. (Granted, in 1992, decompressing a JPEG would easily take half a minute or more.)
• rANS is pretty neat; it gives you almost all the advantages of arithmetic coding, but does without divisions in the decoder (and if you have static probabilities and fast high multiplies, also in the encoder). I still don't feel like I understand it fully, though. And it has the odd property of having to encode and decode in different directions, which may or may not be a problem to you. But ryg's interleaving tricks to get SIMD are pretty nice; I won't claim to fully understand those either. Maybe I will need to eventually.
• GPU performance models continue to bend my head. GPUs may have managed to make a parallel programming model that people actually manage to use, but getting really high performance out of compute shaders is still hard to get use to.
• It really annoys me that Quick Sync doesn't support luma-only encoding (ie. 4:0:0); the best you can do is seemingly 4:2:0 and then have empty chroma planes, which wastes resources. It would have been a nice way to hack around the limitation that you can't have 4:2:2 or alpha.
• Time-frequency switching is the solution to a problem that sounds trivial if you're a DSP beginner and nearly impossible if you've actually done some DSP. I'm not intuitively convinced it's worth it if you have fast float muladds, but it's pretty neat. Allowing multiple block sizes to an intraframe codec has a cost in that you'll need a size decision heuristic, though, which pretty much instantly complicates the encoder.

Work in progress :-) Maybe something more coherent will come out eventually. Edit: Forgot about TF switching!

Dear conference organizers, In this day and age, people stream conferences and other events over the Internet. Most of the Internet happens to be in a different timezone from yours (it's crazy, I know!). This means that if you publish a schedule, please say which timezone it's in. We've even got this thing called JavaScript now, which allows you to also convert times to the user's local timezone (the future is now!), so you might want to consider using it. :-) (Yes, this goes for you, DebConf, and also for you, Assembly.)

Solskogen 2017 is over! What a blast that was; I especially enjoyed that so many old-timers came back to visit, it really made the party for me. This was the first year we were using Nageru for not only the stream but also for the bigscreen mix, and I was very relieved to see the lack of problems; I've had nightmares about crashes with 150+ people watching (plus 200-ish more on stream), but there were no crashes and hardly a dropped frame. The transition to a real mixing solution as well as from HDMI to SDI everywhere gave us a lot of new opportunities, which allowed a number of creative setups, some of them cobbled together on-the-spot:

• Nageru with two cameras, of which one was through an HDMI-to-SDI converter battery-powered from a 20000 mAh powerbank (and sent through three extended SDI cables in series): Live music compo (with some, er, interesting entries).
• 1080p60 bigscreen Nageru with two computer inputs (one of them through a scaler) and CasparCG graphics run from an SQL database, sent on to a 720p60 mixer Nageru (SDI pass-through from the bigscreen) with two cameras mixed in: Live graphics compo
• Bigscreen Nageru switching from 1080p50 to 1080p60 live (and stream between 720p50 and 720p60 correspondingly), running C64 inputs from the Framemeister scaler: combined intro compo
• And finally, it's Nageru all the way down: A camera run through a long extended SDI cable to a laptop running Nageru, streamed over TCP to a computer running VLC, input over SDI to bigscreen Nageru and sent on to streamer Nageru: Outdoor DJ set/street basket compo (granted, that one didn't run entirely smoothly, and you can occasionally see Windows device popups :-) )

It's been a lot of fun, but also a lot of work. And work will continue for an even better show next year after some sleep. :-)

I've released version 1.6.1 of Nageru, my live video mixer. Now that Solskogen is coming up, there's been a lot of activity on the Nageru front, but hopefully everything is actually coming together now. Testing has been good, but we'll see whether it stands up to the battle-hardening of the real world or not. Hopefully I won't be needing any last-minute patches. :-) Besides the previously promised Prometheus metrics (1.6.1 ships with a rather extensive set, as well as an example Grafana dashboard) and frame queue management improvements, a surprising late addition was that of a new transcoder called Kaeru (following the naming style of Nageru itself, from the japanese verb kaeru ( ) which means roughly to replace or excahnge iKnow! claims it can also mean convert , but I haven't seen support for this anywhere else). Normally, when I do streams, I just let Nageru do its thing and send out a single 720p60 stream (occasionally 1080p), usually around 5 Mbit/sec; less than that doesn't really give good enough quality for the high-movement scenarios I'm after. But Solskogen is different in that there's a pretty diverse audience when it comes to networking conditions; even though I have a few mirrors spread around the world (and some JavaScript to automatically pick the fastest one; DNS round-robin is really quite useless here!), not all viewers can sustain such a bitrate. Thus, there's also a 480p variant with around 1 Mbit/sec or so, and it needs to come from somewhere. Traditionally, I've been using VLC for this, but streaming is really a niche thing for VLC. I've been told it will be an increased focus for 4.0 now that 3.0 is getting out the door, but over the last few years, there's been a constant trickle of little issues that have been breaking my transcoding pipeline. My solution for this was to simply never update VLC, but now that I'm up to stretch, this didn't really work anymore, and I'd been toying around with the idea of making a standalone transcoder for a while. (You'd ask why not the ffmpeg(1) command-line client? , but it's a bit too centered around files and not streams; I use it for converting to HLS for iOS devices, but it has a nasty habit of I/O blocking real work, and its HTTP server really isn't meant for production work. I could survive the latter if it supported Metacube and I could feed it into Cubemap, but it doesn't.) It turned out Nageru had already grown most of the pieces I needed; it had video decoding through FFmpeg, x264 encoding with speed control (so that it automatically picks the best preset the machine can sustain at any given time) and muxing, audio encoding, proper threading everywhere, and a usable HTTP server that could output Metacube. All that was required was to add audio decoding to the FFmpeg input, and then replace the GPU-based mixer and GUI with a very simple driver that just connects the decoders to the encoders. (This means it runs fine on a headless server with no GPU, but it also means you'll get FFmpeg's scaling, which isn't as pretty or fast as Nageru's. I think it's an okay tradeoff.) All in all, this was only about 250 lines of delta, which pales compared to the ~28000 lines of delta that are between 1.3.1 (used for last Solskogen) and 1.6.1. It only supports a rather limited set of Prometheus metrics, and it has some limitations, but it seems to be stable and deliver pretty good quality. I've denoted it experimental for now, but overall, I'm quite happy with how it turned out, and I'll be using it for Solskogen. Nageru 1.6.1 is on its way into Debian, but it depends on a new version of Movit which needs to go through the NEW queue (a soname bump), so it might be a few days. In the meantime, I'll be busy preparing for Solskogen. :-)

Nageru 1.6.1 is on its way, and what was intended to only be a release centered around monitoring improvements (more specifically a full set of native Prometheus] metrics) actually ended up getting a fairly substantial change to how Nageru manages its frame queues. To understand what's changing and why, it's useful to first understand the history of Nageru's queue management. Nageru 1.0.0 started out with a fairly simple scheme, but with some basics that are still relevant today: One of the input cards was deemed the master card, and whenever it delivers a frame, the master clock ticks and an output frame is produced. (There are some subtleties about dropped frames and/or the master card changing frame rates, but I'm going to ignore them, since they're not important to the discussion.) To this end, every card keeps a preallocated frame queue; when a card delivers a frame, it's put into the queue, and when the master clock ticks, it tries picking out one frame from each of the other card's queues to mix together. Note that mix here could be as simple as picking one input and throwing all the other ones away; the queueing algorithm doesn't care, it just feeds all of them to the theme and lets that run whatever GPU code it needs to match the user's preferences. The only thing that really keeps the queues bounded is that the frames in them are preallocated (in GPU memory), so if one queue gets longer than 16 frames, Nageru starts dropping it. But is 16 the right number? There are two conflicting demands here, ignoring memory usage:

• You want to keep the latency down.
• You don't want to run out of frames in the queue if you can avoid it; if you drop too aggressively, you could find yourself at the next frame with nothing in the queue, because the input card hasn't delivered it yet when the master card ticks. (You could argue one should delay the output in this case, but for how long? And if you're using HDMI/SDI output, you have no such luxury.)

The 1.0.0 scheme does about as well as one could possibly hope in never dropping frames, but unfortunately, it can be pretty poor at latency. For instance, if your master card runs at 50 Hz and you have a 60 Hz card, the latter will eventually build up a delay of 16 * 16.7 ms = 266.7 ms clearly unacceptable, and rather unneeded. You could ask the user to specify a queue length, but the user probably doesn't know, and also shouldn't really have to care more knobs to twiddle are a bad thing, and even more so knobs the user is expected to twiddle. Thus, Nageru 1.2.0 introduced queue autotuning; it keeps a running estimate on how big the queue needs to be to avoid underruns, simply based on experience. If we've been dropping frames on a queue and then there's an underrun, the safe queue length is increased by one, and if the queue has been having excess frames for more than a thousand successive master clock ticks, we reduce it by one again. Whenever the queue has more than this safe number, we drop frames. This was simple, effective and largely fixed the problem. However, when adding metrics, I noticed a peculiar effect: Not all of my devices have equally good clocks. In particular, when setting up for 1080p50, my output card's internal clock (which assumes the role of the master clock when using HDMI/SDI output) seems to tick at about 49.9998 Hz, and my simple home camcorder delivers frames at about 49.9995 Hz. Over the course of an hour, this means it produces one more frame than you should have which should of course be dropped. Having an SDI setup with synchronized clocks (blackburst/tri-level) would of course fix this problem, but most people are not so lucky with their cameras, not to mention the price of PC graphics cards with SDI outputs! However, this happens very slowly, which means that for a significant amount of time, the two clocks will very nearly be in sync, and thus racing. Who ticks first is determined largely by luck in the jitter (normal is maybe 1ms, but occasionally, you'll see delayed delivery of as much as 10 ms), and this means that the 1000 frames estimate is likely to be thrown off, and the result is hundreds of dropped frames and underruns in that period. Once the clocks have diverged enough again, you're off the hook, but again, this isn't a good place to be. Thus, Nageru 1.6.1 change the algorithm around yet again, by incorporating more data to build an explicit jitter model. 1.5.0 was already timestamping each frame to be able to measure end-to-end latency precisely (now also exposed in Prometheus metrics), but from 1.6.1, they are actually used in the queueing algorithm. I ran several eight- to twelve-hour tests and simply stored all the event arrivals to a file, and then simulated a few different algorithms (including the old algorithm) to see how they fared in measures such as latency and number of drops/underruns. I won't go into the full details of the new queueing algorithm (see the commit if you're interested), but the gist is: Based on the last 5000 frames, it tries to estimate the maximum possible jitter for each input (ie., how late the frame could possibly be). Based on this as well as clock offsets, it determines whether it's really sure that there will be an input frame available on the next master tick even if it drops the queue, and then trims the queue to fit. The result is pretty satisfying; here's the end-to-end latency of my camera being sent through to the SDI output: As you can see, the latency goes up, up, up until Nageru figures it's now safe to drop a frame, and then does it in one clean drop event; no more hundreds on drops involved. There are very late frame arrivals involved in this run two extra frame drops, to be precise but the algorithm simply determines immediately that they are outliers, and drops them without letting them linger in the queue. (Immediate dropping is usually preferred to sticking around for a bit and then dropping it later, as it means you only get one disturbance event in your stream as opposed to two. Of course, you can only do it if you're reasonably sure it won't lead to more underruns later.) Nageru 1.6.1 will ship before Solskogen, as I intend to run it there :-) And there will probably be lovely premade Grafana dashboards from the Prometheus data. Although it would have been a lot nicer if Grafana were more packaging-friendly, so I could pick it up from stock Debian and run it on armhf. Hrmf. :-)

Bem vindo ao Debian Stretch! Ontem, 17 de junho de 2017, o Debian 9 (Stretch) foi lan ado. Eu gostaria de falar sobre alguns procedimentos b sicos e regras para migrar do Debian 8 (Jessie). Passos iniciais

• A primeira coisa a fazer ler a nota de lan amento. Isso fundamental para saber sobre poss veis bugs e situa es especiais.
• O segundo passo atualizar o Jessie totalmente antes de migrar para o Stretch. Para isso, ainda dentro do Debian 8, execute os seguintes comandos:

# apt-get update
# apt-get dist-upgrade

Migrando

• Edite o arquivo /etc/apt/sources.list e altere todos os nomes jessie para stretch. A seguir, um exemplo do conte do desse arquivo (poder variar, de acordo com as suas necessidades):

deb http://ftp.br.debian.org/debian/ stretch main
deb-src http://ftp.br.debian.org/debian/ stretch main
   
deb http://security.debian.org/ stretch/updates main
deb-src http://security.debian.org/ stretch/updates main

• Depois, execute:

# apt-get update
# apt-get dist-upgrade

Caso haja algum problema, leia as mensagens de erro e tente resolver o problema. Resolvendo ou n o tal problema, execute novamente o comando:

# apt-get dist-upgrade

Havendo novos problemas, tente resolver. Busque solu es no Google, se for necess rio. Mas, geralmente, tudo dar certo e voc n o dever ter problemas. Altera es em arquivos de configura o Quando voc estiver migrando, algumas mensagens sobre altera es em arquivos de configura o poder o ser mostradas. Isso poder deixar alguns usu rios pedidos, sem saber o que fazer. N o entre em p nico. Existem duas formas de apresentar essas mensagens: via texto puro em shell ou via janela azul de mensagens. O texto a seguir um exemplo de mensagem em shell:

Ficheiro de configura o '/etc/rsyslog.conf'
 ==> Modificado (por si ou por um script) desde a instala o.
 ==> O distribuidor do pacote lan ou uma vers o atualizada.
 O que deseja fazer? As suas op es s o:
 Y ou I : instalar a vers o do pacote do maintainer
 N ou O : manter a vers o actualmente instalada
 D : mostrar diferen as entre as vers es
 Z : iniciar uma shell para examinar a situa o
 A a o padr o   manter sua vers o atual.
*** rsyslog.conf (Y/I/N/O/D/Z) [padr o=N] ?

A tela a seguir um exemplo de mensagem via janela: Nos dois casos, recomend vel que voc escolha por instalar a nova vers o do arquivo de configura o. Isso porque o novo arquivo de configura o estar totalmente adaptado aos novos servi os instalados e poder ter muitas op es novas ou diferentes. Mas n o se preocupe, pois as suas configura es n o ser o perdidas. Haver um backup das mesmas. Assim, para shell, escolha a op o "Y" e, no caso de janela, escolha a op o "instalar a vers o do mantenedor do pacote". muito importante anotar o nome de cada arquivo modificado. No caso da janela anterior, trata-se do arquivo /etc/samba/smb.conf. No caso do shell o arquivo foi o /etc/rsyslog.conf. Depois de completar a migra o, voc poder ver o novo arquivo de configura o e o original. Caso o novo arquivo tenha sido instalado ap s uma escolha via shell, o arquivo original (o que voc tinha anteriormente) ter o mesmo nome com a extens o .dpkg-old. No caso de escolha via janela, o arquivo ser mantido com a extens o .ucf-old. Nos dois casos, voc poder ver as modifica es feitas e reconfigurar o seu novo arquivo de acordo com as necessidades. Caso voc precise de ajuda para ver as diferen as entre os arquivos, voc poder usar o comando diff para compar -los. Fa a o diff sempre do arquivo novo para o original. como se voc quisesse ver como fazer com o novo arquivo para ficar igual ao original. Exemplo:

# diff -Naur /etc/rsyslog.conf /etc/rsyslog.conf.dpkg-old

Em uma primeira vista, as linhas marcadas com "+" dever o ser adicionadas ao novo arquivo para que se pare a com o anterior, assim como as marcadas com "-" dever o ser suprimidas. Mas cuidado: normal que haja algumas linhas diferentes, pois o arquivo de configura o foi feito para uma nova vers o do servi o ou aplicativo ao qual ele pertence. Assim, altere somente as linhas que realmente s o necess rias e que voc mudou no arquivo anterior. Veja o exemplo:

+daemon.*;mail.*;\
+ news.err;\
+ *.=debug;*.=info;\
+ *.=notice;*.=warn  /dev/xconsole
+*.* @sam

No meu caso, originalmente, eu s alterei a ltima linha. Ent o, no novo arquivo de configura o, s terei interesse em adicionar essa linha. Bem, se foi voc quem fez a configura o anterior, voc saber fazer a coisa certa. Geralmente, n o haver muitas diferen as entre os arquivos. Outra op o para ver as diferen as entre arquivos o comando mcdiff, que poder ser fornecido pelo pacote mc. Exemplo:

# mcdiff /etc/rsyslog.conf /etc/rsyslog.conf.dpkg-old

Problemas com ambientes e aplica es gr ficas poss vel que voc tenha algum problema com o funcionamento de ambientes gr ficos, como Gnome, KDE etc, ou com aplica es como o Mozilla Firefox. Nesses casos, prov vel que o problema seja os arquivos de configura o desses elementos, existentes no diret rio home do usu rio. Para verificar, crie um novo usu rio no Debian e teste com ele. Se tudo der certo, fa a um backup das configura es anteriores (ou renomeie as mesmas) e deixe que a aplica o crie uma configura o nova. Por exemplo, para o Mozilla Firefox, v ao diret rio home do usu rio e, com o Firefox fechado, renomeie o diret rio .mozilla para .mozilla.bak, inicie o Firefox e teste. Est inseguro? Caso voc esteja muito inseguro, instale um Debian 8, com ambiente gr fico e outras coisas, em uma m quina virtual e migre para Debian 9 para testar e aprender. Sugiro VirtualBox como virtualizador. Divirta-se!

I've released version 1.6.0 of Nageru, my live video mixer, together with dependent libraries Movit 1.5.1 and bmusb 0.7.0. The primray new feature this time is integration with CasparCG, the dominating open-source broadcast graphics system, which opens up a whole new world of possibilities for intelligent overlay graphics. (Actually, the feature is a bit more generic than that; any FFmpeg file or stream will do as input. Audio isn't supported yet, though.) You can see a simple HTML5 CasparCG setup in the ultimate tournament stream test we did in April, in preparation of a larger event in September; CasparCG generates a stream with alpha, which is then fed into Nageru and used on top of the three camera sources. Apart from that, there's a new frame analyzer that helps with calibrating your signal chain; there are lots of devices that will happily mess with your signal, and measuring is the first step in counteracting that. (There's also a few input interpretation tweaks that will help most common issues.) 1.6.0 is on its way to Debian experimental, along with its dependencies (stretch will release with Nageru 1.4.2); there are likely to be backports when stretch releases and the backport queue opens up.

The last week, I found no less than three pet bugs I have hope that will be allowed to go in before stretch release:

• #863286: lua-http: completely broken in non-US locales
• #843448: linux-image-4.8.0-1-armmp-lpae: fails to boot on Odroid-Xu4 with rootfs on USB (actually my problem is that the NIC doesn't work, but same root cause this makes stretch basically unusable on XU4)
• #863280: cubemap: streams with paths exactly 7 characters long get broken buffering behavior

I promise, none of these were found late because I upgraded to stretch too late just a perfect storm. :-)

Following up on my previous post, I have only a few small nuggets of extra information: The power draw appears to be 230 240 mA at 5 V, or about 1.15 W. (It doesn't matter whether they have a signal or not.) This means you can power them off of a regular USB power bank; in a recent test, we used Deltaco 20000 mAh (at 3.7 V) power banks, which are supposed to power a GoPro plus such a converter for somewhere between 8 12 hours (we haven't measured exactly). It worked brilliantly, and solved a headache of how to get AC to the camera and converter; just slap on a power bank instead, and all you need to run is SDI. Curiously enough, 230 mA is so little that the power bank thinks it doesn't count as load, and just turns itself off after ten seconds or so. However, with the GoPro also active, it stays on all the time. At least it ran for the two hours that we needed without a hitch. The SDI converters don't accept USB directly, but you can purchase dumb USB-to-5.5x2.1mm converters cheap from wherever, which works fine even though USB is supposed to give you only 100 mA without handshaking. Some eBay sellers seem to include them with the converters, even. I guess the power bank just doesn't care; it's spec-ed to 2.1 A on two ports and 1 A on the last one. Update: Sebastian Reichel pointed out that USB 2.0 ups the limit to 500 mA, so you should be within spec in any case. :-) And here's a picture of the entire contraption as a bonus:

I often need to convert signals from HDMI to SDI (and occasionally back). This requires a box of some sort, and eBay obliges; there's a bunch of different sellers of the same devices, selling them around $20 25. They don't seem to have a brand name, but they are invariably sold as 3G-SDI converters (meaning they should go up to 1080p60) and look like this: There are also corresponding SDI-to-HDMI converters that look pretty much the same except they convert the other way. (They're easy to confuse, but that's not a problem unique tothem.) I've used them for a while now, and there are pros and cons. They seem reliable enough, and they're 1/4th the price of e.g. Blackmagic's Micro converters, which is a real bargain. However, there are also some issues:

• For 3G-SDI, they output level A only, with no option for level B. (In fact, there are no options at all.) Level A is the most sane standard, and also what most equipment uses, but there exists certain older equipment that only works with level B.
• They don't have reclocking chips, so their timing accuracy is not superb. I managed to borrow a Phabrix SDI analyzer and measured the jitter; with a very short cable, I got approximately 0.85 0.95 UI (varying a bit), whereas a Blackmagic converter gave me 0.23 0.24 UI (much more stable). This may be a problem at very long cable lengths, although I haven't tried 100m runs and such.
• When converting to and from RGB, they seem to assume Rec. 601 Y'CbCr coefficients even for HD resolutions. This means the colors will be a bit off in some cases, although for most people, it will be hard to notice without looking at it side-by-side. (I believe the HDMI-to-SDI and SDI-to-HDMI converters make the same mistake, so that the errors cancel out if you just want to use a pair as HDMI extenders. Also, if your signal is Y'CbCr already, you don't need to care.)
• They don't insert SMPTE 352M payload ID. (Supposedly, this is because the SDI chip they use, called GV7600, is slightly out-of-standard on purpose in order to avoid paying expensive licensing fees to SMPTE.) Normally, you wouldn't need to care, but 3G-SDI actually requires this, and worse; Blackmagic's equipment (at least the Duo 2, and I've seen reports about the ATEMs as well) enforces that. If you try to run e.g. 1080p50 through them and into a Duo 2, it will be misdetected as 1080p25, no signal . There's no workaround that I know of.

The last issue is by far the worst, but it only affects 3G-SDI resolutions. 720p60, 1080p30 and 1080i60 all work fine. And to be fair, not even Blackmagic's own converters actually send 352M correctly most of the time I wish there were a way I could publish this somewhere people would actually read it before buying these things, but without a name, it's hard for people to find it. They're great value for money, and I wouldn't hesitate to recommend them for almost all use but then, there's that almost. :-)