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April 2013

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ffmpeg-devel.log.20130424
by burek 25 Apr '13

25 Apr '13

[00:30] <ubitux> yepee i almost have my 2d fft api with ffmpeg [00:30] <ubitux> i guess i'll be able to write some cool filters soon [02:09] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:fc6903337188: avcodec/x86/sbrdsp_init: disable using the noise code in x86_64 MSVC, Try #2 [03:02] <ubitux> advise number #1: never ever use only the low 8-bits of a prng to generate a data stream [03:03] <ubitux> this shit is cycling relatively fast, and thus was fucking my fft results... and of course i thought it was my fft code, while it was just a redundancy in the random i wasn't able to notice at first glance [03:03] <ubitux> best bug ever. [03:04] <ubitux> (i had zero bands in my fft results, dafuck) [03:14] <ubitux> anyway, here is my naive implementation: https://github.com/ubitux/FFmpeg/compare/fft2d-experiments#L2R75 [03:14] <ubitux> the branch include the exact fftw equivalent, results are the same, tested with mismatching w/h [03:15] Action: ubitux can now start writing funny filters [03:57] <Compn> ubitux : bright flash remover filter ? [03:57] <Compn> for the poor epileptics [04:33] <highgod> Hi, I want to ask a question, is there any filters in ffmpeg can tailor the edge of video [10:02] <cone-797> ffmpeg.git 03Nicolas Bertrand 07master:2e2d24667a2c: jpeg2000: Add mutlti-threading support to decoder [10:02] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:03b81df9e673: Merge commit '2e2d24667a2c5cc7af94cfa7b63bb9ec1c04224b' [10:12] <cone-797> ffmpeg.git 03Martin Storsjö 07master:6a8561dbd7c0: x86: Factorize duplicated inline assembly snippets [10:12] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:c2a0833c0962: Merge commit '6a8561dbd7c078eb75985f7011ad1ad3fda9e223' [10:17] <cone-797> ffmpeg.git 03Hendrik Leppkes 07master:feec9349d35b: mpegvideo: unref cur/next/prev frames when flushing [10:17] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:8d86cf4cfa2e: Merge commit 'feec9349d35b3a46d0c6a05e3b23626050b76a77' [10:24] <cone-797> ffmpeg.git 03Martin Storsjö 07master:486f76f0298a: x86: Get rid of duplication between *_rnd_template.c [10:24] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:2e789d165b2b: Merge remote-tracking branch 'qatar/master' [10:34] <ubitux> highgod: vf crop? [10:34] <ubitux> unless you mean something else? [10:34] <ubitux> Compn: nope [10:36] <highgod> thanks [10:37] <highgod> I will try it [10:47] <highgod> @ubitux:yes, this filter, thanks [11:02] <highgod> Hi, I want to ask a quesion, can I use ffprobe to get how long time is the video [11:22] <ubitux> highgod: -show_format [11:23] <ubitux> highgod: and you can play with the different ffprobe output (xml, csv, json, ...) [11:24] <ubitux> you can also just select the field [11:36] <highgod> OK, thanks [12:58] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:62d944539683: vc1: simplify code use INIT_LUT() [13:04] <ubitux> cvslog dead? [13:05] <ubitux> j-b: it's not because Skyler_ tried to mailbomb everyone yesterday with x264 that you had to disable ffmpeg cvslog :( [13:10] <ubitux> oh, finally sent :) [14:07] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:1d0f817b17d7: vc1dec: Fix mv_f shuffling [14:07] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:3fdd0979acd3: vc1dec: Fix mv_f shuffling [14:10] Action: michaelni hides in the corner for failing to realize the 2 commits wherent stashed together as they where supposed to [14:12] Action: av500 hands michaelni the Narrenkappe [14:13] <durandal_1707> michaelni: do your work in repo and ask me to merge it when its finished [14:13] <ubitux> :D [14:16] <michaelni> durandal_1707, iam happy to send my changes to the vc1 maintainer for merging in his tree, now if only that wasnt defacto myself :( [14:55] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:72e5d9190843: vc1dec: factorize picture pointer selection code [15:08] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:236b0c408426: vc1dec: factorize picture pointer selection in vc1_mc_4mv_chroma() [15:08] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:c5669f3c5996: vc1dec: fix current ptr selection in vc1_mc_4mv_chroma() [15:27] <ubitux> http://b.pkh.me/lenafft.png [15:27] <ubitux> i have no idea what i'm doing @_@ [15:28] <durandal_1707> whats that supposed to do? [15:29] <ubitux> i'm just trying to filter the re/im values of each blocks with an eval [15:29] <ubitux> here it's "fft='val*lt(hypot(x,y),0.1)'" [15:29] <ubitux> most likely there is a bug somewhere anyway [15:30] <ubitux> (x and y being block x/y position in [0,1] scale) [15:30] <ubitux> i just pushed the filter in my fft2d-experiments branch experiment, if you're interested [15:34] <ubitux> well maybe that's just because i'm trying to filter them as a circle [15:36] <michaelni> ubitux, block based transforms lead to blocking artifacts [15:37] <ubitux> i guess i need some windowing [15:37] <ubitux> or overlap maybe [15:37] <michaelni> yes overlapp + window is a possible solution [15:37] <ubitux> but i believe my eval formula is wrong anyway [15:37] <ubitux> in several ways :) [15:37] <durandal_1707> what it shoud do? [15:37] <michaelni> using a block bigger than your image is another [15:38] <ubitux> durandal_1707: filter some frequencies :p [15:38] <ubitux> (drop some) [15:38] <michaelni> applying the transform with every block position and averaging is yet another but that is slow [15:39] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:c560437e957c: vc1: make INIT_LUT() self contained [15:53] <ubitux> BuxiNess: is your mxf sample freely available somewhere? [15:56] <BuxiNess> yes and no, yes its free but not avlble yet I'll upload it [15:57] <ubitux> did you try with the threading in ffmpeg if it changes anything? [15:59] <BuxiNess> Not yet [15:59] <BuxiNess> ubitux, waht sample do you mean ? The one i'm working in or the fate one? [16:00] <ubitux> moonrise kingdom sample [16:00] <ubitux> which you used for benching [16:00] <kierank> durandal_1707: did you use a specific test sequence to test pullup? [16:01] <BuxiNess> ubitux, it huge 1.7 Gio. [16:02] <ubitux> BuxiNess: no worry then [16:02] <ubitux> i was just curious [16:04] <durandal_1707> kierank: mp=pullup? [16:04] <kierank> yes [16:04] <kierank> i mean did you use a test clip to check the pullup worked [16:04] <durandal_1707> it does not work for you? [16:05] <kierank> no, i'm just looking for a hard telecined clip [16:05] <kierank> to test vivtc with [16:06] Action: ubitux mainly used http://chromashift.org/kyoani_cm_mechagirl.ts [16:06] <kierank> thanks [16:06] <kierank> hopefully that's sfw [16:06] <ubitux> yes [16:07] <durandal_1707> kierank: telecine,mp=pullup looks fine [16:07] <durandal_1707> did not checked other patterns [16:07] <kierank> oh i was unclear [16:07] <Compn> hopefully it ISNT [16:07] <Compn> aww [16:07] <Compn> cm means commercial [16:07] <Compn> ... usually [16:07] <kierank> durandal_1707: i meant i didn't try pullup [16:07] <kierank> i was just looking for a testclip [16:07] <kierank> sorry [16:08] <ubitux> -vf telecine on any clip then [16:08] <ubitux> :P [16:08] <kierank> yeah but i wanted a real one [16:08] <Compn> we should make a set of telecine / telesync / progressive / interlaced clips [16:09] <durandal_1707> tried blended ones? [16:10] <nevcairiel> wtf is telesync [16:10] <Compn> someone who points a camera at a movie screen and records the image with a direct line-in ? [16:10] <Compn> :P [16:11] <ubitux> screener? [16:11] <Compn> erm records the audio [16:11] <nevcairiel> seems like the odd man out in your list there :p [16:12] <JEEB> ubitux, lol you are still using that commercial :D [16:12] <ubitux> JEEB :D [16:12] <ubitux> i have a few other telecined sources, but not that much [16:12] <JEEB> kyoani's commercials are nice, though [16:12] <ubitux> and i like this one :) [16:12] <JEEB> yeah [16:12] <JEEB> there are a couple of others, too [16:12] <ubitux> feel free to share :) [16:12] <JEEB> lemme see if I can find them around :) [16:13] <JEEB> ...if I have the caps any more [16:13] <JEEB> they usually came in between of kyoani shows [16:13] <JEEB> (pretty much the only company that doesn't have to sell off its ad space) [16:15] <ubitux> what is this commercial for btw? [16:15] <ubitux> a teaser? or just some kind of demo/intro for the company? [16:16] <ubitux> i btw remember seeing a commercial like this one in nichijou [16:16] <ubitux> (the wtf one with the dolphins) [16:17] <JEEB> yeah, it's pretty much an ad for kyoani itself. At some point they made a little note towards kyoani shop (their web shop), but even that commercial was pretty much them demo'ing their skills [16:17] <JEEB> (one with a flying bus driven by a cat) [16:19] <JEEB> https://www.youtube.com/watch?v=gEQbgWp2kAs [16:19] <JEEB> this one [16:19] <ubitux> heh [16:20] <ubitux> cute graphics [16:23] <durandal_1707> cute? thats for kids .... [16:23] <ubitux> durandal_1707: https://www.youtube.com/watch?v=x7NYrwmlqnU ;) [16:26] <JEEB> https://www.youtube.com/watch?v=Pel3lW4ZFBQ <- all of kyoani's ads for spring 2012 [16:28] <ubitux> :) [16:28] <JEEB> and... yeah, it seems like I've removed hyouka, chuu2koi and tamako market caps from my boxes [16:28] <JEEB> but I bet someone archived those :) [16:28] <JEEB> because that ts file I cut off my cap for another archiver, too [16:46] <cone-797> ffmpeg.git 03Nicolas Bertrand 07master:69c02b520bce: MAINTAINERS: add jpeg2000 maintainer [17:51] <cone-797> ffmpeg.git 03Stefano Sabatini 07master:892af3eb47de: doc/filters: split luma/chroma descriptions for unsharp option [17:51] <cone-797> ffmpeg.git 03Stefano Sabatini 07master:18144950d46a: doc/filters: introduce example in vflip filter [17:51] <cone-797> ffmpeg.git 03Stefano Sabatini 07master:6f1716c4b6d4: doc/filters: amend documentation for the hue filter [17:53] <zimbatm> hi [17:54] <zimbatm> what tool do you use to debug the structure of an mp4 container ? [17:54] <zimbatm> apart from the usual ffprobe [17:54] <zimbatm> i'm trying to figure out what is the content of big headers at the start of a file [17:55] <zimbatm> that slow down the start of http streaming [17:55] <nevcairiel> boxdumper (part of l-smash) can give you a detailed dump of the mp4s contents [17:57] <zimbatm> sweet, thx [18:34] <Diogo> hi this is a option to loop a video input using ffmpeg? [18:34] <Diogo> -loop? [18:34] <durandal_1707> not just vide but audio ... [18:47] <cone-797> ffmpeg.git 03Georg Martius 07master:4364e1f15097: lavfi: add video stabilization plugins using vid.stab library [18:47] <cone-797> ffmpeg.git 03Clément BSsch 07master:017051266b37: lavfi/vidstab: miscelenaous cosmetics and typo fixes. [18:47] <cone-797> ffmpeg.git 03Clément BSsch 07master:cdc48775aa8e: lavfi/vidstab: consistent descriptions. [18:48] <cone-797> ffmpeg.git 03Clément BSsch 07master:c857c21a8c1c: doc/filters: reword vidstab options intro. [18:48] <cone-797> ffmpeg.git 03Clément BSsch 07master:9273cc8eb59f: lavfi/vidstab: add missing flags for 2 options. [18:48] <cone-797> ffmpeg.git 03Clément BSsch 07master:111ac10b60cc: lavfi/vidstab: make use of direct variable. [18:48] <cone-797> ffmpeg.git 03Clément BSsch 07master:c82f6e58daf6: lavfi/vidstab: remove mention of defaults for consistency with other filters. [18:48] <cone-797> ffmpeg.git 03Clément BSsch 07master:c7d2a3abf037: lavfi/vidstabdetect: fix direct path. [19:38] <cone-797> ffmpeg.git 03Clément BSsch 07master:1f6aa7de3ecc: lavfi: add vidstabutils.h to SKIPHEADERS. [20:01] <saste> ubitux: time to close https://ffmpeg.org/trac/ffmpeg/ticket/1062? [20:02] <ubitux> oh :) [20:02] <ubitux> let me do that then [20:02] <ubitux> delogo has not the flag though [20:03] <ubitux> let me check if it's possible.. [20:03] <saste> ubitux, that's a separate issue [20:03] <saste> since this is now generically possible [20:03] <ubitux> it looks possible [20:03] <ubitux> there is no frame caching [20:04] <ubitux> or anything from the previous frames [20:04] <ubitux> so i'll add that in a moment and close the ticket [20:04] <saste> ubitux, don't forget removelogo [20:07] <ubitux> saste: are you sure context is constant in removelogo? [20:07] <saste> ubitux, i don't know [20:07] <saste> i think so [20:10] <ubitux> the direct path in remove logo doesn't make much sense :p [20:10] <ubitux> it copies all planes anyway [20:13] <ubitux> it even copies it 2x [20:14] <cone-797> ffmpeg.git 03Clément BSsch 07master:73050df2402c: lavfi: add timeline support in {de,remove}logo. [20:14] <nevcairiel> there are two such filters? [20:16] <ubitux> nevcairiel: one use an external picture as reference [20:16] <ubitux> while the other well... it just seems to blur the area to fake an interpolation [20:17] <durandal_1707> what is worst thing ever invented? [20:18] <ubitux> god [20:18] <saste> the world [20:19] <durandal_1707> planar sample format [20:19] <ubitux> close enough [20:21] <nevcairiel> planar audio makes sense, if most decoders actually decode to this and then interleave, and a encoder deinterleaves before encoding, being able to skip the step seems like a good idea [21:54] <kierank> nevcairiel: +1 [21:55] <durandal11707> tell that to users [21:56] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:782ebd611824: vc1dec: redesign the intensity compensation [21:56] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:84c0ec92ae79: vc1dec: Implement intensity compensation for vc1_interp_mc() [21:56] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:100184ccffbf: vc1: use use_ic instead of shuffling mv_mode around [21:59] <durandal_1707> michaelni: i'm hacking encoder which use put_bits, if put_bits leave uninitialized bytes, ffmpeg encoding becomes very slow and start spamming 0000, enabling assert in put bits does not show anything [22:00] <durandal_1707> something becomes broken, but i dunno what.... [22:25] <michaelni> i cant guess from that information what the problem might be [22:31] <tfoucu> I have the coefficient for ebur128 for different sampling rate, like 44.1khz, 32kHz, etc.. looking at the code, right now, the coef are just some define. What will be the right approach to add them? [22:32] <durandal_1707> tfoucu: table/switch/whatever [22:32] <tfoucu> no preference? cool.. I will look at it then. [22:33] <nevcairiel> should probably talk to the author of that filter, which would be ubitux [22:33] <tfoucu> that's why i came to IRC to see if he was online and has some idea. [22:36] <ubitux> tfoucu: iirc there are some algorithm to compute find the coeff for any freq [22:37] <durandal_1707> michaelni: you can try it, in dcaenc branch [22:37] <ubitux> there was a paper about that, i can find it back if necessary [22:38] <ubitux> also, if you're on this, you can certainly merge the two filters [22:38] <ubitux> i didn't try to optimize this at all; just made a naive implementation of what the specs said [22:38] <tfoucu> I have the paper and i even have the compliant audio 48kHz stream if you need them. [22:39] <tfoucu> ubitux: will you prefer the algorithm to compute them for any freq or just some ahrd coded one? [22:40] <ubitux> i'd prefer the algorithm [22:40] <durandal_1707> its would be nice if filter would support any sample rate, so resampling isn't needed at all [22:40] <ubitux> with the two FILTER() merged into one if possible :) [22:40] <ubitux> durandal_1707: resampling will be needed anyway when i will add the true peaks [22:41] <tfoucu> ok.. Let me find again the algorithm and see if I can add it to the filter. [22:41] <ubitux> cool :) [22:41] <tfoucu> and do you want the reference audio file for Fate? only 48kHz [22:42] <ubitux> ebur128 is covered by fate afaik, with one of the reference 48kHz file [22:42] <tfoucu> They are from ITU BS 2217 [22:42] <ubitux> i used one from the ebu set [22:42] <ubitux> isn't it ok? [22:43] <tfoucu> don;t you need the 5.1 one as well? [22:44] <ubitux> yeah well i picked a small and simple one to not have too big samples [22:44] <ubitux> is it problematic in practice for testing this? [22:46] <tfoucu> the 5.1 sample from the set is 16MB. I guess it is too big for testing. [22:46] <ubitux> even in flac? [22:47] <tfoucu> the one i have is PCM. Let me encode it to FLAC [22:47] <ubitux> the current sample is 4.2M [22:47] <ubitux> the other one was too big [22:47] <ubitux> but i believe it covers a relatively good part of the filter [22:48] <ubitux> eventually, we could take 2-3s from that 5.1 sample if you absolutely want to try some other channel layouts [22:48] <kierank> hi tfoucu [22:49] <kierank> i thought ebur128 was only 48khz [22:49] <tfoucu> the 5.1 in flac will be 2.2MB. [22:49] <ubitux> kierank: yeah the specs is defined for 48khz only [22:49] <ubitux> kierank: but some guys were able to get compute coeff for other frequencies [22:50] <tfoucu> it's just that in 5.1, the weight for each channel is not just 1. so having a test for 5.1 will validate the weight per channel is correct. [22:50] <ubitux> i guess 2.2MB is ok [22:51] <tfoucu> ok.. Let me check how i can had the test. [22:54] <ubitux> ok got the two links [22:54] <ubitux> but he left :( [22:59] <kierank> everything should be resampled to a multiple of 48khz :) [23:00] <kierank> saves a lot of trouble syncing with ntsc [23:01] <durandal_1707> 6khz [23:04] <cone-797> ffmpeg.git 03Michael Niedermayer 07master:12e3c1bee487: vc1dec: drop old use_ic code from vc1_b_mc [23:08] <durandal_1707> michaelni: you compare with reference code or out of head? [23:19] <michaelni> durandal_1707, the code looked wrong, i did look at the reference sw & spec many times today but not for this one [23:20] <durandal_1707> michaelni: i'm just interested in high quality decoder [23:20] <cehoyos> As opposed to what kind of decoder ? [23:22] <durandal_1707> cehoyos: dirac/j2k/... and incomplete ones (much longer list) [23:28] <cehoyos> I believe the dirac decoder works pretty well (at least I don't know of any bug reports atm. [23:29] <durandal_1707> its not bitexact with ref [23:32] <durandal_1707> cehoyos: nooooo [23:33] <cehoyos> durandal_1707: You could add a "inclusion of" or similar to the commit message (I misunderstood originally) [23:34] <durandal_1707> hmm, fps filter should be auto insterted [23:34] <cehoyos> The relevant commit was not merged [23:35] <cehoyos> Maybe 74b961d [23:37] <saste> durandal_1707, in noise docs: q higher quality (slightly better looking, slightly slower) [23:37] <saste> but it is not implemented [23:37] <durandal_1707> cehoyos: it just replaces, instead of autoinsert for muxers when needed [23:38] <durandal_1707> saste: so, i'm ignorant, remove it, or add flag to code that does nothing [23:38] <saste> durandal_1707, was it ever implemented? [23:38] <durandal_1707> yes, git log is your friend [23:38] Action: durandal_1707 afk [23:40] <saste> also it's weird, in mplayer it was "h", not "q" [23:52] <saste> why all uses of av_dynarray_add() are not checked? [23:58] <saste> indeed it will crash badly in case of realloc failure, am I right? [23:58] <saste> so the question is, should we fix it? [23:59] <saste> ^^ michaelni? [00:00] --- Thu Apr 25 2013

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ffmpeg.log.20130424
by burek 25 Apr '13

25 Apr '13

[00:24] <Sashmo_> anyone know if there is a way to use external sync when transcoding to avoid PTS errors? [00:28] <Mavrik> well, there is a setpts filter [00:50] <trose> Hey so I'm compiling video from individual frames. Is it possible to get ffmpeg to duplicate frames such that i could get 24 fps by rendering one image 24 times each second? [03:10] <TeruFSX> So, I work on StepMania, which is essentially a PC DDR clone [03:11] <TeruFSX> we've been plagued with FFMpeg issues for about 3 years [03:12] <TeruFSX> And I think this is mainly because the person who maintained the video code is gone now, and we've basically been upgrading the existing interface enough to work [03:12] <TeruFSX> Many video types either don't decode or decode improperly, and under certain circumstances the game can crash outright. [03:43] <Mista_D> http://www.mediafire.com/?53qx6rszmtnrea2,dibsabp9zdzo7s9 -- two sample files that fail "-f concat" and also fail "MP4Box -cat"... Both encoded identically, ffprobe reports same specs for both. 48 hours looking at them... Plz advise. [07:57] <Mista_D> 1 pass encoded files concat just fine. 2 pass fail. Any ideas? [08:55] <Mista_D> https://ffmpeg.org/trac/ffmpeg/ticket/2498 [09:07] <MrKim> hi . can anybody tell me about planar audio format? [09:08] <MrKim> or give a data sheet or something for reference.. [09:13] <MrKim> JEEB : hi [10:32] <iKriz> anyone here have a working VS2010 project with libswscale in it? [10:34] <iKriz> i'm stuck on a unresolved external symbol "struct SwsContext... whereas i definitly have the lib files and the headers included... i'm missing something? [11:04] <iKriz> i love the ffmpeg faq :) forgot to extern "C" the include :) [11:39] <MrKim> what is planar audio. [11:39] <MrKim> ... [11:40] <MrKim> i have 2 raw audio one is interleaved and another is planar [11:43] <durandal_1707> how you got planar one? [11:53] <MrKim> whell [11:53] <MrKim> well [11:53] <MrKim> um.. [11:53] <MrKim> got from avi [11:53] <durandal_1707> you can convert planar to interleaved with swresample [11:54] <durandal_1707> interleaved audio have all samples stored in data[0] [11:55] <durandal_1707> planar have each channel samples stored in separate array: extended_data[ X ], where x is 0 - (number_of_channels - 1) [11:55] <MrKim> it means..data[0] -> LRLRLRLRLRLRLR [11:56] <durandal_1707> yes, for interleaved [11:56] <MrKim> planar is..? data[0] - LLLLLLLLL data[1] - RRRRRRR ? [11:56] <durandal_1707> yes [11:56] <durandal_1707> but use extended_data if you use more than 8 channels [11:57] <MrKim> i have some question [11:57] <durandal_1707> me too.... [11:58] <MrKim> From the point of view of Memory.. [11:58] <durandal_1707> can you finally say it? [11:58] <MrKim> planar is data[0]- LLLLLLLLRRRRRRRRRRR ? [11:58] <durandal_1707> nope [11:59] <MrKim> isn it Consecutive? [11:59] <durandal_1707> no its not [11:59] <MrKim> oh.... [12:00] <durandal_1707> there are holes between them [12:00] <MrKim> holy.. [12:00] <MrKim> that why i got many crash [12:00] <JEEB> yes, they are completely separate memory planes :P [12:01] <JEEB> separately malloc'd [12:01] <JEEB> they /might/ be consecutive under some random circuimstances, but you should not at any point plan your code that it would be [12:01] <MrKim> ok i see [12:02] <MrKim> another question.. [12:02] <MrKim> sorry [12:02] <MrKim> avcodec_decode_audio3(pAudioCodecCtx, pAudioBuffer, &nAudioDataSize, &audio_avpacket); [12:02] <MrKim> My code use this func [12:03] <MrKim> 2nd param is output buffer, [12:04] <MrKim> I use that buffer to convert interleaved when input audio fmt is planar [12:05] <MrKim> how can i get data[0] and data[1] about planar fmt? [12:05] <MrKim> it seems that 2nd param means data[0] .. [12:08] <MrKim> sorry i cant speak english well.. can you understand my question.. :0 [12:10] <JEEB> I recommend you switch to using http://ffmpeg.org/doxygen/trunk/group__lavc__decoding.html#ga834bb1b062fbcc… [12:10] <JEEB> that one will give you an AVFrame and the memory allocation will be handled by libavcodec [12:11] <MrKim> that will issue many questions to me lol [12:11] <MrKim> ok i'll try [12:14] <MrKim> is there no way using function3 to solve this matter? cuz i'm not understand this code .. [12:14] <MrKim> well; [12:16] <JEEB> using decode_audio3 /is/ possible, but I have a feeling that losing the AVFrame is gonna be a huge PITA [12:16] <JEEB> not to mention that IIRC 3 works by using 4 in the version you have [12:16] <JEEB> in the internals [12:17] <MrKim> sorry what is PITA? [12:17] <JEEB> Pain In The Ass [12:18] <MrKim> haha [12:18] <MrKim> ok i'll try it [13:15] <freesia> hello [13:15] <freesia> i am building ffmpeg on android [13:16] <freesia> got error : /open_source_archive/linux/toolchains/gcc-4.4.3_android/bin/arm-eabi-gcc is unable to create an executable file. [13:16] <freesia> anybody know why? [13:17] <JEEB> see config.log in fullness [13:17] <JEEB> pastebin it somewhere [13:17] <JEEB> and link here [13:33] <xintron> What software would you guys recommend for relay-streaming over HTTP to multiple clients (raw data)? ffserver doesn't seem to handle relay streaming (it needs to transcode). [14:53] <Guest2161> hi [15:41] <zimbatm> hi [15:42] <zimbatm> any idea how I could debug the content of a .mp4 header ? [15:42] <zimbatm> i've got this file that doesn't stream well over http [15:43] <zimbatm> after playing with ffprobe a while I noticed that it finished reading the headers after ~8MB [15:43] <zimbatm> File position before avformat_find_stream_info() is 8820685 [15:43] <zimbatm> File position after avformat_find_stream_info() is 8821848 [15:43] <zimbatm> the file is 592M big so it's not the atom at the end [15:45] <zimbatm> when hexediting the content looks like mapping tables [15:52] <zimbatm> this is the command that I ran: https://gist.github.com/zimbatm/5452268 [15:52] <zimbatm> and i'm going to provide with a header dump if interested [16:03] <zimbatm> here is the start of the video: http://ffmpeg-debug.s3.amazonaws.com/big-header-video.mp4 [16:10] <zimbatm> JEEB: do you mind taking a look ? [16:10] <zimbatm> sorry if i'm interrupting [16:21] <zimbatm> is it possible that it's the quantization tables ? [16:53] <tefid> hello friends, ffmpeg with x11grab only records black video. What's wrong? I have searched more than 50 search pages of google and then I'm here [16:54] <tefid> in fullscreen only. [16:54] <tefid> for normal desktop applications it is fine but for fullscreen games it just produces black video [16:55] <zimbatm> tefid: probably because OpenGL takes over the screen buffer [16:55] <tefid> thanks zimbatm. is there any way to fix it? [16:55] <tefid> How do other people screencast games like minecraft? [16:55] <zimbatm> i'm not really an expert [16:55] <tefid> I am using kde [16:56] <zimbatm> i would try to turn composition on/off and see if it makes a difference [16:56] <tefid> composition is off [16:56] <tefid> is it related to native qt rendering system? [16:57] <zimbatm> tefid: http://blog.smr.co.in/linux/graps-capture-opengl-frames/ [16:57] <zimbatm> maybe that helps [16:57] <zimbatm> or try turning composition on and see if ffmpeg x11grab works at all [16:58] <tefid> Thanks for the link zimbatm. I had already had glc and it only crashed so still I was not able to record [16:58] <tefid> thought the project is dead [16:58] <tefid> and it is. graps is dead [16:59] <Sashmo_> does anyone know if I can add an external sync to avoid PTS errors while transcoding? [16:59] <zimbatm> tefid: well the issue is with OpenGL for sure, try looking "OpenGL linux record screen" on google [17:00] <zimbatm> Sashmo_: did you try the -async command ? [17:01] <Sashmo_> that just moves the timing around +/- I want to use a tottaly external sync to restamp it during transcoding, I find that over time my source (ip transport stream) goes out of sync from errors coming down from the satellite [17:05] <zimbatm> Sashmo_: is the stream in realtime ? [17:05] <Sashmo_> yes [17:05] <zimbatm> maybe you can use your system's clock [17:05] <zimbatm> http://ffmpeg.org/ffmpeg-all.html#setpts [17:05] <zimbatm> there's the TB -> time base [17:06] <zimbatm> not sure if it's it but might be worth giving a shot [17:06] <Sashmo_> hmm, ok I'll give that a try, thanks for the input! [17:11] <zimbatm> check out the examples below and let us know how it goes :) [17:17] <Sashmo_> I like the command setpts='(RTCTIME - RTCSTART) / (TB * 1000000)' [17:17] <Sashmo_> this is what I am looking for [17:17] <Sashmo_> I need to test it [17:17] <pingufan> Hello. I recorded for my frien a TV movie where his son is an artist. Now I want to convert the HD movie (.ts file from a dm800 box) into a useful format and then I want to cut out advertizing. [17:17] <pingufan> Can somebody please, help? [17:18] <pingufan> ProjectX does not support HD, I stuck. [17:20] <pingufan> Currently I try to use ffmpeg to convert the original .ts file, hope it does not get out of sync. As this needs long time, can somebody, please, tell me if this commandline is ok? "ffmpeg -i 20130419_1805_-_ZDF_HD_-_SOKO_WIEN.ts -c:v libx264 -preset fast -crf 21 soko.mp4" [17:42] <Diogo> hi one question this is possible use ffmpeg using loop mode? [17:42] <Diogo> i can't get this options working...i'm streaming to rtmp server [17:43] <Diogo> command: /servers/ffmpeg/bin/ffmpeg -re -i "rod.mp4" -vcodec libx264 -f flv -acodec libfaac rtmp://IP/streamname [17:43] <Diogo> ?? [17:43] <Diogo> any friend can help me please [17:52] <saschagehlich> hey guys, does anybody know of a good way to properly convert .gif images to other video formats? ffmpeg always gives me a black image [17:52] <zimbatm> pingufan: seems good to me. you can add ` -t "00:00:30" ` to just encode 30 seconds of the video to see if it's the quality you want [17:55] <Mysterytrain> I have a list of about a hundred video files that I want to extract the audio from. will ffmpeg support a batch operation ie. for i in *.mkv;do ... or do I have to do it all by hand. [17:56] <zimbatm> saschagehlich: can you paste your command and the output on pastie.org [17:56] <zimbatm> ? [17:57] <zimbatm> Mysterytrain: by hand seems the good solution [17:57] <saschagehlich> zimbatm: https://gist.github.com/saschagehlich/378fe272c09d230c48a0 [18:00] <zimbatm> saschagehlich: sorry, I don't see any clue from the output [18:00] <zimbatm> ah maybe it's the color space [18:01] <zimbatm> try adding "-pix_fmt yuv420p" to your command [18:01] <zimbatm> not all players suppot the yuv444p color space [18:02] <zimbatm> saschagehlich: also try using "ffplay" to play the video and see if it works [18:02] <zimbatm> if you're using homebrew you need to brew install ffmpeg --with-ffplay [18:02] <saschagehlich> ah okay, one sec [18:02] <saschagehlich> -pix_fmt fixed it, but the timing is wrong [18:03] <saschagehlich> there is like one frame that is supposed to be 5 seconds but it's more like 0.1s [18:06] <zimbatm> ffmpeg will make an average, if you have different timings between frames I think you're toast [18:07] <saschagehlich> :( [18:07] <saschagehlich> any hints on where to look at the code if i want to implement this? [18:08] <zimbatm> the gif demuxer ? [18:08] <saschagehlich> okay [18:08] <zimbatm> libavcodec/gif* [18:09] <saschagehlich> okay. I'll have a look at it [18:09] <saschagehlich> haven't done C in years, but hey& challenge accepted [18:09] <zimbatm> hehe, good luck [18:09] <zimbatm> ffmpeg is not any C program [18:10] <saschagehlich> yeah i dived into the gif implementation couple of months ago [18:10] <zimbatm> if you want to avoid duplicating frames I would look at fixing the PTS [18:12] <zimbatm> saschagehlich: it's a bit hard to get help from the ffmpeg devs because of all the noise [18:12] <saschagehlich> the code noise? :D [18:12] <zimbatm> once you feel confident enough with the code you might get better help from #ffmpeg-devel [18:12] <zimbatm> no, the lambda user's noise :D [18:12] <saschagehlich> hehe [18:13] <zimbatm> the ones who don't even read the docs [18:13] <saschagehlich> alright, thanks so far. if i won't be back in 5 days, call the police [18:13] <saschagehlich> i might have died in the codebase [18:13] <zimbatm> ok, i'll set a notification [18:14] <zimbatm> you're sure 5 days ? it's a bit long without food [18:14] <saschagehlich> i'm a big boy, i have enough for 5 days [18:16] <zimbatm> lol [18:34] <Diogo> hi this is possible to loop a video x times? [18:36] <spaam> yes and no [18:36] <spaam> with a hack, yes [18:36] <spaam> concat the same file X times [18:38] <spaam> Diogo: https://ffmpeg.org/ffmpeg-all.html#concat-1 ^^ [18:38] <saschagehlich> zimbatm: so looks like the gif demuxer works fine, it reads the duration / delay correctly. now I have to find out what causes the issue :( [18:44] <zimbatm> saschagehlich: i'm really not an expert [18:44] <zimbatm> maybe you can try to play with http://www.ffmpeg.org/ffmpeg-all.html#asetpts_002c-setpts [18:44] <zimbatm> and also use "-loglevel debug" for more output [18:52] <saschagehlich> it looks like some kind of index issue& the frames get the PTS of the previous GIF frame [19:14] <durandal_1707> saschagehlich: what libavformat version you are using? [19:14] <saschagehlich> 55.3.100 [19:14] <saschagehlich> pulled it from master i believed [19:15] <saschagehlich> -d [19:18] <durandal_1707> ubitux: ^ [19:19] <ubitux> sounds like a problem in the demuxer [19:19] <ubitux> no? [19:20] <MrKim> hi ubitux [19:20] <ubitux> the problem is from gif to <something-else>, right? [19:20] <saschagehlich> yes ubitux [19:21] <ubitux> saschagehlich: can you share the gif and the cmd line to reproduce? [19:21] <ubitux> i'll have a look in a moment [19:21] <saschagehlich> sure, hang on [19:21] <ubitux> hi MrKim [19:22] <saschagehlich> ubitux: https://gist.github.com/saschagehlich/5ab45f376f649075851b [19:23] <saschagehlich> first frame is rendered too short (looks like default delay), second frame is rendered with the duration of the first frame, third one is rendered with the duration of the second one etc.. [19:26] Action: ubitux saw that gif on imgur a while ago [19:27] <saschagehlich> hehe [19:28] <ubitux> looks like a bug in the mp4 muxer [19:28] <ubitux> it works with mkv afaict [19:31] <saschagehlich> same issue with mpg [19:46] <ubitux> saschagehlich: i don't have time to dig much into the issue right now, can you open an issue so it's not forgotten? [19:46] <saschagehlich> sure [19:47] <ubitux> thank you [21:25] <MrKim> Ubitux: [21:26] <MrKim> durandal_1707: [21:26] <MrKim> JEEB: [21:26] <MrKim> Thank you very much for your help [21:26] <ubitux> stop it please [21:26] <ubitux> yeah right :) [21:27] <MrKim> oh. sorry if it annoying you; [21:27] <ubitux> if it's to thank us we can tolerate :) [21:27] <MrKim> It works.. [21:27] <JEEB> I guess you just used audiodec4 in the end? [21:28] <JEEB> as it allocates the buffers and all for you [21:28] <MrKim> yes.. haha [21:29] <MrKim> I never heard planar fmt ever [21:29] <MrKim> learn very many thing this time [21:30] <MrKim> so thankyou.. thankyou :) [21:32] <MrKim> my program inspect movie , music files.. [21:32] <MrKim> and detect what the file is .. [21:33] <MrKim> using its audio's spectrum.. [21:34] <MrKim> the more i worked hard the more my girl friend go far from me [21:35] <MrKim> and said goodbye to me today [21:35] <MrKim> so sad :< [21:44] <guns> Hello. Does ffmpeg have a builtin audio notch filter (for filtering ac hum)? [21:44] <guns> I am currently exporting audio to audacity, applying the filter there, then remuxing for the final output [21:44] <guns> I would like to automate this process if possible. [21:49] <durandal_1707> guns: bandpass/bandreject? [21:50] <guns> durandal_1707: thank you. I'll look into those [23:23] <AnimeFreak> I used ffmpeg to convert mkv vid to 3gp but there are no subtitles in output video! How can I make ffmpeg keep subtitles which are part of mkv file? [23:26] <klaxa> i think 3gp doesn't support embedded subtitles, so i guess you want to hardsub them AnimeFreak? [23:26] <KTO> Are binary distributions which use FFMPEG to decode MP3s required to pay the Fraunhofer license? [23:26] <AnimeFreak> klaxa: yep [23:27] <klaxa> that's a bit tricky with ffmpeg, you will have to extract the subtitle files from the mkv, as far as i know ffmpeg can't hardsub embedded subtitles straight from an mkv [23:27] <klaxa> let me take a look at the wiki though... [23:29] <klaxa> hmm... yes... [23:29] <saste> klaxa, subtitles filter [23:29] <klaxa> well yes, but it doesn't work with embedded subtitle files [23:29] <klaxa> that's why it's a bit tricky [23:29] <saste> "embedded"? [23:29] <klaxa> in mkv files [23:29] <saste> why not? [23:30] <klaxa> oh? [23:30] <klaxa> how? [23:30] <klaxa> never worked for me [23:30] <klaxa> i always had to extract the subtitles [23:30] <klaxa> or rather [23:30] <klaxa> i always did, it would be great if it worked without extracting them [23:30] <saste> what do you mean by "extract the subtitles"? [23:31] <saste> afaik you can just pass the file as is to subtitles, and it will demux (and thus extract) the subs [23:31] <klaxa> ffmpeg -i myfile.mkv -vn -an -c:s copy somefile.ass [23:32] <klaxa> and then i can do ffmpeg -i myfile.mkv -c:v $codec_v -c:a $codec_a -vf subtitle=somefile.ass hardsub.mp4 [23:33] <saste> klaxa, what's wrong with subtitles=myfile.mkv? [23:33] <klaxa> oh... OH... [23:33] <klaxa> i never tried that, will that work? [23:34] <saste> i think so [23:37] <AnimeFreak> Aaaand I'm lost. [23:38] <klaxa> i will pull the latest git, the version i'm running right now produces double-free corruption [23:39] <klaxa> (for the record: https://gist.github.com/klaxa/5455806 ) [23:40] <klaxa> AnimeFreak: don't worry, we'll ge-- [23:40] <klaxa> well... [23:42] Action: ezekiel is on edge of seat [23:48] <Ilias95> Hello. In a command like "-acodec libmp3lame -v 2" what does the -v option means? I can't find where it is mentioned in the documentation. [23:51] <saste> Ilias95, grep for -v in ffmpeg(1) [23:54] <Ilias95> saste, I already did but it seems in a wrong way. I see it's about loglevel so it has nothing to do with the final file, right? [23:55] <saste> right [23:55] <Ilias95> okay, thank you [00:00] --- Thu Apr 25 2013

1 0

ffmpeg-devel.log.20130423
by burek 24 Apr '13

24 Apr '13

[00:53] <ubitux> http://www.iquilezles.org/blog/?p=2177 [01:08] <cone-735> ffmpeg.git 03Clément BSsch 07master:b8a5c7613194: lavfi: add frame counter into AVFilterLink and use it in filters. [01:08] <cone-735> ffmpeg.git 03Clément BSsch 07master:fdd93eabfb26: lavfi: add timeline support. [01:08] <cone-735> ffmpeg.git 03Clément BSsch 07master:38853169ce49: lavfi: add 'enable' command injection to filters supporting timeline. [01:08] <cone-735> ffmpeg.git 03Clément BSsch 07master:15d2f2699890: lavfi/overlay: support timeline through the new system. [01:18] <ubitux> michaelni_: your opinion on vid.stab? [01:47] <michaelni_> ubitux, my oppinion is if theres a maintainer and the code has no major issues then its probably a good idea, but you reviewed it i just took a quick look, you should know it better [01:47] <michaelni_> than ,e [01:47] <michaelni_> me [01:48] <ubitux> ok [01:48] <ubitux> thank you :) [03:24] <highgod> Hi, how can I get the download statistics of ffmpeg? [03:37] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:a9a537b154aa: avfilter/drawtext: add pict_type support [08:33] <highgod> Hi, all, I want to ask a question. how can I get the download statistics number of ffmpeg? [09:47] <durandal_1707> we gonna keep another jpeg2000 decoder? [10:25] <cone-623> ffmpeg.git 03Janne Grunau 07master:94660c35249e: threads: always call thread_finish_setup for intra codecs [10:25] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:7c49c8b0ac30: Merge commit '94660c35249e11c53b9a9f0c2f85c4335b21eda7' [10:37] <cone-623> ffmpeg.git 03Janne Grunau 07master:10f1a4d9bd82: jpeg2kdec: output is native endian AV_PIX_FMT_XYZ12 [10:37] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:5056cff12597: Merge commit '10f1a4d9bd8239ac64f364e0d3b6423c28230d6d' [10:48] <michaelni_> durandal_1707, depends on features/bugs/speed/maintaince activity of the 2. If one is clearly better in all categories the other will be droped [10:49] <durandal_1707> michaelni_: j2kdec is not bitexact at all at any level [10:54] <michaelni> only the 9/7 dwt should be non bitexact as it uses floats [10:56] <michaelni> also theres are 2 integer 9/7 dwts in our codebase [11:11] <highgod> Hi, all, I want to ask a question. how can I get the download statistics number of ffmpeg? [11:12] <cone-623> ffmpeg.git 03Diego Biurrun 07master:c1ad70c3cb05: x86: Move some conditional code around to avoid unused variable warnings [11:12] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:0a73803c86f6: Merge remote-tracking branch 'qatar/master' [11:13] <michaelni> highgod, i dont think theres a statistic, why do you ask ? [11:14] <michaelni> you could look at the download statistics of other projects using ffmpeg though, like chrome, mplayer, vlc, ... if they have a public statistic [11:15] <j-b> chrome and VLC should give you around 500 millions users [11:16] <highgod> OK, thanks [11:16] <j-b> then all your SmartTVs [11:21] <highgod> OK, thanks [11:24] <j-b> Hello BuxiNess [11:27] <BuxiNess> Hello, I'm the culpable for the new jp2K decoder :-) [11:28] <michaelni> Hi BuxiNess :) [11:31] <BuxiNess> Hello [11:34] <michaelni> BuxiNess, can you send a patch that adds you to MAINTAINERS ? (iam assuming you still want to maintain it) [11:35] <BuxiNess> michaelni, yes this afternoon. I'm on DWT optimsation right now [11:35] <michaelni> ahh, ok, float or integer dwt ? [11:36] <BuxiNess> float 9/7 dwt in the begining [11:36] <BuxiNess> Than int [11:36] <michaelni> with int be careful, there are some pitfalls with precission and coefficient quantization IIRC [11:37] <michaelni> that is simple scalar quantization of the float wavelet lifting coefficients to integers does not work very well [11:38] <BuxiNess> michaelni, yes. I'm not sure bitexact method for 9/7 DWT is reaaly useful for playback . [11:38] <BuxiNess> Usefull for regression test indeed [11:38] <michaelni> yes [11:39] <michaelni> also snow has a 9/7 integer dwt, might be a good idea to look at that [11:39] <michaelni> maybe it could even be used with little changes for regression testing [11:40] <BuxiNess> dirac also, but not with the same lifting steps AFAIK [11:40] <michaelni> IIRC the snow DWT also uses a more efficient line permutation than there was in the j2k dwt but i might misrememeber [11:41] <michaelni> yes but if its just for regresison testing slightly different lifting coefficients might not be important [11:41] <michaelni> for display to the user it would be a problem though (likely) [11:43] <michaelni> but overall the snow and j2k 9/7 dwts should be quite similar, also see: http://guru.multimedia.cx/the-snow-wavelet/ [12:34] <ubitux> BuxiNess: if node is NULL tag_tree_decode will access stack[-1] [12:34] <ubitux> maybe that's not possible in practice, but gcc complains [12:34] <ubitux> libavcodec/jpeg2000dec.c:126:22: warning: stack[-1] may be used uninitialized in this function [-Wmaybe-uninitialized] [12:34] <ubitux> maybe add a av_assert0(node) at the beginning of the function or something [12:48] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:7a617d6c17d0: avcodec/x86/sbrdsp_init: disable using the noise code in x86_64 MSVC [14:19] <ubitux> can anyone confirm me that this is not possible with avfft api? https://gist.github.com/ubitux/5442675 [14:20] <ubitux> (just a 2d fft) [14:21] <ubitux> (output: https://gist.github.com/ubitux/5442713) [14:27] <merbanan> ubitux: I confirm [14:27] <ubitux> thanks :( [14:28] <ubitux> well i'll write my filter with fftw for now [14:28] <michaelni> ubitux, are you doing something different than 2x 1d ffts ? [14:29] <ubitux> michaelni: not yet, it should just be row & cols 1d fft [14:29] <michaelni> so where is the problem ? [14:29] <ubitux> well i can make a "slow" version with av fft api though if necessary [14:30] <michaelni> why slow ? [14:30] <ubitux> i guess the memcpy/transpose might be a bit slow, or i'm wrong? [14:30] <michaelni> yes likely [14:30] <ubitux> that's what i had in mind, unless you think of something else? [14:31] <merbanan> if the transform is separable then a 1d is faster [14:31] <michaelni> it should be seperable if i understand correctly what ubitux does [14:32] <michaelni> but a dedicated optimized transform would likely be faster [14:33] <ubitux> i'm mostly just trying experiments; the goal is to allow eval() filtering on 2d-transformed blocks [14:33] <ubitux> i've no idea if it's relevant or not [14:33] <ubitux> but i need 2d fft anyway, and don't plan to mess up much with fft optims :p [14:33] <pengvado> for instance, a single 1d fft requires some shuffles in the inner loop, whereas 4 simultaneous 1d ffts can avoid that by packing 1 value from each block into a xmmreg. [14:35] <ubitux> (also, i will possibly need a 3d fft at some point) [14:38] <ubitux> michaelni: i'll write the filters with fftw (because that's simpler for my purpose), and will eventually write a naive fft 2d with 2x 1d fft before submitting [14:38] <ubitux> but don't count on me to work on a fast version.. :P [14:38] <michaelni> fine with me ... [15:51] <durandal_1707> michaelni: ^ that commit cehoys mentioned is ridiculous [15:56] <ubitux> 50M should be enough for anyone © [15:56] <cone-623> ffmpeg.git 03Paul B Mahol 07master:ad56535dd1d7: avformat: fix orig_pos type to match pkt->pos [15:58] <durandal_1707> that is 2gb of input ... and i could be this fixes #2494 [15:58] <durandal_1707> *bet [16:01] <ubitux> don't you mean another ticket? [16:12] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:8f04220df805: avfilter/geq: rgb support [16:19] <burek> there is some kind of a bug in docs http://ffmpeg.org/download.html [16:19] <burek> "0.10.7 was released on 2013-04-10." [16:19] <ubitux> is that wrong? [16:19] <burek> the date doesn't sit in between of 0.9.2 and 0.11.3 [16:19] <durandal_1707> thats not doc [16:20] <ubitux> burek: it's the revision [16:20] <ubitux> not the original 0.10 release [16:20] <ubitux> check the next sentence [16:21] <burek> i see, so it's the last stable version from that branch [16:21] <ubitux> yes [16:21] <burek> ok, my bad, sorry [16:24] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:488c760091c2: doc/filters: document geq rgb mode [16:41] <burek> [16:40:50] <b_jonas> man, this was fast, thank you (I reported a bug and a developer has fixed it within an hour) :))) #ffmpeg [16:41] <burek> +1 [17:08] <cone-623> ffmpeg.git 03Carl Eugen Hoyos 07master:9526c9ccc3dd: Read block_align from extradata when decoding qclp in aiff. [17:08] <cone-623> ffmpeg.git 03Michael Niedermayer 07master:b21b2ca95b33: Merge remote-tracking branch 'cehoyos/master' [20:51] <ubitux> there is no particular alignment or padding requirement for the fft code, right? [22:34] Action: michaelni is trying to find the remaining bugs in vc1 [22:35] <michaelni> i hadnt realized we have random pieces of code with 0 && in there :) [22:50] <cone-623> ffmpeg.git 03Stefano Sabatini 07master:638ffb24136d: lavfi/hue: rework logic for setting commands [22:50] <cone-623> ffmpeg.git 03Stefano Sabatini 07master:dfdee6cab323: doc/filters: sort multimedia filters by name [22:56] <saste> ubitux, how can I install locally a python module? [22:57] <ubitux> with pypi possibly... i guess [22:58] <ubitux> try something like pip2 search zmq [22:58] <ubitux> pip2 install pyzmq might do the trick [22:59] <ubitux> (i have pip for python3 here, and pip2 for python2) [23:00] <saste> pip? in which package should I find it? [23:01] <saste> python-pip [23:01] <ubitux> i guess :) [23:01] <ubitux> make sure it's the python2 version if you're interested in python2 packages :p [23:07] <saste> ubitux, can I install the package in my home, so that i don't mess with the system? [23:08] <ubitux> saste: good question; you might be able to just download the thing and put it in the script dir (the import should work) [23:09] <saste> ubitux, script dir? [23:09] <ubitux> yeah the zmq script i guess in your case [23:09] <saste> note that I don't know the first thing about python installation paths [23:09] <ubitux> otherwise there are some env path [23:10] <ubitux> saste: iirc pip basically just download and unzip the package into /usr/lib/python-xxx/site-packages/ [23:10] <ubitux> which is part of the python import env path [23:10] <saste> ubitux, and compiles it [23:10] <ubitux> ah, well yeah for import to work :) [23:10] <saste> and i don't think it was using my locally installed libzmq when compiling [23:18] <ubitux> i have no idea :p [23:18] <ubitux> saste: what distro are you using btw? [23:18] <saste> ubitux, ubuntu [23:18] <saste> 12.04 [23:18] <ubitux> oh. [23:18] <ubitux> okay [23:19] <saste> anyway it's mostly working but the script it's printing garbage on return [23:19] <saste> did you try it? [23:19] <ubitux> i didn't go up to the read() :D [23:20] <saste> what should we ship, the C client or the python one? [23:20] <ubitux> it was written in a few minutes and then i had to review your hue patch [23:20] <saste> or both? [23:20] <ubitux> dunno, i really like the shell helpers of cmd [23:20] <ubitux> are you already done with the filter? [23:20] <saste> btw we may make use of some readline utils [23:20] <ubitux> gpl? :) [23:21] <saste> or we hack our own variant? [23:21] <saste> or we use a non-gpl lib, if exists [23:21] <ubitux> for what? still the zmq inject c example? [23:21] <saste> there is also ffeval [23:21] <ubitux> saste: btw, can't you use something like netcat to send commands? [23:22] <ubitux> which would simplify everything... [23:23] <saste> ubitux, no, the user is supposed to use libzmq, so makes sense to provide an example script/client [23:24] <ubitux> whatever you prefer [23:24] <saste> we can have both [23:24] <saste> now look at the eval read loop [23:25] <saste> i'm duplicating most of the code reallocating the buffer size [23:25] <saste> don't we have a function for that? [23:27] <ubitux> bprint? :P [23:27] <ubitux> ah for recv. [00:00] --- Wed Apr 24 2013

1 0

ffmpeg.log.20130423
by burek 24 Apr '13

24 Apr '13

[00:17] <lullabud> it appears that ffmpeg isn't seeing the meatadata at all. runing "ffmpeg -i 20371231-23-59-59-1.mov -f ffmetadata metadata.txt" produceds only two lines, comment and comment-eng [00:17] <lullabud> *extra metadata [00:24] <pudgypaw> I'm currently on Ubuntu Studio, 12.04. I've beta tested both ubuntu and blender for 5+ years now and right now I want to assemble a guide on how to completely strip avconv out of my system and go pure 100% ffmpeg. However, does such a howto already exist?? [00:26] <otters> my ffmpeg doesn't have alac support. how do I get it? (latest version from mac homebrew, 1.2) [00:27] <ubitux> pudgypaw: that might not be a good idea unfortunately, because a lot of applications depends on the libraries; we should be binary compatible thought [00:28] <ubitux> pudgypaw: you can use static build, or build it yourself without installing it (that's relatively straightforward), or install it in a user directory [00:28] <ubitux> you can also complain to your linux distro [00:28] <ubitux> you can try some packages such as deb-multimedia (not sure if that will work with with ubuntu) [00:28] <ubitux> or you can... just switch distro [00:29] <ubitux> otters: afaik it's natively supported; [00:29] <ubitux> % ./ffmpeg -v 0 -codecs|grep -i alac DEA..S alac ALAC (Apple Lossless Audio Codec) [00:30] <otters> okay, that's weird. [00:30] <otters> "[NULL @ 0x7fb4ca025600] Unable to find a suitable output format for 'alac'" [00:30] <ubitux> lullabud: the exif tags don't seem exported, maybe open a ticket with a feature request [00:31] <ubitux> otters: alac is a codec, not a format [00:31] <otters> mb [00:31] <ubitux> the container is a mov/mp4 based [00:32] <ubitux> ffmpeg -i ... -c:a alac foobar.m4a [00:32] <otters> https://gist.github.com/joelteon/b1c4973797dc3d39386e [00:33] <ubitux> flag ? [00:33] <ubitux> flac* [00:33] <otters> yeah, flac [00:33] <ubitux> oh, shell weirdness, sorry i'm tired [00:33] <otters> yeah, that's just an expansion of a function I defined for my zsh [00:34] <otters> there, I added it to the gist [00:34] <ubitux> remove -threads [00:34] <otters> https://gist.github.com/joelteon/b1c4973797dc3d39386e [00:34] <otters> ok [00:34] <ubitux> -threads takes an argument [00:34] <ubitux> currently you're passing "-acodec" to -threads [00:34] <ubitux> and "alac" stands alone [00:34] <otters> heh [00:34] <ubitux> so it thinks it's the output file [00:34] <otters> well, that used to work, what changed? [00:34] <ubitux> -threads auto [00:34] <ubitux> but it's the default [00:34] <ubitux> just remove it. [00:35] <ubitux> i doubt it ever worked [00:35] <otters> it definitely worked for me in the past [00:35] <ubitux> it was wrong then [00:35] <ubitux> maybe tolerated for some random parsin reason [00:36] <otters> ok well that works, tyvm [00:36] <ubitux> yw [00:51] <pudgypaw> ubitux, Back when I did alot of blender I did compile from source thanks to fakeoutdoorsman, and used FFMPEG/FFPLAY at the command line too. Having Blender suddenly referencing avconv threw me off, lol. I'm just dismayed that, even if I do compile from source, the entire OS is still stapled to what I consider a less stable FLOSS project [00:52] <pudgypaw> I guess I'll see what I can do and report back later =/ [00:53] <ubitux> all you can do is complain to the packager [00:53] <ubitux> note that we tried to do it for years now [00:53] <ubitux> but unfortunately we are the evil developers [00:53] <ubitux> as a user, you have more power [00:53] <ubitux> (for that kind of stuff) [01:02] <pudgypaw> ubitux, from my research the package maintainer is actively involved in avconv, closing objectivity imho. Thanks for confirming the scope of what I'm setting out to do. [01:50] <gp> Is -movflags faststart equivalent to using qt-faststart? The docs are vague with "will not work in various situations". If I am encoding files for progressive download via the html video element or flash playback will I encounter these various situations? [02:07] <MrKim> . [02:12] <MrKim> Hi . I wanna convert sample fmt with libswresample , I decoded audio stream, and i gave the output buffer to the SWR. but it print error msg like "Input channel count and layout are unset" [02:14] <MrKim> but I set it with av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout,0) [02:14] <MrKim> what's wrong .. [02:39] <quidnunc> Is there any public documentation of the mp4 container? [03:14] <iive> quidnunc: it is public standard. look for iso/iec 14496-12 (part 12 is the container, aka .mp4) [03:15] <iive> hum... scratch that. it is replaced by part 14... [03:15] <iive> so 14496-14 [05:13] <rcombs> https://ffmpeg.org/trac/ffmpeg/ticket/1582#comment:8 <-- so, banding issues [06:28] <monokrome> Hey. Is ffmpeg smart enough that if I were to ask it to transcode stdin to the same format that it's already in, it wont actually change the file contents? [06:29] <monokrome> (I mean, the same audio data that went in will come out) [06:57] <defaultro> hey folks, need some help. This param is giving me a very low quality jpeg. ffmpeg -i video.mp4 -r 30000/1001 -s hd720 -f image2 image-%3d.jpeg [06:57] <defaultro> resulting jpg file is only 42k [07:00] <defaultro> i got it, -qscale 0 [07:14] <defaultro> what param for extracting audio and keeping the audio quality? [07:15] <defaultro> is -ab 320k good? [07:16] <dv--> -acodec copy? [07:16] <defaultro> cool [07:18] <defaultro> this is failing, ffmpeg -i video.mp4 -vn -ac 2 -ar 44100 -acodec copy -f mp3 audio.mp3 [07:20] <defaultro> i changed it to aac and m4a, it worked [07:20] <dv--> that will transcode the audio, rather than copy it [07:23] <defaultro> k [07:23] <defaultro> ffmpeg -i video.mp4 -vn -acodec copy audio.aac [07:24] <defaultro> it worked. However, it was mono [07:24] <defaultro> how can I make it 2 channel? [07:25] <matt_keys> I'm getting this error message but I don't know what it means : Could not write header for output file #0 (incorrect codec parameters ?) [07:25] <dv--> huh, it shouldn't be mono if the original is stereo [07:25] <defaultro> my original is mono [07:26] <spaam> -ac 2 ? [07:26] <spaam> for two channels [07:26] <matt_keys> I'm trying to "ffmpeg -i $cam1 -i $cam2 -i $cam3 -i $cam4 -i $cam5 -vcodec copy -f rtsp -muxdelay 0.1 rtsp://localhost:5454/live/cammux.sdp" [07:27] <defaultro> spaam, original source is mono [07:29] <matt_keys> cam1='rtsp://user:pass@192.168.1.31:554/cam1.sdp' [07:30] <defaultro> which is better to use -ar 44100 or -ar 48000 [07:32] <defaultro> this worked, ffmpeg -i video.mp4 -vcodec copy -acodec libfaac -ab 320k -ar 48000 -ac 2 -map_channel 0.1.0 -map_channel 0.1.0 output.mp4 [07:33] <defaultro> the command worked but produced invalid file, lol [07:37] <matt_keys> that error message was with it set to debug by the way :) [07:40] <defaultro> you are correct spaam, I got it working :) [07:41] <defaultro> the command that worked => ffmpeg -i video.mp4 -map 0:1 -vn -acodec libfaac -ab 320k -ar 48000 -ac 2 output.aac [07:41] <defaultro> :) [08:38] <vijaym123> Hello, I am getting the following error "/root/Desktop/bin/gcc is unable to create an executable file. c compiler test failed" ..when I tried to compile the ffmpeg with --cc options [08:38] <vijaym123> how do i resovle this? [08:38] <vijaym123> Thanks in advance for any help. [08:45] <vijaym123> defaultro: can you help me? : [09:02] <vijaym123> bahar: benpro cbreak cbdev Chikuzen [09:02] <vijaym123> fasta: fatpony [09:03] <cbreak-work> what's up moron? [09:03] <vijaym123> Hello, I am getting the following error "/root/Desktop/bin/gcc is unable to create an executable file. c compiler test failed" ..when I tried to compile the ffmpeg with --cc options [09:03] <vijaym123> how do i resovle this? [09:04] <cbreak-work> by stop being a moron [09:04] <cbreak-work> duh [09:04] <vijaym123> Okay...sorry! [09:06] <vijaym123> ubitux: I have a little problem compiling ffmpeg with cc option..can you help me? [09:07] <vijaym123> kick cbreak-work [09:08] <cbreak-work> you've still not learned? Honestly, stop being dumb. It'll help you in the future. [09:08] <cbreak-work> spamming isn't going to make people want to help you. [09:08] <vijaym123> please help me with that issue [09:09] <cbreak-work> neither is begging people to kick others that expose you for what you are. [09:18] <MrKim> Hi . I wanna convert sample fmt with libswresample , I decoded audio stream, and i gave the output buffer to the SWR. but it print error msg like "Input channel count and layout are unset" [09:18] <MrKim> Hi . I wanna convert sample fmt with libswresample , I decoded audio stream, and i gave the output buffer to the SWR. but it print error msg like "Input channel count and layout are unset". [09:18] <MrKim> but I set it with av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout,0) [09:23] <MrKim> when s16 -> s16 convert for test. it shows that error message . [09:23] <MrKim> and try fltp->s16 , it crash. [09:24] <MrKim> I think swr_ctx is not inited ..what is my fault.. [09:57] <Cherry_G> Excuse me, I have a question about ffmpeg fate test, I've now created an SSH key pair, how can I send the public key to the FATE server? [10:24] <xintron> I've been looking at ffserver to just use as a simple relay server for a stream but I can't find any solution to just pipe the raw input out over HTTP with ffserver. Is there any other good software for my situation? [10:36] <MrKim> Hi . I wanna convert sample fmt with libswresample , I decoded audio stream, and i gave the output buffer to the SWR. but it print error msg like "Input channel count and layout are unset". [10:36] <MrKim> but I set it with av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout,0) [10:38] <MrKim> My problem is I don't know what i don't know... i know.. [11:18] <MrKim> somebody tell me why nobody tell me anything to me [11:26] <ubitux> you can guess a channel layout based on the number of channels [11:26] <ubitux> i think swr does that automatically though [11:26] <ubitux> did you set both input and output channel layouts? [11:26] <MrKim> yes i did [11:26] <ubitux> can i see the code? [11:27] <MrKim> src_ch_layout=(int64_t)pAudioCodecCtx->channel_layout; [11:27] <ubitux> Cherry_G: please ask on #ffmpeg-devel :) [11:27] <MrKim> dst_ch_layout = AV_CH_LAYOUT_STEREO [11:27] <ubitux> no i mean, please use pastebin and show me the code [11:28] <MrKim> ok i try [11:28] <Cherry_G> ok Thanks :) [11:33] <MrKim> http://pastebin.com/7xGxtZS2 [11:36] <MrKim> it fails line:61 swr_init() . [11:37] <ubitux> you need to check if the channel layout you get is valid [11:38] <MrKim> when i printed it , output is "3" [11:38] <ubitux> MrKim: what swr version are you using btw? [11:39] <MrKim> wait i'll check it [11:41] <MrKim> swresample_version() returns 4454 [11:41] <ubitux> just run "ffmpeg" [11:41] <ubitux> libswresample 0. 17.102 / 0. 17.102 [11:41] <ubitux> you should see something like this [11:42] <MrKim> ok wait [11:42] <MrKim> actually i use only libraries [11:42] <ubitux> ok no worry [11:42] <ubitux> try to set the channel count [11:42] <ubitux> when you have no channel layout [11:43] <MrKim> umm.. [11:44] <MrKim> like.. [11:44] <ubitux> if (!src_ch_layout) av_opt_set_int(swr_ctx, "ich", pAudioCodecCtx->channels, 0); [11:44] <MrKim> av_opt_set_int(swr_ctx, "in_channel_count", src_nb_channels, 0); ? [11:44] <ubitux> and same for output [11:44] <Guest75813> bro [11:45] <MrKim> ok thanks i'll check it [11:45] <Guest75813> Can someone tell me how to compile FFMPEG for ARM-v7 ? [11:45] <Guest75813> Is it possible? [11:47] <Guest75813> MrKim: [11:48] <MrKim> sorry i can't help you [11:48] <Guest75813> cbreak-work: [11:49] <Guest75813> chiral: [11:49] <Guest75813> cbreak: [11:49] <Guest75813> c10ud: [11:49] <cbreak-work> what's up idiot? [11:49] <cbreak-work> practicing as spammer? [11:49] <Guest75813> [15:15] <Guest75813> Can someone tell me how to compile FFMPEG for ARM-v7 ? [11:49] <cbreak-work> seems like you are practicing as spammer [11:49] <cbreak-work> do that somewhere else [11:50] <relaxed> Guest75813: look here and stop highlighting random people. http://fate.ffmpeg.org/ [11:50] <Guest75813> F*** YOU [11:51] <cbreak-work> F? [11:51] <relaxed> click on the far right link for your arch [11:51] <cbreak-work> FFMPEG YOU? [11:51] <cbreak-work> what the hell is that supposed to mean? [11:51] <ubitux> may the FORCE be with YOU [11:51] <brontosaurusrex> relaxed, do you know if it is possible to make static compile with x11grab? [11:52] <relaxed> I don't think it's possible [11:52] <brontosaurusrex> ok, thanks [11:54] <MrKim> ubitux: i got a crash. [11:54] <MrKim> addr src buffer: 0x13813e0 [11:54] <MrKim> addr dst buffer: 0x1381340 [11:54] <MrKim> it means.. [11:56] <ubitux> MrKim: compare your code with the doxy example [11:56] <ubitux> also you can look at libswresample/swresample-test.c [11:56] <durandal_1707> ubitux: give me force! [11:56] <ubitux> MrKim: oh and check doc/examples/resampling_audio.c [11:57] <ubitux> hey durandal_1707 :) [11:57] <MrKim> ok I'll check [11:57] <MrKim> hi durandal [11:59] <MrKim> A few days ago, i asked same thing here and sir durandal answered me . [12:00] <MrKim> I was thrilled. and i tried ask more precisely [12:01] <MrKim> and..the web irc client ....was stopped.. [12:01] <durandal_1707> usually people answer when they have free time [12:02] <MrKim> I think so.. [12:03] <MrKim> so i'm using mIRC now.. web IRC client sux.. [12:13] <x_> ffmpeg -f jack -i ffmpeg -f x11grab -r 24 -s 1280x720 -i :0.0 -c:v libx264 -y test.avi When doing this I get Audio packet xrun, what is going wrong? [12:14] <x_> Does it have to do with ffmpeg and cpu usage? [12:14] <durandal_1707> it happens with any encoder? [12:16] <x_> Trying another one. [12:25] <x_> Yes. [12:25] <durandal_1707> what encoder you tried? [12:38] <x_> Tried mpeg4, strangely the video is locked at 4 fps when I use mpeg4... [12:38] <x_> wtf [12:39] <x_> if I remove -f jack -i ffmpeg then the fps is no longer limited. [12:40] <x_> why would the encoder matter? [12:42] <x_> if I use pulse I don't get the xrun but the audio cuts out which is the same? [12:46] <dv--> linux audio is braindead [13:14] <x_> There is no work around for this until someone fixes Linux Audio? :P [13:21] <x_> But why does an Xrun totally bork the sync of audio and video? [13:26] <x_> http://pastebin.com/JY5y1X6s [13:26] <Sashmo> Hey guys, does anyone know if there is a way to use a external sync to fix PTS problems when transcoding? [13:39] <xk0bi> hi need help when using Adobe Flash Media Live Encoder i can stream in h264 but when i am using ffmpeg i get black screen [13:41] <xk0bi> ffmpeg -f dshow -i video="Microsoft LifeCam Cinema" -c:v libx264 -r 29.97 -b 800k -s 640x480 -f flv rtmp:///live/f [13:41] <xk0bi> any halp wil be great [13:43] <dmi> hey all [13:44] <dmi> I've got a video that I think has the blue and red channels swapped [13:44] <dmi> I tried using the lutrgb filter, but that fairly obviously didn't work [13:44] <dmi> what's the best way to rebuild the video properly? [13:45] <dmi> I'd love to essentially do: ffmpeg -i video.mp4 -vf "lutrgb=r=b:b=r" [13:55] <x_> back [13:57] <durandal_1707> x_: make sure your machine is really fast to real-time encode audio & video stream [13:57] <dmi> aha, looks like I want the colorchannelmixer filter from trunk [13:58] <dmi> which appeared in the last week or so [13:58] <durandal_1707> dmi: its rgb input or yuv? [13:59] <dmi> durandal_1707: yuv420p [13:59] <dmi> might there be an easier way then? [14:00] <durandal_1707> swapuv ? [14:01] <dmi> durandal_1707: ah, perfect! [14:01] <dmi> thanks :) [14:03] <dmi> I didn't know enough about yuv to know that was what I needed [14:04] <dmi> thanks a lot, durandal_1707 -- you've made my day :) [14:08] <xk0bi> i wqant to stream with ffmpeg in h264 flv can someone know how? [14:28] <xk0bi> my Q is if i am using -vcodec h264 is it ok to send it as flv? [14:33] <durandal_1707> xk0bi: flv supports h264 codec [14:34] <xk0bi> but i see black screen [14:35] <xk0bi> ffmpeg -f dshow -i video="Microsoft LifeCam Cinema" -r 20 -b 500k -f flv rtmp: [14:35] <xk0bi> this works great [14:36] <xk0bi> but if i had -vcodec h264 i got black screen [14:39] <xk0bi> its just that i want to stream in h264 and its not worknig to me [14:41] <durandal_1707> xk0bi: add -pix_fmt yuv420p [14:41] <xk0bi> ok cool i will try [14:42] <durandal_1707> your decoder have limited support for h264 [14:42] <xk0bi> that meen that? [14:44] <durandal_1707> like its only support baseline profile [14:44] <xk0bi> ok cool i will try it thx [14:45] <Sashmo> Hey guys, does anyone know if there is a way to use a external sync to fix PTS problems when transcoding? [14:55] <xk0bi> thx durandal_1707 working great [14:57] <durandal_1707> xk0bi: this is most frequently asked question, and it also on FAQ [15:02] <xk0bi> ok cool [15:02] <xk0bi> i will read the faq now [15:15] <burek> does any of you know of an open source software for counting people using computer vision and a web camera? [15:16] <cbsrobot> burek: use opencv .... [15:17] <burek> im reading about it [15:17] <burek> it woudl be cool if it could be somehow merged with ffmpeg as a filter or something ^^ [15:17] <burek> http://www.youtube.com/watch?v=u7N1MCBRdl0 [15:17] <burek> i know i sound funny, but still :) [15:18] <MozartsGhost> lol, why do you want to count people ? .. are you working for the NSA burek ? [15:21] <burek> no, i was writing an article of all the possible uses of web cameras and video processing tools, like ffmpeg (and opencv) [15:21] <burek> and it seemed like a cool feature :) [15:24] <gmag_> Hello. I wonder if there is a way of decoding a container to YUV file. Maybe a reference to some guide or command? [15:46] <burek> gmag_, did you try something like: ffmpeg -i inputfile.mp4 -c:v huffyuv outputfile.raw [15:47] <burek> or use "ffmpeg -codecs | grep yuv" for more exotic yuv variants [15:54] <gmag_> burek, thanks, I am gonna try that [15:57] <abhilashtt01> I was trying to install ffmpeg 0.10 using atrpms-repo [15:58] <abhilashtt01> previously it was running 0.6.5 [15:58] <abhilashtt01> After installation , I am getting /usr/local/bin/ffmpeg-bin: relocation error: /usr/lib64/libavformat.so.52: symbol ff_raw_pix_fmt_tags, version LIBAVCODEC_52 not defined in file libavcodec.so.52 with link time reference [15:58] <abhilashtt01> Any idea how to fix this? [15:59] <Mavrik> that looks like a version mismatch [15:59] <Mavrik> the libavformat.so is probably provided by a different package [16:00] <abhilashtt01> # rpm -qa | grep -i libavformat [16:00] <abhilashtt01> libavformat52-0.7.11-52.el5 [16:00] <abhilashtt01> libavformat53-0.10.2-54.el5 [16:00] <burek> abhilashtt01, you probably forgot to refresh your ld with ldconfig [16:00] <abhilashtt01> i already done ldconfig [16:00] <abhilashtt01> still having the same issue [16:01] <gmag_> burek, I am getting an "Unrecognized option 'c:v'" [16:01] <gmag_> any idea [16:02] <Mavrik> gmag_, youre using old ffmpeg [16:02] <burek> abhilashtt01: http://brettwalenz.org/2011/08/17/ffmpeg-shared-libraries-relocation-errors… [16:02] <gmag_> Mavrik, I see [16:02] <gmag_> gonna upgrade then [16:02] <burek> gmag_, replace c:v with vcodec [16:02] <gmag_> hmm ok [16:03] <gmag_> burek, yup, it works :) [16:03] <gmag_> thanks mate [16:03] <burek> :beer: thank Mavrik too :) [16:03] <gmag_> thanks mates :) [16:04] <gmag_> btw [16:04] <gmag_> I also tried ffmpeg -i cont.mp4 out.yuv [16:04] <gmag_> does this make any sense? [16:04] <gmag_> it actually creates something, I havent checked yet if it is a yuv... [16:06] <abhilashtt01> it didnt fixed :( [16:06] <vjiaym123> configure: error: i686-pc-linux-gnu/libgomp [16:07] <vjiaym123> I am getting this error [16:07] <burek> gmag_, you're better off specifying the vcodec directly [16:07] <burek> abhilashtt01, i pointed you to the direction where to look for issues [16:07] <burek> that thing is not so general that i could narrow it down exactly where your problem is [16:08] <burek> you need to figure out what is being linked against what and what's causing your issues [16:08] <burek> btw, did you compile your ffmpeg or what? [16:08] <abhilashtt01> ok [16:08] <abhilashtt01> I was trying to install using yum [16:14] <burek> probably you have somehow installed two versions of ffmpeg (probably a fork libav and ffmpeg) or something similar, which caused that confusion.. [16:15] <burek> if you can, uninstall all ffmpegs that you have (and libav) [16:15] <burek> and compile your ffmpeg, since it doesnt take that long to do it anyway and its not complicated [16:16] <abhilashtt01> I want to intsall the latest version of ffmpeg [16:16] <abhilashtt01> 0.10 [16:16] <abhilashtt01> do you have any download links / [16:18] <burek> 0.10 is old [16:30] <abhilashtt01> Ok, I want to run hwdmediashare in this server and it needs 0.7.1+ [16:33] <vijaym123> Hello, when I tried to build the ffmpeg as show here http://pastebin.com/gMnrfM3C [16:34] <vijaym123> I got a error and I pasted it in the same paste bin..please help :) [16:35] <Mavrik> vijaym123, hmm, that looks like a compatibility problem [16:35] <Mavrik> which device are you targeting? [16:35] <vijaym123> I am building it for cpu=arm7 and arch=arm [16:35] <Mavrik> hmm, it compiles perfectly for me on --cpu=armv7-a [16:36] <vijaym123> Oh thanks will try that ! [16:36] <Mavrik> for android that is [16:36] <Mavrik> so --arch=arm --cpu=armv7-a [16:36] <vijaym123> I want to build it for armv7 [16:38] <vijaym123> What should I do if I need to build it for armv7? [16:39] <vijaym123> make [16:40] <b_jonas> man, this was fast, thank you (I reported a bug and a developer has fixed it within an hour) [16:40] <burek> abhilashtt01, then git clone ffmpeg version 0.7.1 and compile it [16:40] <abhilashtt01> ok [16:40] <abhilashtt01> what is the latest version of ffmpeg [16:40] <burek> b_jonas, btw, one of the letters in 'ffmpeg' stands for 'fast' :) [16:41] <b_jonas> really? [16:41] <b_jonas> I mean, I know it runs fast [16:41] <b_jonas> that's why I'm using it [16:41] <b_jonas> but I didn't know development was also fast [16:41] <burek> :) [16:42] <vijaym123> Can anyone help me with compiling ffmpeg source with armv7 and linux? thanks in advance. I tried with the following configure statement http://pastebin.com/gMnrfM3C, but in vain [16:42] <burek> vijaym123, did you check our wiki [16:42] <burek> there is a tutorial for raspberry pi, which might give you some clues [16:43] <vijaym123> yeah...even googled it but couldn't find any source. :) [16:43] <b_jonas> maybe that's why there are two f-s in the name [16:43] <abhilashtt01> when I tried to compile , I am getting this error [16:43] <abhilashtt01> yasm not found, use --disable-yasm for a crippled build [16:43] <abhilashtt01> If you think configure made a mistake, make sure you are using the latest [16:43] <abhilashtt01> version from Git. If the latest version fails, report the problem to the [16:43] <abhilashtt01> ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. [16:43] <abhilashtt01> Include the log file "config.log" produced by configure as this will help [16:43] <abhilashtt01> solving the problem. [16:44] <b_jonas> abhilashtt01: well install yasm [16:45] <vijaym123> But rassberry pi is not arm v7 architecture [16:46] <b_jonas> abhilashtt01: see http://yasm.tortall.net/ [16:46] <abhilashtt01> got it.. Now getting this error ERROR: libfaac not found [16:51] <abhilashtt01> gsm is already installed, but duing compile getting error ERROR: libgsm not found [16:56] <burek> abhilashtt01, for all errors like: ERROR: <insert_library_here> not found [16:56] <burek> google for the library and see how to install it on your arm device [16:56] <burek> and then repeat configure for ffmpeg [16:57] <burek> btw, there is an internal aac decoder/encoder which you can use also, if you can't install libfaac for any reason [17:00] <MozartsGhost> hi guys .. if I transcode an input stream with ffmpeg and output it to flv with rtmp my audio works, but if I transcode a stream, and then pipe the output to a second ffmpeg with -c copy and output to rtmp, it gives me an error saying malformed bitstream detected, and that I should use -bsf:a aac_adtstoasc, .. and if I do that, then the stream works but audio disapears from the stream [17:01] <abhilashtt01> ok [17:01] <MozartsGhost> anyone have an idea why that could be ? [17:03] <MozartsGhost> nevermind, got it working. [18:51] <xlinkz0> how do i find out if a video has constant or variable bitrate? [18:51] <xlinkz0> and will seek by bytes be accurate on variable bit rate? [19:09] <Mavrik> xlinkz0, there's really no such thing as CBR video for most formats [19:09] <Mavrik> and no, seek by bytes won't be accurate [19:09] <xlinkz0> thanks [19:24] <xlinkz0> what does isnan() do? [19:25] <teratorn> xlinkz0: presumably checks for NaN (not-a-number), a special floating-point value. [20:18] <xlinkz0> http://codepad.org/3Hyvd8Ly [20:19] <xlinkz0> this supposedly takes 0 miliseconds and succedes [20:20] <xlinkz0> on a raw video file of 2 hours, i doubt that's the case :\ [20:22] <xlinkz0> this is more illustrative http://codepad.org/rNxlmrIp [20:25] <xlinkz0> however if i don't substract 100k from pos i get Operation not permitted and it takes 12 seconds [20:36] <blez> hello [20:36] <blez> I'm trying to convert a file to flac -ac 1 -ab 8000 [20:36] <blez> but it seems it's skipping the flags [20:37] <LithosLaptop> you can't set bitrate on flac... [20:37] <blez> why [20:37] <LithosLaptop> it's lossless :) [20:37] <blez> er. [20:37] <blez> how does vlc does it then? [20:37] <blez> it does set a bitrate [20:37] <LithosLaptop> it will take as much as it needs to be lossless [20:38] <blez> I should record to wav and then flac? [20:38] <LithosLaptop> no you can directly record to flac I think [20:39] <LithosLaptop> VLC can set bitrate for FLAC? [20:39] <LithosLaptop> if you want to set a bitrate then select a lossy codec [20:39] <relaxed_> levels of compression [20:41] <relaxed_> blez: ffmpeg -h encoder=flac [20:41] <LithosLaptop> yes, levels of compression makes sense for flac [20:42] <blez> no no, I need it for a web api that needs flac with bitrate 8000, whatever that means [20:42] <blez> else it refuses it [20:42] <LithosLaptop> maybe it needs a sample rate of 8000Hz? [20:43] <blez> maybe [20:43] <blez> how to set that ? [20:43] <LithosLaptop> think -ar [20:43] <LithosLaptop> lemme check [20:45] <LithosLaptop> -ar 8000 [20:45] <blez> thanks [20:45] <blez> that was it [20:45] <LithosLaptop> no prob :) [20:57] <x_> http://pastebin.com/cgcye2Nr Audio will not be synced with video depending. [20:57] <x_> Something I can do? [21:08] <Sashmo_> anyone know if there is a way to use external sync when transcoding to avoid PTS errors? [22:04] <midiane> hello all. [22:05] <midiane> I've built ffmpeg from source (latest version). when i run -version it still says 0.5.1 even though I've downloaded and built the latest version. I'm not sure what I'm doing wrong. [22:06] <midiane> http://pastebin.com/abxCEMa4 [22:07] <sacarasc> Did you have one in /usr/bin and one in /usr/local/bin? [22:09] <midiane> let me check. i was running it from a separate directory but i guess it goes to /usr/bin first... [22:12] <midiane> weird. there's nothing in /usr/bin but the latest version is in /usr/local/bin. [22:12] <midiane> the one being picked up is from macports. ill uninstall from there. [22:12] <midiane> do i then just copy the one in /usr/local/bin to /usr/bin [22:12] <midiane> ? [22:13] <durandal_1707> you did not installed it? [22:13] <midiane> the 0.5.1 version was installed by macports. uninstalling it now. [22:20] <midiane> sorry, battery died. after uninstalling 0.5.1 version via macports, i get this error. macbookpro:tools midiane$ ffmpeg -version [22:20] <midiane> -bash: /opt/local/bin/ffmpeg: No such file or directory [22:22] <ackjewt> midiane: stop using macports and start using homebrew.... [22:23] <midiane> ackjewt: i just uninstalled the macports version. I'm not sure how to change the path now to point to homebrew... [22:24] <midiane> this is weird& http://pastebin.com/Ljv7HNQZ [22:25] <ackjewt> ln -s /opt/local/bin/ffmpeg /usr/bin/ffmpeg [22:25] <ackjewt> or something.. .) [22:25] <midiane> oh.. ok. [22:29] <midiane> didn't work. looks like i have to start from scratch. [22:29] <ackjewt> or just use /usr/bin/ffmpeg [22:32] <midiane> ok. [22:38] <cbsrobot> midiane: http://ffmpeg.org/trac/ffmpeg/wiki/MacOSXCompilationGuide [22:38] <cbsrobot> it's copy pastable ! [22:39] <midiane> oh. thanks. :) i thot homebrew meant compile from source. lol got the command working now... [22:39] <midiane> thanks to all for your help. [22:46] <midiane> the command line is powwa. [22:54] Action: midiane leave [00:00] --- Wed Apr 24 2013

1 0

ffmpeg-devel.log.20130422
by burek 23 Apr '13

23 Apr '13

[00:47] <ubitux> saste: i'm going to test your filter :) [00:49] <saste> ubitux, hue is still borken [00:49] <saste> (see patch) [00:51] <ubitux> ok, just a moment [00:51] <ubitux> i'm writing something cool :) [00:53] <saste> ubitux, what? [00:53] <ubitux> saste: http://pastie.org/7693879 [00:53] <ubitux> can you try this? [00:54] <ubitux> (now you should have various shell features, including history etc, and you can easily add some completion routines) [00:57] <ubitux> i'm looking at the hue patch now [00:59] <saste> uhm i wonder it could mess with the 3.x/2.x thing [00:59] <ubitux> :* [00:59] <ubitux> :( [00:59] <saste> i have python binding for 2.x, but locally installed the 3.x API [01:00] <saste> i suppose i'll downgrade the API [01:00] <ubitux> i have zmq3 api, and python2 and 3 bindings here [01:00] <ubitux> i can adjust the code if necessary [01:00] <ubitux> (to python3) [01:00] <saste> but should we prefer 2.x over 3.x? [01:01] <ubitux> you mean for zmq api? [01:01] <saste> 3.x is relatively new, and thus not included in a lot of distros [01:01] <saste> yes [01:01] <ubitux> i have only zeromq3 packaged here [01:01] <saste> ubitux, what distro? [01:01] <ubitux> archlinux [01:01] <ubitux> http://pastie.org/7693937 [01:02] <saste> well i don't know [01:02] <ubitux> it's new code, you can request bleeding edge technos since you don't have to care about retro compat [01:02] <ubitux> and if necessary, you can add some additionnal compatible code [01:03] <ubitux> but who cares about debian anyway ;) [01:03] <saste> ubitux, no i don't like that [01:03] <saste> i mean both versions, it's a mess to test [01:03] <ubitux> the latest stable only, 3. [01:03] <saste> my tendency is to favor 3.x and be done with that [01:04] <ubitux> same. [01:04] <saste> now I don't know what I have to do in order to install the required python modules [01:04] <saste> that's why i wrote a c client [01:05] <ubitux> :( [01:05] <ubitux> but my script is cool :( [01:06] <saste> :) [01:06] <saste> well i'll sleep and let you test the shiny zmq code [01:07] <ubitux> i'm on the hue review now [01:07] <ubitux> 'night :) [01:07] <saste> the main problem right now is that it used hardcoded buffer size [01:07] <saste> the zmq message API should allow to overcome that limitation I guess [01:08] <saste> ok, thanks for reviewing [02:34] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:de9fbf3dc41a: indeo4: implement haar 8x1 and 1x8 transforms [02:43] <Compn> where is maksym and his indeo patches ? [02:52] <highgod> Hi, I want to ask a question, what the usage of mux and demux? [02:52] <highgod> what can they do?thanks [02:53] <Compn> mux creates the container , demux takes apart the container and sends the data to codecs [02:56] <highgod> Compn:thanks [02:57] <highgod> another question, how can I do the benchmark, you have told me, but I forgot it, sorry. [03:09] <BBB-> someone broke msvc again [03:10] <ubitux> it's a conspiracy [03:12] <ubitux> i guess it's 76c7277385120741914168d02b57a7a1acf87a9b [03:27] <Compn> highgod : something with oprofile and start_timer stop_timer in the code. but i dont remember exactly. i think its written down in the docs somewhere [03:28] <highgod> OK [03:28] <Compn> its in timer.h file [10:55] <cone-735> ffmpeg.git 03Luca Barbato 07master:1c88617b9421: riff: K&R formatting cosmetics [10:55] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:e381cc3e4ea6: Merge commit '1c88617b9421c9af8ae961c05d2381b19c585026' [11:51] <cone-735> ffmpeg.git 03Luca Barbato 07master:8e329dba378c: riff: check for eof if chunk size and code are 0 [11:51] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:369e5dcfc530: Merge commit '8e329dba378cef0ff6400c7df9c51da167d5a1f0' [12:03] <dekarl> hi, has this fix ever been commited? I'm not seeing it in master http://ffmpeg.org/trac/ffmpeg/ticket/1240 [12:25] <durandal_1707> michaelni: are you in middle of something? [12:26] <av500> otherwise he would be on the edge of the world [12:33] <cone-735> ffmpeg.git 03Peter Ross 07master:e544782623ca: riff: Move guid structs and helper functions into riff [12:33] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:52e40155a8e1: Merge commit 'e544782623caf1eb4b2429e9c4f35d6b627accba' [12:34] <durandal_1707> av500: something like that [12:41] <durandal_1707> michaelni: what you think about sh4 code removal? [12:41] <cone-735> ffmpeg.git 03Peter Ross 07master:a3bf80a5aea5: riff: Perform full lookup on WAVEFORMATEXTENSIBLE subformat guid [12:41] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:05deb05fe9ce: Merge commit 'a3bf80a5aea577663354bd538e345b283bcb73de' [12:46] <J_Darnley> dekarl: Yes, if got log is to be believed [12:46] <J_Darnley> 364c71c80e9124a2624e3bfeb8e84c5cddeda222 says it fixes 1240 [12:47] <J_Darnley> ... that was supposed to be "git log" [12:52] <durandal_1707> dekarl: if bug is still present, reopen bug and upload file with steps how to reproduce bug [12:52] <michaelni> durandal_1707, do we have someone who wants to maintain sh4 ? [12:52] <cone-735> ffmpeg.git 03Peter Ross 07master:516089d5d86b: riff: Add ACTRAC3+ guid [12:52] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:51286aaffcad: Merge commit '516089d5d86b6809a9458f9cf171f2bc9a8e2a25' [12:55] <ubitux> btw, i'll apply the lavfi timeline tonight [12:57] <durandal_1707> michaelni: so no maintainer -> rm -rf * [12:58] <av500> sh4 is dead [12:58] <av500> renesas told me [12:58] <av500> they went arm [13:01] <durandal_1707> av500: citation needed [13:01] <cone-735> ffmpeg.git 03Luca Barbato 07master:b845f5e97b65: riff: Factor out WAVEFORMATEX parsing [13:01] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:645d0384e85b: Merge commit 'b845f5e97b655de0a191f736594777fec9754cf5' [13:02] <michaelni> durandal_1707, no maintainer -> code might break, and when it breaks without anyone willing to fix then removal is the only option [13:02] <michaelni> otherwise ffmpeg wouldnt build/work anymore on sh4 ... [13:05] <av500> durandal_1707: renesas guy to me at last embedded world [13:05] <av500> when I asked him for HW to keep sh4 alive [13:08] <cone-735> ffmpeg.git 03Anton Khirnov 07master:9bfc6e02bae9: afifo: fix request_samples on the last frame in certain cases [13:08] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:e4c0f258d38a: Merge commit '9bfc6e02bae9de354fb9ba09a8a140e83eeadf7d' [13:10] <dekarl> J_Darnley: thanks for finding the commit [13:10] <dekarl> durandal_1707: looks like I have to rework our transcoder to look for the frame_size/duration in the new place. http://code.mythtv.org/trac/ticket/2077#comment:42 [13:13] <cone-735> ffmpeg.git 03Diego Biurrun 07master:311a592dfcfc: x86: Remove some duplicate function declarations [13:13] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:2288c7768970: Merge remote-tracking branch 'qatar/master' [14:28] <cone-735> ffmpeg.git 03Paul B Mahol 07master:0327400ee31f: doc/filters: add examples for stereo3d [14:28] <cone-735> ffmpeg.git 03Paul B Mahol 07master:2f11aa141a01: lavfi: remove mp=down3dright [15:07] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:80d4d3a41de6: fate: re-enable af_join test [16:05] <cone-735> ffmpeg.git 03highgod0401 07master:6d7ec4fc2ef7: lavu/opencl:optimize compile kernel operation [17:40] <burek> hi guys :) [17:41] <burek> i don't want to bug you, but does anyone know what to answer to this guy: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=897 [18:34] <cone-735> ffmpeg.git 03Diego Biurrun 07master:78fa0bd0f706: x86: cavs: Put mmx-specific code into its own init function [18:34] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:f84e373797eb: Merge commit '78fa0bd0f7067868943c0899907e313414492426' [18:38] <cone-735> ffmpeg.git 03Diego Biurrun 07master:fce99322b0cc: h264: Drop unused variable [18:38] <cone-735> ffmpeg.git 03Diego Biurrun 07master:a0d5ca4f0a8e: mxfenc: Use correct printf format specifier for int64_t [18:38] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:74e1aa17df99: Merge commit 'a0d5ca4f0a8e2c34d784d503a12af6303424ac6a' [18:45] <cone-735> ffmpeg.git 03Diego Biurrun 07master:b4ad7c54c878: x86: cavs: Refactor duplicate dspfunc macro [18:45] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:430d69c9421f: Merge commit 'b4ad7c54c878dead7dfa4838b912a530c1debe85' [19:05] <cone-735> ffmpeg.git 03Diego Biurrun 07master:0c15a9aa7e16: sh4: Remove dubious aligned dsputil code [19:06] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:7a556ebccfd6: Merge commit '0c15a9aa7e1654a19144eb594f9639a57fd47482' [20:14] <cone-735> ffmpeg.git 03Nicolas Bertrand 07master:c81a70638116: JPEG 2000 decoder for DCinema [20:14] <cone-735> ffmpeg.git 03Diego Biurrun 07master:f13888afcc5e: avcodec: Bump minor for JPEG 2000 decoder [20:14] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:c1c2b0b33904: avcodec: rename jpeg2000 decoder to j2k [20:15] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:e2e9bee2daef: Merge remote-tracking branch 'qatar/master' [20:36] <mateo`> hello guys :) [20:37] <mateo`> testing the brand new jpeg2000 decoder, is there a way to convert xyz colorspace to other format ? [20:37] <ubitux> mateo`: there is a pending hack filter to do the convert [20:37] <ubitux> it doesn't apply anymore though [20:37] <ubitux> the other side may be working on it, but they prefer not to answer me when i asked [20:38] <mateo`> my sudden excitement has just vanished :( [20:40] <mateo`> anyway the decoder does not like the SD samples i have [20:40] <nevcairiel> its only optimized for DCinema jpeg2000 profiles [20:43] <j-b> indeed [20:44] <j-b> and it's currently slow [20:44] <michaelni> slower than the old j2k decoder ? [20:44] <j-b> probably, because it has 0 asm in it [20:45] <j-b> but, it will be faster [20:45] <j-b> and it is in the code, so easier to leverage ASM skillz [20:48] <nevcairiel> its a brand new decoder, clean and working implementation is the first goal, optimizing can start now [20:48] <nevcairiel> the dwt code can probably be simded [20:50] <j-b> indeed [20:50] <j-b> And yes, so far, it is XYZ DCP profiles, because we need it... [20:51] <ubitux> we don't yet optim for libavcodec/j2k_dwt.c i guess? [20:51] <michaelni> its based on kamil nowosads decoder like ours, so iam not sure "brand new" is a completely correct description [20:51] <ubitux> it's the exact same basis, the guy wasn't aware that is was upstream here [20:52] <ubitux> and of course no one told him when he start working on it :p [20:53] <j-b> Well, this one works. Sorry to be harsh, but I can't say the same for the other implementations [20:54] <michaelni> j-b, no harshness here, we want the best code ... [20:55] <j-b> michaelni: and for this work, we care about DCP. [21:00] <michaelni> also nicolas code should contain all bugfixes from ffmpeg according to him [21:01] <michaelni> so the "other" decoder can possibly be droped soon [21:49] <j-b> michaelni: Yeah, I told him to look at it [21:54] <cone-735> ffmpeg.git 03Michael Niedermayer 07master:2ae91c86f386: avcodec_get_context_defaults3: set codec_id [22:11] <cone-735> ffmpeg.git 03Piotr Bandurski 07master:f4596e8bb6f7: mjpegdec: Add support for 4:1:1 YUV [23:17] <ubitux> http://prog21.dadgum.com/170.html [23:53] <michaelni> Is someone looking into the MSVC x86-64 aac-sbr noise compile failure ? [23:54] Action: ubitux isn't [23:54] <michaelni> iam no windows or msvc expert and dont have the compiler setup ... [00:00] --- Tue Apr 23 2013

1 0

ffmpeg.log.20130422
by burek 23 Apr '13

23 Apr '13

[04:57] <green_deep> I found some packet's pos return by ffmpeg is -1, pkt->pos=-1, why this could happen? [08:28] <teptanic> Can FFMPEG be used to check a video file for errors? How? [08:28] <teptanic> --without writing another file, preferably. [08:53] <hazardous> hi [08:53] <hazardous> quick question, i am using ffmpeg -ss 00:50:13.01 -t 00:52:27.00 -i a.flv -acodec copy -vcodec flv -async 1 out.flv [08:53] <hazardous> it splits successfully [08:53] <hazardous> but the flv when played is near instantly fast forwarded through [08:53] <hazardous> and shows up with a timestamp like 35175:41 / 2:13 [08:54] <hazardous> is there any way to fix [10:40] <relaxed> teptanic: ffmpeg -i input -f null - || echo "error" [10:41] <teptanic> relaxed: thanks. Does that just copy the streams to null, reporting if error in packaging is encountered? [10:43] <teptanic> I see, it decodes it. (so it says) [10:43] <relaxed> right, to raw audio/video [10:45] <relaxed> you could also encode to a specific format and send it to /dev/null, ffmpeg -i input -f matroska ->/dev/null [10:46] <relaxed> of course -f null should be much faster [11:04] <MrKim> Hello, I'm going to resample audio with libswresample, and SWR gives error msg to me "Input channel count and layout are unset" [11:05] <MrKim> How can I set them? example code is helpless.. [11:13] <durandal_1707> MrKim: with av_opt_set_int() [11:13] <durandal_1707> MrKim: and that is in example, what example you are using? [12:02] <delicado> hi, what library do ffmpeg uses to decode the shockwave flash files (.swf)? [12:03] <delicado> i mean, what library does libavcodec uses to decode the swf files, im sure it isn't from macromedia or adobe right? can you guys help me know what it is? [12:09] <durandal_1707> delicado: libavcodec uses libavcodec for swf [12:12] <delicado> yeah, but libavcodec is the "whole", i mean the thing is made up of open-source external libraries. so im guessing the swf decoder came from somewhere else and not hardcoded into libavcodec. im wanting to use swf as a GUI frontend for my app, but i want the open-source alternatives. [12:13] <delicado> i found gameswf, but it is dead as of 2003. [12:14] <durandal_1707> swf in libavformat&libavcodec supports only video & audio streams [12:14] <ubitux> swf is format only, so libavformat [12:15] <ubitux> and indeed the support is partial (no support for all the vector stuff) [12:17] <delicado> durandal_1707: i tried playing swf games in it (mplayer classic), mplayer displayed the thing correctly, it even responded to keyboard and mouse input, but i know mplayer handled the input handling, still, buttons, text, sprites and stuff was displayed correctly. [12:17] <delicado> i mean ubitux, [12:17] <ubitux> mplayer classic? wut? [12:17] <ubitux> media player classic you mean? [12:17] <delicado> yeah. [12:18] <ubitux> it might embed a flash player [12:18] <ubitux> you can check if the FFmpeg libs support it by just ffplay the file [12:21] <delicado> maybe, but i tried removing my adobe flash player, and its still working. im guessing its using SWFDEC cause it is what's used in firefox, but i might have to take a look [12:24] <ubitux> i assume it's on windows [12:24] <ubitux> on windows, afaict, there is copy of stuff everywhere [12:25] <ubitux> so it's not unlikely you have a copy of the flash plugin in the mpc directory or even statically linked somewhere [12:25] <ubitux> swfdec? used in firefox? what are you talking about [12:25] <ubitux> swfdec isn't maintain for a long time now [12:25] <ubitux> and firefox supports the official flashplugin [13:26] <xk0bi> hi i am tring to ffmpeg to fms h264 like ffmpeg -f dshow -i video="Logitech QuickCam Pro 9000" -c:a copy -c:v libx264 -r 20 -b 800k -s 640x480 -preset baseline -f flv rtmp:... and all i get is black screen can someone help me with that [13:46] <viric> Hello [13:46] <MozartsGhost> xk0bi: I am actually just testing now with RTMP from ffmpeg to wowza [13:47] <viric> I've some broken mpeg2 files... trying to move them to mkv, ffmpeg complains: [13:47] <viric> Can't write packet with unknown timestamp [13:47] <MozartsGhost> seems to be working nicely, except my display size is wrong from wowza, versus rtp which is working [13:48] <MozartsGhost> not sure if rtmp is fully working properly with ffmpeg, I don't know, at least I am getting video and audio, but video is the wrong size, sort of stretches [13:52] <viric> mh genpts :) [13:52] <MozartsGhost> o_O heh [13:57] <Gozzy> good afternoon, i am trying to flattening down a 16 channel prores audio stream to stereo; anyone who has some tips to do this with ffmpeg, sox or somesort of cmd line tool? [13:57] <xebra> hey, I solved the problem yesterday. With x256 (libx264) it forces some kind of automatic (and variable) frame rate. To really change the frame rate and have a fixed one you need to add "-vsync 1" [14:08] <xebra> I have another question though. If I am encoding a video at 1fps, should I change the settings for the gop? I see the settings in the presets are like g=250, keyint_min=25, but I guess those are good values for higher frame rates (25fps) [14:18] <xebra> ok, I'll answer myself: I guess I don't need to mess with those parameters, because ffmpeg will adjust them automatically according to the frame rate. With ffprobe I checked a file I encoded at 1fps, and it says gop=12. So I guess that's ok [14:21] <MozartsGhost> xebra .. I think it should probably adjust be itself :) hehe [14:25] <MozartsGhost> I guess if your fps is 1, you could just divide your keyint and keyint_min by 25 ? .. [14:30] <Bor0> when building ffmpeg on mingw with --enable-openssl, I get: #error "ws2tcpip.h is not compatible with winsock.h. Include winsock2.hinstead.". however, without openssl it compiles fine. what can I do? [14:31] <MozartsGhost> Question: has anybody experienced any strange display size issues while send RTMP to Wowza or something similar ? .. when I use RTP(udp) to send a stream (720x576) from ffmpeg to wowza with RTMP wowza receives it and says its 720x576, but when u view the stream with JWPlayer it stretches it likes its 4:3 instead of 16:9, however when u use RTP its fine ?, .. I have sent wowza support an email, [14:31] <MozartsGhost> waiting to hear what they say, so not sure if its wowza or ffmpeg giving the issue. trying to figure it out, anyone have some advice ? [14:32] <MozartsGhost> I have litteraly tried almost every -aspect -s size settings, and everything that I can .. but still, display issue. [14:37] <MozartsGhost> I might add, this is a really serious issue for me, I have 32 encoders running 4 channels each, now all of them running ffmpeg (previously vlc) ;) thank you ffmpeg ;) however, RTP is giving network problems, lost frames and such. so want to switch to RTMP which is tcp, more reliable. but now have this display size issue .. [14:51] <chattychatty> Hello guys, I'm trying to compile ffmpeg in CentOS 6 [14:52] <chattychatty> and when I use the configure command for ffmpeg following this guide: http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide [14:52] <chattychatty> I get this error back: [14:52] <chattychatty> libfdk_aac is incompatible with the gpl and --enable-nonfree is not specified. [14:53] <chattychatty> Any ideas? Should I just add the nonfree option? Any conflicts with that or is it just an informality about licensing ? [14:53] <divVerent> the latter [14:53] <divVerent> it basically means you cannot give away the ffmpeg package you then made [14:54] <divVerent> so you can't use this combination for any distributed binary packages, but it's fine for your own system [14:54] <chattychatty> Thanks, should I just remove the previous --enable-gpl directive? [14:54] <chattychatty> Or include both? [14:56] <MozartsGhost> just use both I guess [14:56] <MozartsGhost> is libfdk much better than libfaac ? .. I've just been using libfaac [15:20] <divVerent> 14:56:46 MozartsGhost | is libfdk much better than libfaac ? .. I've just been using libfaac [15:20] <divVerent> y14:56:46 MozartsGhost | is libfdk much better than libfaac ? .. I've just been using libfaac [15:20] <divVerent> yes, clearly [15:26] <MozartsGhost> cool, thanks, I will try it out :) [15:43] <Razorblade> Hi! I've got a probably very simple question but it seem I can't get together with the CLI. I'd like to upscale a Video (mov) from 720p to 1080p, add a watermark (png) and save it to a new file (mp4), but even if I try resizing to 1080p it produces an output of 720p. Could someone help me with the syntax please? [15:46] <MozartsGhost> -s 1920x1080 -aspect 16x9 ? .. [15:46] <Razorblade> I was using -vf "scale=h=1080:-1" . I'm using windows btw. [15:50] <Razorblade> sry http://pastebin.com/QShcPwJy [16:05] <DonGnom> in which borders could i specify the quality setting for vorbis audio? (-q:a/-aq) [16:06] <klaxa> if i'm not mistaken, 1-9 float [16:06] <DonGnom> klaxa: thx i found that in the mean time http://ffmpeg.org/trac/ffmpeg/wiki/TheoraVorbisEncodingGuide it says 0-10 [16:07] <klaxa> ah... heh i'll try to remember that [16:19] <MozartsGhost> Razorblade: I'm no ffmpeg expert, Yet, but I would try to use "-s 1920x1080 -aspect 16x9" instead of -vf .. [16:22] <Mavrik> Razorblade, hmm, I just think your scale parameters are off [16:22] <Mavrik> -vf scale=-1:1080 [16:23] <Razorblade> MozartsGhost: Yeah, that did the trick. But only if I skip the -aspect parameter. Mavrik: Will try that also [16:23] <Mavrik> also [16:23] <Mavrik> please don't encode videos without specifying parameters for quality [16:24] <Mavrik> it'll break your quality :) [16:26] <Razorblade> Well I haven't used any parameters jet because I watched both videos and it doesn't seem to lack some quality. Maybe a little bit. But the source recording was allready somwhat shaky so it could be. I'll consider that when I'm using a mounted camera next time. [16:26] <exutux> hi guys, how can I encode a video for to see it on iphone? [16:26] <MozartsGhost> cool man [16:26] <exutux> I mean in streaming with opera and iphone [16:27] <MozartsGhost> exutux: once again, i'm no expert, but I can give u the settings I am using for iphone users ? .. paste me your current command in pastebin .. then I will modify it for you [16:27] <Razorblade> Mavrik: Your parameter worked as good as Mozarts. Thanks for that. [16:28] <Razorblade> I'll take a look again about overlaying the watermark and come back if I can't figure it out. Thanks Guys :) [16:29] <exutux> MozartsGhost: uhm I don't have any command now... [16:29] <MozartsGhost> Razorblade: I saw something in the documentation about adding images with filters .. but can't remember now, pretty sure its in the ffmpeg documentation tho .. [16:30] <exutux> MozartsGhost: I will know which kind of codec can I use [16:30] <exutux> mpeg4? or what and which type of container ... mp4? [16:30] <Razorblade> Yeah, I've seen it. I believe it was Filters 8:55 but the wrong scale syntax probably messed it up. [16:30] <MozartsGhost> oh, mpegts container, and x264 codec with aac audio codec [16:31] <MozartsGhost> exutux: I will give you my x264 codec settings for old apple devices [16:31] <exutux> thanks [16:34] <MozartsGhost> http://pastebin.com/vBcrSWD8 [16:35] <MozartsGhost> sorry, lol, change keyint=25 to 250, I made a typo [16:36] <exutux> MozartsGhost: I will try [16:36] <exutux> MozartsGhost: the output.mp4 true? [16:37] <MozartsGhost> output -f mpegts or output.ts, I will be honest I have not tried .mp4 but I guess it would work .. I re-stream via wowza .. [16:39] <exutux> MozartsGhost: ok I will try and will send you a feedback :D [16:40] <MozartsGhost> cool man, hope it works ;) [17:03] <Razorblade> Allright, im back from testing and I figured it out so far. I've got now a syntax which resizes the video from 720 to 1080 using Mavriks Syntax and added a Watermark (http://pastebin.com/FzT5QudU) Unfortunately, the quality suffered (as expected) do you people know good quality settings for youtube uploads? [17:30] <exutux> MozartsGhost: it does'n works for safari and html5 video :( [17:46] <Razorblade> Hey mavrik, I think you haven't seen it before. Just want to let you guys know that I was able to figure the right syntax out (http://pastebin.com/FzT5QudU) And yeah, the quality decreased... [17:46] <Mavrik> Razorblade, yep, use "-crf" to control quality :) [17:46] <Mavrik> (when using libx264 to encode H.264) [17:48] <Razorblade> ty :) may you have any suggestions for a decent youtube video? [17:50] <Mavrik> Razorblade, uhm& "23" is a good baseline [17:50] <Mavrik> Razorblade, the lower you go, the better quality (and larger filesize) [17:51] <Razorblade> thanks. 23 is default, isn't it? [17:51] <Mavrik> Razorblade, um, largely depends on which x264/ffmpeg combo you have [18:03] <Razorblade> allright, thanks for your help guys. gtg. Bye :) [19:22] <Gozzy> good evening, is it possible to output to 2 files? something like ffmpeg -i inputfile -ss 00:00:00 -t 00:30:00 first30minoutputfile -ss 00:30:00 -t 01:00:00 second30minoutputfile [19:30] <Mavrik> Gozzy, it is, but I think time control parameters work on input so you can't do what you want [19:33] <Gozzy> hmm ok thx Mavrik [19:36] <Gozzy> when you play a file in ffplay you can see some other kind of timing mechanism (4206 instead of 01:10:x) do you know whats that is? (it does work when you use it in ffmpeg instead of a timestamp) [19:38] <Mavrik> Gozzy, those are probably seconds :) [19:41] <Gozzy> lol right [19:41] <Gozzy> :$ graci [20:18] <grobda24> I've been recommended this command from multiple places ... "ffmpeg -vcodec copy -acodec copy -i djbarney_hydrogen_qmidiarp_yoshimi_to_ardour_mixing_desk_sync_test.ogv -ss 0 -t 360 djbarney_hydrogen_qmidiarp_yoshimi_to_ardour_mixing_desk_sync_test_CUT.ogv" [20:19] <grobda24> Ffmpeg reports ""Unkown decode 'copy'" [20:19] <grobda24> This is on ffmpeg version 1.04 [20:21] <ubitux> move -vcodec copy -acodec copy as output option (aka after the -i ...) [20:21] <ubitux> also, you can just -c copy [20:21] <ubitux> ffmpeg 1.0 is old, you should upgrade [20:29] <Mavrik> yp [20:29] <grobda24> ubitux: ok, thanks. I'll try that [22:49] <rainmaker1> Hi, does anyone know if ffmpeg or any other tool can detect streams that have audio/video out of sync? [22:51] <klaxa> how would you detect that? [22:53] <rainmaker1> If I know how to detect that I would not ask :) [23:02] <Diogo> hi what is the command to loop a input file for rtmp streaming? [23:02] <Diogo> command : /servers/ffmpeg/bin/ffmpeg -re -i "4914731_vIhLy.mp4" -vcodec libx264 -s 1280x720 -f flv -acodec libfaac "rtmp://server" [23:02] <Diogo> -loop works? [23:08] <Diogo> please this is possible? [23:54] <lullabud> question about metadata. i'm trying to have ffmpeg copy all metadata, including Camera tags (as shown by "exiftool -G2 filename"), but can't find a way to do it. i've tried "-map_metadata 0:s:0" which i read was supposed to copy all metadata, but it's not working. [23:54] <lullabud> anybody know how to do it? [23:57] <lullabud> here's some terminal output showing what i'm doing - http://sprunge.us/gTdY [00:00] --- Tue Apr 23 2013

1 0

ffmpeg-devel.log.20130421
by burek 22 Apr '13

22 Apr '13

[00:02] <michaelni> looks cute [00:13] <ubitux> btw, the documentation of the project is insanely good [00:27] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:63090842df76: av_samples_alloc_array_and_samples: Fix sizeof type. [00:27] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:578d9cf7ccd3: mxf_timestamp_to_str: dont leave uninitilaized fields in time. [00:37] <ubitux> saste: i will apply the fade patch maybe tomorrow [00:37] <ubitux> unless you have some comments [00:38] <saste> ubitux, I didn't read the patch but I trust you [00:38] <ubitux> you shouldn't [00:38] <ubitux> i don't even trust myself [00:38] <saste> assuming it doesn't break the syntax [00:38] <saste> well i don't trust me as well [01:56] <ubitux> anyone on hydrogenaudio to request that guy to share a sample? http://www.hydrogenaudio.org/forums/index.php?showtopic=100448&hl= [02:38] <cone-132> ffmpeg.git 03Roberto Togni 07master:8017683647f5: qdm2: initialize coeff_per_sb_select from bit_rate [10:06] <cone-720> ffmpeg.git 03Martin Storsjö 07master:6d0fbebf9493: ppc: hpeldsp: Include attributes.h [10:06] <cone-720> ffmpeg.git 03Michael Niedermayer 07master:2d234930209a: Merge remote-tracking branch 'qatar/master' [11:19] <cone-720> ffmpeg.git 03Pavel Koshevoy 07master:5a2a0603780a: libavfilter/af_atempo: Fix uninitialized memory access [11:19] <cone-720> ffmpeg.git 03Pavel Koshevoy 07master:0c77cdb491de: libavfilter/af_atempo: Avoid round-off error build-up, ticket #2484 [12:00] <saste> ubitux: i'm trying to write a POC for zmq messaging [12:01] <ubitux> hehe :) [12:01] <saste> basically i want to write a filter/server which brokes messages queued when filter_frame() is called [12:02] <saste> or should it be called a client, which subscribe to a server which sends the commands? [12:02] <saste> also ideally we could have more than one publishing servers, and thus the "client"/filter may need to subscribe to more than one [12:02] <saste> but for the moment I think 1 client/1 server should be ok [12:03] <saste> do you have specific ideas in mind? [12:04] <ubitux> i don't understand your first sentence [12:04] <ubitux> :( [12:04] <saste> with the filter/subscriber approach we can have several filters subscribed to different publishers [12:04] <ubitux> but maybe that's because i just woke up [12:05] <saste> the publisher sends a command, [12:05] <saste> the filter/subscriber receives the command, queues it and process it in filter_frame() [12:05] <cone-720> ffmpeg.git 03Michael Niedermayer 07master:83330cf5fae6: init_vlc_sparse: fix leak on error [12:06] <cone-720> ffmpeg.git 03Michael Niedermayer 07master:14369f59c5ad: ffmpeg_filter: fix leak on error [12:06] <saste> then it should probably send the answer back (e.g. in case of error) [12:06] <saste> since we have a single publisher to which to communicate, this is going to be a bit easier [12:07] <saste> the alternative is to have a server, which receives commands from the clients, process them and replies with an error status/message [12:08] <saste> i'd go with the filter/subscriber approach and see if it works [12:09] <saste> (the publisher logic needs to be handled outside of ffmpeg of course) [12:09] <ubitux> just a minute, for the context, the idea is to allow third app to have a more advanced communication with filters than the send command thing, right? [12:10] <saste> yes [12:13] <ubitux> so you want a subscriber filter in the filtergraph which is used as the gateway? [12:13] <ubitux> how will that filter do something with the other filters? [12:13] <ubitux> use our internal command system? [12:13] <saste> it will send commands to other filters [12:14] <saste> like sendcmd [12:14] <ubitux> right, ok [12:15] Action: saste is reading man zmq_socket [12:18] <saste> the problem with the subscriber/publisher approach is that it is unidirectional [12:19] <saste> so no way to send back status/feedback [12:40] <saste> looks like i need to do threading in any case... [12:41] <ubitux> paul was talking about threading a few days ago [12:41] <ubitux> it might be relevant to have some frame threading... [12:42] <ubitux> not sure how that can help you though :) [12:43] <saste> uhm no after all maybe i don't need it [12:43] <saste> since I can just query from a socket in filter_frame() in non-blocking mode [13:22] <ubitux> saste: i wonder if it would be relevant to have a n-1 thread for a filter with n outputs [13:23] <ubitux> given the design of lavfi, every branch should be independant from each other [14:22] <ubitux> fun, there is sws_getCoefficients() [14:22] <ubitux> but it's not used at all [14:22] <ubitux> at least internally [14:22] <ubitux> mmh the table is accessed directly, ok [16:21] <cone-539> ffmpeg.git 03Nicolas George 07master:b57c23f8c85c: lavfi: add asetrate filter. [16:48] <xlinkz0> is av_seek_frame with no flags the fastest way to seek? [16:54] <michaelni> xlinkz0, depends on the demuxer [16:54] <michaelni> with some (that support it) seeking per byte position is faster [16:55] <xlinkz0> Video: h264 (Baseline) (avc1 / 0x31637661) [16:55] <xlinkz0> but how would i know what byte i want to seek to? [16:56] <xlinkz0> don't frames have different sizes? [17:01] <xlinkz0> well the bottleneck seems to be disk speed anyway [17:02] <xlinkz0> seek to the end of a 2h 25fps movie is 350ms if the movie is cached [17:03] <xlinkz0> and 12 seconds if it's not :( [17:13] <michaelni> xlinkz0, try something else than raw h264 [17:14] <xlinkz0> i have to work on this particular codec [17:14] <cone-539> ffmpeg.git 03Nicolas George 07master:fd7a7e11b94b: lavfi/buffersink: fix possible leaks after allocation failures. [17:15] <nevcairiel> seeking in raw files is always impossible, i would suggest to mux it into a container easier for seeking [17:22] <xlinkz0> that overhead would be bigger than actual seeking [17:22] <xlinkz0> anyway i can adjust the video length to something as small as 10 minutes [17:22] <xlinkz0> i tried it on extreme cases to see how bad it could be [17:32] <cone-539> ffmpeg.git 03Stefano Sabatini 07master:5e893df7ae15: lavfi/sendcmd: fix various "in in" rerepepetitions in messages [17:32] <cone-539> ffmpeg.git 03Stefano Sabatini 07master:df766673e577: lavfi/sendcmd: drop now pointless sendcmd/asendcmd_init() wrappers [18:25] <cone-539> ffmpeg.git 03Stefano Sabatini 07master:3e40b85683e6: lavfi: add interleave filters [19:41] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:047716bbd8f5: ffplay: check av_samples_get_buffer_size() return [19:44] <michaelni> ubitux, did you see CID991855 Resource leak (vf_curves) ? [19:47] <michaelni> or CID991854 (ffprobe.c) [19:47] <ubitux> oh i forgot to mark it as false positive [19:48] <ubitux> (if it's the same that last time) [19:51] <michaelni> it will not disappear until you mark it ... [19:53] <ubitux> yes i know [19:53] <ubitux> i'm going to do it for real [19:53] <ubitux> i'm looking again at it [19:58] <ubitux> done [20:13] <ubitux> about ffprobe, i'd better let saste deal with it, it's his code mostly :) [20:14] <ubitux> at least this part [20:25] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:bc7330a84029: tools/ismindex: dont ignore return code of write_fragments() [20:25] <cone-539> ffmpeg.git 03Michael Niedermayer 07master:ebea370dc390: ismindex: check mkdir()s return code [20:57] <cone-539> ffmpeg.git 03Andy Martin 07master:4ac2da379383: lavfi/fade: Added ability to do video fade based on timestamp [00:00] --- Mon Apr 22 2013

1 0

ffmpeg.log.20130421
by burek 22 Apr '13

22 Apr '13

[00:07] <brontosaurusrex> so is there something to show me how many consecutive 0 dB samples there was at specific point in time? [00:32] <Fjorgynn> Good night [02:57] <THoRIX_> hi, if you are decoding a video using FF_THREAD_FRAME - do you need to subtract (thread_count * single frame delay) in order to generate the correct pts? [03:16] <Plorkyeran> you do have to compensate for the delay from frame threading yourself [03:16] <Plorkyeran> but that is unlikely to be a correct way to do it [03:35] <defaultro1> good evening folks, I would like to convert a .flac to mp3 with the same quality as possible. What would be the parameters? [03:43] <defaultro1> is 320k highest bitrate for audio? [03:49] <THoRIX_> Plorkyeran: what would be the right way to do it? [03:55] <THoRIX_> i tried subtracting ((thread_count-1) * 1.0 / fps) and it seems to fix the incorrect pts issue. And what i mean by this is when i subtract this constant (when thread_count > 1 - e.g. multi-core machines), my player starts spitting out identical pts to when the player's affinity is set to a single logical core. [05:47] <roboman2444> so, i am trying to stream yuv420i video and raw pcm audio through pipes to ffmpeg [05:48] <roboman2444> how should i set up the command? i have been trying a while and keep getting Ignoring attempt to set invalid timebase for st:0 [07:46] <cryptopsy> how to remove this clicking? http://www.filedropper.com/clicking [07:46] <cryptopsy> i tried the click removal in audacity too, it didn't work [07:46] <cryptopsy> file is 10 second mp3 [11:09] <jacobs1> hi, how can I configure mpeg4 to do fast encoding ( like x264 ultrafast ) ? [11:22] <cryptopsy> how to force a video that goes to youtube to stay 1080p quality (at least for the audio) - resize to a high resolution before uploading? [11:22] <cryptopsy> my command for mp3 to avi is $FFMPEG -loop 1 -i $1 -i $2 -vcodec mpeg4 -threads 4 -acodec copy -y -t $TIME $3 [11:22] <cryptopsy> my mp3's are 320k, but when uploading to yt video is in 360p, which makes the audio less than that [11:22] <cryptopsy> there are two -i because one is just a picture [11:24] <cryptopsy> should i uxe x264 instead of mp4 ? [11:24] <cryptopsy> ah i needed -s hd1080 [11:24] <cryptopsy> thanks [11:24] <LithosLaptop|2> Youtube determines what bitrate to use based on the resolution of the input video [11:24] <cryptopsy> LithosLaptop|2: tnx [11:24] <cryptopsy> fucking jews [11:25] <LithosLaptop|2> :) [11:25] <cryptopsy> LithosLaptop|2: does that work with mp4 or do i need to use libx264? [11:27] <cryptopsy> what does -vpre do? can't find it in ffmpeg manpage [11:29] <LithosLaptop|2> works with anything [11:30] <LithosLaptop|2> youtube will still transcode the audio [11:31] <cryptopsy> File Preset for max not found when doing -vpre=max [11:31] <LithosLaptop|2> but at a higher resolution video (720p/1080p), youtube will use a higher bitrate for audio than what it would of used for 360p input [11:31] <cryptopsy> what's this file preset error, i get it for 'hq' too [11:32] <cryptopsy> is -vpre deprecated? [11:32] <LithosLaptop|2> I don't know [11:33] <LithosLaptop|2> maybe somebody else here could help [11:38] <cryptopsy> LithosLaptop|2: had i known your advice month ago, wouldnt have to transcode over 40 albums and reupload it all [11:39] <LithosLaptop|2> ffmpeg -loop 1 -i 720p.png -i test.mp3 -shortest -c:v libx264 -c:a copy -r 24000/1001 -profile:v baseline -pix_fmt yuv420p -crf 18 testing.mkv [11:39] <LithosLaptop|2> thats what I used once [11:39] <LithosLaptop|2> just make sure that 720p.png is 1280x720 or 1920x1080 [11:41] <cryptopsy> that's not practical [11:42] <cryptopsy> use -s hd1080 [11:43] <LithosLaptop|2> well I didn't want to resize a 360p image to 1080p :) [11:45] <LithosLaptop|2> anyway the best would be to upload raw pcm instead of 320kbps MP3s and let youtube encode the audio only once, but that will take a long time to upload [11:45] <LithosLaptop|2> FLAC doesn't work :( [11:51] <cryptopsy> you're right, gonna put that on my todo lists [11:51] <cryptopsy> i usually upload from university [11:51] <cryptopsy> so download r fast [11:51] <cryptopsy> uploads* [11:52] <cryptopsy> do you kno whow to upload to yt from the command line? [11:53] <cryptopsy> there's a youtube-dl and a youtube-upload but youtube-upload isn't in the gentoo tree [11:53] <cryptopsy> even though the'yre made by the same software writer [11:57] <cryptopsy> LithosLaptop|2: i met a guy in amsterdam who was uploading full albums in flac every 3 minutes [11:58] <cryptopsy> almost 24/7 [11:58] <cryptopsy> LithosLaptop|2: going to sleep, peace out [11:58] <LithosLaptop|2> hmm, I guess he knows something that I don't [11:58] <saste> yes, for example the name of his own pet [11:58] <LithosLaptop|2> haha [14:12] <_julian> hi [14:12] <_julian> is there a way to flush data that was read by avformat_find_stream_info? [14:13] <_julian> actually when playing live streams I would prefer to skip some data at start instead of having a bigger delay [16:30] <g0dfrey> I'm going to write a small web-controled-only player. Which lib do i need to learn? Or is there any simple api usage examples? [16:47] <xlinkz0> i have yet to find something simple about api usage [17:40] <lnd7> anyone know how to escape the path to the font correctly (win7 .bat). tried a lot of different things but couldn't get it to work: [17:40] <lnd7> C:\video\ffmpeg -i video.avi -vf "scale=iw/5.35:-1,drawtext=fontfile="C:\Windows\Fonts\Arial.ttf":Fonts: timecode='00\:00\:00\.00': rate=30000/1001: x=(w-tw)/2: y=h-(1.5*lh): fontcolor=white: box=1: boxcolor=0x00000000@1" temp_ss_%%06d.jpg [17:49] <lnd7> ah got it, nvm [18:41] <xebra> hi, does anybody know how to convert a video from one framerate to another? Is it even possible with ffmpeg? (since I see several people online complain it doesn't work) [18:42] <xebra> the -r option doesn't seem to work well. I have a video at 1fps, I want to change it to 24fps without changing the speed [18:44] <xebra> and sometimes it seems to work, but it's "fake", which means I get 24fps reported by one player, but in reality it's still 1fps when imported in a video editor like blender for example [18:47] <LithosLaptop> -r should work [18:51] <brontosaurusrex> xebra, most video editors have "interpret footage" thingy hidden somewhere [18:54] <divVerent> xebra: how do you want it converted [18:54] <divVerent> by duplicating each frame 24 times? [18:59] <xebra> divVerent, I guess so. [19:09] <xebra> I just checked with blender, it's so weird. I converted the video this way: ffmpeg video.mp4 -vcodec libx264 -vpre default -b 3000k -s 1280x720 -r 24 temp.mp4 ...well, the ubuntu movie player (gstreamer?) reports it to be 24fps, but vlc says it's still 1fps, and in blender it's treated like it's 6fps or so [19:10] <xebra> and if I'm not mistaken, youtube also converts 1fps videos to 6fps when you upload them [19:21] <xebra> however, I just tried to export that same video with blender, at 24fps (but accelerated), and the output REALLY is at 24fps then. Doesn't blender use ffmpeg to export in h264? If so, then it should also be possible to do this directly from ffmpeg, but I don't seem to be able to [19:24] <brontosaurusrex> maybe ffmpeg -r 1 video.mp4 -vcodec libx264 -vpre default -b 3000k -s 1280x720 -r 24 temp.mp4 [19:35] <xebra> brontosaurusrex, I think I've tried that... but with an additional -i in front of video.mp4 [19:36] <xebra> now I'm tring to check what blender does when it exports a video, reading the messages in the console, what parameters it uses [19:57] <xebra> I don't see anything there that even mentions frame rate. I guess blender encodes starting from a totally different format (frames are images) and that's why it works, while if I use ffmpeg directly it doesn't [19:57] <xebra> maybe the solution at this point is to use pipelining... [20:30] <xebra> it's a problem with mp4 though. If I output the file in avi, -r 24 works normally [20:30] <xebra> I wonder why [21:48] <knhalp> I know about the concat: thing in order to join videos of the same type, but is there a way to reliably join any format/encoding of video together as long as the resolution is the same? [21:49] <knhalp> Or will I need to uncompress it before sticking it into a container [21:52] <ubitux> knhalp: https://www.ffmpeg.org/faq.html#How-can-I-join-video-files_003f [21:53] <ubitux> you might be looking for the concat demuxer [21:55] <knhalp> woah what [21:55] <knhalp> certificate error on ffmpeg domain [21:55] <knhalp> It's fine but, strange [21:56] <knhalp> ubitux: I love you. this is EXACTLY what I want. Will it 'demux' any video file? Like a FRAPS AVI or an X264 MKV [21:57] <ubitux> it should [21:57] <knhalp> And (Sorry to keep bothering) how does it handle videos with different framerates and resolutions? [21:57] <knhalp> I don't need it to handle other resolutions, but differing framerates might be useful [21:58] <ubitux> it doesn't care about frame rate but timestamps; and that should be explained in the demuxer documentation (http://www.ffmpeg.org/ffmpeg-formats.html#concat) [21:59] <knhalp> Thank you [21:59] <brontosaurusrex> so is it possible to compile ffmpeg with x11_grab statically? [22:03] <brontosaurusrex> relaxed, do you have your static-build commands handy? [22:03] <brontosaurusrex> i need x264, fdk-aac and x11_grab (if possible) version [00:00] --- Mon Apr 22 2013

1 0

ffmpeg-devel.log.20130420
by burek 21 Apr '13

21 Apr '13

[00:13] <durandal_1707> michaelni: i need frame->nb_samples to follow exactly same pattern as described in specification [00:15] <ubitux> michaelni: are you in the middle of a merge? [00:19] <durandal_1707> probably, because stack is full [00:20] <durandal_1707> ... more that 10 entries.... [00:28] <cehoyos> Hi! Is there a bug in ticket #2484 or do I miss someting? [00:28] <ubitux> cehoyos: options not transmitted [00:28] <cehoyos> options not transmitted? [00:29] <ubitux> known limitations to our playlist-like demuxer [00:29] <ubitux> oh wait [00:29] <cehoyos> Sorry, I don't understand... [00:29] <ubitux> wrong bug [00:29] <ubitux> sorry forget what i said [00:29] <cehoyos> Is that #2485 you are talking about? [00:29] <ubitux> yeah [00:29] <cehoyos> Since I don't understand #2485, it would be great if you could comment [00:29] <ubitux> keyboard derp [00:30] <ubitux> yeah i'll do eventually [00:30] <cehoyos> Is is a duplicate of another ticket? [00:30] <cehoyos> s/Is is/Is it [00:30] <ubitux> not really [00:30] <cehoyos> But it is reproducible? [00:31] <ubitux> if there is a related ticket, it's hls options thing [00:31] <cehoyos> Will the fix be the same? [00:31] <ubitux> anyway, it's just a guy from my previous company, so i'm a bit lazy to get active on the ticket [00:31] <ubitux> that's more a design limitation, maybe a hack can be added but well.. [00:31] <cehoyos> I see. Am I correct that #2484 is a bug, or do I miss something? [00:32] <cehoyos> (My question is mostly; Was atempo never tested?) [00:32] <ubitux> i never used it, can't help you [00:33] <cehoyos> I wonder if the most common use-case isn't PAL-NTSC conversion, and it appears that it can't do that. [00:33] <cehoyos> (But as said: I may absolutely misunderstand something.) [00:34] <durandal_1707> the guy who write it is still active [00:34] <cehoyos> Can video filters buffer frames? [00:34] <durandal_1707> sure [00:34] <cehoyos> So is the comment in ticket #1430 wrong? [00:34] <durandal_1707> i read that vid. thing, and i decided to stay away from it [00:35] <cehoyos> You have a point... [00:36] <durandal_1707> but don't push it... [00:38] <michaelni> ubitux, no iam not in the middle of a merge [00:39] <ubitux> ok i can push then [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:0f1250b7e52f: lavc/gif: make possible to disable offsetting. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:e1b35bdde2fc: lavc/gif: add flag to enable transparency detection between frames. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:5927ebab513d: doc/general: animated GIF are now compressed. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:7004cad36df2: Changelog: notify GIF improvements. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:f5ede48fbb41: lavc/gif: miscellaneous cosmetics. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:a16c20569db6: lavf/gifdec: add loop support. [00:39] <cone-429> ffmpeg.git 03Clément BSsch 07master:67cc31d6c74b: lavf/gif: add final_delay option. [00:39] <ubitux> here we go, i'm done with gif [00:40] <durandal_1707> but 2pass [00:40] <ubitux> whatever [00:40] <ubitux> it's good enough for me [00:40] <cehoyos> Did you mention somewhere (Changelog?) that the gif muxer does not support rawvideo anymore? [00:40] <ubitux> nope.. [00:40] <durandal_1707> i seriosly doubt anyone used that encoder .... [00:41] <ubitux> my gif sample went from 3M to < 400kB [00:41] <ubitux> and that's not violent optim [00:41] <ubitux> using the rawvideo muxer previously was definitely a bad idea :P [00:42] <cehoyos> What used the rawvideo muxer previously? [00:42] <durandal_1707> /* better than nothing gif encoder */ :))))) [00:43] <durandal_1707> cehoyos: gif muxer [00:51] <durandal_1707> ugh, i must add more pixel formats to stereo3d so i can dump down3dright [00:53] <durandal_1707> ubitux: funny. but does ffmpeg -i in.gif -codec copy out.gif works? [00:55] <ubitux> mmh good question [00:56] <ubitux> might need some adjustments :) [00:56] <ubitux> it doesn't seem to work previously anyway [00:57] <durandal_1707> the only case it would be useful is to change delay thing... [01:09] <durandal_1707> ubitux: gonna add yourself to (c)? [01:12] <ubitux> i don't really care [01:12] <ubitux> should i? [01:12] <durandal_1707> dunno... [01:14] <ubitux> btw [01:14] <ubitux> what do i do about timeline? [01:14] <ubitux> i need that feature :( [01:16] <durandal_1707> outvote? [01:17] <ubitux> well, i just have nicolas that isn't really excited about that [01:18] <ubitux> i got reviews from stefano, so well the code is ready [01:18] <ubitux> but... no explicit lgtm or anything [01:18] <ubitux> durandal_1707: not interested in the feature? [01:19] <durandal_1707> feature - yes, implementation - so, so [01:19] <ubitux> what would you prefer? [01:20] <durandal_1707> that everyone is happy [01:20] <ubitux> of course [01:23] <durandal_1707> nevcairiel: what is this BLZ0 fourcc? [02:45] <cone-429> ffmpeg.git 03Michael Niedermayer 07master:8ebfd7c49e44: h264: remove unused variable [02:45] <cone-429> ffmpeg.git 03Michael Niedermayer 07master:a0fbc28c3881: vc1dec: Fix non pullup tff [02:45] <cone-429> ffmpeg.git 03Michael Niedermayer 07master:6c9d28a2294d: vc1dec: Fix tff == 0 handling in init_block_index() [02:58] <Compn> highgod is writing a vp8 gpu decoder, thats pretty awesome :) [02:58] <Compn> i wonder for what video card? [03:13] <kierank> amd i think [03:13] <kierank> amd seem to be sending a ton of money to that company [03:58] <highgod> Hi,I want to ask a question, how can I set the value in HAVE_LIST, such as that I add a CL_CL_H to detect whether there is a cl/cl.h exist [03:59] <highgod> @Compn:about the vp8 decoder,hehe, the performance is very low, and the result is not correct [04:01] <iive> you've implemented vp8 decoder in opencl? [04:02] <highgod> we want to, but it is so hard to use GPU to implement [04:04] <highgod> and we are doing the work, it is very hard [04:05] <iive> why is it hard? gpu are not good when code needs to branch a lot? [04:06] <oneal> Yes, gpu doesn't good at the codes with more branch [04:07] <oneal> this is limited by the gpu architecture. [04:08] <highgod> GPU is not suitable for the data which has a strong correlation [04:09] <highgod> So, can anyone give me some help on the configure thing? hehe, I am not familiar with the configure,hehe I reference the openjpeg, but still can't set the value [04:11] <drv> add a check_header line down by check_header direct.h, then add an entry in HAVE_LIST [04:16] <highgod> OK, I will try it, thanks [04:17] <Skyler_> eesh, doing a video decoder on a gpu sounds very difficult [04:19] <highgod> @drv:sorry, I can't find the direct.h file [04:20] <iive> maybe a hybrid one would be more feasible. doing MC, quant/idct/pp look like tasks that could easily be done in parallel. [04:21] <drv> highgod: sorry, that is a line in 'configure' :) [04:21] <drv> as is the HAVE_LIST variable [04:21] <highgod> @iive:what is pp? [04:22] <iive> post processing [04:22] <iive> like loop filter [04:22] <Skyler_> iive: idct+predict can at best be done in wavefront order, it's very hard to get enough parallelism (because of the dependencies with intra pred) [04:22] <oneal> Yes, MC, quanti/idct can be one in parallel. [04:23] <Compn> what about using the driver api itself [04:23] <Compn> without going into opencl [04:23] <Compn> not even sure ati has open api on windows... [04:23] <Compn> just to offload the mdct or cabac [04:23] <Compn> (or whatever vp8 uses) [04:23] <Compn> that shares with h264 [04:23] <oneal> but gpu can have a speedup just in the case of enough data to be processed by GPU. [04:23] <Skyler_> um... both nvidia and amd use ASIC decoders that do the whole process basically [04:24] <Skyler_> and are programmed by microcode on the graphics card [04:24] <Skyler_> vp8 doesn't use the same idct as h264, nor the same entropy coder [04:24] <Skyler_> nor the same prediction (very /similar/, but not the same) [04:24] <Skyler_> nor the same MC, nor the same deblock... [04:24] <oneal> in the pipeline of decoder, there is a strict serial process procedure. [04:25] <Compn> Skyler_ : darn, i thought something was similar :\ [04:25] <highgod> @drv:OK, find it, thanks [04:25] <Skyler_> they're similar, but not in ways that are really useful [04:27] <oneal> I have tried to optimze the inter-predication and MC. but there is a dependency between the MC in a frame. [04:27] <highgod> @drv: it works, thanks [04:29] <oneal> the design I have done is all blocks which to be process by inter predicton is done in one function(like OpenCL kernel). [04:30] <leo2013> hello [04:30] <leo2013> ? [04:31] <Compn> yo [04:31] <leo2013> I met one problem in vp8 [04:31] <oneal> we assume the blocks inter-predicted just dependent the reference block in the reference frame. in fact, there is a dependency from the adjacent mb. [04:32] <leo2013> My problem is that the edge processing. [04:33] <leo2013> for the input image whose size is 1920*1080 [04:33] <leo2013> the 1080 can't be dived well by 16 [04:33] <leo2013> so that it'll be the 67.5 blocks. [04:33] <leo2013> and for the last block 68 [04:33] <leo2013> we have half block in dst. [04:34] <leo2013> but the edge processing will fill it by using 16*16 [04:34] <oneal> in this case, it should to be paded to 16 multiple [04:34] <leo2013> so that the overflow for more than 8 lines. [04:34] <leo2013> I checked that mc_fun[][] [04:35] <leo2013> this function in one mode is that only copy the data to dst. [04:35] <Skyler_> cropping is done after decoding [04:35] <Skyler_> e.g. the buffer is allowed to be 1920x1088; the decoder should not have to special-case that internally [04:35] <Skyler_> (h264 and all similar formats have the same kind of thing) [04:36] <leo2013> but the height of dst is 1080. [04:36] <Skyler_> then there's something wrong; you should be allocating a frame of the size of the decoded video, not the size of the displayed video [04:38] <weixuan> the dst comes from the decoded video, not the displayed video the height of it is 1080 [04:39] <Skyler_> um... you decide in your decoder what frame size to allocate, unless I'm totally missing something [04:39] <Skyler_> because that's how every decoder works [04:39] <Skyler_> look at how vp8.c does cropping, or how h264.c does cropping [04:40] <leo2013> #define PUT_PIXELS(WIDTH) \ [04:40] <leo2013> static void put_vp8_pixels ## WIDTH ##_c(uint8_t *dst, ptrdiff_t dststride, uint8_t *src, ptrdiff_t srcstride, int h, int x, int y) { \ [04:40] <leo2013> int i; \ [04:40] <leo2013> for (i = 0; i < h; i++, dst+= dststride, src+= srcstride) { \ [04:40] <leo2013> memcpy(dst, src, WIDTH); \ [04:40] <leo2013> } \ [04:40] <leo2013> } [04:40] <Skyler_> ? [04:40] <leo2013> in ffmpeg,vp8 will use this func. [04:40] <Skyler_> yes...? [04:40] <leo2013> yes. [04:40] <Skyler_> that's a C implementation of the 00 MC position [04:40] <leo2013> yes. [04:40] <leo2013> that the c implementation for 00 [04:40] <Skyler_> but what does this have to do with post-decode cropping? [04:41] <leo2013> that'll overflow for the last line. [04:41] <Skyler_> erm, no it won't [04:41] <leo2013> if the last block value can't get dived well. [04:41] <Skyler_> No, again, it won't. [04:41] <leo2013> why? [04:42] <leo2013> dst height is 1080, and the block after padded is 1088 [04:42] <Skyler_> dst height is 1088 [04:42] <Skyler_> Your video has a coded size of 1920x1088. [04:42] <Skyler_> Your video has a DISPLAY size of 1920x1080. [04:43] <Skyler_> Your video could, in theory, have a display size of 532x362, with a coded size of 1920x1088 (this is possible with h264, for example) [04:43] <leo2013> no,the codes size is 1984*1088 [04:43] <leo2013> so it's 124 block in horizonal. [04:44] <Skyler_> Um.... [04:44] <Skyler_> how did it go from 1920 to 1984? [04:45] <leo2013> I used the av_log to check the real width. [04:46] <leo2013> s->mb_width [04:46] <weixuan> 1984 is the linesize of the y plane [04:47] <weixuan> and it is the decoder video linesize [04:47] <Skyler_> Linesize is not width! [04:47] <weixuan> yes [04:47] <Skyler_> They are two different things. [04:47] <leo2013> yes. [04:48] <leo2013> but it's calculated in mc_fun [04:48] <leo2013> in the func:mc_func[0][0](dst, linesize, src + y_off * linesize + x_off, linesize, block_h, 0, 0); [04:48] <leo2013> just like you see,for block 68,the block_h is 16 [04:48] <Skyler_> ???? that function has nothing to do with what you're talking about [04:48] <Skyler_> that's the height of a macroblock, which has nothing to do with cropping frames [04:49] <leo2013> where does the other data(more than 8 lines) come from ? [04:49] <iive> i actually don't remember how MC handled the down/right edges. does it emu_edge/pad on the visible image boundary or on MB boundary? [04:49] <Skyler_> You're confusing the coded and display resolution again. [04:49] <Skyler_> iive: the spec itself does decoding to the coded size [04:49] <leo2013> the other data(more than 8 lines) will come from ref-frames. [04:50] <Skyler_> Your decoder literally does not know it is possible to create frames that are not divisible by 16. [04:50] <Skyler_> If your decoder knows this, your decoder is written incorrectly. [04:50] <Skyler_> Cropping is done after decoding. [04:50] <iive> fair enough. [04:50] <Skyler_> The actual specification dictates that decoding is done on the coded frame, not the display frame. [04:50] <iive> i assume that encoders should pad the extra pixels before encoding. [04:51] <Skyler_> actually. I'm 98% sure of that. Need to confirm it's true for VP8, I'd assume it is, but [04:51] <leo2013> yes.but before the cropping, need loopfilter. [04:51] <leo2013> the loopfilter use the filled data from dst in the last block line,right? [04:51] <Skyler_> Yes, loopfilter is part of decoding. It is not something that happens after decoding. [04:51] <Skyler_> This is why it is called a loopfilter. [04:52] <Skyler_> okay, confirmed, the code in MC does this: [04:52] <Skyler_> int width = 16*s->mb_width, height = 16*s->mb_height; [04:52] <Skyler_> VP8 operates the same way as H.264 then. [04:52] <Skyler_> From the perspective of the decoder, all decoding is done on a plane of macroblocks whose width and height are divisible by 16 pixels. [04:53] <iive> things are much simpler this way :) [04:53] <Skyler_> After the decoder is done, it says "oh, and by the way, you should only display 1920x1080 of this, not 1920x1088 of this." [04:53] <Skyler_> But for the decoder, decoding a 1920x1080 and 1920x1088 stream is exactly, precisely, identical. [04:53] <Skyler_> In fact, if the decoder doesn't do it that way, its output won't be correct. [04:54] <leo2013> yes.so the width and the hieight will be cropping results.but for the last block line,it depends the ovreflow parts,right? [04:54] <Skyler_> What do you mean? [04:54] <Skyler_> Yes, the decoder will use parts of the frame that are not displayed. This is by design, and the decoder does not know about this or care. [04:54] <leo2013> we decoded the last block line. [04:55] <leo2013> in fact, the last block line is 8 lines for input(1080p) [04:55] <iive> only half of the last block line would be shown. [04:55] <leo2013> how about the overflow 8 lines in dst[0]? [04:55] <Skyler_> *sigh* how many times do I have to repeat myself before you will read what I am saying? [04:55] <leo2013> yes.you're right. [04:55] <iive> but it is decoded as a whole [04:55] <leo2013> but how to fill it? [04:55] <Skyler_> The same way you fill any other chunk of memory, by writing to it? [04:56] <iive> it is not decoder problem. [04:56] <leo2013> only filled the last 8 lines according to 8 lines in dst[0]? [04:57] <Skyler_> *sigh* [04:57] <iive> leo2013: imagine this. you get 1920x1080 before encoding. you expand it to 1920x1088 and encode it. [04:57] <leo2013> we haven't more than 8 lines data to fill it. [04:57] <Skyler_> What do you mean? [04:57] <Skyler_> Your reference frame is 1920x1088. [04:57] <Skyler_> Your current frame is 1920x1088. [04:57] <Skyler_> All your frames are 1920x1088. [04:57] <Skyler_> If they're not, your code is incorrect and will not decode VP8 correctly. [04:57] <iive> you always decode 1920x1088 and then cut the last 8 lines. [04:57] <leo2013> I want to know where the data comes from for the last 8 lines? [04:58] <Skyler_> Um, you create it. You're the decoder. [04:58] <Skyler_> Your job is to decode the video [04:58] <Skyler_> which means decoding the pixels coded in the video file... [04:58] <iive> leo2013: the data is padded (filled) before encoding. [04:58] <Skyler_> which includes those last 8 lines. [04:58] <Skyler_> for a total of 1920x1088. [05:01] <oneal> leo2013: the encoded video stream contains 1920x1088, not the 1920x1080. [05:02] <Skyler_> exactly ! [05:03] <iive> Skyler_: btw does x264 repeat the edge pixels, when padding like this? [05:04] <Skyler_> yup [05:04] <Skyler_> I tried some other approaches (e.g. mirroring), but inter prediction makes mirroring a lot worse than repeating [05:04] <Skyler_> in pure intra mode it might help? I'm not sure if I tried that (I think JPEG generally does repeating) [05:05] <iive> ok, i'm off [05:06] <leo2013> your mean my reference frame is the 1920*1088 too? [05:06] <Skyler_> correct [05:07] <leo2013> and the key frame is 1920*1088 to fill the refrence frame. [05:07] <oneal> Skyler, what's mean of "I tied some other apporache(e.g. mirroring), but inter prediction makes mirroring a lot worse than repeating" [05:08] <Skyler_> oh, it's not related to what you're doing, but the encoder gets to decide how to fill those padded pixels (to pad up to divisible-by-16) [05:08] <Skyler_> the easy way is padding, but in theory there could be something smarter [05:08] <leo2013> so the padded data(8 lines) comes from key frame which more than 1080. [05:08] <Skyler_> JPEG encoders typically use a technique called mirroring where they "mirror" the pixels over the edge, instead of repeating the last line over and over [05:08] <Skyler_> but I found this was pretty useless for video. [05:09] <oneal> oh, I understand what's you mean [05:09] <Skyler_> so basically, the encoder gets 1920x1080, and needs to turn it into 1920x1088 [05:09] <Skyler_> in the way that takes the fewest possible extra bits. [05:09] <Skyler_> the decoder then decodes it to 1920x1088, and throws away the last 8 lines when it's done, to get 1920x1080 again. [05:23] <leo2013> In emulated_edge_mc,for(; y<end_y; y++){ [05:23] <leo2013> memcpy(buf, src, w*sizeof(pixel)); [05:23] <leo2013> src += linesize; [05:23] <leo2013> buf += linesize; [05:23] <leo2013> } [05:25] <leo2013> the end_y can be 21, and the src is from 1071, so when copy src to buf, it'll be more than 1088. [05:47] <Skyler_> leo2013: end_y = FFMIN(block_h, h-src_y); [05:47] <leo2013> I met another problem in vp8 is the edge processing in inter_predict.Especially for the last block line in emulated_edge_mc.There will be the possibility of overflow when fill the td->edge_emu_buffer. [05:48] <leo2013> yes. [05:48] <leo2013> end_y = FFMIN(block_h, h-src_y); [05:48] <leo2013> this min value can be 21 [05:48] <leo2013> 21 for Y [05:49] <leo2013> 8+8+5 [05:50] <Skyler_> What is block_h? [05:50] <leo2013> block_h is 16 [05:51] <leo2013> h-src_y [05:51] <Skyler_> What is the maximum value of MIN(16,x)? [05:55] <leo2013> block_h + subpel_idx[1][my] [05:55] <leo2013> but if the block_h is 21,how about it? [05:55] <weixuan> block_h + subpel_idx[1][my] [05:56] <weixuan> and the block_h = 16 and the subpel_idx[1][my] = 5 [05:56] <weixuan> so the final value of the block_h = 21 [05:57] <weixuan> so it will overflow the range of the src [05:57] <leo2013> static const uint8_t subpel_idx[3][8] = { [05:57] <leo2013> { 0, 1, 2, 1, 2, 1, 2, 1 }, // nr. of left extra pixels, [05:57] <leo2013> { 0, 3, 5, 3, 5, 3, 5, 3 }, // nr. of extra pixels required [05:57] <leo2013> { 0, 2, 3, 2, 3, 2, 3, 2 }, // nr. of right extra pixels [05:57] <leo2013> }; [05:57] <Compn> you guys have to talk to BBB- [05:57] <Compn> hes the vp8 guy :) [05:58] <Compn> i am not, and its time for me to sleep. night and good luck :) [05:58] <Compn> probably michaelni knows something about it as well [05:58] <leo2013> extra pixels required [05:58] <Skyler_> except, no. [05:58] <Compn> and of course Skyler_ :) [05:58] <Skyler_> end_y = FFMIN(block_h, h-src_y); [05:58] <Skyler_> put simply: [05:59] <weixuan> good night [05:59] <Skyler_> The loop you pointed to copies the pixels that do not need to be emulated. [05:59] <Skyler_> The MIN() statement says "we don't copy more pixels than we need rows of pixels (obviously)" [05:59] <Skyler_> that's the h-src_y part [05:59] <Skyler_> er, correction [05:59] <Skyler_> that's the block_h part [05:59] <Skyler_> h-src_y is the constraint on the number of rows available. [05:59] <Skyler_> But the inner workings of emulated_edge_mc aren't actually important; it's probably easier to understand _what_ it does than how it works. [06:00] <Skyler_> All it does is "if a pixel is off the frame, calculate it to be equal to the closest pixel that is on the frame". That's it. [06:00] <Skyler_> It's a bit obfuscated, yes <.< [06:02] <leo2013> I found there are sevral blocks fit that condition extra.the over the bottom of scr,what's the nearest value could be gotten? [06:03] <Skyler_> The phrasing I used is equivalent to "any pixel past the edge is equivalent to what you'd get if you padded the edge by repeating the last line over and over" [06:03] <Skyler_> So this generally leaves you with two options in a codec: [06:04] <Skyler_> 1) pad the edges of the frame with the last line/column, repeated ~16 times or however many is necessary to guarantee correct behavior. Don't implement emulated_edge_mc. [06:04] <Skyler_> Advantage: faster MC, no extra special case [06:04] <Skyler_> Disadvantage: requires, well, padding every fram [06:04] <Skyler_> *frame [06:04] <Skyler_> 2) emulated_edge_mc, which eliminates the need for padding. [06:04] <Skyler_> I think technically you need about 19 lines of padding because of subpel. [06:05] <Skyler_> I think vp8 used to support both, but 1) was eventually removed due to not being faster? I'm not sure. [06:05] <Skyler_> But which one you do is totally your choice. [09:53] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:68d8238cca52: hpeldsp: Add half-pel functions (currently copies of dsputil) [09:53] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:23de9e91df1c: Merge commit '68d8238cca52e50e8cc81bf2edcaf8088c52d4c0' [10:08] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:3bd062bf7f81: vp3: Use hpeldsp instead of dsputil for half-pel functions [10:08] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:cb7ecb75635d: vp56: Use hpeldsp instead of dsputil for half-pel functions [10:08] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:8f992dc8c7c5: indeo3: Use hpeldsp instead of dsputil for half-pel functions [10:08] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:0f0a11d5768e: bink: Use hpeldsp instead of dsputil for half-pel functions [10:08] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:8071264f2196: interplayvideo: Use hpeldsp instead of dsputil for half-pel functions [10:08] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:3c6621708b93: Merge commit '8071264f2196d71ff49c3944c33f8d3d83f548f1' [10:22] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:c10470035e60: mimic: Use hpeldsp instead of dsputil for half-pel functions [10:23] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:6caa44aa7df0: svq1: Use hpeldsp instead of dsputil for half-pel functions [10:23] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:f4fed5a2f97e: mpegvideo: Use hpeldsp instead of dsputil for half-pel functions [10:23] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:2f6bc5f7c193: svq3: Use hpeldsp instead of dsputil for half-pel functions [10:23] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:ab4ba6b74d9e: Merge commit '2f6bc5f7c193477c2ebc0acce8f2d5551445e129' [12:50] <saste> ubitux: comments on interleave? [13:05] <ubitux> saste: ok, will do in a moment [13:05] <ubitux> thanks for the timeline reviews btw [13:28] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:1277dc07fbe6: svq1enc: Use hpeldsp instead of dsputil for half-pel functions [13:28] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:28bc406c84b0: mjpeg: Use hpeldsp instead of dsputil for half-pel functions [13:28] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:3fee9fa02232: Merge commit '28bc406c84b04a5f1458b90ff52ddbec73e46202' [13:44] <durandal_1707> michaelni: so any ideas for s302m? [13:46] <saste> ubitux: ffplay IN -vf "select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], edgedetect, [tmp] interleave" [13:48] <durandal_1707> michaelni: if not, i will push it with experimental flag and be done with it... (hoping others can fix it...) [13:50] <michaelni> durandal_1707, iam not sure how to solve it best, a s302 muxer would be an option instaead of a encoder, or a private option that contains the frame rate but that would need a change in ffmpeg*.c to set it, i also dont mind if you push with the flag ... [13:51] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:8db00081a37d: x86: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [13:51] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:d0aa60da1022: Merge commit '8db00081a37d5b7e23918ee500bb16bc59b57197' [13:53] <durandal_1707> michaelni: s302m muxer is useles, how would you than put it in ts? [13:54] <michaelni> the s302 muxer would have to pass the stuff into a ts muxer, not sure how that would look [13:54] <durandal_1707> whats wrong with internal approach that when variable frame size flag is set than interleave thing sets frame size that s320m expects? [13:55] <durandal_1707> or additional flag could be introduced [13:55] <michaelni> what interleave code exactly do you speak of ? [13:56] <durandal_1707> whatever code that sets frame->nb_samples when its 0 in init [13:57] <michaelni> my concern is just that complexity should not be moved into the user application [13:57] <michaelni> that is the encoder or avcodec or avformat should do thze work [13:58] <michaelni> needing every user app to have code to get the interleaving right would be annoying, thats why i asked which code you meant ... [13:59] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:47e5a98174eb: ppc: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [13:59] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:055e5c8e01c5: Merge commit '47e5a98174eb9c07ad17be71df129719d60ec8b7' [13:59] <durandal_1707> so variable frame flag just means user app can set wheterever it wants? [14:03] <durandal_1707> saste: buffer queue overflows do not happens with interleave anymore? [14:04] <michaelni> durandal_1707, about var-frame within limits id say yes [14:05] <durandal_1707> michaelni: and it can't be changed after set in init, because for some fps, it cant be same for every frame, so it goes 161,162,161,162 or similar [14:05] <durandal_1707> so the only way how it can be currently handled is from user app [14:07] <durandal_1707> hmm, it can be changed ..., from decoder, but from encoder and without flag? [14:08] <durandal_1707> it simply can't, or i'm missing something? [14:08] <durandal_1707> so its API limitation [14:09] <durandal_1707> i could do internaly buffering ... [14:10] <ubitux> saste: ok, will try soon, but first: since i'm lazy, do you have a out-of-box working command injection test? [14:10] <ubitux> (i've restored the enable command, and now for all filters) [14:10] <ubitux> (supporting the timeline of course) [14:14] <michaelni> durandal_1707, yes internal buffering may be an option [14:21] <saste> ubitux: out-of-box working command injection test?? [14:21] <saste> I use sendcmd to test process_command() callbacks [14:22] <ubitux> yeah it's ok i've finally made one [14:25] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:7384b7a71338: arm: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [14:25] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:c4010972c4a7: Merge commit '7384b7a71338d960e421d6dc3d77da09b0a442cb' [14:26] <durandal_1707> kierank: could you provide useful file with s302m codec for coverage test? (the one i found is 0 only) [14:27] <kierank> not sure if i can make any s302m public [14:27] <kierank> i think vlc have some [14:29] <kierank> you have the spec folder access, right? [14:29] <durandal_1707> where? [14:29] <kierank> oh apparently not [14:29] <kierank> do you have dropbox [14:31] <kierank> https://dl.dropboxusercontent.com/u/2701213/Specs/SMPTE/SMPTE%20Standards/s… [14:32] <kierank> whatever you end up doing in ffmpeg, i'll probably have to make a small fork of s302m because i need to set the frame size manually [14:34] <durandal_1707> kierank: why? [14:36] <durandal_1707> in that case i just leave code as is, (it will obviosly need changes in ffmpeg ....) [14:36] <durandal_1707> and where are those vlc's s302m samples? [14:39] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:bfb41b5039e3: bfin: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [14:39] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:c5a11ab6d106: Merge commit 'bfb41b5039e36b7f873d6ea7d24b31bf3e1a8075' [14:55] <durandal_1707> hmm, what about w3fdif deinterlacer from FFmbc? [14:55] <kierank> durandal_1707: because 302m is used for low latency and so i get frames with 1601, 1602, 1601 etc. I will need to pass whether i am 1601 or 1602 straight from hardware [14:56] <kierank> durandal_1707: in vlc samples repo i guess [14:57] <durandal_1707> sorry, i dont know where that is [15:01] <kierank> http://streams.videolan.org/ somewhere but i can't seem to find it [15:09] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:278bd2054ca6: sh4: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [15:09] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:4bdec0e71edc: Merge commit '278bd2054ca61ab70dfe38f1774409cda2da5359' [15:13] <durandal_1707> kierank: well private option to not use interneal buffering at could be added, leaving it to the calling code [15:20] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:78ce568e43a7: sparc: hpeldsp: Move vis half-pel assembly from dsputil to hpeldsp [15:20] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:fdb1f7eb7a80: Merge commit '78ce568e43a7f3993c33100aa8f5d56c4c4bd493' [15:27] <ubitux> saste: hey the blur effect is awesome :) [15:27] <ubitux> i believe you can do the same with enable now though :) [15:27] <durandal_1707> how you repeat Nth frame X times? [15:30] <ubitux> ./ffplay ~/samples/matrixbench_mpeg2.mpg -vf "boxblur=enable='if(gt(random(0), 0.2), 0, 1)'" [15:30] <ubitux> hehe [15:30] <ubitux> durandal_1707: from the cmdline? [15:30] <durandal_1707> how that can be useful? [15:30] <durandal_1707> ubitux: with lavfi [15:31] <ubitux> maybe we could ajudst copy to re-inject the same frame [15:31] <ubitux> re-inject each frame copied N times [15:31] <ubitux> (and use enable to make the copy only once) [15:31] <durandal_1707> FFmbc have repeatframe [15:32] <ubitux> (durandal_1707: the boxblur thing make some kind of wave effect) [15:32] <ubitux> (so that's awesome) [15:33] <ubitux> durandal_1707: how can this dup thing be useful? [15:34] <durandal_1707> dup thing? [15:34] <ubitux> repeatframe [15:34] <ubitux> what you are talking about [15:34] <durandal_1707> filter from FFmbc [15:34] <ubitux> yes but what is the purpose? [15:35] <durandal_1707> to repeat frame [15:35] <ubitux> what for? [15:36] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:2957d29f0531: alpha: hpeldsp: Move half-pel assembly from dsputil to hpeldsp [15:36] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:6ec26157b96b: Merge commit '2957d29f0531ccd8a6f4378293424dfd92db3044' [15:42] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:54cd5e4f92de: dsputil: Remove hpel functions (moved to hpeldsp) [15:42] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:d2d2c309e8f3: Merge commit '54cd5e4f92de6bd0fb8e24069153b0156c8136bc' [15:42] <durandal_1707> ubitux: to repeat single image with blend filter.... [15:50] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:c443117f25e0: dsputil: Remove dct_bits [15:50] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:c3624cfe7638: Merge commit 'c443117f25e034c1e3ef35334b61b937e1e208ff' [15:54] <ubitux> http://lucy.pkh.me/youtube-free.webm youtube and my ISP, yepee. [15:55] <ubitux> 0kb/sec regularly ffs [15:55] <ubitux> (i can in practice download up to 1.6MBytes/sec) [15:59] <Compn> ubitux : dang [16:00] <Compn> ubitux : throttling ? [16:00] <ubitux> it's been like this since about 2 years [16:00] <ubitux> it's particularly painful nowadays [16:01] <ubitux> Compn: it's a ping pong about who will pay between google & my isp [16:01] <Compn> too bad you cant use youtube over https [16:01] <Compn> the video urls still seem to be http [16:01] <Compn> html over https tho [16:01] <ubitux> it's the same with https [16:02] <Compn> hows the speed on googlevideo or vimeo ? [16:03] <ubitux> it's fine most of the time for vimeo [16:03] <ubitux> (googlevideo still exists?) [16:03] <Compn> yeah they were going to kill it, but then didnt [16:03] <Compn> but the last video i tried to watch there was missing or unavailable [16:03] <Compn> they were thinking about migrating it to youtube, but i dunno current status [16:04] <Compn> they pestered me about moving my single video to youtube, which i finally did [16:04] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:619e0da19119: dsputil: Remove unused 32-bit functions [16:04] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:f6dcd844ee65: Merge commit '619e0da19119bcd683f135fe9a164f37c0ca70d1' [16:06] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:c9f5fcd08c3a: dsputil: Merge 9-10 bpp functions for get_pixels and draw_edge [16:06] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:bf66016e4c03: Merge commit 'c9f5fcd08c3a33bfb1b473705c792ab051e7428d' [16:20] <cone-132> ffmpeg.git 03Ronald S. Bultje 07master:d4d186d185df: dsputil: Remove non-8bpp draw_edge [16:20] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:9ae56b85b664: Merge commit 'd4d186d185df98492d8935a87c5b5cf369db9748' [16:24] <cone-132> ffmpeg.git 03Martin Storsjö 07master:287c8db39e71: cosmetics: bfin: Fix indentation in the dsputil init function [16:24] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:0e3d2b2c8d02: Merge commit '287c8db39e71af7047e551bbfd1264d771cccbc9' [16:28] <saste> btw there will be socis this year? [16:29] <saste> in case we are going to propose something very much space oriented (like the star mapper proposed by michaelni) [16:29] <j-b> saste: ping [16:29] <saste> j-b: pong? [16:32] <cone-132> ffmpeg.git 03Martin Storsjö 07master:a60136ee570c: vc1: Remove now unused variables [16:32] <cone-132> ffmpeg.git 03Martin Storsjö 07master:b71a0507b01e: x86: Remove unused inline asm instruction defines [16:32] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:0dd25e46996d: Merge remote-tracking branch 'qatar/master' [16:59] <BBB-> ? [17:06] <Compn> just gpu devs looking for vp8 help [17:11] <kierank> durandal_1707: who's Darryl Wallace? [17:13] <BBB-> hm I'll respond to the ML again [17:14] <BBB-> next time they're on IRC, please ask them for samples. I don't like the thought of a bug at all, better to confirm quickly or just confirm that it's a bug on their end [17:16] <Compn> sounds like it was a bug on their end with edge emulation and display res [19:24] <durandal_1707> kierank: the guy who added pull request on ffmpeg github mirror [19:24] <kierank> ah [19:27] <durandal_1707> michaelni: how fast is subsampled yuv to unsubsampled yuv conversion in sws? [19:37] <xlinkz0> can i ask library usage questions here? #ffmpeg seems to be pretty dead [19:39] <michaelni> durandal_1707, someone would have to benchmark [19:40] <funman> win 15 [19:49] <cone-132> ffmpeg.git 03Paul B Mahol 07master:a56fd051ee64: lavfi/stereo3d: support more formats for non-anaglyph outputs [19:57] <cone-132> ffmpeg.git 03highgod0401 07master:fdad04e75628: avfilter/deshake_kernel: fix reset value bug of deshake kernel [21:08] <michaelni> j-b, "http://download.videolan.org/pub/contrib/c99-to-c89/" seems dead (https://ffmpeg.org/trac/ffmpeg/ticket/2487) [21:09] <michaelni> s/dead/404/ [21:11] <funman> there's https://github.com/mstorsjo/c99-to-c89 [21:11] <funman> dunno if we should make a tarball [21:12] <funman> ah apparently it was binaries [21:12] <michaelni> yep, the link from github.com/mstorsjo/c99-to-c89 points also to videolan for binaries [21:12] <michaelni> same 404 [21:13] <funman> i can't find that folder somewhere else on the ftp, i guess it has been deleted.. [21:13] <funman> j-b: was it you? [21:13] <j-b> no, but I think it was a weird old script [21:14] <j-b> I'll re-put that in place [21:15] <cehoyos> j-b: Hi, is there a reason why the c99-wrap.exe binaries disappeared from http://download.videolan.org/pub/contrib/c99-to-c89/ [21:15] <cehoyos> ? [21:16] <cehoyos> nevcairiel: Could we provde the binaries you use on ffmpeg.org ? [21:16] <funman> cehoyos: you just missed the conversation ^_^ [21:16] <cehoyos> (Iirc, this is the third download location that disappeared) [21:16] <funman> 21:14 <@j-b> I'll re-put that in place [21:16] <funman> 21:14 -!- cehoyos [~cehoyos(a)chello080108089202.30.11.tuwien.teleweb.at] has joined #ffmpeg-devel [21:16] <cehoyos> Enlighten me please [21:16] <cehoyos> ok, thank you! [21:17] <cehoyos> Ah, you mean "just" as in "really just" =-) [21:36] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:4824aea7afd4: avcodec/mpegvideo: change asserts to av_asserts [22:09] <cone-132> ffmpeg.git 03Marton Balint 07master:40693ba3ac3f: ffplay: simplify aspect ratio calculation [22:09] <cone-132> ffmpeg.git 03Marton Balint 07master:d148339d19c6: ffplay: use AV_NOPTS_VALUE video frame pts instead of using 0 [22:09] <cone-132> ffmpeg.git 03Marton Balint 07master:b8facbeecb66: ffplay: only do early frame drop if video queue is not empty [22:09] <cone-132> ffmpeg.git 03Michael Niedermayer 07master:eda61abc846b: Merge remote-tracking branch 'cus/stable' [22:15] <cehoyos> ubitux: I never tried to reproduce ticket 2446, could you test and close the ticket if it is fixed? [23:03] <ubitux> cehoyos: sure, will do [23:06] <cehoyos> Merci [23:51] <ubitux> hey btw [23:51] <ubitux> just came accross this: http://cr.i3wm.org/ [23:51] <ubitux> it looks awesome for contributions [00:00] --- Sun Apr 21 2013

1 0

ffmpeg.log.20130420
by burek 21 Apr '13

21 Apr '13

[01:13] <msmithng> can anyone explain this to me? [01:13] <msmithng> GPAC ISO Hint Handler [01:13] <msmithng> sorry [01:13] <msmithng> Stream #0:2(und): Data: none (rtp / 0x20707472), 66 kb/s [01:13] <msmithng> Metadata: [01:13] <msmithng> creation_time : 2013-04-13 21:27:28 [01:13] <msmithng> handler_name : GPAC ISO Hint Handler [01:14] <larrikin> gpac.sourceforge.net ? [01:16] <msmithng> ok, I'll start there& [01:23] <msmithng> seeing these in ffprobe output as [01:23] <msmithng> Unsupported codec with id 0 for input stream 2 [01:23] <msmithng> Unsupported codec with id 0 for input stream 3 [01:25] <StFS> Hi. I'm having audio sync problems when I record my desktop. The audio is always "downpitched" and lags. I've tried -async 1 and that fixes the sync but introduces very annoying "clipping" [01:29] <StFS> here is the command I'm using: http://pastebin.com/ZSZDa3Ef [01:38] <durandal_1707> StFS: perhaps your machine is not fast enough to encode it in realtime... [01:39] <larrikin> msmithng: http://records.sigmm.ndlab.net/2012/12/open-source-column-gpac/ [01:42] <larrikin> msmithng: seems like you might need something called MP4Box to decode your stream [02:13] <msmithng> larrikin: thank you sir& I'll look into that for sure [03:33] <Max-P> Hi, does by chance anyone here knows how to force 2 live inputs (x11grab+alsa) keep in sync. Or is there a way to adjust buffers so it holds the same amount of data? Searching for a week now, still haven't found anything relevant :/ Thanks! [04:31] <leo2013> hello? [04:34] <Max-P> leo2013: There are people here. We just don't respond under 3 minutes... Meh. [04:52] <THoRIX_> hi, i've been working with dranger's ffmpeg tutorials (im building under win64 using zeranoe builds) and im noticing something strange. Depending on how many logical cores are detected the video sync timings seem to vary. For example if i set the program affinity to a single core then the video frames match perfectly with the audio, the more cores i enable the more the video is off sync (always behind the audio). I can replicate the behaviour by programmical [04:52] <THoRIX_> ly setting the thread_count under AVCodecContext. Can anyone shed any light on why this is happening? [12:26] <hendry> what's the best audio codec for playback on the Web? Just voice? webm? And how do I target that? [12:27] <elkng> can ffmpeg generate some waves that makes brain relaxed ? [12:27] <ubitux> yes [12:28] <hendry> wondering if speex was still a thing [12:28] <elkng> hendry: speex ? [12:28] <hendry> audio codec IIRC [12:28] <elkng> ubitux: how ? [12:28] <ubitux> elkng: https://ffmpeg.org/ffmpeg-formats.html#sbg [12:29] <klaxa> yes, audio codec targeted at voice compression [12:29] <klaxa> for voip and such [12:31] <elkng> ubitux: what that link about, how to generate sound waves with ffmpeg ? I need some specific audio waves for brain [12:32] <saste> elkng, audio waves for brain, why? Now I'm curious [12:32] <saste> but yes sbg looks interesting from that point of view [12:32] <ubitux> elkng: "SBaGen -- Binaural Beat Brain Wave Experimenter's Lab" [16:56] <xlinkz0> is it possible to get information about frames(pts, dts, is keyframe, frame type) from an encoded files without actually decoding the frames? [16:58] <xlinkz0> even if ffmpeg doesn't have that functionality.. is it theoretically possible? [16:59] <JEEB> in sane containers pts/dts and frame type might be get'able [16:59] <xlinkz0> any functions in ffmpeg that can do that? [17:00] <xlinkz0> i don't need it to work in general [17:23] <zap0> anyone know of a way to determine the number of cuts in a movie? [17:24] <xlinkz0> pkt_pts is in AV_TIME_BASE units right? [17:25] <xlinkz0> i don't get it, the difference between every frame pts is on average 3600 and the stream is 25fps [17:26] <xlinkz0> 25*3600 = 90k [17:26] <xlinkz0> shouldn't it be 100k so that it makes a second? [19:29] <xlinkz0> is there some way to iterate through keyframes and getting basic info about them like pts and dts? [20:18] <esing> Hey [20:18] <esing> Is avconv also supported in #ffmpeg? [20:19] <klaxa> no [20:19] <klaxa> #libav is the channel you are seeking [20:19] <esing> Thanks [20:20] <esing> (I try to record from /dev/video0 and when I tried it with ffmpeg I got the message: This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. [20:20] <durandal_1707> because you are using ffmpeg from older Libav release [20:21] <klaxa> you can get a recent static ffmpeg build or compile the latest version from source [20:23] <esing> Oh, I see [20:23] <iive> the message is a lie [20:23] <iive> libav doesn't support ffmpeg [20:24] <iive> ffmpeg supports ffmpeg. [20:24] <esing> I use the kubuntu package and I worry if I compile the newest ffmpeg source myself that it doesn't updates automatically anymore [20:25] <esing> Shall I ignore the message and work with the "older" ffmpeg? [20:27] <klaxa> right now you're not using ffmpeg [20:34] <iive> there should be debian package on the ffmpeg download page [20:40] <esing> klaxa, Hm? If I write ffmpeg in the terminal I suppose Iam using ffmpeg [20:43] <xlinkz0> esing: http://blog.pkh.me/p/13-the-ffmpeg-libav-situation.html [20:44] <xlinkz0> apparently ubuntu is lying to you [20:44] <esing> xlinkz0, thanks for the link [20:45] <xlinkz0> good thing i compiled from source, didn't even know about this until now :\ [20:51] <xlinkz0> if codec_context->time_base is 1/180000 and fps is 25 shouldn't the pts difference between frame i+25 and i be 180000? [20:55] <xlinkz0> is this a weekend thing or is the channel usually so quiet? [20:57] <zap0> everyones baked [22:31] <Fjorgynn> :D [22:36] <xlinkz0> why isn't ticks_per_frame built into time_base? [22:36] <xlinkz0> i mean what's the purpose of having two variables [22:37] <xlinkz0> is it for legacy compatibility or another reason? [22:43] <Fjorgynn> :D [22:43] <Fjorgynn> Dunno [22:43] <Fjorgynn> sorry [22:51] <brontosaurusrex> is -aspect 16:9 still valid option? [22:53] <xlinkz0> channel is dead come back monday [22:56] <Fjorgynn> :) [22:56] <Fjorgynn> brontosaurusrex: -vf aspect=16:9 [22:56] <Fjorgynn> I think [22:57] <xlinkz0> brontosaurusrex: ffmpeg -h | grep aspect [22:59] <brontosaurusrex> right, its still valid [23:00] <brontosaurusrex> probably due to a fact that audio files can't really have aspect :P [23:09] <brontosaurusrex> burek, do you have any cool notes on how to built completely static ffmpeg on debian? [23:10] <brontosaurusrex> i need x11_grab and fdk_aac [23:11] <esing> I worry if I compile ffmpeg that I have compatibilty issues with libav's libaries [23:12] <esing> Or maybe I read it wrong: "However, both Libav and FFmpeg periodically break backward compatibility in order to more easily provide new features, so applications built against one version will not always work with the next. " [23:12] <esing> (From: https://launchpad.net/~jon-severinsson/+archive/ffmpeg ) [23:14] <galtgendo> is there a way to reindex an avi file using ffmpeg alone ? [23:15] <galtgendo> also, is reindexing possible without reencoding ? [23:18] <brontosaurusrex> -c:v copy -c:a copy < galtgendo [23:20] <galtgendo> brontosaurusrex: that doesn't work - the video seems OK, but the audio is shot [23:21] <brontosaurusrex> shot? [23:31] <galtgendo> it's ...I don't know...garbled ? [23:55] <xlinkz0> esing: just remove all of libav and compile ffmpeg [23:56] <esing> xlinkz0, and if I'd wanted to use libav again I would do the same reverse, but install the libav ubuntu package, right? [23:57] <xlinkz0> idk, they conflict so choose one or the other [23:57] <xlinkz0> or install ffmpeg into a non shared folder [23:58] <esing> thanks for the tip [23:58] <xlinkz0> better yet do that, compile ffmpeg without installing [23:58] <xlinkz0> and alias to its location [00:00] --- Sun Apr 21 2013

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