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January 2013

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ffmpeg-devel.log.20130130
by burek 31 Jan '13

31 Jan '13

[00:16] <saste> is Rob Sykes on IRC? [00:23] <michaelni> no idea [00:49] <Compn> name doesnt sound familiar saste , but i could be wrong [00:49] <Compn> there used to be a sykes i think that came in here [00:50] <Compn> dunno if same person [00:50] <saste> Compn, he helped with the soxr thing in libswr [00:50] <saste> he also contributed to the wiki IIRC [00:50] <saste> at first I believed he was Skyler_ 'cause the assonance [00:51] <cone-913> ffmpeg.git 03Carl Eugen Hoyos 07master:3c3d68a97677: Fix 1bpp palettized png with width not a multiple of 8. [00:57] <wm4> wow png is complex [01:03] <Compn> the real question is, does anyone use ffmpeg for image processing :P [01:03] <Compn> not talking about film tiffs either [01:08] <wm4> Compn: I bet converting image<->video is pretty common [01:11] <saste> Compn, the question is, why not? [01:11] <saste> video processing is just a generalization of image processing [01:12] <Compn> i know :) [01:15] <cehoyos> saste: I still believe it would be much more important to get hands on a failing sample than more unreproducible reports about aresample... [01:18] <llogan> saste: what about Rob Sykes? i was in contact with him about a libsoxr package in Arch user repo [01:19] <saste> llogan, i asked if he was on IRC, he could review the filters from durandal since he's the author of the ported code [01:20] <llogan> oh. i see. you could always email him via the address in his code. he seemed nice enough. [01:33] <llogan> why did fflogger re-repeat my wiki edits? [01:34] <michaelni> llogan, ask burek [01:44] <llogan> ot: anyone here like/use linode? [01:47] <llogan> my old server is old and i don't need a dedicated/physical server anymore... [02:52] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:4d3d36254959: dirac/x86: fix compile without inline asm [04:48] <Compn> https://aws.amazon.com/elastictranscoder/ [04:49] <Compn> anyone contact amazon bout ffmpeg? :P [04:51] <wm4> Compn: can ffmpeg use the cloud? [05:02] <Compn> wm4 : thats the point. i've bet they've modified it to use the cloud [05:06] <wm4> oh, so you just want to play GPL police [05:26] <Compn> lol [05:27] <Compn> wm4 : since amazon isnt distributing thier encoder, there isnt much to police gpl wise [05:27] <Compn> asking nicely for changes never hurt anyone [05:28] <wm4> switch to AGPL [06:53] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:71f8d7045638: dirac/x86: fix compile without yasm [07:14] <leo2013> hello [07:14] <leo2013> I'm trying to write one encoder for h264 on hardware in ffmpeg. [07:14] <leo2013> and confused by the time stamp as the ffmpeg flow. [07:14] <leo2013> I use the decoder's value for outputs: "pkt->pts = frame->pkt_pts; pkt->dts = frame->pkt_dts;" [07:14] <leo2013> but not correct.Who could give some suggentions,thanks! [07:15] <leo2013> so confused about how to add the time stamp one frame after one frame. [10:06] <cone-913> ffmpeg.git 03Peter Ross 07release/0.11:155a0bed9722: wtvdec: demux thumbnail picture to AVStream.attached_pic [10:06] <cone-913> ffmpeg.git 03Peter Ross 07release/1.0:4f8b7eb87f64: wtvdec: demux thumbnail picture to AVStream.attached_pic [10:07] <cone-913> ffmpeg.git 03Peter Ross 07release/1.1:54e19092fd60: wtvdec: demux thumbnail picture to AVStream.attached_pic [10:31] <cone-913> ffmpeg.git 03Carl Eugen Hoyos 07master:91f359292a52: Correctly mark non-default streams when muxing matroska. [11:00] <kierank> Compn: they distribute ffmpeg to customers [11:00] <kierank> oh wait they don't [11:00] <kierank> never mind [11:00] <kierank> i thought you get a vm in the cloud [11:00] <kierank> affero gpl would be rather comical for me [13:04] <ubitux> hey [13:04] <ubitux> got some little free time, any stuff i should look at in priority? [13:08] <wm4> ubitux: just now there were some subtitle crash tickets [13:11] <ubitux> doesnt sound fun :( [13:11] <ubitux> but ok [13:12] <ubitux> 404, and it's not a crash [13:13] <Compn> ubitux : i have a subtitle that mplayer cannot display properly , i think its a strange encoding if you want to take a look :P [13:13] <Compn> or uh [13:13] <Compn> theres a stereo3d patch [13:13] <Compn> to add another red/blue color [13:13] <ubitux> utf16? [13:13] <Compn> possibly [13:13] <ubitux> can i have a look to the sub? [13:14] <Compn> ubitux : http://www.datafilehost.com/download-9885ce69.html [13:14] <Compn> or i can put it in incoming [13:15] <Compn> "Cloud Atlas (2012).srt" [13:15] <Compn> is in incoming [13:15] <ubitux> wait wat is this [13:15] <ubitux> :D [13:15] <ubitux> hahaha the tags are encoded too [13:15] <Compn> i have no idea, but its so broken i couldnt figure it out [13:15] <ubitux> that's awesome [13:16] <Compn> i just told the user to download a different .srt file :) [13:16] <ubitux> that's actually very interesting :) [13:17] <Compn> can iconv handle it? i dont have it installed ... [13:17] Action: wm4 should try to get rid of subreader.c [13:17] <Compn> wm4 : get libass to take all that code? :P [13:17] <Compn> wm4 : or use ffmpeg parsers ? [13:18] <wm4> Compn: I doubt ibass wants to do it [13:18] <wm4> so, ffmpeg [13:18] <Compn> obviously libass wont do it [13:18] <wm4> but what about libav users... [13:18] <wm4> well it could [13:18] <ubitux> Compn: don't you get something with -utf8 with mplayer? [13:19] <Compn> wm4: you could submit patch to libav from ffmpeg [13:19] <ubitux> Compn: there is an utf8 bom in the file [13:19] <av500> yep [13:19] <Compn> ubitux : oh that works [13:19] <av500> so assume utf8 [13:19] <wm4> Compn: backporting all that stuff is a lot of effort [13:19] <ubitux> Compn: you'll get some trouble with the tags i believe though [13:19] <Compn> be nice if it were autodetected [13:20] <Compn> wm4 : wont work as just one big patch huh? [13:20] <ubitux> haha like ffv1? [13:20] <wm4> definitely not [13:20] <ubitux> it works if it's from ffmpeg [13:21] <ubitux> it won't if it's original work [13:21] <ubitux> (see BBB fighting with Diego recently) [13:21] <Compn> lol [13:21] <ubitux> if it's some stuff from ffmpeg you can squash them altogether [13:21] <wm4> lol? [13:21] <ubitux> "it's crap anyway" [13:22] <ubitux> wm4: don't you remember all the ffv1 commits squashed in libav? [13:22] <wm4> so libav is less strict with massive backport patches from ffmpeg? [13:22] <wm4> no [13:22] <ubitux> see 0f13cd3187192ba0cc2b043430de6e279e7b97c3 [13:22] <wm4> wasn't aroind [13:22] <ubitux> that commit is awesome [13:23] <ubitux> there is like 20 commits in one [13:23] <Compn> thats my understanding that libav doesnt do 'histories' from ffmpeg ... [13:23] <Compn> as long as you clean up a patch to get it in libav (unless they decide to just rewrite the whole thing and make another api) [13:23] <wm4> nice [13:24] <Compn> which is what they did instead of taking libswsresample iirc [13:24] <ubitux> you just have to make sure the codng style is correct [13:24] <ubitux> if the coding style is correct, then it's certainly good code [13:24] <Compn> enough trolling! [13:24] <Compn> :P [13:24] <ubitux> yeah [13:24] <ubitux> sorry :)) [13:25] <wm4> lolol [13:25] <av500> code is read way more often than written [13:25] <av500> so style matters [13:25] <Compn> too bad everyone hates code comments :P [13:26] <ubitux> av500: so libswresample was rewritten from scratch because of the unreadable coding style? [13:26] <av500> did I say that? [13:26] <ubitux> i never implied this [13:26] <ubitux> i'm asking [13:26] <wm4> god praise uncrustify [13:26] <av500> ubitux: I never looked into it [13:26] <av500> so dont ask me [13:27] <ubitux> av500: yeah, none of the libav dev did :( [13:27] <wm4> it's a survival tool when dealing with mplayer code [13:27] <ubitux> uncrustify has a lot of issues [13:27] <ubitux> especially with the bad config libav is using [13:28] <wm4> produces some funny constructs [13:28] <av500> ubitux: but yes, somebody should run indent over it [13:28] <ubitux> i don't think it matters [13:28] <av500> for (ch=0; ch<srcs->ch_count; ch++) { [13:30] <burek> just to paste the url here before i forget about it, i dont know if the guy has submitted a bug report or not, but he did find some issue related to negative ctts values in ffmpeg: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=778 [13:30] <ubitux> i see no real problem with it, except the inconsistency with some other part of the code base [13:30] <ubitux> it's not less readable, just different [13:31] <av500> opinions differ on that I guess [13:31] <av500> I find it hard to read [13:31] <av500> but ymmv [13:31] <av500> still, making an effort to style the code is not wasted [13:32] <Compn> nevcairiel : is the dxva2 patch still on your list to review ? [13:32] <ubitux> oh nice the filters from Paul [13:32] <Compn> just curious, i dont want to bug anyone about it. [13:35] <burek> just curious about mentioned coding style above, does ffmpeg have any "official" coding style, that could just be applied at entire source code (just like a lot of software companies do), making it irrelevant who writes the code in which fashion [13:35] <ubitux> that's what Diego has been working on for 10 years [13:35] <ubitux> ask him [13:36] <av500> http://ffmpeg.org/developer.html#Coding-Rules-1 [13:36] <ubitux> short answer: yes we have a recommended coding style, no it's not that simple to apply it everywhere [13:36] <av500> ...The main priority in FFmpeg is simplicity and small code size in order to minimize the bug count. [13:36] <ubitux> especially when it sometimes makes sense to violate these rules to give more sense to the code [13:36] <av500> some people mistake "fewer spaces" with "small code size" [13:40] <Compn> ehe [13:40] <Compn> http://lists.libav.org/pipermail/libav-devel/2013-January/042729.html [13:40] <Compn> neat [13:40] <Compn> basically 'port avisynth filter' request [13:41] <Compn> and probably 'make ffmpeg filters work in vlc' [13:44] <burek> ubitux, i see, well, in Eclipse, there is a simple file (a template) that describes all the stuff related to the coding-style, like indent in functions, loops, breaking long lines, etc. So, when you copy paste a bunch of crap in the editor it styles it even automatically for you. Now, I don't know if there is any standalanone (batch) tool that does the similar thing, but it might be worth considering using it [13:45] <ubitux> it's not that simple [13:45] <burek> also, those special cases can be marked with special kind of comments, that would tell the tool not to touch that part of the code [13:45] <ubitux> try to use it with templated code, or simply macro [13:46] <ubitux> also, you'll start the enternal debate on "a*b + c" more readable than "a * b + c" sometimes [13:46] <ubitux> also, various vertical align on a lot of stuff [13:46] <Compn> "pretty printing" [13:46] <ubitux> sometimes done because it's related data, sometimes not, which an automatic script can't guess [13:46] <ubitux> and will obviously destroy [13:46] <ubitux> or make insane [13:46] <Compn> dont forget, there were cosmetics patches that changed benchmarks/speed :D [13:46] <Compn> long long ago [13:47] <burek> well, you agree upon one "standard" coding style pattern and all the small parts of that coding styles, that you could disagree on, could be solved with your own such local tool which will style it like you want [13:47] <ubitux> yes, and some that broke stuff like we saw in libav several times [13:47] <Compn> unless i made that up [13:47] <burek> so everybody is happy [13:47] <Compn> :D [13:47] <ubitux> (and which are still unfixed because they don't watch ffmpeg history) [13:48] <Compn> i miss the days of '10l' [13:48] <Compn> :P [13:48] <ubitux> that still happens at time in ffmpeg history [13:48] <burek> well ok, but not willing to change things, in order not to break anything, will not get you anywhere except it will stall you [13:49] <Compn> burek : it will stall the person trying to unify the code, yep [13:49] <burek> ok, those were just my 2 cents, nothing more :) [13:49] <Compn> how are you going to not break architechtures you dont own [13:49] <av500> if you restyle code, the generated asm should stay identical [13:49] <Compn> like bfin, mips, etc [13:49] <av500> thats easy to test [13:49] <Compn> emulated? [13:50] <ubitux> av500: not always that easy [13:50] <Compn> its a slow process, and i'm not against it anyhow, if you want to do it :P [13:50] <av500> ubitux: ? [13:50] <Compn> burek : we just like to bikeshed about it :P [13:51] <ubitux> av500: hint: macro with stuff like __line__, mixed preproc, etc [13:51] <av500> that will still be compiled [13:51] <av500> and you can compare asm [13:51] <ubitux> you need to do various compilations config [13:51] <ubitux> sometimes [13:51] <av500> sure [13:51] <ubitux> which you might not be able to do depending on your config [13:51] <ubitux> or at least, it can be a pain [13:51] <ubitux> and most people never do it [13:52] <ubitux> i'm pretty sure even diego doesn't btw [13:52] <Compn> ffmpeg (and libav) are getting lots of patches from various companies now [13:52] <Compn> thats neat [13:52] <Compn> intel copyrights, google copyrights, nvidia ... [13:52] <Compn> mips [13:53] <ubitux> the mips guys are awesome :) [13:53] <ubitux> they are very productive [13:53] <ubitux> funny that libav reject them because of "beee inline asm eviiil" btw [13:53] <ubitux> oups sorry i'm trolling again [13:53] <JEEB> I think many people within libav facepalmed at that as well [13:53] <Compn> the mips code isnt in libav ? [13:54] <JEEB> since afaik there is no other way of doing it >_> [13:54] <ubitux> Compn: yup :) [13:56] <j-b> good morgen [13:57] <Compn> j-b : where are you with libavfilter support? :P [13:57] <j-b> Compn: I am in Italy :) [13:57] <j-b> Compn: maybe this is not the right answer, though :D [13:57] <Compn> doesnt sound ideal for coding... [13:57] <Compn> :P [13:57] <Compn> hows the weather ? [13:58] <Compn> today will be about 20C and tonight will be 0C for me :) [13:58] <Compn> crazy weather [14:05] <j-b> Compn: :) I am with Luca, so, maybe we can discuss about this topic. [14:10] <Compn> ubitux : btw we tried -subcp utf8 not -utf8 . thats why it didnt work in mplayer [14:10] <Compn> why these options do different things i have no clue :( [14:11] <Compn> wm4 : maybe you could fix it in your fork :P [14:19] <michaelni> burek, can you add a feature to fflogger ? [14:19] <michaelni> i mean something that alerts us when a fate client stoped sending results ? [14:20] <michaelni> the machines have the tendency to get stuck occasionally [14:23] <burek> i see [14:24] <burek> like a timeout or something [14:30] <burek> how long interval should i set [14:32] <Compn> burek : 24hours [14:32] <Compn> if i had to guess :) [14:32] <burek> ok :) [14:32] <Compn> could be 12 hours , i dont know how often some machines do builds [14:38] <michaelni> it differs between machines how often they do builds [14:40] <Compn> yeah [14:42] <michaelni> some run it just twice a day some run the tests in a loop continiously but the loop may contain many tests so still can have significant delay for each individual configuration [14:42] <michaelni> and most machines wait when theres no new commit since last time [14:45] <Compn> sounds like this will be hard to code [14:45] <Compn> unless you put it on the client side to ping 'alive' messages or something [14:46] <Compn> or make it like rss ? [14:46] <nevcairiel> just make it a conservative timeout, dont think any machines need more then 24 hours for a new build [14:46] <nevcairiel> doesnt have to be immediate [14:46] <ubitux> durandal_1707: nit: be consistent with the usage of struct & typedef [14:46] <nevcairiel> just so you find machines easier which hang for a prolonged time [14:46] <ubitux> durandal_1707: typically chan_cache [14:47] Action: durandal_1707 ignores nits [14:47] <ubitux> :) [14:48] <durandal_1707> you mean to caps it? [14:48] <ubitux> struct chan_cache, typedef ChanCache [14:49] <ubitux> use only struct, or typedef with anonymous struct, or whatever [14:49] <ubitux> but if you use the typedef, ChanCache should be use [14:49] <ubitux> there is a leak in your realloc if it fails [14:50] <ubitux> the assert proposed by stefano in the switch case makes the code more obvious, and also makes sure the compiler won't warn in some cases (not here, yet) [14:51] <ubitux> durandal_1707: is it ok to only copy the the pts and not all the other props with the new buffer? [14:52] <ubitux> (sorry for not doing a proper ml review) [14:55] <durandal_1707> ubitux: other filters do it [14:55] <ubitux> ok [14:55] <ubitux> in the same kind of scenario? [14:56] <ubitux> anyway, that's a nice patch you sent there [14:56] <ubitux> no more comment from me [14:57] <ubitux> i should compare with the biquad filter from ebur128, but i'm too lazy to do it now :) [14:57] <cone-913> ffmpeg.git 03Luca Barbato 07master:4e0bc996d995: bfin: unbreak compilation [14:57] <cone-913> ffmpeg.git 03Martin Storsjö 07master:61d36761efda: movenc: Simplify code by using avio_wb24 [14:57] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:afb4bc3d291f: Merge remote-tracking branch 'qatar/master' [15:02] <durandal_1707> ubitux: how/what would you compare? [15:03] <ubitux> the implementation, to see if one is more optimal than the other [15:04] <durandal_1707> nonsense, it is simple formula [15:04] <durandal_1707> which cant be SIMD optimised AFAIK [15:05] <ubitux> seems i used a packed method [15:06] <durandal_1707> feel free to improve it ... [15:07] <ubitux> ebur128 is a bit particular since you have 2 successive filters [15:07] <ubitux> some optim can be done in it to merge them into one, but i didn't look [15:07] <ubitux> durandal_1707: sure, i was just curious, not blocking in any way [15:12] <durandal_1707> ubitux: well, you can always optimize specific situation [15:14] <durandal_1707> the other thing i wondered is to not have separate filters but just one filter with option to change type, but this may not be any more user friendly [15:14] <ubitux> (nit: the enum should be in caps) [15:15] <durandal_1707> yes, but i would need to add 3rd arg to my macro [15:15] <durandal_1707> i hate macros [15:15] <ubitux> i don't really know [15:16] <durandal_1707> i think from ms office days ... [15:16] <ubitux> sounds fine to me with multiple filters [15:16] <ubitux> the other way around is fine as well :p [15:17] <durandal_1707> DEFINE_BIQUAD_FILTER(BIQUAD, biquad, ... [15:18] <durandal_1707> or i would add another macro to map biquad to BIQUAD? [15:18] <durandal_1707> when i code something, i just ignore such nits, and move on... [16:31] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:d8a7c4958e5a: mpegvideo_enc: factor expression out [16:47] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:c2992b705381: msrledec: move output pointer test up [16:47] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:dbaae33c2c71: msrledec: move loop into switch [16:47] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:d2e0a276d593: msrledec: merge switches [16:59] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:32de2831036a: avstring: fix "warning: return discards const qualifier from pointer target type" [18:21] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:b926cc7834d5: mss3: prevent AC state from becoming invalid in rac_normalise() [19:51] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:4a2da83a787b: dnxhddec: fix integer overflow / index check [20:04] <saste> do we support image sizes where size is not a multiple of the chroma subsampling power? [20:05] <saste> does it even make sense? [20:06] <durandal_1707> it makes sense to me [20:06] <durandal_1707> it is just that IIRC we drop last collumn of pixels [20:08] <saste> would be possible to support odd-sized cropping/padding? [20:08] <saste> i tend to believe this would not be possible (losslessly) [20:13] <cone-913> ffmpeg.git 03PaweB Hajdan, Jr 07master:0451ff295a9f: oggparsevorbis: use av_realloc consistently [20:13] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:033f1644b59a: fixup_vorbis_headers: add missing malloc failure check [20:29] <barque> when I use ffmpeg to encode theora/OGG videos AvFormatContext->duration comes out bad [20:29] <barque> Any idea on how to correct this? [20:30] <barque> is there some other way I can approximate time [20:30] <barque> more or less [20:30] <barque> even to the second it's fine [20:30] <barque> I've used ffmpeg2theora.exe on top of ffmpeg before, but then I get a LOT of "errors" on packet_reads [20:30] <michaelni> barque, submit a bugreport [20:30] <barque> and no EOF [20:31] <barque> michaelni, right but, is there some work around? [20:31] <barque> I've seen the -t option saying it can transcode time [20:31] <barque> would that help or something? [20:32] <michaelni> you can decode the whole video and count the frames [20:32] <barque> :S [20:33] <barque> No really would need it initially [20:33] <barque> timestamps on the packets are spot on [20:33] <michaelni> bugreport <--- [20:33] <barque> so decoding the video and reading last time stamp would work [20:35] <barque> but yeah can't do that [20:35] <barque> how do you carry over the same header info from the other file? [20:35] <barque> maybe it can extract from there or something? [21:21] <durandal_1707> michaelni: is'nt av_realloc when fails leaks memory in 033f1644b59a [21:22] <durandal_1707> nvm... [23:16] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:984add64a41c: wma: check byte_offset_bits [23:21] <cone-913> ffmpeg.git 03Paul B Mahol 07master:10e4905dd9b7: auenc: remove put_au_header() and merge its code into au_write_header [23:21] <cone-913> ffmpeg.git 03Paul B Mahol 07master:0dcfccaa691b: auenc: strict check for supported codec [23:24] <durandal_1707> wtf, why suddenly ffmpeg lost bunch of + on g+ [23:27] <wm4> maybe you trolled too much [23:28] <durandal_1707> i did not trolled at all [23:51] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:a084884b628f: flashsv: clear blocks array on reallocation [00:00] --- Thu Jan 31 2013

1 0

ffmpeg.log.20130130
by burek 31 Jan '13

31 Jan '13

[00:47] <obiwahn> ffmpeg -sameq -i ./input_file.mp4 -aspect 16:9 ./output_file.mp4 i found this line on the internet an dit hleps me to fix the aspect ratio of a movie [00:47] <obiwahn> but i can not find the -sameq option in the man [00:47] <obiwahn> what does it mean? [00:53] <obiwahn> is there a way to set the decoding aspect ratio without reencoding? [00:53] <obiwahn> after setting it with makemkv mplayers initila box has the right size but then snaps back to the wrong aspect [00:57] <saste> obiwahn, -aspect and -codec copy, it might work [00:57] <obiwahn> i have tried that as well as chaning some properties in the file [00:57] <obiwahn> like the with but it did not help at all [00:57] <saste> width? [00:57] <saste> no you can't change that without re-encoding [00:58] <saste> but you usually can set the aspect with -aspect [00:58] <obiwahn> + Display width: 750 [00:58] <obiwahn> ill do it [00:59] <obiwahn> it sadly blew the encoded avi up to 3 times it size [00:59] <obiwahn> now i convert the original first [00:59] <obiwahn> and then encode it with 2 passes again [01:00] <obiwahn> i need to switch form mencoder to ffmpeg:) [01:03] <obiwahn> http://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide <-- where is the output file for the first pass? [01:06] <obiwahn> ah got it:) [01:11] <teratorn> anyone could tell me the ffplay invocation for playing raw mono audio file, S16 samples? [01:19] <obiwahn> mh the file gets a lot bigger even thoug i just copy it using anohter aspect [01:19] <obiwahn> why does it get bigger there is not more information [01:21] <obiwahn> http://paste.debian.net/230289/ [01:22] <obiwahn> i am still not sure waht sameq does it does not stand for quality and it is obsolete:) but without the option the copy looks very bad [01:22] <obiwahn> it grew from 2. [01:23] <llogan> that paste is worthless [01:23] <obiwahn> from 4.1 to 6.1 gb [01:23] <llogan> seems like you're not using copy. [01:23] <obiwahn> then the output is worthless [01:23] <llogan> you need to include the command and the complete console output, not just a few lines [01:24] <obiwahn> http://paste.debian.net/230290/ [01:24] <llogan> do not use sameq. [01:25] <obiwahn> what should i use instead? [01:25] <llogan> you're re-encoding, so of course the size will be different [01:25] <obiwahn> without smaeq the resulting file is 380mb big ... [01:26] <llogan> if "-codec copy" did not work with -aspect then you will probably have to re-encode. [01:28] <obiwahn> ffmpeg -codec copy -i ./16.video.m2v -aspect 16:9 ./16.video-fix2.m2v gives me unknown decoder copy [01:28] <llogan> because you're using -codec copy as an input option [01:28] <llogan> option placement matters [01:30] <obiwahn> ah ok lets see:) it is my last try for today its too late here i was just waiting for my girl to come home:) thank you saste and llogan! [01:32] <llogan> teratorn: ffplay -f s16le -ar 48000 input.pcm [01:32] <teratorn> llogan: thanks yeah, turned out i needed s16be to not hear garbage [01:36] <klaxa> can anyone tell me how to build docs for libass? i can't find it in the makefile and i also can't find where to contact devs on irc [01:36] <klaxa> are there even docs? [01:40] <saste> klaxa, how is libass related to ffmpeg? [01:41] <klaxa> probably in no way, like i said i can't find where to contact the devs and i figured someone in here might also work on libass [01:41] <saste> klaxa, maybe ask on #mplayer [01:41] <klaxa> will do thanks [01:41] <saste> there is just one main author of libass, you may contact him by email [01:42] <klaxa> ah... okay i should be able to get that mail from the sourcefiles then [03:56] <praveenmarkandu> hi guys. is there any info on performance limitations of ffmpeg [03:56] <praveenmarkandu> like how many transcodes can be run simultaneously [03:56] <praveenmarkandu> or is it purely limited by the hardware [04:06] <epifanio> hi All [04:06] <epifanio> i'm tring to make an animation for a sequence of png's images .. i'm trying with the following command : ffmpeg -f image2 -i *.png -r 25 "output.mov [04:09] <llogan> epifanio: ok [04:09] <epifanio> but i receive this error from the command line : http://paste.debian.net/230330/ [04:09] <epifanio> sorry .. i tried to reduce the lines in the paste [04:10] <llogan> try adding "-pattern_type glob" as an input option [04:11] <llogan> -r 25 is default for inputting images, so it is redundant, and "-f image2" may not be needed either [04:11] <epifanio> like : ffmpeg -i *.png -pattern_type glob "output.mov" ? [04:12] <llogan> no [04:12] <llogan> ffmpeg [global options] [input options] -i input [output options] output [04:13] <epifanio> i tried : ffmpeg -pattern_type glob -i *.png "output.mov" [04:14] <llogan> did it work this time? [04:14] <epifanio> i got the same error log [04:14] <epifanio> pasting it [04:15] <epifanio> http://paste.debian.net/230336/ [04:17] <llogan> maybe libavformat was compiled with globbing support [04:17] <epifanio> i don't know if this help .. i'm on osx using ffmpeg installed with homebrew [04:18] <llogan> you can try cat instead [04:18] <llogan> cat *.png | ffmpeg -y -f image2pipe -c:v png -i - output [04:19] <epifanio> terrific! [04:20] <llogan> i meant "maybe libavformat was *not* compiled with globbing support" [04:20] <epifanio> it worked .. super fast [04:20] <llogan> what are you using the output in? [04:20] <epifanio> it is an animation of wave height in the gulf of main [04:20] <llogan> i mean are you going to edit it in FCP or something? [04:21] <epifanio> no, i'll embed it in a html page [04:21] <epifanio> without editing it [04:22] <llogan> why mov? [04:23] <epifanio> i used mpeg [04:23] <epifanio> no ? [04:23] <epifanio> i means : cat *.png | ffmpeg -y -f image2pipe -c:v png -i - output.mpeg produces an mpeg i guess [04:23] <epifanio> correct ? [04:23] <llogan> oh, it's bigger than i thought. 1200x1200. i assumed you were wanting it to play in borwser, but i guess viewers will just download it? [04:24] <epifanio> no i was supposed to publish it in browser [04:24] <epifanio> humm .. [04:24] <epifanio> the final size is 3.4 mb [04:25] <epifanio> do you have any hints to decrease the size without loosing too much resolution ? [04:25] <llogan> mpeg probably won't play in a browser. H.264 in mp4 is generally used (with a flash player). [04:25] <llogan> file size? [04:25] <llogan> or frame size? [04:26] <epifanio> filesize in mpged is 3.4 mb [04:26] <llogan> i don't know what you mean by "size" [04:28] <epifanio> how "big" is the video in megabyte [04:29] <llogan> see https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [04:29] <epifanio> thanks, i was looking on how to change the output format to : H.264 mp4 [04:29] <llogan> then add "-movflags +faststart" as an output opyion [04:29] <llogan> now i must go [04:31] <epifanio> llog thanks! i'm trying [04:32] <mattt55> hi all :-) i'm using the deb-multimedia version of ffmpeg on debian squeeze to encode some audio tracks... i'm having trouble getting reliable ogg's (with libvorbis) - specifically, ffprobe shows (for some files) a rediculous duration, and ff shows NaN in html5 players despite that apache is serving byte-ranges... any ideas? [04:32] <mattt55> the files do play, btw... but frequently only for a few (seemingly random amount of) seconds [04:32] <mattt55> also encoding same files to mp3, which seems to be going swiimingly :-) [04:34] <mattt55> (errr... i mean same origin files - a mixture (usually flac, mp3 or ogg)) [05:58] <leo2013> hello [05:59] <leo2013> I want to add one encoder for h264 on hardware but not x264. [05:59] <leo2013> I'm meeting one problem about how to calculate the time stamp for h264 in ffmpeg, [05:59] <leo2013> in which one after one frame comes continously.I try to use [05:59] <leo2013> pkt->pts = frame->pkt_pts; [05:59] <leo2013> pkt->dts = frame->pkt_dts; [05:59] <leo2013> but it's not correct. [05:59] <leo2013> Who could help me about this,thanks! [07:23] <praveenmarkandu> does FFMpeg support encoding WEBVTT subtitles into mpegts streams yet? [07:23] Action: praveenmarkandu referencing https://datatracker.ietf.org/doc/draft-pantos-http-live-streaming/?include_… [07:24] <praveenmarkandu> section 4. Media Segments [07:27] <tomahawk> I am new to FFmpeg. I want to know how to set sample aspect ratio (SAR) in cli [07:29] <tomahawk> Please someone help me. Currently I am working in a Vindoz machine. Using ffmpeg-20130103-git-43adc62-win32-static [07:31] <tomahawk> I have read the docu already. Tried the same commands in console. None working. Here is the command I tried. [07:31] <tomahawk> http://pastebin.com/CmZgNVHn [07:38] <wakoinc> I am working on a project that involves processing audio from multiple microphones and cancelling background noise. Is FFmpeg a good fit for processing the audio streams and outputting a single stream? [07:43] <praveenmarkandu> tomahawk, whats the output [07:43] <praveenmarkandu> pastebin [07:49] <tomahawk> Praveenmarkandu, here is the output: http://pastebin.com/NcLuhRBV [08:05] <K-Rich> hi all, i am having an issue with well, i guess tearing, blue and white flashes when recording my desktop. I am using linux mint 13, kernel 3.2.0-36, ffmpeg 0.8.5-4:0.8.5-0ubuntu0.12.04.1, ndivia driver 304.64.... i am using the command line: ffmpeg -acodec pcm_s16le -f alsa -i pulse -s 1280x800 -qscale 1 -r 30 -vcodec rawvideo -f x11grab -i $DISPLAY -s 640x356 -vf pad=640:360:0:2:000000 ~/Desktop/`date +%F_%T`.avi' --- Any ideas? [08:07] <K-Rich> cinnimon desktop [08:16] <tomahawk> praveenmarkandu, I can't even determine the source video's par with -i option. [08:32] <K-Rich> http://www.youtube.com/watch?v=sphlXIJp-vU sample of my issue [08:33] <K-Rich> anyone? [08:33] <K-Rich> i mean i'd hate to make a screencast and some epileptic have a seizure [08:35] <K-Rich> brb, please memo if you kno a solution in the mean time [08:38] <K-Rich> same with vsync on or off in nvidia drive, same with cinnamon2d [08:42] <praveen> tomahawk, par? you mean sar or dar? [08:42] <tomahawk> I mean SAR [08:43] <praveen> i normally use ffprobe [08:44] <praveen> try ffprobe -v quiet -print_format json -show_format -show_streams <inputfile.mp4> [08:44] <tomahawk> Let me try that [08:44] <praveen> you should pipe that to less or something [08:45] <tomahawk> That works. [08:45] <praveen> or | grep aspect [08:45] <K-Rich> praveen or tomahawk would you know anything about my issue? [08:46] <tomahawk> Thanks. The source video's SAR and DAR is 0:1. So can I change it to 1:1? Which command to use? [08:46] <praveen> im a noob at ffmpeg. but wouldnt your -aspect command conflict with your setsar [08:47] <tomahawk> So should I remove the -aspect command [08:48] <praveen> not only that, the way setsar is used seems wrong. it says invalid argument [08:49] <tomahawk> Can you tell me how to use the setsar command? I have read the whole docu several times. But nothing works... [08:54] <praveen> try -vf setsar="1:1" [08:54] <praveen> im just guessing [08:54] <praveen> probably wrongg [08:54] <tomahawk> Thanks. Let me try that. [08:55] <praveen> i just googled ffmpeg setsar and thats how people have written it [08:55] <praveen> seem to get more output than you [08:57] <tomahawk> I have been googling about this for a couple of days. Reading the docu. Trying examples. Ok. [08:58] <tomahawk> My main issue is that I have to upload some files to vimeo. Vimeo has the clear spec that the video should have 1:1 Pixel Aspect Ratio (aka Sample Aspect Ratio). [09:02] <tomahawk> praveen, here is the output: http://pastebin.com/qGxULavs [09:18] <praveen> tomahawk, sorry mate. way beyond me [09:33] <tomahawk> Ok. Thanks for your help. I think I'll find out the rest myself. Thanks alot [09:48] <tomahawk> praveen, through your lead, I could finally fix it. Here is the command: http://pastebin.com/a0Tc99FF / thanks alot [10:42] <praveen> Hi. quick question. does ffmpeg support full multithreading for decoding and encoding? [10:45] <Mavrik> yes [10:45] <Mavrik> it depends on encoder / decoder [10:45] <Mavrik> but ffmpeg does support multithreaded encoders / decoders :) [10:47] <praveen> Mavrik, specifically the x264 encoder? [10:47] <praveen> yes? [10:47] <Mavrik> x264 supports multithreaded encoding very well :) [10:54] <praveen> Mavrik, thanks [10:55] <praveen> actually, another question. is the performance of FFmpeg just limited to the amount of hardware i have [10:55] <praveen> does it scale well? [11:01] <Mavrik> praveen: um [11:01] <Mavrik> praveen: depends on what you're doing [11:01] <Mavrik> x264 scales well to about 4-6 cores [11:01] <Mavrik> after that more cores don't actually give more performance [11:01] <Mavrik> so having as high CPU clock as possible is important [11:01] <Mavrik> everything else is secondardy [11:01] <Mavrik> *secondary [11:04] <praveen> Mavrik, but what happens if I have multiple transcodes going on? [11:05] <praveen> would more than 6 cores help? [11:06] <Mavrik> probably [11:06] <Mavrik> but usually buying more machines for that use case is cheaper [11:06] <praveen> hmmm yep. company looking to buy hardware for our transcodes [11:07] <praveen> btw, are you a dev? [11:07] <JEEB> Mavrik, that depends on the settings you're using btw @ x264 scaling, and lately'ish after D_S made the lookahead multithreaded as well [11:07] <Mavrik> JEEB: possibly [11:07] <Mavrik> I've tested that about 3 months ago [11:07] <JEEB> also the development tree has new optimizations now in general :3 [11:07] <Mavrik> and it stopped scaling at about 6 cords [11:07] <Mavrik> *cores [11:07] <Mavrik> for "slow" preset 720 transcodes [11:07] <JEEB> make sure you're not restricted by the decoding [11:08] <JEEB> and yes, I remember at one point the slower presets didn't scale as much [11:08] <JEEB> but after lookahead multithreading it should be better off [11:09] <JEEB> I would like to test at some point with a beefy machine :V [11:09] <JEEB> too bad people generally don't have 12-core CPUs available for some testing [11:09] <JEEB> and/or 4xquad core machines [11:10] <praveen> JEEB, would help test, but dont plan on buying such high specs. [11:11] <JEEB> well, yeah -- in general a nice ivy quad core would already handle what you need nicely, and then just buy more of them when needed [11:11] <Mavrik> yeah, I test on 6-core Xeon [11:12] <Mavrik> 8-cores are fricking expensive [11:12] <Mavrik> cheaper to just get another 1U unit [11:12] <JEEB> yes [11:12] <JEEB> haswell optimizations are also incoming IIRC, although some of it ended up being rather... derpy from the instruction set specifications [11:13] <JEEB> (and no, the x264 folk don't get new CPUs before hand, the only case of them getting some was when a rather big company poked Intel -- and even then the machine was gotten very close to release date for testing) [11:14] <Mavrik> mhm [11:21] <praveen> lol [13:51] <cousteau> in which version were filters (-vf) introduced? [13:52] <burek> did you check git log [13:53] <burek> http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=-vf [13:53] <burek> i think this one is the one you are looking for: "rename -vfilters cli option to -vf" [13:57] <cousteau> So what version is that? [13:57] <burek> it says there [13:57] <cousteau> mine is so incredibly old that it doesn't even have a -vfilters option [13:57] <burek> the exact commit version [13:58] <burek> what does your ffmpeg say in the version text? [13:58] <cousteau> 0.6 or so [13:58] <cousteau> (yeah, so old it hurts) [13:59] <cousteau> I think I'll just use imagemagick for the resizing operations [14:00] <burek> :) [14:00] <cousteau> I was just converting a video to frames and then resizing the frames; I can use imagemagick for that (which involves adding extra commands and a for loop, but anyway) [14:01] <cousteau> wait, no need for a for loop if I use mogrify [14:03] <burek> cousteau [14:03] <burek> could you try static builds of ffmpeg? [14:07] <cousteau> could be an option [14:09] <cousteau> "static" = "no lib dependencies", right? [14:10] <cousteau> anyway, I have already fixed it using imagemagick mogrify. It's an ugly workaround but it works. Anyway, good to know I can get static builds. [14:11] <jeje34> Hi to all;-) [14:12] <jeje34> I have a problem when I use av_image_alloc function, it return me error -22 "Invalid data found when processing input" [14:13] <jeje34> In my code, I use avcodec_alloc_frame to get my AVFrame* pointer [14:14] <jeje34> and jsut after, I call av_image_alloc to allocate the AVFrame data buffer [14:14] <jeje34> but it return me -22 [14:17] <jeje34> the calling code is av_image_alloc(m_lpFrame->data, m_lpFrame->linesize, AV_PIX_FMT_RGB32, m_lpCodecCtx->width, m_lpCodecCtx->height, 32); [14:18] <jeje34> with width=720 and height=576 [14:24] <burek> did you check the docs for avcodec_alloc_frame [14:26] <jeje34> burek > yes: Allocate an AVFrame and set its fields to default values. [14:27] <burek> http://ffmpeg.org/doxygen/trunk/group__lavc__core.html#gad5f9212dec34c9fff0… [14:28] <jeje34> burek > yes and so, it return me a valid pointer [14:28] <jeje34> burek > but it doesn't allocate the buffer of the AVFrame [14:29] <jeje34> burek > so I need to call av_image_alloc just after it seems [14:29] <burek> it allocates AVFrame struct, not buffer [14:29] <burek> zes [14:29] <burek> yes [14:31] <jeje34> burek > but if I make my call to av_image_alloc(m_lpFrame->data, m_lpFrame->linesize, AV_PIX_FMT_RGB32, m_lpCodecCtx->width, m_lpCodecCtx->height, 32); just after, the av_image_alloc return me an error and the buffer (data pointers in my AVFrame) are not allocated [14:32] <burek> jeje34, did you check examples [14:32] <burek> related to that function [14:32] <burek> (on that page) [14:33] <jeje34> burek > when looking this example:http://ffmpeg.org/doxygen/trunk/doc_2examples_2decoding_encoding_8c… [14:34] <jeje34> In the decoding video part (it's my case, decoding H264 from an IP camera) I just see the avcodec_alloc_frame and never see he av_image_alloc [14:37] <burek> just follow the examples [14:37] <burek> People spent time creating time, now you spend time understanding them :) [14:38] <jeje34> yes but if I'm true, I never have to call the av_image_alloc [14:38] <jeje34> I can't understand this [14:40] <jeje34> or the avcodec_decode_video2 allocate them for me... [14:43] <suzaru> ffmpeg supporting concat for m4a files? [14:43] <suzaru> or have to demux [14:48] <burek> suzaru, isn't it faster to just test it? :) [14:54] <zmode> hi. i'm unable to play .flac files with mplayer2 and to my understanding this has to do with ffmpeg1 on my system. i get this output: http://bpaste.net/show/4olrx4S4mEUNeeubl9oS/ . is there something i need to change in my make options? [14:55] <durandal_1707> there is no ffmpeg1 [14:56] <durandal_1707> this is not mplayer2 support channel [14:57] <durandal_1707> you are using old mplayer2 with new lavf/lavc [14:57] <zmode> i meant ffmpeg-1.0.1 [15:00] <ubitux> does it play with ffplay? [15:00] <ubitux> if so, you should ask #mplayer2 [15:00] <zmode> ubitux: no, i get a "could not open codecs" error [15:00] <jeje34> burek> if I do the same thing than in decoding video example, at the end, it just call avcodec_free_framebut in the documentation, there's a warning:this function does NOT free the data buffers themselves. So, the buffer themselves are never free? [15:01] <zmode> i also couldn't get mpd to play .flac files which is why i asked here first [15:01] <ubitux> zmode: pastebin the full output of ffplay [15:02] <zmode> ubitux: http://bpaste.net/show/A9SfVjYuHht8RpONMVOm/ [15:02] <ubitux> that is very old [15:03] <ubitux> 0.7& we release 0.8, 0.9, 0.10, 0.11, 1.0 and 1.1 since then [15:05] <burek> jeje34, im not a developer, so im not of much help :S [15:05] <burek> you might ask in ffmpeg-devel, but be patient, because people there dont have much spare time for chit-chat [15:06] <someone-noone1> Hello! I'm writing video player based on libav* (Yes, I know about ffplay). Currently, I'm implementing a\v syncing (Yes, I saw how it's implemented in ffplay). [15:06] <someone-noone1> I have strange thing while decoding video and calculating PTS values with next code: [15:06] <someone-noone1> http://ideone.com/75nWG5 [15:07] <durandal_1707> burek: do not direct people that ask for user help to dev channel [15:07] <durandal_1707> i may kick you next time there if I'm in really bad shape... [15:07] <burek> ok.. [15:08] <someone-noone1> If for each frame I will find prev_pts - current_pts (value is directly in milliseconds) sometimes it's 41ms, which is exactly equal 1000ms\24fps [15:08] <jeje34> durandal_1707>I'm ok with you but where can I find an answer to my question [15:08] <someone-noone1> But sometimes, it's more then 100ms and more over, sometimes it's NEGATIVE/ [15:08] <someone-noone1> How can you explain it? [15:08] <jeje34> I think I'm in the right chatroom [15:09] <someone-noone1> btw, stream is mpeg-ts (h264+aac) [15:09] <burek> someone-noone1, does prev_pts - current_pts have to be constant? [15:09] <durandal_1707> jeje34: did you carefuly read documentation and read examples? [15:10] <someone-noone1> burek, no but why it can be negative? [15:10] <durandal_1707> jeje34: i can provide my limited hely to you in my spare time [15:10] <someone-noone1> Sorry, not prev_pts-current_pts, but current_pts-prev_pts [15:10] <durandal_1707> s/hely/help [15:10] <jeje34> durandal_1707> yes but there is the memory part of the AVFrame I don't understand... [15:11] <burek> someone-noone1, frame ordering is arbitrary, that's why there are pts/dts [15:11] <someone-noone1> Looks like frames are ordered in decoding order, but doesn't avcodec_decode_video2 should order them in play order? [15:11] <burek> not really [15:11] <burek> it makes sense in some cases [15:11] <burek> to decode some future frames first [15:12] <burek> in order for other (delta) frames to have their needed data available [15:12] <burek> (some delta frames might reference the frames in the future, not only in the past) [15:12] <someone-noone1> burek, I know about I,P and friends frames [15:13] <someone-noone1> But again, shouldn't avcodec_decode_video order them in play order? [15:13] <burek> isn't it faster and more convenient if you order them in decode order? [15:13] <burek> (doesn't require seeking) [15:14] <someone-noone1> burek, then how should I draw those frames? Pre-buffer and make valid order? [15:14] <burek> if you have I1,B1,B1,P1,I2 [15:14] <burek> B2* [15:14] <burek> isn't it obvious that you might need I2 decoded (together with I1) in some point in time [15:16] <someone-noone1> burek, yes it's obvious. But why is there in avcodec_decode_video2(& ,int *got_picture,&) parameter? If decoding fails(need more frames), I just should be notified and continue decoding, but frames should be in play order. Isn't it? [15:17] <burek> couldn't you check how did ffplay do it? [15:17] <Mavrik> because you're not always giving full frames to decoder [15:18] <Mavrik> so if you have certain formats [15:18] <Mavrik> you'll feed decode_video2 with data but it won't be able to return you a frame [15:18] <Mavrik> and no, decode_video2 will not reorder frames for you, it wouldn't make sense on that level [15:19] <someone-noone1> Mavrik, that is what I wanted to hear. Thanks [15:19] <someone-noone1> I thought, it makes reordering.. [15:19] <Mavrik> nope, it just decodes frames as soon as it can [15:19] <Mavrik> usually you get them out in the same order as you add them to the queue... that's in DTS order most cases [15:19] <Mavrik> so you'll need a queue to reorder frames by PTS if your format has B frames [15:21] <someone-noone1> Mavrik, thanks [15:29] <Aziroshin> Hello. :o [15:30] <Aziroshin> I am trying to stream to twitch.tv using ffmpeg. My problem is that the audio gets horribly out of sync. When using -async 1, the audio goes away completely. [15:30] <Aziroshin> (there might be some audio fragments left, however) [15:30] <suzaru> i cant seem to join .h264 files into 1 [15:31] <suzaru> ffmpeg -i 1_Output.h264 -i 2_Output.h264 -i 3_Output.h264 -i 4_Output.h264 -i 5_Output.h264 -i 6_Output.h264 -c copy -bsf h264_mp4toannexb final.h264 [15:31] <suzaru> that's wrong i guess? [15:31] <Mavrik> suzaru: yeah, that creates 6 streams in file :) [15:31] <suzaru> only needed one -i ? [15:32] <Aziroshin> The command that I use goes somewhat like this: ffmpeg -async 1 -f x11grab -s <inres> -r 30 -i :0.0 -f alsa -ac 2 -i <card> -vcodec libx264 -s <outres> -acodec libmp3lame -ab 128k -ar 44100 -threads 0 -f mp4 test.mp4 [15:32] <Mavrik> ffmpeg really can't handle concating files well :\ [15:32] <Mavrik> I suggest you find a better tool [15:32] <Mavrik> Aziroshin: you should probably move async parameter after "-async" [15:32] <ubitux> < Mavrik> ffmpeg really can't handle concating files well // ??? [15:32] <ubitux> it has at least 3 ways of concatening files [15:33] <durandal_1707> protocol, demuxer and filter [15:33] <Mavrik> yes, and most of them don't work well [15:33] <Aziroshin> Mavrik: What exactly do you mean? [15:33] <Mavrik> unless there have been significant code changes in latest gits [15:34] <ubitux> what doesn't work? [15:34] <ubitux> https://ffmpeg.org/faq.html#How-can-I-concatenate-video-files_003f [15:34] <durandal_1707> Aziroshin: you sure your machine is really fast so it can encode h264 real time? [15:34] <durandal_1707> btw how fast it should be to encode it realt time? [15:35] <Mavrik> ubitux: that was not there a stable version ago, no need to get hostile -_- [15:35] <durandal_1707> where is prey? [15:35] <ubitux> how am i hostile? [15:36] Action: durandal_1707 likes predators (movies) [15:36] <Aziroshin> durandal_1707: To be honest, I am not sure how fast a machine should be for that. The idea is, however, that in case the machine should slow down temporarily during a stream, that this would not damage synchronicity anyway. [15:37] <Aziroshin> So, I have to find a way to force it to be synchronous without the audio quality suffering, which means that I would like to make ffmpeg drop frames when required. [15:37] <Aziroshin> The audio has to stay in sync and top quality at all costs. [15:37] <durandal_1707> Aziroshin: i dunno how you machine is fast, did you try ultra fast preset? [15:38] <suzaru> i seem to lose sync when i mux .aac files back to m4a with ffmpeg. i demuxed them from m4a [15:38] <suzaru> when i mux it with the video it is out of sync somewhat [15:39] <suzaru> if it possible when it joins the .aac files it introduces some sort of minor delay at the points where it joins [15:39] <suzaru> is it* [15:40] <durandal_1707> pastebin commands [15:41] <Aziroshin> durandal_1707: AMD Phenom II 955, it's a quad core processor. It has some of the best single core performance AMD CPUs offer today. Not sure whether that should be enough. [15:41] <durandal_1707> Aziroshin: i dunno either, you are only one that can test it.... [15:41] Action: durandal_1707 brb [15:42] <Aziroshin> Yeah, but as said, that's not that much the problem. If the processor should be too slow, either naturally, or because other processes take up too much CPU, the idea is that the audio does not go out of sync. [15:42] <Aziroshin> That's what I am trying to figure out. [15:42] <Aziroshin> What I forgot to say is, that the audio goes out of sync when recording to a file as well. [15:42] <Aziroshin> That was when I tried with flv, though. Let's do that with mp4... [15:43] <Aziroshin> (-async kills the synchronicity in both cases, flv and mp4) [15:43] <Aziroshin> kills the sound, sorry. Mistake. [15:44] <suzaru> i did something like this i believe [15:44] <suzaru> ffmpeg -i 1_Output.m4a -c:a copy -bsf:a aac_adtstoasc 1.aac [15:44] <suzaru> ffmpeg -i 2_Output.m4a -c:a copy -bsf:a aac_adtstoasc 2.aac [15:44] <suzaru> ffmpeg -i concat:"1.aac|2.aac" -c:a copy -absf aac_adtstoasc final2.m4a [15:48] <Aziroshin> Okay, it seems it doesn't record my audio properly anymore at all. I've already invested so much time in finding a solution to these issues some days before, it's ridiculous. I am taking a break here. [15:48] <Aziroshin> Probably I'll have to somehow shoehorn in a third sound interface, maybe via USB, to run a dedicated pulseaudio process. I hear it syncs things properly. [15:51] <Aziroshin> At least someone reported that, if he uses -i pulse, things get in sync. Also, all help available through google on that subject is purely pulse centric. [15:52] <Aziroshin> The problem is, though, that that will likely help me little with streaming wine based games. But maybe with some loopback trickery, something can be done. [16:23] <someone-noone> Hello! I'm developing video player based on libav* (Yes, I know about ffplay). I'm trying to reorder(by pts value) decoded video frames with next code: http://ideone.com/P0FN5G [16:23] <someone-noone> But looks like they're misordered! But if I do not make any reording, they're coming in right(playing) order. Why is it happening? [16:33] <Mavrik> someone-noone: use "av_frame_guess_best_effort_pts" to get frame pts [16:33] <Mavrik> it's not always the same as packet pts [16:33] <Mavrik> not to mention the fact that decoder may buffer data [16:33] <Mavrik> check the decoding samples [16:34] <someone-noone> Mavrik, btw, I just realized, thet avcodec_decode_video2 ALWAYS returns video in PTS order! [16:34] <someone-noone> https://github.com/mpenkov/ffmpeg-tutorial/issues/7 here is explained [16:34] <someone-noone> you have confused me :( [16:36] <Mavrik> there's nothing in the source of that function that would assure that [16:37] <someone-noone> Mavrik, I don't see any reorder stuff in ffplay.c [16:37] <someone-noone> pict_windex is always incrementing by 1 [16:57] <gmag> hi, is it possible to decode a stream from a container and output one yuv file for each frame? [17:11] <navaismo> Hi, im trying to stream my webcam using ffmpeg and ffserver but when I ran the command: ffserver -f /etc/ffserver.conf & ffmpeg -v verbose -r 5 -s 640x480 -f video4linux2 -i /dev/video0 http://10.0.1.103:8000/webcam.ffm [17:11] <navaismo> i get the error: Unknown input format: 'video4linux2 [17:17] <durandal_1707> gmag: yes, see image2 muxer and rawvideo encoder [17:23] <zmode> hi. i was here a while ago, unable to play .flac files. i was running and older ffplay version, but i'm having no luck with 1.0.1 either http://dpaste.com/900471/ . the folks over at #mplayer2 asked me to try compiling ffmpeg with avresample, but i still get no sound [17:23] <durandal_1707> mplayer2 indeed need avresample [17:23] <durandal_1707> zmode: why not use mpv instead [17:24] <durandal_1707> you know, mplayer2 main developer develops on libav only [17:24] <zmode> zmode: because i use mplayer2, but this doesn't seem to be an mplayer2 issue now, since ffplay can't play it [17:24] <durandal_1707> ffplay can play flac just fine [17:25] <zmode> not on my box [17:26] <durandal_1707> looks like you actully have some audio driver problem [17:26] <durandal_1707> what OS are you using and what audio driver? [17:27] <durandal_1707> or you have very old sdl version which doesnt play well... [17:27] <durandal_1707> because ffplay use sdl to play audio .... [17:30] <zmode> i have sdl 1.2.15 which is the latest stable from what i gather. and indeed ffplay has trouble with .mp3 files (just tested actually). i'm using FreeBSD 9.1 and 'cat /dev/sndstat' returns 'FreeBSD Audio Driver (newpcm: 64bit 2009061500/amd64)' [17:32] <durandal_1707> zmode: does any other audio app works/plays ok? [17:32] <zmode> durandal_1707: yeah, i do have audio [17:33] <durandal_1707> perhaps sdl is miscompiled/misconfigurated [17:33] <durandal_1707> or you compiled ff* without sdl support... [17:35] <durandal_1707> or if other sdl apps on your system works ok then you miscompiled ffmpeg [17:40] <zmode> ffmpeg doesn't seem to even make ffplay if sdl is disabled; but yes, i compile it with SDL. [17:41] <zmode> doesn't seem to depend on sdl_mixer or sdl_sound packages though [17:44] <durandal_1707> why you have some strange stuff in your configuration? [17:44] <durandal_1707> try to disable all that and see if ffplay works [17:44] <zmode> disable all what? [17:45] <durandal_1707> your ./configure --arguments [17:46] <zmode> i was saying it _doesn't_ depend or pull any of these [17:48] <durandal_1707> so if you compile static ffplay it have same problem? [17:57] <zmode> durandal_1707: i'm sorry, it's been about 4 hours on this now. i might try later. thanks for your time [18:15] <gmag> durandal_1707, I am having a hard time extracting yuv frames from a container with ffmpeg. I can only extract one single file (ffmpeg -i x_video.mp4 -r 25 -pix_fmt yuv420p ./outs/foo-%03d.yuv). Any idea how to change this to output every single frame? [18:15] <Diogo> hi i need to compile libxvp8 [18:15] <Diogo> http://pastebin.com/DFD0g70q [18:15] <Diogo> appear this error [18:16] <Diogo> my SO is debian.. [18:16] <gmag> SO? [18:16] <gmag> where you from Diogo [18:17] <gmag> country [18:19] <antonello> hi, I want Know the differenceerence betwen sample format and pcm .... Is it the same ? [18:28] <durandal_1707> antonello: sample_fmt says what kind of pcm is it about [18:36] <antonello> ok , So .. if i Want obtain a stream pcm i can stop at the decoding and possibly make a swr_convert() .Is it correct? [18:42] <gmag> durandal_1707, the output is a single file called "foo-%03d.yuv" [18:46] <durandal_1707> gmag: i cant help you if you ignore me, read what ffloger (bot) said [18:50] <epifanio> hi All, do you kniw if pyffmpeg is able to make videos from an image sequence ? at the moment i'm using a system call to ffmpeg from inside a python script [18:50] <epifanio> the command i'm using is : os.system('cat *.png | ffmpeg -y -f image2pipe -c:v png -i - -c:v libx264 -preset ultrafast -qp 0 -movflags +faststart time.mkv') [20:07] <barque> When I encode videos with ffmpeg the duration of FormatContext comes out wrong/bad [20:08] <barque> any idea how to force/workaround this issue? [20:08] <barque> ffmpeg and vlc give the right numbers [20:08] <barque> when executed [20:08] <barque> however with libavcodec the duration part of formatcontext comes out wrong [20:29] <barque> Hello? [22:24] <edmund_> hello -- i am trying to a 'data' stream from an ASF container. i have tried `ffmpeg -i stream.asf -map 0.2 -f rawvideo sub.txt` and `ffmpeg -i stream.asf -map 0.2 -f srt sub.txt`, both of which fail for different reasons.. can anyone help? [22:24] <edmund_> this what what the stream (that i want) is reported as: "Stream #0.2(eng), 0, 1/1000: Data: [0][0][0][0] / 0x0000, 10 kb/s" [22:25] <edmund_> trying to +extract a data stream [22:27] <edmund_> in the case of `-f srt`, i get the error "Output file #0 does not contain any stream". [22:28] <edmund_> and in the case of `-f rawvideo`, i get "Codec type mismatch for mapping #0.2 -> #0.0" (understandably -- but i was following some suggestions online for subtitle extraction) [22:29] <edmund_> and this is what ffprobe says: http://pastebin.com/3GnJ1UEe [22:35] <edmund_> should i go to the mailing list instead? :) [22:45] <llogan> edmund_: you're not using ffmpeg from FFmpeg. [22:46] <llogan> we support ffmpeg from FFmpeg here. [23:22] <ashes> hello [23:23] <ashes> i want to do 2 pass xvid encoding to reduce the size of videos, to be displayed on a 7 inch tablet. is it possible for me to set the display size when i'm encoding? [23:37] <ashes> nm [23:45] <edmund_> this article is not very illuminating. [23:45] <edmund_> but, i get it. [00:00] --- Thu Jan 31 2013

1 0

ffmpeg-devel.log.20130129
by burek 30 Jan '13

30 Jan '13

[00:00] <durandal_1707> ff_raw_pix_fmt_tags should be renamed to something like avpriv_raw_pix_fmt_tags [01:19] <cone-597> ffmpeg.git 03Piotr Bandurski 07master:f9a8eeb08c4c: iff/deep: fix rle32 on big-endian [04:59] <cone-394> ffmpeg.git 03Michael Niedermayer 07master:11c99c78bafa: h264: check the pixel format directly and force a reinit on mismatches. [11:42] <durandal_1707> why is context missing when av_log is called from .write_header in muxer? it should be present and in distinct color. [12:11] <cone-913> ffmpeg.git 03Paul B Mahol 07master:8a6ae87b9915: lavc: move deprecated audio_resample* bellow [13:47] <cone-913> ffmpeg.git 03Luca Barbato 07master:f550583c00e2: bfin: update VP3 idct [13:47] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:8265c0f43a44: Merge commit 'f550583c00e231b587d8ef98451cfbb6b6561eb6' [13:55] <cone-913> ffmpeg.git 03Diego Biurrun 07master:438ea561ade1: bfin: Separate VP3 initialization code [13:55] <cone-913> ffmpeg.git 03Diego Biurrun 07master:c59211b437aa: x86: Simplify some arch conditionals [13:55] <cone-913> ffmpeg.git 03Anton Khirnov 07master:f1c395944ce9: eatgv: use fixed-width types where appropriate. [13:55] <cone-913> ffmpeg.git 03Anton Khirnov 07master:098eed95bc1a: mdec: merge mdec_common_init() into decode_init(). [13:55] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:14aa358c20c2: Merge commit '098eed95bc1a6b2c8ac97f126f62bb74699670cf' [14:05] <cone-913> ffmpeg.git 03Anton Khirnov 07master:f713411d4cfb: mdec: cosmetics, reformat [14:05] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:f02033b98b75: Merge commit 'f713411d4cfbd9c467aeda77b16ca6bc4db55d10' [14:18] <cone-913> ffmpeg.git 03Anton Khirnov 07master:30d62507cd9c: mdec: return meaningful error codes. [14:18] <cone-913> ffmpeg.git 03Anton Khirnov 07master:e6da5d215b1f: mimic: remove a pointless cast. [14:18] <cone-913> ffmpeg.git 03Anton Khirnov 07master:aec50f79e746: rawdec: use AVPALETTE_SIZE instead of magic constants. [14:18] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:31d8d61f590e: Merge commit 'aec50f79e7460340a148a3096fe212d67edc2c64' [14:24] <cone-913> ffmpeg.git 03Anton Khirnov 07master:729b37149c9c: mvi: set framerate [14:24] <cone-913> ffmpeg.git 03Anton Khirnov 07master:e6b1c3bbe708: pthread: make ff_thread_release_buffer idempotent. [14:24] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:3c8085dc4250: Merge commit 'e6b1c3bbe7082c71ea8ee8ac83698c156c9e4838' [14:39] <cone-913> ffmpeg.git 03Anton Khirnov 07master:231fd1ed3932: utvideoenc/v410enc: do not set AVFrame.reference. [14:39] <cone-913> ffmpeg.git 03Anton Khirnov 07master:47318953ddfe: mpegvideo: remove some unused variables from Picture. [14:39] <cone-913> ffmpeg.git 03Anton Khirnov 07master:76e74e4831f0: h264: remove obsolete comment. [14:39] <cone-913> ffmpeg.git 03Anton Khirnov 07master:f81c37e40fe3: vf_delogo: fix an uninitialized read. [14:39] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:d1bbd304bf60: Merge commit 'f81c37e40fe3236d54da12aef9cdba48ba70ec31' [15:19] <durandal_1707> ubitux: ping [15:20] <cone-913> ffmpeg.git 03Anton Khirnov 07master:7194330bcd6d: vf_delogo: fix copying the input frame. [15:20] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:5068bcda95c2: Merge remote-tracking branch 'qatar/master' [15:21] <durandal_1707> ubitux: pin3 [15:35] <ubitux> durandal_1707: pong [15:35] <mateo`> ubitux: \o/ [15:36] <ubitux> not for long ;) [15:39] <durandal_1707> ubitux: i was just wondering about making showspectrum more useful [15:40] <durandal_1707> more channels, etc like what SoX spectrogram have [15:41] <ubitux> the point of showspectrum was to port the feature from ffplay, feel free to do whatever you want to it as long as a "compat" mode is still available for when ffplay will actually use it [15:41] <durandal_1707> i could write new filter... [15:41] <ubitux> what for? [15:42] <durandal_1707> compat mode? where all channels are calculated but only 2 displayed [15:42] <durandal_1707> ubitux: spectrogram [15:42] <ubitux> by compat mode i mean something that is close enough to the original [15:43] <ubitux> if you actually improve the output i don't think anyone will complain [15:45] <durandal_1707> well idea is to render all channels [15:45] <durandal_1707> and custom/better colors [15:46] <durandal_1707> and also different algos that calculate spectrum: kaiser/rectangular/hamming... [15:46] <durandal_1707> also to output single image of whole song [15:47] <ubitux> sounds like a good idea [15:48] <mateo`> great idea :) [15:49] <ubitux> just keep a playback mode like currently, i don't mind better and more advanced channel colors [15:50] <ubitux> i guess you might want to make a color-by-intensity instead of channels, but please keep a channel one :) [15:50] <ubitux> (optional i mean) [15:51] <mateo`> http://blog.dubspot.com/files/2011/02/Spectrum1.png < this kind of spectrum rendering is on my todolist [15:55] <durandal_1707> mateo`: that is improved showwaves [15:57] <mateo`> durandal_1707: right :) [16:20] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:036b9ee1c959: oggenc: fix "oggstream may be used uninitialized in this function" warning [16:20] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:ebe368d5d82c: ac3enc: fix 'warning: block0 may be used uninitialized in this function' [16:20] <divVerent> durandal_1707: personally, I say: good idea [16:20] <divVerent> the "compat mode" even sounds quite simple [16:21] <divVerent> option 1: b/w (or still colored), different channels go to different screen areas [16:21] <divVerent> option 2: all channels overlap, colors get added (like now) [16:22] <divVerent> outputting single image for whole song - I'd love that feature, just doesn't sound like something a filter would do. Maybe if the filter has an output file arg, it could additionally do that [17:10] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:3cd9849d9c24: eval: fix 'warning: ignoring return value of strtod, declared with attribute warn_unused_result' [17:10] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:df92ac18528b: r3d: fix division by 0 with 0 sample rate [18:07] <durandal_1707> divVerent: it is just matter of stream duration, you set one and stop filter when it reaches right end... [18:23] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:99b1b2b1c659: r3d: check that sampling rate is non negative. [18:23] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:f67a0d115254: huffyuvdec: Check init_vlc() return codes. [18:23] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:4420b414420e: huffyuvdec: check for and propagate failures from inside generate_joint_tables() [18:45] <durandal_1707> thing is that improving showspectrum is much more work that writing new filter from scratch [19:21] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:0dfc01c2bbf4: huffyuvdec: Skip len==0 cases [19:49] <divVerent> durandal_1707: hehe, right... if you give the duration in advance, the filter can operate normally [19:56] <durandal_1707> divVerent: currently it displays up to two channels at same time [19:57] <durandal_1707> divVerent: how would you display >2 channels in same row? [20:02] <michaelni> if you have 8 channels one could draw something similar to ffplays rdft in 8dimensional colorspace instead of the common 3dim RGB and could then project it to 3 dimensions to be able to display it ... [20:04] <michaelni> such projection should likely preserve luminance (as overall loudness) [20:04] <divVerent> durandal_1707: color overlap [20:04] <divVerent> like, for 3 channels, r g b [20:04] <divVerent> for more, I'd distribute them along hue evenly [20:05] <divVerent> this would be method 1 [20:05] <divVerent> method 2 would be what sox does [20:05] <durandal_1707> really? my ffplay plays only mono or stereo and output is same as if I play only 2 channels [20:05] <divVerent> first 6th of height is channel 1 [20:05] <divVerent> then next 6th is channel 2 [20:05] <divVerent> etc. [20:05] <divVerent> durandal_1707: currently it only supports 2 channels [20:05] <divVerent> and usually the spectrum is so similar for both channels [20:05] <divVerent> that you see no coloring [20:05] <divVerent> try Bohemian Rhapsody to see it ;) [20:05] <durandal_1707> divVerent: sox does what i posted patch now [20:06] <divVerent> thing is, it doesn't SOUND very hard to both support sox method and old ffplay method at the same time [20:06] <divVerent> you always build an array of screen height size [20:06] <divVerent> for sox method, you divide it into n parts [20:06] <durandal_1707> divVerent: i see (a + b) /2 colors ... nothing spectactular .. [20:06] <divVerent> for ffplay method, you use a single part for all [20:06] <durandal_1707> and what you get? [20:06] <divVerent> but otherwise, you always color the spectrum value by the channel color [20:06] <divVerent> and add up [20:07] <durandal_1707> i just add numbers and say here is color? [20:07] <divVerent> "sort of" [20:08] <divVerent> right [20:08] <divVerent> currently, R is a, G is b, B is (a+b)/2 [20:08] <divVerent> so the color of left channel is #ff0080 (reddish-pink) [20:08] <michaelni> for side by side display it might make sense to put the left channels left, the right channels right and the center/lfe in the middle [20:08] <divVerent> the color of blue channel is #00ff80 (greenish-cyan)# [20:08] <michaelni> front up and back down on the screen ... [20:08] <divVerent> michaelni: agreed [20:08] <divVerent> the same rotated by 90 degrees may actually be better [20:08] <divVerent> but most programs display spectrum horizontally [20:09] <divVerent> so time axis goes from left to right [20:09] <divVerent> personally I'd actually prefer it the other way, time axis from top to bottom and scrolling up [20:09] <divVerent> as that means more accuracy for the pitch [20:10] <durandal_1707> you just have wrong monitor [20:10] <divVerent> hehe [20:10] <divVerent> actually, I could rotate mine [20:10] <divVerent> it does support it [20:10] <divVerent> and I have xrandr too... [20:10] <divVerent> but I am used to wider-than-high [20:11] <divVerent> but anyway, even though I personally prefer the rotated view [20:11] <divVerent> I would rather recommend that ffmpeg use the standard view [20:11] <divVerent> as most audio guys are used to that view [20:11] <divVerent> given 99% of audio editors use it [20:11] <divVerent> i.e. frequency from bottom to top, time from left to right [20:11] <divVerent> for my personal preference, I can always add a rotate filter [20:12] <durandal_1707> patch welcome, i implement for what i have motivation and other too if code is soo good that small changes.... [20:12] <divVerent> sure [20:12] <divVerent> actually, this may be something I would want to do... have a screenshot of what you currently have? [20:13] <durandal_1707> same as sox but mono and sox have channels listed in different order [20:13] <divVerent> #define IM(ch) data[ch][2*y + 1] [20:13] <divVerent> this BTW is hideous ;) [20:13] <durandal_1707> so next thing to do is colors... [20:13] <divVerent> in the old code [20:13] <divVerent> basically, a hidden macro parameter "y" [20:14] <durandal_1707> i did not wrote that code [20:14] <divVerent> not saying you did [20:15] <divVerent> probably nobody knows anymore who wrote it originally [20:15] <divVerent> as it stems from ffplay [20:15] <divVerent> as for colors, a constructive idea... what if the filter output yuv444p instead? ;) [20:15] <durandal_1707> its like: lets implement some nice feature but on end you implement only 10% of it [20:15] <divVerent> then you could use Y for intensity, and calculate U and V based on color [20:16] <divVerent> advantage versus RGB would be that different colors would have roughly the same intensity [20:16] <divVerent> unlike e.g. RGB, where #00ff00 is a lot brighter than #0000ff [20:16] <divVerent> or did you already do that, as it also lets you be lazy for grey output? ;) [20:17] <divVerent> disadvantage is that I don't currently know a good hue -> YUV conversion [20:17] <divVerent> as for what I have in my head, I'd evenly space out the colors in hue [20:18] <durandal_1707> sox use pal8, so only 256 colors [20:18] <divVerent> ah, yes [20:18] <divVerent> that's the other way to do colors [20:18] <divVerent> to show intensity by color too [20:18] <divVerent> right, I forgot - sox does that [20:18] <durandal_1707> i would prever something more poverful [20:18] <durandal_1707> *prefer [20:18] <durandal_1707> like 16bit bps [20:19] <divVerent> BTW, out of curioisity... would it make any sense to show the phase (direction of re, im)? [20:19] <divVerent> probably not [20:19] <durandal_1707> for double sample format... [20:19] <divVerent> sox kinda "solves" it by having a different scale [20:20] <divVerent> ffmpeg uses sqrt() currently, didn't check what sox does, but it seems logarithmic [20:20] <divVerent> which kinda solves the accuracy issue [20:25] <durandal_1707> ubitux wants it to mimic in some mode, what ffplay does, which limits what i can do without making code even more convoluted [20:26] <divVerent> I didn't see your code, but to me it looks possible to unify it [20:26] <divVerent> if you want, I can try bringing the old mode back "as cleanly as possible" [20:26] <divVerent> tomorrow [20:26] <durandal_1707> so what is best formula to map X-channels into 24 bit rgb value? [20:28] <durandal_1707> divVerent: if you are subscribed to devel ml you can see it .... [22:55] <cone-913> ffmpeg.git 03PaweB Hajdan, Jr 07master:1d81f7448c8a: dict.c: use av_mallocz instead of av_realloc [22:55] <cone-913> ffmpeg.git 03Michael Niedermayer 07master:dc8dd2f6e972: sanm: Check MV before using them. [23:07] <saste> michaelni, about my eval doc patches? should I push them? [23:07] <saste> nobody's going to review them anyway (but - maybe - you) [23:25] <durandal_1707> michaelni: the sanm commit looks wrong [23:27] <durandal_1707> how are you sure that there in nothing between prev and frm2? [23:31] <durandal_1707> there is already code that check motion vectors... [23:43] <durandal_1707> michaelni: are you aware of this? [23:47] <michaelni> durandal_1707, what can be between prev2 and frm2 ? [23:47] <durandal_1707> anything [23:47] <durandal_1707> it is not real fix [23:47] <durandal_1707> frm0/1/2 do not be conitinous in memory/ one after another [23:48] <michaelni> prev2 points in frm2 [23:48] <durandal_1707> so you could make frmX to come one after another [23:49] <michaelni> uint8_t *prev2 = (uint8_t*)ctx->frm2; [23:50] <michaelni> the MV could be checked differently of course but i dont see a problem with how its checked now, this doesnt mean there is none, if theres an issue please explain [23:52] <durandal_1707> you are indeed right, sorry for wasting time [00:00] --- Wed Jan 30 2013

1 0

ffmpeg.log.20130129
by burek 30 Jan '13

30 Jan '13

[01:59] <njbair> am I doing something wrong or is libvpx waaaaay slower than libx264? [02:05] <juanmabc> isn't vpx = very poor experience ? [02:05] <juanmabc> ;P [04:06] <Miesco> Chrome doesn't support ogg, how do I convert a .ogg to a webm vorbis? [04:11] <Miesco> Wait I think I know: ffmpeg -i file.ogg -acodec vorbis file.webm [04:16] <Miesco> I think wikipedia and w3c got Chrome's codec support wrong for <audio>. Chrome doesn't support ogg... [05:55] <saju> how to convert 2k video file to 4k [06:09] <praveenmarkandu> hi. how come this command doesnt make my file twice as large? or twice as long? [06:09] <praveenmarkandu> ffmpeg -i "concat:bigbuckbunny1080p.mov|bigbuckbunny1080p.mov" -c copy bigbuckbunny1080p_double.mov [06:30] <relaxed> praveenmarkandu: ffmpeg's concat is limited. Try mkvmerge, then copy back to mov if you need that container. [07:21] <praveenmarkandu> ok. i tried mencoder. seems to work [07:21] <praveenmarkandu> relaxed, what do you mean by limited? [08:15] <LorentzFactor> is there a way to use the original video stream while reencoding the audio stream? It'll take forever if it has to reencode both with all the files I'm trying to convert :\ [08:16] <klaxa> you can copy the video stream with -c:v copy [08:16] <LorentzFactor> ah [08:19] <LorentzFactor> thanks that's like 300% faster [08:19] <LorentzFactor> 3000%* [08:21] <Youka> Is there a way to use ffmpeg in msvc9? building it is just possible with msvc10 ('wrong C compiler' error) and using this library in msvc9 causes errors because of double definitions of f.e. _sopen [08:36] <Miesco> Hi [08:37] <Miesco> Anyone want to know the beautifulness that ffmpeg just trancoded: ftp://miesco.is-a-geek.org/soundcloud/dont%20know%20shawnsvocals.webm [08:37] <Miesco> Dats a song I wrote [09:12] <frostywolf> hi [09:13] <frostywolf> i need some help with a ffmpeg command [09:14] <CruX|> h265 is supported by ffmpeg ? [09:15] <frostywolf> i'm trying to get padding to work on a file. Downscaling widescreen to 4:3, from 1280x720 to 720x540 (therefore the actual video should be 720x405, so the padding command is 720:540:0:67:black right? [09:16] <JEEBsv> CruX|: smarter from the libav project is working on a hevc / H.265 decoder [09:16] <JEEBsv> the specification isn't even finished yet [09:31] <praveen> wait, so libav libaries are still used in FFMPEG? [09:31] <praveen> eventhough they have forked? [09:33] <JEEB> praveen, libav's changes are being merged into ffmpeg, yes? [09:34] <praveen> JEEB, ok. thanks. [09:34] <praveen> this may seem like a flame bait question [09:35] <praveen> but i want to know, why would one use libav over ffmpeg or vice versa [09:35] <JEEB> there are plenty of write-ups from either side on that (Ž [09:35] <JEEB> ubitux has a blog entry, lu_zero from gentoo has another... [09:35] <praveen> im guessing since im on #ffmpeg it is a loaded question [09:36] <praveen> ok. will search [09:36] <praveen> are changes from FFMPEG being merged into libav? or are they generally ahead? [09:37] <JEEB> some changes from ffmpeg are cherry-picked into libav [09:37] <JEEB> like lately the gbrp stuff [09:38] <praveen> ok cool [09:56] <jeje> Hi to all ;-) [09:56] <jeje> I have a question about using swscale lib [09:58] <jeje> when calling sws_getContext, the function can have a sws_flags parameters but I don't understand ther difference between each one . We also can set it to NULL. If someone can explain me please. [09:59] <jeje> for informations I use swscale to convert and resize an YUV420 image (H264 decoding by FFMPEG) to an RGB32 image displayed on screen [10:02] <jeje> I think it can change the speed of the swscale operation, but is one sws_flags is better than another one? [10:26] <praveen> is there a maximum file size ffmpeg can handle? [10:27] <Mavrik> hmm... not really [10:27] <Mavrik> it's more of a container support question [11:36] <frostywolf> i'm trying to get padding to work on a file. Downscaling widescreen to 4:3, from 1280x720 to 720x540 (therefore the actual video should be 720x405, so the padding command is 720:540:0:67:black right? [12:45] <venom200> Anyone knows how to set pixel aspect ratio in FFmpeg? [12:56] <venom200> Please answer - how to set PAR - Pixel Aspect Ratio in FFmpeg. [12:58] <durandal_1707> read documentation first [13:00] <relaxed> hmm, ffmpeg has -sar but not the setsar filter? [13:00] <durandal_1707> ffmpeg _have_ setsar filter [13:01] <durandal_1707> setsar V->V Set the pixel sample aspect ratio. [13:04] <venom200> I tried with setsar=1, but did not work. [13:05] <venom200> Here is the complete cli I used: http://pastebin.com/B9cxpqMC [13:08] <durandal_1707> venom200: do you understand what word complete mean? [13:08] <durandal_1707> paste full uncut output [13:10] <venom200> Here it is: http://pastebin.com/ALuX6yqM [13:11] <durandal_1707> venom200: please read documentation [13:11] <durandal_1707> how to use setsar filter [13:11] <venom200> I am sorry. I am not familiar with setsar. [13:12] <durandal_1707> can you read documentation? [13:12] <venom200> Yes. Where can I find that? [13:12] <durandal_1707> at ffmpeg.org [13:12] <venom200> Thanks If setsar is fully documented, I will read that. Thanks [13:22] <guest361> does -aspect option work for input file? [13:26] <durandal_1707> depends, usualy it is read from container/stream [13:30] <klaxa> http://aws.amazon.com/elastictranscoder/ <-- i wish they would post more than "720p and not 720p" [13:31] <klaxa> oh wait they do [13:33] <klaxa> mh not much more than "we use H264/AAC/MP4" i guess one would have to sign up to take a closer look [14:02] <brontosaurusrex> when muxing two streams (one video + one audio) i'd like that the final stream has duration of the shortest one, was there a command for that? [14:03] <mateo`> brontosaurusrex: -shortest ? [14:04] <brontosaurusrex> mateo`: really? cool [14:18] <guest361> do I have to re-encode to modify DAR ? [14:43] <JEEB> in ffmpeg, do -maxrate and -bufsize only limit the video or all tracks within the encode? [15:12] <foxlover> Hello everyone! Just want to know which ffmpeg version the CLI option -print_format appeared, was it 0.9 with json, compact and csv support or was yet before that? Readed the changelogs, but they don't helped much... thank you in advance! [15:15] <durandal_1707> foxlover: with JSON output [15:16] <durandal_1707> so 0.9 [15:37] <foxlover> durandal_1707: Thank you for your help! :) [15:56] <Guest32954> Hi, I use av_new_packet to allocate and initialize my AVPAcket with the PADDINGSIZE (IE: dwImageSize+FF_INPUT_BUFFER_PADDING_SIZE) [15:56] <Guest32954> but when I do a av_free_packet, it doesn't free the buffer of the AVPacket [15:56] <Guest32954> How can I free the buffer of the AVPacket? [15:57] <Guest32954> Do I need to call a av_freep (packet.data[0]) before? [16:09] <Guest32954> even if I call av_destruct_packet, my memory grow up [16:39] <Demon_Fox> Can you mux multiple files together with ffmpeg easily? [16:39] <Demon_Fox> I am going to reword that [16:40] <Demon_Fox> Can you easily mux multiple files together with ffmpeg?* [16:40] <Demon_Fox> or [16:40] <Demon_Fox> Can one easily mux multiple files together with ffmpeg?* [16:40] Action: Demon_Fox should have got that right the first time. [16:42] <sacarasc> Demon_Fox: So you have multiple streams or make the video longer? [16:42] <Demon_Fox> more like [16:42] <Demon_Fox> One audio stream and one video stream [16:42] <JEEB> that's simple [16:42] Action: Demon_Fox did not know [16:42] <sacarasc> http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20concatenate%20%28join,%20merg… [16:43] <JEEB> ffmpeg -i input1 -i input2 -map 0:v -map 0:a -c copy out.file [16:43] <JEEB> two inputs [16:43] <JEEB> map the video from the first one, audio from the second one [16:43] <sacarasc> Oh, right. [16:43] Action: sacarasc goes back to bed. [16:44] <Demon_Fox> thanks [16:44] <JEEB> -map input_nr:track_nr or track_type [16:45] <JEEB> selects a track [16:45] <JEEB> and adds it to output (overrides any automatic selection before then) [16:50] <Demon_Fox> thanks once again [16:52] <someone-noone> Hello! When I want to read data from bufffer using read callback putted in avio_alloc_context(), my program is always crashing exactly here: http://cekirdek.pardus.org.tr/~ismail/ffmpeg-docs/aviobuf_8c-source.html#l0… [16:53] <someone-noone> why can it be? [16:55] <someone-noone> found problem, my buffer wasn't allocated with av_malloc [16:56] <teratorn> if it bad to use timestamps with negative values? [16:57] <Demon_Fox> JEEB, This may sound rude, but are you from japan. I want to know because I would like to know how polite to be before I get rude. [16:59] <Demon_Fox> In some cultures trying to be as polite as possible can make me rude. [16:59] <JEEB> I have lived in Japan but I am from Finland >_> [17:00] <Demon_Fox> JEEB, Thanks, I needed how polite I needed to be about one of the most helpful IRC users [17:01] <Demon_Fox> for one of the* [17:04] <paltman> anyone know what this means? [libx264 @ 0x9c5a720] err{or,}_recognition separate: 1; 1 [17:06] <Demon_Fox> Strange error, maybe the people in #x264 can help. [17:12] <paltman> also getting some errors with dvvideo, but it looks like it might have been a regression that was fixed back in march https://ffmpeg.org/trac/ffmpeg/ticket/1042 [17:12] <paltman> as my build is previous to that [17:13] <paltman> as i am getting Unsupported bit depth: 0 as well [17:13] <paltman> in trying to transcode dvvideo to x264 [17:13] <paltman> going to recommend my client upgrade ffmpeg to latest [17:45] <sam_> ffmpeg doesnt accept profile options passing to libxx264 [17:46] <sam_> i had tried profile:v at command line it works but passing through configuration doesnt [18:37] <beastd> heh, good i looked into the archive. ffserver-user mailing list will be shutdown and merged with ffmpeg-user [19:44] <Miesco> Do you have to pay royalties for mp3 encoding/decoding in software that is free? [19:46] <JEEBsv> Miesco: software licenses and possible licenses for formats etc. are completely separate. If you see that a company wants you to get a license to deal with a certain format, and a lawyer tells you that it is valid in your country then it's better to follow it :P [19:46] <JEEBsv> I think MPEG-1 Layer 3 audio was licensed by fraunhofer? [19:47] <Miesco> http://mp3licensing.com/royalty/ [19:47] <Miesco> Im looking at that [19:49] <JEEBsv> oh, technicolor? Noted [19:49] <JEEBsv> anyways, IANAL and I can't tell you if you *have* to pay those royalties [19:49] <JEEBsv> that's better to be left to the lawyers [19:49] <JEEBsv> or just do a decision by yourself [19:53] <Miesco> its 1.25 for decoding mp3 in your app, what if its a $1 app on the android market, doesn't make sense [19:54] <llogan> shows .75 for mp3 [19:54] <llogan> who uses mp3pro? [20:03] <Miesco> llogan: Okay so I would have to pay $0.75? [20:28] <llogan> Miesco: probably, but apparently licensing agencies are still in the year 2000 and don't have an "app" pricing structure. [20:28] <JEEBsv> usually e-mailing them about it helps, at least it seems to have helped some people when licensing AAC [20:29] <JEEBsv> if they really don't want to bulge then you'll have to decide whether you want to pay or not [20:57] <barque> ((unsigned int)pFormatCtx->duration)/(float)AV_TIME_BASE is not giving me the correct duration of a video [20:57] <barque> is there some better way of getting the duration? [20:57] <barque> or do I have to maybe re-encode it to get a better duration? [20:58] <barque> I've also tried multiplying by time_base but I think that also gives me bad values [20:58] <barque> AVFractional time_base [20:59] <JEEBsv> in most such cases you haven't dealt with timestamps correctly, although I have no idea how well libavformat can give you the length with various container formats [20:59] <JEEBsv> some have timestamps for all frames [20:59] <barque> timestamps? [20:59] <JEEBsv> others just plain don't and stuff gets very roughly approximated [20:59] <barque> on packets? [21:00] <barque> no I mean I'm trying to get the duration before reading packets [21:00] <barque> do I have to read packets to get a decent duration? [21:00] <barque> the timestamps are actually on the mark [21:00] <barque> very accurate [21:00] <barque> I've logged them [21:02] <JEEBsv> no idea, but I wouldn't expect libavformat's checks always give you the correct length... but I will bet that you most probably are doing something weird if the format is such where libavformat can give you a relatively straightforward duration, for example forgetting to normalize the timestamps (duration and its time_base) [21:02] <barque> the packet timestamps are pefect.... what I'm trying to do to get duration is : ((unsigned int)pFormatCtx->duration)/(float)AV_TIME_BASE [21:03] <barque> here's how I get packet timestamps : (float)(packet.pts * av_q2d(pCodecCtx->time_base)); [21:03] <barque> gives me fractional seconds on the mark [21:03] <barque> I've tried multiplying duration by av_q2d(pCodecCtx->time_base) [21:03] <JEEBsv> I have no idea what AV_TIME_BASE is but it sounds like some nonchanging constant, not sure if that's a good idea [21:03] <barque> but gives me a rather large number [21:04] <barque> 47k some such [21:04] <barque> as *if* it was in milliseconds [21:04] <barque> actual video length is 43 seconds [21:05] <JEEBsv> what's your format btw? [21:05] <barque> so I don't know if just dividing things by 1000 on my own is a good idea [21:05] <barque> Theora/OGG [21:05] <JEEBsv> uh-oh [21:05] <JEEBsv> have fun is all I can say [21:05] <barque> hah, there's no way eh? [21:05] <JEEBsv> does command line ffmpeg give you a length? [21:05] <barque> let me see [21:05] <JEEBsv> nah, I'm going to expect a PEBKAC first :P [21:08] <barque> it gives me perfect duration [21:09] <JEEBsv> ok [21:09] <barque> I'm doing pre-reads and using custom seeks and stuff [21:09] <barque> callbacks [21:09] <JEEBsv> not sure if libavformat by default does the exact same checks with the exact same amount of file... but at least it's a hint towards the fact that you might be having a PEBKAC [21:10] <barque> So I'm not computing things correctly? [21:10] <JEEBsv> the most general thing is that the user forgets to normalize his timestamps (in this case, duration) to whatever he's expecting or whatever :V [21:10] <barque> ok [21:10] <JEEBsv> I'm pretty sure you're not supposed to be using any define like AV_TIME_BASE [21:11] <barque> yeah I used the other one [21:11] <barque> gives me a large number [21:11] <barque> (still) [21:14] <JEEBsv> ok... [21:14] <JEEBsv> http://www.ffmpeg.org/doxygen/trunk/structAVFormatContext.html#ad0ea78ac48f… [21:18] <barque> yeah apparently http://www.sebastianroll.de/TableTopDoc/video_texture_8cpp-source.html is doing the same\ [21:21] <JEEBsv> also since I guess both the timestamp and timescale are int64_t, I recommend you make sure whatever you cast the values into can hold the end result values [21:21] <barque> ok [21:21] <barque> maybe I'll cast as the very very end [21:22] <barque> because it obviously finally needs to fit [21:22] <barque> I don't have godly large videos [21:38] <barque> for duration I'm getting 477051922 [21:38] <barque> AV_TIME_BASE is 1000000 [21:38] <barque> but the length of the video is 43.3 seconds [21:38] <barque> not 477 seconds [22:06] <sam__> how to force ffmpeg to decode using baseline profile by default it goes with high and add following parameter dont know how [22:06] <barque> bleh re-encoding of the same thing fixed the issue :/ [22:06] <sam__> rc=abr mbtree=1 bitrate=479 ratetol=1.0 / [00:00] --- Wed Jan 30 2013

1 0

ffmpeg-devel.log.20130128
by burek 29 Jan '13

29 Jan '13

[00:16] <cone-76> ffmpeg.git 03Stephan Hilb 07master:0501d0646849: lavd/v4l2: use avcodec_find_decoder in list_formats [02:18] <cone-76> ffmpeg.git 03Michael Niedermayer 07release/1.1:59f7d583a3e4: mpeg1enc: Disable threads for resolutions too large for multi-threading [02:18] <cone-76> ffmpeg.git 03Xi Wang 07release/1.1:ea2d44503f08: rtpenc: fix overflow checking in avc_mp4_find_startcode() [02:18] <cone-76> ffmpeg.git 03Xi Wang 07release/1.1:b54c155f5b18: rtmp: fix multiple broken overflow checks [02:18] <cone-76> ffmpeg.git 03Xi Wang 07release/1.1:c2d11275f7cf: rtmp: fix buffer overflows in ff_amf_tag_contents() [02:18] <cone-76> ffmpeg.git 03Michael Niedermayer 07release/1.1:f4fb841ad13b: sanm: check image dimensions before using them [02:18] <cone-76> ffmpeg.git 03Michael Niedermayer 07release/1.1:5c316acaa08e: ffmpeg: copy tmcd track timebase parameters [02:18] <cone-76> ffmpeg.git 03Michael Niedermayer 07release/1.1:5589549c1d9e: movenc: Calculate fps for tmcd without intermediate step. [02:18] <cone-76> ffmpeg.git 03Michael Niedermayer 07release/1.1:bfd586577cbe: movenc: check that fps for tmcd is within encodable range. [05:09] <vtorri> hey [05:09] <vtorri> what's the status of h265/hevc in ffmpeg ? [05:10] <Compn> ask #libav-devel , they're working on it for gsoc iirc [05:10] <Compn> its status is ... nothing in ffmpeg yet :P [05:11] <vtorri> ok [05:11] <vtorri> thanks [05:11] <Compn> probably another hugely bloated piece of crap standard [05:11] <Compn> specification, i mean [05:11] <Compn> with lots of crap that only one channel in one country will ever use [05:11] <wm4> actually they say that h265 is less complex than h264 [05:11] <Compn> making it the largest decoder , again, for no reason [05:11] <Compn> hahaha [05:11] <vtorri> well, they certainly took h264 spec and added plenty of other stuff [05:11] <Compn> you believe it tho wm4 ? [05:12] <wm4> Compn: yes, because it wasn't the publishers who said that [05:12] <wm4> just people talking about the codec [05:12] <Compn> anyone compare the specs size of both of them ? [05:12] <vtorri> and about daala in ffmpeg ? [05:12] <wm4> daala doesn't even exist [05:12] <Compn> daala is vp9 ? [05:13] <wm4> http://xiph.org/daala/ [05:13] <Compn> vp9 support in libvpx was added to ffmpeg in a merge ... [05:13] <vtorri> i saw that in xiph homepage [05:13] <wm4> basically theora second try [05:13] <Compn> oh wow [05:13] <Compn> i dont even [05:13] <wm4> but I don't think they have anything working [05:13] <vtorri> well, it exists a bit, there's a git repo :) [05:14] <wm4> it's in the early stages of research probably [05:16] <Compn> google owns on2 now [05:16] <Compn> wonder if they got properly integrated with the rest of googles 'brain drain' [05:17] <Compn> googles web search is turning unusable for me. returning barely related results now [05:25] <michaelni> Compn, theres a gsoc student working on h265/hevc and some other people but thanks for pointing everyone to libav [05:27] <michaelni> i just wanted to help that student and review some of the code for security issues [05:27] Action: Compn grumbles about bunch of users going to be asking questions about h265 [05:27] <Compn> are you still banned from the lists ? [05:28] <Compn> ...and irc ? [05:28] <michaelni> i guess if i help him review the code iam libav devel then [05:28] <Compn> ehe [05:28] <michaelni> kinda ROTFL [05:28] <Compn> well, as long as libav keeps removing your copyright author you wont be :P [05:28] <vtorri> that's crazy to ban people in foss... [05:29] <wm4> vtorri: ffmpeg and co is a good example how foss can be crazy [05:29] <Daemon404> i dont recall any authorship removal. [05:29] <wm4> just ask Daemo... oh here he is [05:29] <michaelni> Compn, lists AFAIK yes iam banned, IRC i was told to leave which i did and did not return so i dont know [05:29] <Compn> Daemon404 : the h264 qpel move from dsputil to its own file ? [05:29] <wm4> why did they ban you? [05:30] <Compn> or whichever copyrights michael fixed [05:30] <Compn> its a blur to me, the merges [05:30] <Compn> why am i talking about libav again [05:30] <Compn> enough of them [05:30] <michaelni> from libav-dev i always was baned from other lists i got instantly baned when i replied to a user telling that the bug he hit was fixed in ffmpeg or so [05:31] <Compn> Daemon404 : you want to review dxva2 patch ? :) [05:31] <Daemon404> fuck. no. [05:33] <Compn> highgod said he wanted to work on opencl / threading of filters, if he can get past this dxva2 patch ... [05:34] <Compn> i'm really surprised china hasnt taken over any big open source projects yet [05:34] <Daemon404> you must have missed china MITMing github earlier [05:35] <Compn> i heard bout githubs id_rsa problem , but i think its unrelated :p [05:35] <Daemon404> nah [05:36] <Daemon404> apparently in china, their glorious government is MITM github's ssl [05:36] <Daemon404> or something [05:36] <wm4> don't they do that with all SSL? [05:36] <Daemon404> i cant remember [05:37] <Daemon404> i do know firefox has had a bug open for eons for allowing a chinese root cert [05:37] <Compn> >using bloated ass firefox [05:37] <Compn> hey look, pdf reader ... that will end well! [05:37] <wm4> everyone should use IE6 - light, smart, bug-free [05:38] <Daemon404> pretty sure all current browsers are bloated [05:38] <wm4> except dillo [05:38] <Compn> remember when chrome was small? :P [05:38] <Daemon404> wm4, i said current [05:42] <vtorri> dillo uses its own rendering engine (no gecko nor webkit) ? [05:43] <wm4> no, and that's why some would call it a toy browser [12:34] <cone-597> ffmpeg.git 03Piotr Bandurski 07master:51e9d2dbc8f8: aasc: fix 16bpp on big-endian [14:19] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:4eb93bed4e9b: swscale: GBRP output support [14:19] <cone-597> ffmpeg.git 03Daniel Kang 07master:05b0998f511f: dsputil: Fix error by not using redzone and register name [14:19] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:bb2f4ae43422: Merge commit '05b0998f511ffa699407465d48c7d5805f746ad2' [14:52] <cone-597> ffmpeg.git 03Luca Barbato 07master:4839fbe2d1b9: shorten: fix array subscript is below array bounds warning [14:53] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:834e9fb05639: x86: hpeldsp: Fix a typo, use the right register [14:53] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:1146bbc5a630: Merge remote-tracking branch 'qatar/master' [15:54] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:4484c722f68e: alsdec/read_specific_config: check for init_get_bits failure [19:44] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:8c4aebb58d00: qdm2: increase noise_table size [20:39] <durandal_1707> michaelni: encoding bgra ljpeg segv here [20:43] <durandal_1707> which is strange because fate is not failing [20:56] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:5c9cae744752: dirac: Only use MMX if MMX is available. [20:56] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:94ef1667bb04: dirac/x86: Fix handling blocksizes that are not a multiple of 4 [21:32] <teratorn> greetings most knowledgeable ffmpeg wizards, pray tell does these stack frames indicate a bug in swresample, or can that not necessarily be inferred? http://hastebin.com/cucaxeciva.txt [21:32] <teratorn> i'm wondering because the pointers look sane, until conv_AV_SAMPLE_FMT_S16_to_AV_SAMPLE_FMT_FLT is called with pi=0x1f40 <Address 0x1f40 out of bounds> [21:33] <teratorn> so if my input was bad I would expect a crash sooner. but it crashes with SIGABRT in frame #6 [21:48] <michaelni> teratorn, how to reproduce this ? [21:48] <teratorn> michaelni: i don't have a minimal example handy. I can reproduce on my end easily, but it would be some work to make something that *you* could run :( [21:49] <teratorn> I was just kinda wondering if the pointer parameters indicated a bug in swresample, or if I should do more work to make sure im passing in valid inputs first [21:50] <michaelni> is that stacktrace complete ? #6 .. #10 ? [21:50] <teratorn> michaelni: sec [21:51] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:1336382c6d4d: avfilter_get_audio_buffer_ref_from_frame: fix handling of >8 channels [21:51] <teratorn> michaelni: http://hastebin.com/nividoleja.txt [21:51] <teratorn> I'm in code called via a JNI call from Java :( [21:52] <teratorn> (im sorry) [21:52] Action: teratorn works on coming up with better test case [21:55] <teratorn> ffmpeg 1.01 - I guess I should get around to upgrading, also :/ [21:55] <teratorn> 1.0.1 [22:03] <teratorn> no difference on ffmpeg trunk - I do imagine this is my fault somehow. the stack trace looked wonky, so I was just wondering if any of you gdb nuts could tell anything from it... [22:05] <michaelni> teratorn, can i see the stacktrace from trunk ? (so the linenumbers make sense to me) [22:05] <teratorn> michaelni: sure. one sec [22:06] <teratorn> michaelni: http://hastebin.com/bomubixajo.txt [22:09] <michaelni> teratorn, are the in/out buffers you pass correct, i mean not pointer to pointer vs just pointer mixup ? [22:10] <michaelni> also i assume you never had swresample working and this is not just one case out of otherwise working code ? [22:11] <teratorn> no, never had it working yet [22:11] <teratorn> they look right on my end. i'm working on a test case though, so I can't be sure [22:12] <teratorn> my brain didn't understand how swresample could have gotten a pointer like 0x1f40 from anything I passed in.... [22:12] <teratorn> which is what prompted my questions :) [22:13] <michaelni> well a few av_log/printf in swr should awnser how it got there [22:13] <teratorn> yeah.. [22:52] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:14c8ee00ffd9: vp3dec: move threads check out of header packet type check [23:25] <cone-597> ffmpeg.git 03Michael Niedermayer 07master:3939b790f2eb: wmavoicedec: use the checked bitstream, reader [23:37] <someone-noone> Hello! I can't redirect ffmpeg output to pipe, where on other end my application is reading with kqueue() (OS X). [23:38] <someone-noone> However I can read data from my application if I, for example, make echo "test" > test.pipe [23:38] <someone-noone> As well as I can write there with simple C application. [23:39] <someone-noone> As well as I can write with ffmpeg to pipe and read with ffplay [23:39] <someone-noone> What can it be? [23:40] <someone-noone> ffmpeg hangs and I even can't exit with Ctrl+c [23:40] <someone-noone> only with killall [23:50] <cone-597> ffmpeg.git 03Paul B Mahol 07master:9efceaf1f788: takdec: switch to init_get_bits8() [00:00] --- Tue Jan 29 2013

1 0

ffmpeg.log.20130128
by burek 29 Jan '13

29 Jan '13

[01:15] <Jake_> Hello [01:15] <Jake_> I'm trying to run ffmpeg and then try to change its audio recording under pulse audio control, but it isn't showing up. [01:16] <Jake_> I don't know what's the problem, maybe I should use pulse syntax under my command? [01:16] <Jake_> But that'll lead it to saying that pulse audio doesn't exist. [01:16] <Jake_> Any help will do. [01:18] <klaxa> what command line are you running? [01:18] <klaxa> does it look anything like this?: ffmpeg -f alsa -i pulse output.wav [01:19] <Jake_> more like this: ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 25 -s 712x366 -i :0.0+229,186 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 output.mov [01:20] <Jake_> klaxa: more detail in this link: http://pastebin.com/iJj66Anm [01:20] <klaxa> pulse is running though? does your soundcard support recording? [01:21] <klaxa> it doesn't really look like an ffmpeg related error to me [01:21] <Jake_> Yes, pulseaudio is running identified by system monitor. It uses my microphone as default recording device. [01:22] <Jake_> klaxa, I thought so , since it's concern around Pulse. But even when i run ffmpeg replacing hw:0,0, it still won't show under the pulse audio control recording tab [01:23] <klaxa> i think that shouldn't even work with pulse running... [01:23] <klaxa> because pulse would be blocking the alsa devices [01:24] <Jake_> I've tested with sound recorder (default gnome applet), it's properly identified with pavucontrol and able to select between input/monitor [01:25] <klaxa> what ffmpeg version are you running? [01:25] <Jake_> It should be in the pastebin, if you haven't read it. [01:25] <Jake_> 0.7.13 [01:25] <Jake_> Not compiled, install pre-packaged [01:25] <klaxa> ah right forgot i could check there, i suggest you get a newer version that one is ancient [01:27] <klaxa> either compile from source (recommended, but takes some time) or download a static build [01:27] <klaxa> you can get static builds either here: http://ffmpeg.gusari.org/static/ or here: http://dl.dropbox.com/u/24633983/ffmpeg/index.html [01:27] <klaxa> but really, if possible compile from source [01:30] <Jake_> Ok, I've considered that option multiple times. But I'm not sure if that'll resolve the issue. [01:31] <klaxa> given that your ffmpeg version is from 2011 or even older, it's very likely [01:35] <Jake_> Sounds to me like it's a bug of somekind that's prominent in my version. [01:35] <klaxa> i'm not sure, but it's not impossible [01:37] <Jake_> So if I manage to compile it successfully, then how would I update if ffmpeg release another version? Is it related to something about git pull? [01:38] <klaxa> hmm... i would install it into a seperate directory [01:38] <klaxa> during ./configure specify a directory for the binaries with --prefix=/some/dir [01:39] <klaxa> there's a lot of flags you'll probably want to activate during configuration [01:44] <Jake_> Ok. Thanks I'll see what I can do. [03:21] <mate364> i have recently changed from ubuntu 12.10 to linux mint Cinnamon 14... now my webcam capture is out of sync... would this have anything to do with the OS? [03:22] <mate364> doing everything exactly the same? [03:36] <mate364> http://pastie.org/5892282 [03:37] <mate364> it is exactly what i did when running ubuntu 12.10 no syncing problems then [03:40] <lodenrogue> hey guys can anyone help me one minute? I am getting an error in a script. It says "Incorrect Frame Size. Failed to set value '"1680x1050"' for option 's'. [04:14] <lodenrogue> which audio codec can I use with ffmpeg? [04:43] <zap0> anyone you want. [06:41] <TAFB> can anyone help me to get ffmpeg to recognize my virtual capture card "AmaRec Video Capture"? Here is my vfw and dshow list: http://pastie.org/pastes/5894139/text [06:45] <zap0> ffmpeg doesn't recognize a "AmaRec Video Capture" [06:49] <TAFB> oh, bummer :( [06:49] <TAFB> I wonder how ffsplit uses ffmpeg to do it? it works fine with AmaRec Video Capture but only encodes in h264 which is too much for my cpu :( [06:54] <TAFB> I'm off to bed. I'll worry about it tomorrow. [06:54] <n3rV3> hi, i am working on building a video streaming server. i am reading the documentation and links online and found that youtube asks for 1080p videos with bitrates ~50 Mbps [06:55] <n3rV3> i know that its needed for the uploads to have 3 times the recommended bitrates [06:55] <n3rV3> but isn't 50 mbps a bit too much? [06:55] <n3rV3> or have i missed out something [08:23] <Ycarene> I want to capture an application window with ffmpeg, but I don't want to write it at the same dimensions, how do I scale it down so that it captures a 1440x900 screen and writes a 720x450 video file? [08:42] <SubJunk> When cross compiling FFmpeg (on Linux for Windows) how can I make the enable-avisynth flag work? It gives an error something like error can not find vfw32 (from memory) [12:40] <antonello> I'm trying to run the resampling_audio example, but I display the following error "The value set by option 'in_sample_fmt' is not a sample formatThe value set by option 'out_sample_fmt' is not a sample format". I want know if this comes from one of my bad installation? [14:51] Last message repeated 1 time(s). [16:10] <TAFB_afk> so I can't use my Blackmagic Intensity Pro in ffmpeg? :( [dshow @ 0000000000324ae0] Unknown compression type. Please report verbose (-v 9) debug information. [16:10] <TAFB_afk> video=Blackmagic WDM Capture: Not yet implemented in FFmpeg, patches welcome [16:33] <sam_> how to setup rc parameter libx264 through ffmpeg by default it takes rc=abr [16:34] <sam_> http://pastebin.com/nn25GzuC [16:35] <Mavrik> sam_, -x264opts rc_lookahead=40 [16:36] <sam_> if you look the commandline i mentioned rc_lookahead=40 [16:37] <sam_> is this not sufficient? [16:37] <sam_> i want to set it like / rc=crf [16:39] <JEEB> just set a crf value via -crf [16:39] <JEEB> that's it [16:39] <JEEB> that will set rc to crf :P [16:39] <JEEB> actually, crf is the default nowadays, and you only get something else if you set a bit rate [16:40] <sam_> means we dont need to set rc_lookahead=40 then? [16:41] <JEEB> no... that's completely unrelated [16:41] <JEEB> that's the lookahead length [16:41] <JEEB> in frames [16:41] <sam_> let me try that and update you [16:41] <JEEB> you should leave that be and just set a preset to control speed/compression related settings [16:41] <JEEB> (and a rate control mode) [17:20] <svt_raiden_> hi all [17:21] <svt_raiden_> where can I find information about the compatibility of mss1 and mss2 codecs with ffmpeg? [17:21] <svt_raiden_> does anyone knows? [17:21] <svt_raiden_> Please help [17:34] <klaxa> svt_raiden_: ffmpeg -codecs | grep mss [17:34] <klaxa> both codecs are supported for decoding only [17:34] <svt_raiden_> ok - I will check it now. [17:34] <svt_raiden_> thank you klaxa [17:42] <praedo> hello [17:43] <praedo> i want to copy an audio stream to another stream which has silence [17:43] <praedo> how should that be done? [17:43] <praedo> using -map [17:47] <svt_raiden_> I have another question... :) [17:47] <svt_raiden_> how can you discover weather a .wmv video uses mss codec or not? [17:48] <svt_raiden_> is there a terminal command that gives you all info on a wmv file? [17:59] <durandal_1707> svt_raiden_: ffprobe/ ffmpeg -i to see what codecs are in streams.... [18:00] <svt_raiden_> durandal_1707: you mean: ffmpeg -i <filename> [18:00] <svt_raiden_> ? [18:01] <durandal_1707> yes [18:01] <svt_raiden_> OK thanks. [18:01] <svt_raiden_> it worked [18:02] <svt_raiden_> but apperantly the file that I have is not with the mss codecs, that I need. [18:02] <svt_raiden_> can anyone could tell me where can I download a sample video file with mss codecs? [18:02] <svt_raiden_> I need to run an experiment [18:04] <praedo> hello [18:04] <praedo> i want to copy an audio stream to another stream which has silence [18:04] <praedo> how should that be done? [18:04] <praedo> using -map [18:09] <Mavrik> um [18:09] <Mavrik> why not replace the other stream? [18:29] <Zeeflo> I use this command ./ffmpeg -i input.avi -sn -aspect 16:9 -vf "scale=1280:720,subtitles=file.srt" -c:v libx264 -b:v 1800k -maxrate 1800k -bufsize 3200k -c:a aac -strict experimental -ac 2 -b:a 128k -movflags +faststart output.mp4. [18:29] <Zeeflo> My question is, will the preset Baseline do any difference? [18:29] <Mavrik> profile you mean? [18:29] <Zeeflo> yes [18:29] <Mavrik> it's gonna make your movie look like garbage [18:29] <Zeeflo> i thought so [18:30] <Mavrik> and it still won't work on devices that don't support Main and High profiles :) [18:30] <Zeeflo> i cant understand why Longtail videos suggest using that profile for encoding video.. [18:30] <Mavrik> Zeeflo, some older devices don't support Main or High profile [18:30] <Mavrik> but also... those certanly won't be able to play 720p baseline video either :P [18:31] <Zeeflo> as you can see I dont even use a profile.. [18:31] <Mavrik> it's being autoselected [18:31] <Mavrik> you see the chosen one at the start of encode [18:31] <Zeeflo> ok [18:31] <Zeeflo> I wont be making any changes today then ;) [18:32] <praedo> Mavrik, yes i want to replace the other stream, how is that done? [18:33] <Mavrik> praedo, just specify streams you want to keep with "-map" [18:34] <praedo> yes [18:35] <praedo> but how? [18:35] <praedo> i tried -map 0:1 -map 0:2 [18:39] <praedo> i want to copy an audio stream to another stream which has silence [18:40] <praedo> how should that be done? [18:40] <praedo> using -map [18:40] <Zeeflo> praedo, [18:41] <Zeeflo> you need to put " around your streams [18:41] <Zeeflo> like this -map "0:1" -map "0:4" [18:47] <praedo> that means 0:1 will be copied to 0:4 [18:47] <praedo> ? [18:47] <praedo> do i understand correctly? [18:48] <Zeeflo> no [18:48] <Zeeflo> you have an input file with streams in it [18:48] <Zeeflo> you tell ffmpeg to take stream number 1 and stream number 4 and put that in your output file [18:51] <Zeeflo> lets say you have a MKV file, with a video stream (stream 0:0 for an example) and have 10 audio streams, but you only want 1 of them (stream 0:3 for an example) in your OUTPUT.MKV/avi/mov what ever.. [18:51] <Zeeflo> Thats where you use the -map commands [18:52] <Zeeflo> if you dont use a map command, ffmpeg will automatically chose the best audio stream if you dont make a MAP selection to tell it which audio/video stream to use in the output.avi/mkv file [18:53] <Zeeflo> got it? [19:05] <praedo> i have a mp4 with sound only on one speaker [19:06] <praedo> i want to double that track to both channels [19:06] <praedo> i don't know if that's a stream, a track or a channel exactly [19:06] <praedo> or it's all the same [19:06] <Zeeflo> you can use ffprobe to find out [19:06] <Zeeflo> ffprobe file.mp4 [19:10] <praedo> only one audio: [19:10] <praedo> Stream #0.1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 62 kb/s [19:11] <praedo> then how can i duplicate that to a second channel? [19:11] <praedo> -map "0:1" -map "0:2" [19:11] <praedo> would work? [19:11] <Zeeflo> you dont need to [19:12] <Zeeflo> its a stereo audio file [19:12] <Zeeflo> perhaps you should fix your speaker instead ;) [19:23] <klaxa> is there a way to autodetect all available libraries? [19:24] <klaxa> for configuration i mean [19:32] <klaxa> nvm [19:51] <praedo> Zeeflo, it's not the speaker [19:51] <praedo> it's the file that has silence on one channel [19:51] <praedo> i want to duplicate the good channel to the bad one [19:51] <praedo> how is that done? [19:51] <Zeeflo> I dont know. I have never done something like that.. I dont even think FFmpeg can do that. You would need some sampling software? [19:56] <durandal_1707> praedo: there are many audio filters that can do that: pan, amerge [19:59] <praedo> ffmpeg must be able to copy streams [19:59] <praedo> from one to replace another [20:02] <Zeeflo> does ffmpeg perform just as well on a windows operating system as it would on a linux? [20:02] <Zeeflo> correction: as it DOES on a linux ;) [20:03] <cbreak> windows has many shortcommings [20:04] <Zeeflo> ok [20:04] <cbreak> and ffmpeg has many pecularities, such as C99 requirement [20:04] <cbreak> if you can get your hands on a binary it should work [20:04] <Zeeflo> hmm [20:04] <cbreak> if you want to compile it yourself, prepare for a puzzle hunt [20:04] <Zeeflo> nono [20:04] <Zeeflo> we are getting a new encoding server tomorrow. [20:04] <cbreak> last time I did that it took me a day or so to get it to compile [20:05] <Zeeflo> I was just wondering if I for once should try a windows o/s [20:05] <cbreak> server? With windows? Is that a joke? :/ [20:05] <cbreak> I do use ffmpeg on windows, so that works [20:05] <cbreak> but... it's windows... [20:05] <Zeeflo> i dont know.. Last time I had a windows server it was 2003 web edition.. [20:06] <Zeeflo> I can only assume its been developed since then. [20:06] <Mavrik> Zeeflo, there's no difference in performance between Linux and Windows [20:06] <Zeeflo> Mavrik, ok [20:06] <Mavrik> if you have it compiled for same arch [20:06] <Zeeflo> ok [20:06] <Mavrik> Zeeflo, grab Zeranoes static builds [20:07] <cbreak> I had to compile ffmpeg myself since I make closed source software [20:07] <Zeeflo> Ill give it a go :) [20:07] <cbreak> so the GPL parts of ffmpeg were a hindrance [20:07] <cbreak> most precompiled ffmpeg come with those though [20:07] <Mavrik> mhm [20:07] <Zeeflo> its no worse then wiping windows and get debian on it if its shite. [20:07] <Mavrik> isn't it possible to actually compile ffmpeg with MSVC now? [20:07] <cbreak> not sure if ffmpeg's GPL parts are GPL 2 or 3 though [20:07] <Mavrik> I had to use mingw-64 last time I did it. [20:08] <cbreak> Mavrik: last time I checked: no way. [20:08] <Mavrik> cbreak, both, there's a special switch for v3 stuff [20:08] <cbreak> it required C99 [20:08] <cbreak> and it required cygwin [20:08] <cbreak> which I had to compile with some custom gcc [20:08] <Mavrik> hmm [20:08] <cbreak> ... I don't even remember :/ [20:08] <cbreak> was ... around last spring or so [20:09] <Mavrik> it doesn't really need cygwin, usually people compile it with mingw to avoid performance penalties and dependencies [20:09] <durandal_1707> ffmpeg now compiles with MSVC just fine, read documentation [20:09] <cbreak> could be. [20:10] <cbreak> then msys required some ancient gcc 3 or so? :) [20:11] <cbreak> what ever. Maybe it's easier nowadays :) [20:11] <durandal_1707> http://ffmpeg.org/platform.html#Microsoft-Visual-C_002b_002b [20:17] <Zeeflo> Does any of you guys encode on a Xeon E5620 cpu ? [20:22] <Peace-> ffmpeg compiled from git [20:22] <Peace-> ./ffmpeg -s 1280x800 -f x11grab -r 25 -i :0.0 -vcodec ljpeg -threads 2 -y /home/sem/output.avi [20:22] <Peace-> core dumped [20:22] <Peace-> compiled now [20:34] <Peace-> btw http://paste.ubuntu.com/1583353 [20:35] <Peace-> this is the debug [20:36] <durandal_1707> reproduced here [22:26] <jiippe> is v4l2 broken in ffmpeg? [22:28] <Mavrik> nop [22:29] <jiippe> im trying to capture video from easycap dc60, it works perfectly in vlc and other players.. but when i try to capture and encode to mpeg2 or mpeg4 ... it just fails [22:34] <jiippe> http://pastebin.com/tX50JRzH [22:34] <jiippe> okay there [22:52] <jiippe> no ideas? [22:57] <Mavrik> jiippe, on some devices you have to manually enable input over ctl command of whatsit [22:57] <Mavrik> also [22:57] <Mavrik> [video4linux2,v4l2 @ 0x1e4d6c0] The v4l2 frame is 391680 bytes, but 829440 bytes are expected [22:57] <Mavrik> is seems your input configuration is wrong [22:57] <Mavrik> and your device is streaming in different format than what you're setting [23:02] <jiippe> YUV rawvideo [23:02] <jiippe> l420 [00:00] --- Tue Jan 29 2013

1 0

ffmpeg-devel.log.20130127
by burek 28 Jan '13

28 Jan '13

[00:36] <michaelni> anyone has any oppinons on "Frantisek Korbe (2.5K) [FFmpeg-devel] [Patch] Update of documentation.html" ? [00:37] <beastd> michaelni: is it about the ffmpeg book? i did not read that patch yet [00:39] <michaelni> beastd, yes, id like to awnser something, either a applied or a "no because ..." [00:49] <beastd> michaelni: imho it is not ok because it is not community contributed in the sense that we own the material. the visitor actually needs to buy it from the author/publisher. [00:50] <beastd> so maybe a new section would be in ordre [00:52] <beastd> i am not sure if it is best to put it on the documentation page either. but i can't think of any better place atm [00:52] <beastd> any other thoughts? i can send a reply on the ML [00:53] <michaelni> so we should add a section for books? [00:53] <michaelni> either way, yes please reply [01:08] <saste> michaelni, i'd like your opinion about my image2 RFC, especially since it affects code written by you [01:10] <beastd> saste: despite i am not michaelni , i found the pattern file similar hard to grasp than updatefirst. maybe even harder, dunno. [01:11] <saste> beastd, really, what's hard to get about that? [01:12] <beastd> saste: really? i don't even know where to start. [01:12] <beastd> saste: my first impression would be i could supply a text file with one name per line [01:13] <saste> well i don't mind changing the name if that helps [01:14] <saste> the idea is that the output is determined by the chosen "pattern" [01:14] <saste> "pattern" is not the right term here, i'm keeping for consistency with the demuxer [01:14] <beastd> saste: not sure if it is enough to tackle this problem. [01:14] <saste> but the point here is to avoid the behavior to depend on the special name of the file [01:15] <beastd> saste: yes, maybe even pattern is not the right option to achieve that effect [01:15] <saste> this helps predictability, especially when you don't have control on the output filename [01:15] <saste> pattern => specify the output filenames pattern [01:15] <beastd> you mean in case the output filename contains % [01:15] <saste> whatever, see the second patch [01:16] <saste> anyway updatefirst (which i find counter-intuitive) is retained [01:16] <beastd> saste: the second is for generating names containing timestamps, right. will reread in a few minutes [01:17] <beastd> saste: i do not find updatefirst easy to guess either. both solutions have problems because they are close to the mechanics [02:10] <beastd> saste: about image2 muxer pattern: i will comment on-list tomorrow. i am way too tired now :( sorry [02:11] <beastd> but i have already read the patches multiple times and part of the source [02:51] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:362271d72fc3: mvdec: Check the frame counter against the correct limit. [02:51] <cone-701> ffmpeg.git 03Michael Niedermayer 07master:5de286ef88be: mvdec: check var_read_string() return value [06:16] <cone-941> ffmpeg.git 03Michael Niedermayer 07master:b16830840eb9: tiff: in add_string_metadata() check the count more completely [06:16] <cone-941> ffmpeg.git 03Michael Niedermayer 07master:167af704ea3d: swr: limit phase_shift to a less insane value. [06:16] <cone-941> ffmpeg.git 03Michael Niedermayer 07master:f9abeecd94cf: swr/ build_filter: use av_calloc() fix buffer overflow [12:02] <cone-76> ffmpeg.git 03Paul B Mahol 07master:07a7145d89fb: lavc/iff: rgbn: fix decoding on big endian [12:37] <cone-76> ffmpeg.git 03Clément BSsch 07master:f2b6aabd3da7: lavf/srtdec: do not try to queue empty subtitle chunks. [12:37] <ubitux> surprise :) [12:38] <michaelni> ubitux :))) [12:43] <durandal_1707> how that possible? [12:43] <ubitux> len was < 0 [12:43] <ubitux> malloc(len) [12:44] <ubitux> and& weird errno :p [12:54] <wm4> lol C string handling [12:54] <wm4> the "classic" kind [13:31] <durandal_1707> vlc - I can't really play everything. [13:41] Action: durandal_1707 trolling on g+ can be fun [13:44] <cone-76> ffmpeg.git 03Luca Barbato 07master:fe1057e017fc: doc: update the reference for the title [13:44] <cone-76> ffmpeg.git 03Ronald S. Bultje 07master:f90ff772e7e3: Move H264/QPEL specific asm from dsputil.asm to h264_qpel_*.asm. [13:44] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:6b2f7fd1c728: Merge commit 'f90ff772e7e35b4923c2de429d1fab9f2569b568' [14:04] <durandal_1707> http://what-if.xkcd.com/28/ [14:43] <Compn> durandal_1707 : has the age/maturity dropped on g+ yet ? like facebook, myspace, etc ... [15:46] <j-b> durandal_1707: still an asshole? [15:46] <j-b> C U never [15:53] <funman> durandal_1707: vlc plays enough things to be able to host ffmpeg git at least [15:58] <cone-76> ffmpeg.git 03Daniel Kang 07master:71155d7b4157: dsputil: x86: Convert mpeg4 qpel and dsputil avg to yasm [15:58] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:91c8921d807b: Merge commit '71155d7b4157fee44c0d3d0fc1b660ebfb9ccf46' [15:58] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:aa3f4499557e: x86/hpeldsp: Fix author attribution [15:58] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:5934be16cc24: x86/mpeg4qpel: Fix author attribution [15:58] <cone-76> ffmpeg.git 03Daniel Kang 07master:d9e62f368d16: dsputil: add missing HAVE_YASM guard [15:58] <cone-76> ffmpeg.git 03Daniel Kang 07master:5327a455529e: dsputil: x86: Correct the number of registers used in put_no_rnd_pixels16_l2 [15:58] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:edde562130fe: AVG_PIXELS8_XY2: fix typo, make code actually work [16:02] <durandal_1707> funman: i can't understand j-b reaction, he got insulted by truth [16:02] <funman> you're a pathetic human being [16:03] <michaelni> durandal_1707, i think your attack against vlc is a bit inappropriate for this channel [16:03] <funman> you should have some respect for his work [16:04] <cbsrobot> guys come on [16:06] <iive> durandal_1707: truth hurts the most. Though in this case I cannot confirm if this is true. [16:07] <iive> I haven't used vlc as a program, but the developers and the project as whole seems quite calm and friendly place. [16:07] <funman> any way we have numerous ways to make bug reports, including "i tried vlc foobar.avi and it doesn't work" [16:08] <iive> at least I haven't heard of any high profile drama... [16:08] <iive> funman: well... at lest this bugreport contains the exact command line :P [16:08] <funman> well we have some drama too .. [16:09] <iive> try dealing with. [16:09] <iive> "It doesn't work. Fix it." [16:09] <michaelni> durandal_1707, i dont really care much about people trolling or not trolling but once it actually ends up hurting people i think then it goes too far [16:10] <michaelni> also if something doesnt work for you, you should write a full and clear bug report [16:11] <iive> funman: can you give some examples for high profile drama? [16:11] <funman> i would even be ok with only a sample [16:11] <funman> iive: well just google rémi denis courmont and you should find some [16:13] <funman> although i hope now the drama is behind us [16:14] <iive> lgpl, no appstore... [16:16] <funman> well lgpl should allow putting vlc in appstore i think [16:19] <iive> witch such kind of drama the project could easily be canadian. Maybe canadians got their politeness from the french :) [16:25] <iive> that was intended as compliment. If it is too insulting, this is because I can't make proper compliments. sorry. [16:26] <funman> iive: didn't feel insulted [16:26] <funman> also i didn't know that canadian politeness was french born [16:28] <funman> i lived in french canada last year but never went to english part of the country [16:33] <cone-76> ffmpeg.git 03Clément BSsch 07release/1.1:fb876e457279: lavf/srtdec: do not try to queue empty subtitle chunks. [16:34] <cone-76> ffmpeg.git 03Rémi Denis-Courmont 07master:ec0e92002b72: vdpau: add missing pixel format for H.264 [16:34] <cone-76> ffmpeg.git 03Rémi Denis-Courmont 07master:78bc4d69ebe6: hwaccel: do not offer unsupported pixel formats [16:34] <cone-76> ffmpeg.git 03Daniel Kang 07master:0eedf5d74dce: dsputil: add missing HAVE_YASM guard [16:34] <cone-76> ffmpeg.git 03Daniel Kang 07master:96753bd00d6d: dsputil: x86: Correct the number of registers used in put_no_rnd_pixels16_l2 [16:34] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:f5c0b9aa6f38: Merge commit '96753bd00d6d4046db6818c0aadc21bf2a11d77b' [16:51] <durandal_1707> vlc - I can really play anything. [16:53] <cone-76> ffmpeg.git 03Martin Storsjö 07master:2026eb1408a7: arm: vp8: Fix the plain-armv6 version of vp8_luma_dc_wht [16:53] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:2b14344ab36e: Merge remote-tracking branch 'qatar/master' [17:00] <cone-76> ffmpeg.git 03Stefano Sabatini 07master:43af18ef8bd9: ffmpeg: implement -force_key_frames expression evalution [17:00] <cone-76> ffmpeg.git 03Stefano Sabatini 07master:27db2bf00093: ffmpeg: remove -crop* and -pad* options [18:02] <cone-76> ffmpeg.git 03Stefano Sabatini 07master:5306976be8a6: lavd/v4l2: sanitize logic of device_try_init(), so that it properly signal errors [18:30] <ulatekh> Not that I mind the brightly-colored static...I saved a minute's worth so that I can use it in my videos :-) [18:41] <cone-76> ffmpeg.git 03Giorgio Vazzana 07master:93d319a5826a: lavd/v4l2: select input immediately after opening the device [18:48] <cone-76> ffmpeg.git 03Matthieu Bouron 07master:5b83b2da08b4: lavc/dnxhddata: add frame_rates field to cid table [18:48] <cone-76> ffmpeg.git 03Matthieu Bouron 07master:9d602a0b0e95: lavc/dnxhdenc: print valid profiles when codec parameters are invalid [21:36] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:96f452ac647d: aacdec: check channel count [21:36] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:deefdf978846: avpriv_mpeg4audio_get_config: check init_get_bits() return code. [21:49] <durandal_1707> that init_get_bits fix is not really fix, overflow can still happen [21:50] <durandal_1707> and there are more paths that use this code so it is more serious work [22:53] <cone-76> ffmpeg.git 03Carl Eugen Hoyos 07master:a6a510165448: Fix some avi rawvideo formats on big endian. [23:42] <cone-76> ffmpeg.git 03Carl Eugen Hoyos 07master:d88d0b6db6bd: Write forced track flag to matroska files. [23:56] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:8888c72fcfe1: mjpegdec: fix memcmp size for *_count [23:56] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:73abc3a634d8: aacdec: check init_get_bits return [23:56] <cone-76> ffmpeg.git 03Michael Niedermayer 07master:4ade824e1f6a: mjpegdec: rgb mode is specific for ljpeg, disable it for others. [00:00] --- Mon Jan 28 2013

1 0

ffmpeg.log.20130127
by burek 28 Jan '13

28 Jan '13

[00:00] <ulatekh> kdenlive didn't recognize .nut [00:01] <ulatekh> huffyuv with .mov produced garbage output! It was kind of entertaining...I might use it later if I need brightly colored static for an effect :-) [00:04] <ulatekh> HuffYUV and a Matroska container produces a clip that identifies itself as progressive, i.e. the interlacing info was lost. [00:07] <beastd> ulatekh: might be easier to tackle the problem the other way around. e.g. try to find a list / set of formats kdenlive accepts, then try to see what of those can be generated by ffmpeg and is sufficiently compressed. [00:08] <beastd> ulatekh: you might need to find out what kdenlive uses to read the files to come to a somewhat complete answer [00:08] <ulatekh> I have an open request on kdenlive-devel for a list of supported formats. I wasn't able to figure out how that was determined by reading the source code. [00:10] <ulatekh> FFV1 and a Matroska container produces a clip that identifies itself as progressive, i.e. the interlacing info was lost. I guess I can add that to my bug report. At least I have one container format & two pixel formats working for my raw video. [00:16] <ulatekh> Score! "-vcodec ffv1" and a .mov container preserved interlacing AND was accepted by kdenlive! Hot dog, we have a weiner! [00:17] <wind_> Hi all, i am trying to burn or add subtitles to an avi file i am doing this: ffmpeg -i video.avi -vf subtitles=subtitle.srt out.avi , but in the output i am not getting any subtitle [00:25] <Keyboard_Warrior> ulatekh, in a more perfect world [00:25] <Keyboard_Warrior> interlacing would never have existed [00:25] <Keyboard_Warrior> and it definitly never would have been including in any modern day digital formats [00:25] Action: Keyboard_Warrior cries his eyes out [00:27] <ulatekh> Yeah, well, that's what my miniDV camcorder produces, and it's what DVD-video uses, so I gotta deal with it...and so should ffmpeg :-) [00:27] <beastd> wind_: can you show us the full output of your command on pastebin? [00:28] <wind_> yeah sure give me 5 min [00:29] <Keyboard_Warrior> ulatekh, dvd can be progressive [00:29] <Keyboard_Warrior> dvd supports soft-interlacing or whatever the technical term i [00:32] <ulatekh> I've never gotten 24000/1001 fps DVDs to work on all players...several of them stutter horribly on such input. [00:33] <ulatekh> AFAIK, interlacing exists because of 1950s TV-camera technology. [00:34] <Keyboard_Warrior> ulatekh, theres a broadcast/bandwidth issue [00:34] <Keyboard_Warrior> aswell [00:34] <Keyboard_Warrior> ulatekh, dvd players SHOULD support soft-telecine [00:34] <wind_> Here it is: http://pastebin.com/embed_js.php?i=wmsEjiTD [00:35] <Keyboard_Warrior> http://en.wikipedia.org/wiki/Telecine#Soft_and_hard_telecine ulatekh [00:36] <Keyboard_Warrior> ulatekh, but yeah, even today, interlacing is a big part of our lives. [00:36] <Keyboard_Warrior> the h264 standard has a TON of provisions for interlacing [00:36] <Keyboard_Warrior> including a special encoding mode to allow interlacing on indiidual macroblocks [00:38] <ulatekh> Ah yes..."soft telecined" is what I produce if I'm encoding the video from raw input. mpeg2enc does 3:2 pulldown. The last time I checked, ffmpeg didn't do soft telecine. [00:39] <ulatekh> Oh...I mean 2:3 pulldown...duh [00:45] <beastd> wind_: seems sth goes wrong. how does the playback look if you replace ffmpeg with ffplay and drop out.avi from that line? [00:45] <Keyboard_Warrior> ulatekh, soft telecine is where the actual video data is progressive but theres flags in the video file explaining how to telecine it [00:45] <Keyboard_Warrior> HARD telecine is where the actual videodata is progressive, but you apply a 2:3 pulldown to make it interlaced before encoding [00:46] <Keyboard_Warrior> ofcourse, both these are different from actual interlaced content [00:46] <Keyboard_Warrior> http://mod16.org/hurfdurf/?p=12 ulatekh [00:47] <ulatekh> "mpeg2enc -p" produces soft telecine...sorry about my sloppiness with terms. [00:50] <wind_> beastd: i run the command on a laptop where the screen is broken and on my local laptop i haven't ffplay [00:52] <beastd> oh [00:53] <wind_> any other tip ? [00:54] <beastd> wind_: sorry. can't think very clearly atm. but wait a bit. maybe sth comes to mind [01:03] <wind_1> i am trying with the mencoder now [01:07] <beastd> wind_1: wind_1 [01:07] <beastd> wind_1: maybe a problem with your fonts/fontconfig ? [01:08] <wind_1> how to troubleshoot ? [01:09] <beastd> wind_1: btw what language is the text contained in your srt file? [01:09] <wind_1> Greek [01:27] <beastd> wind_1: could be the chars you need are not in the font that got selected. maybe try to edit a copy of the subtitle file to contain some lating characters in e,g, the first sub. then use that as input and see if the latin part got rendered [01:35] <wind_1> ok i ll try this tomorrow thank you , good night [10:55] <Fjorgynn> what is H.265? [10:55] <Fjorgynn> http://techcrunch.com/2013/01/25/h265-is-approved/ [11:01] <Fjorgynn> käften [11:01] <Fjorgynn> So what should we think about this? [12:55] <antonello> hallo i must use swr_convert but my RUN FAILED ..... I don't know why . I have encoder and decoderContex and i set encoder parameters by decoder parametrs , after alloc the SwrContext using the two context . My code is http://pastebin.com/GXjjcWNP line 440-458 [15:47] Last message repeated 1 time(s). [16:24] <saste> antonello, did you have a look at doc/examples/resampling.c? [16:24] <saste> you use of the API is fishy to me [16:25] <saste> for example you open the encoder, *then* you set fields on it [16:25] <saste> that should be avoided, unless you know what you're doing [16:26] <saste> also audio samples buffers are tricky to deal with, that's why i pointed to the example [17:08] <wind_> Trying to burn subtitles using ffmpeg didn't work: http://pastebin.com/G5xPmzCp#. However trying to play it with ffplay it complains "fontconfig: Selected font isnot the requested one: 'Liberation Sans != 'Arial' " and the movie playes with subtitles [17:08] <wind_> Any idea ? [17:12] <saste> wind_, that seems regular font-mess, not strictly related to ffmpeg code [17:12] <klaxa> the error message is actually just a warning [17:12] <klaxa> ... or not? [17:13] <saste> also as klaxa wrote it is a warning, so you *may* be able to burn the subs, more or less [17:13] <wind_> Yeah it could be a warning but the final output avi fle doesn't contain any subtitles... [17:14] <klaxa> i get those messages a lot with mplayer2, however the fonts appear correct nonetheless [17:14] <wind_> i want to burn the subtitles in order to play the movie on the TV from a USB disk and not using any player like VLC or mplayer [19:11] <killown> actually I am using this ffmpeg -f alsa -ac 2 -i pulse -f x11grab -s hd1080 -r 24 -i :0.0 -vcodec libx264 -preset ultrafast -an -y video.mkv -acodec pcm_s16le -b:a 48000k -vn -y audio.mkv to grab x11, what could I do to encode it in 1080p and making ffmpeg requires less I/O and CPU? [19:16] <killown> I guess the problem is the preset [19:17] <killown> using -preset fast would requires less I/O and more CPU [19:17] <killown> rather than ultrafast [22:33] <simon___1> exit [00:00] --- Mon Jan 28 2013

1 0

ffmpeg-devel.log.20130126
by burek 27 Jan '13

27 Jan '13

[00:12] <cone-154> ffmpeg.git 03Michael Karcher 07master:dcbb920f1587: Fix atrac3 decoder broken in e55d53905f34f8e8747f6d321e9a695dc02ebb2f [00:18] <beastd> ^^^ e55d53905f34f8e8747f6d321e9a695dc02ebb2f is "atrac3: cosmetics: pretty-printing and renaming" "also does some minor refactoring." [00:18] <beastd> That is why I dislike this heavily squashed cosmetic patches [00:19] <JEEBsv> if there's refactoring, that's no longer cosmetics IMNSHO [00:19] <beastd> JEEBsv: yeah, kind of fuzzy line but i tend to agree [00:21] <beastd> also the error that introduced was as folloes: rename var in outer loop from cnt to i + rename variable from inner loop from i to j . In the 2nd rename the loop body was forgotten so it still contained i which was still there because of the 1st rename. [00:29] <beastd> IMHO it is best to just make a 1st patch that does all whitespace cosmetics that will produce zero output if diffed with "git diff -b" . then add a series with small consistent changes that are easy to review on top of that. this way there is a chance you will get a minimum eyeballing from e.g. patch readers or commit reviewers. [00:31] <j-b> just use tig -b to review [00:32] <beastd> Despite IMHO one should never apply cosmetics on a file one doesn't intent to work on *soonish*. With open source software and especially with recent popularity of Git there is a (high) chance someone is working on that very file you pointlessly change. [00:32] <burek> llogan, did you try to check the git hash from that tessus ffmpeg (from the forum) [00:32] <beastd> Happened to me multiple times and I am not a frequent patch contributor... [00:33] <burek> it seems the git hash does not exist in ffmpeg's git [00:57] <RamajanSan> HI [01:19] <llogan> burek: no, i didn't check. [01:29] <burek> maybe he cloned it to his own git or something, dunno [02:33] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:53a3fdbfc56d: 4xm: Check available space in read_huffman_tables() [02:33] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:d73b65ed0ebc: 4xm: add assert to check that the pointer from read_huffman_tables is within the array [02:33] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:cfc7b9cfff65: 4xm: remove avcodec_get_frame_defaults() calls [03:43] <cone-154> ffmpeg.git 03Michael Karcher 07release/1.1:302094e1d2ea: Fix atrac3 decoder broken in e55d53905f34f8e8747f6d321e9a695dc02ebb2f [04:35] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:66daebc9d50d: indeo4: check for invalid transform_size blk_size combinations [04:35] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:c8f25cafd2f2: atrac3: fix buffer size for get_bits. [05:21] <Daemon404> that ami... always amazes me what he finds [05:21] <llogan> Daemon404: same. i don't know what in hell he does to do that. [05:24] <llogan> burek: i didn't forget about your fate stuff. i intend to take a look after things slow here. [05:26] <Darkarnium> Hey all, quick, and somewhat strange question regarding url_open2. Looking through a patch, there's an option added to an avdict that's passed as the 'options' attribute into ffurl_open, which appears to cascade down to url_open2. How can I determine the supported values that url_open2 will accept? [05:27] <Darkarnium> As in, there's a 'user-agent' added, which works, however, there's also a 'cookies' option set, which doesn't end up in the HTTP request. If I rewrite to snprintf into a 'headers' option and pass that, the cookie does appear in the HTTP request. [05:28] <Darkarnium> Which makes me suspect that 'cookies' in not a valid option in an AVDict / Option to url_open2. Unless I'm losing the plot, which is entirely possible :) [05:28] <wm4> I've used "cookies" successfully lately [05:30] <wm4> probably all a matter of using it correctly, maybe I can help... [05:31] <wm4> though url_open2 is not a public API so you're probably asking from an implementer's perspective or something [05:31] <Darkarnium> Ahh, I thought as much [05:32] <Darkarnium> What format were you required to pass the cookie in? A "Set-Cookie" HTTP header style attribute - value and options "foo=bar; path=/ .. etc" or just attribute value ("foo=bar") [05:32] <Darkarnium> ? [05:32] <wm4> Set-Cookie [05:32] <wm4> the reason is probably that it allows specifying cookies per-domain [05:33] <wm4> http://curl.haxx.se/rfc/cookie_spec.html [05:33] <Darkarnium> Alright, I'll give it a crack :) [05:33] <Darkarnium> Cheers [06:04] <Darkarnium> wm4: Any chance of an example in your usage? As per the spec, only the attribute / value is required. Even so, if I also set the path, expires and domain fields, the cookie doesn't make it to the HTTP header. In this instance, I'm adding an option to override via CLI arguments to allow for a cookie to be set by the client when ffmpeg / ffplay is called - due to an appliation generating the cookie client side from an XML HTTP response, rather than by [06:05] <wm4> Darkarnium: your message was cut off at "than by" [06:05] <Darkarnium> a Set-Cookie HTTP header from the remote web service. [06:05] <wm4> I don't quite remember the code, but maybe setting the domain is required [06:08] <wm4> I'm basically doing sprintf(result, "%s=%s; path=%s; domain=%s; \n", name, value, path, domain) (yeah don't use sprintf) [06:09] <wm4> and pass that as "cookies" option [06:09] <Darkarnium> Alright, I'll wrap it up in an snprintf and give it a shit [06:09] <Darkarnium> ...shot rather [06:09] <Darkarnium> Thanks again for your help :) [06:35] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:61884d198521: sws: GBRP output support [06:35] <cone-154> ffmpeg.git 03Michael Niedermayer 07master:e4033d89f172: sws: GBRP9, GBRP10 GBRP12 GBRP14 output support [06:37] <Daemon404> michaelni, ill fix up my old patch tomorrow [06:37] <Daemon404> and re-send it [06:38] <michaelni> Daemon404, ok thx [10:43] <cone-258> ffmpeg.git 03Paul B Mahol 07master:254e11cc019d: lavf/gifdec: fix typo [10:43] <cone-258> ffmpeg.git 03Paul B Mahol 07master:d1d159d31ecb: lavf/gifdec: cosmetics: remove extra whitespace [10:43] <cone-258> ffmpeg.git 03Paul B Mahol 07master:9648e1495bf6: lavc/gif: remove some obsolete/irrelevant chunks [11:03] <durandal_1707> michaelni: i get [swscaler @ 0x295f9000] 42bpp not supported by yuv2rgb [11:07] <durandal_1707> when doing yuv420p -> gbrp14 [11:20] <cone-258> ffmpeg.git 03Nicolas George 07master:fccd8c21c40c: ffmpeg: add -guess_layout_max option. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:b6afb2dde1aa: lavfi: support unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:7bb98b753b45: lavfi: implement ff_all_channel_counts(). [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:b6b2f3433c21: lavfi: implement ff_query_formats_all(). [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:19506af788b6: lavfi/sink_buffer: accept unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:ea645e90a1ea: lavfi/buffersrc: accept unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:6d962aec8dd3: lavfi/af_aformat: accept unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:b00502457ac3: lavfi/af_aresample: accept unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:699b286a2145: lavfi/af_anull: accept unknown channel layouts. [11:21] <cone-258> ffmpeg.git 03Nicolas George 07master:41f025dff0a3: ffmpeg: support filtering of unknown channel layouts. [11:23] <durandal_1707> michaelni: why isAnyRGB have grbp in it? [11:26] <cone-258> ffmpeg.git 03Nicolas George 07master:255ae9f38096: lavfi: version bump and change log entries after the last commits. [11:40] <cone-258> ffmpeg.git 03Nicolas George 07master:5361f4958fbf: fate: add an attachment to the Matroska test. [11:42] <saste> ^^ please mention git hash when closing bugs [12:21] <durandal_1707> michaelni: should isAnyRGB list all gbrp formats and not just gbrp? [13:00] <cone-258> ffmpeg.git 03Nicolas George 07master:42c6f2a645a8: lavfi/vf_drawtext: default to expansion=normal. [13:03] <cone-258> ffmpeg.git 03Paul B Mahol 07master:25c75525bf1d: lavc/iff: ilbm: unbreak decoding on big endian [13:43] <cone-258> ffmpeg.git 03Luca Barbato 07master:5ea5ffc9cee1: doc: support multitable in texi2pod [13:44] <cone-258> ffmpeg.git 03Justin Ruggles 07master:38c1466ca41c: dict: add av_dict_parse_string() [13:44] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:e7e14bc69a60: Merge commit '38c1466ca41c73c7ce347da702362cb69c151716' [13:47] <durandal_1707> does anyone know link of j2k repo that certain dev is working on? [14:23] <cone-258> ffmpeg.git 03Gavriloaie Eugen-Andrei 07master:29b553c1a663: libx264: introduce -x264-params private option [14:23] <cone-258> ffmpeg.git 03Luca Barbato 07master:ded3673d7794: doc: document libx264 options and mappings [14:23] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:d235d240d8ba: Merge commit 'ded3673d77943c376d94e8157b1238bbd1eeca2d' [14:37] <cone-258> ffmpeg.git 03Michael Karcher 07master:0e3afacd4d8f: atrac3: use correct loop variable in add_tonal_components() [14:37] <cone-258> ffmpeg.git 03Diego Biurrun 07master:033a86f9bb6f: x86: h264qpel: Move stray comment to the right spot and clarify it [14:37] <cone-258> ffmpeg.git 03Anton Khirnov 07master:69c25c928464: dnxhdenc: fix invalid reads in dnxhd_mb_var_thread(). [14:37] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:446d62f0cfea: Merge commit '69c25c9284645cf5189af2ede42d6f53828f3b45' [14:43] <cone-258> ffmpeg.git 03Anton Khirnov 07master:6837bd6e49d5: txd: return meaningful error codes. [14:43] <cone-258> ffmpeg.git 03Anton Khirnov 07master:0859eaa0122b: cyuv: return meaningful error codes. [14:43] <cone-258> ffmpeg.git 03Anton Khirnov 07master:ade402804a0e: eatgv: return meaningful error codes. [14:43] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:3a9f48f0337d: Merge commit 'ade402804a0e811cd00862c90559121a793054a6' [14:51] <saste> uhm... multitable HTML rendering sucks [14:58] <cone-258> ffmpeg.git 03Anton Khirnov 07master:4b7598e2fe07: eatgv: cosmetics, reformat [14:58] <cone-258> ffmpeg.git 03Anton Khirnov 07master:f337c29017b1: eatgq: return meaningful error codes. [14:58] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:325ee4ed7ad2: Merge commit 'f337c29017b10c98ccb4dce20efced4c74b665f6' [15:07] <cone-258> ffmpeg.git 03Paul B Mahol 07master:27cadb9ce3f0: remove av_strnstr from Changelog [15:11] <cone-258> ffmpeg.git 03Paul B Mahol 07release/1.1:c2d2bf1d6bf1: lavc/iff: ilbm: unbreak decoding on big endian [15:16] <cone-258> ffmpeg.git 03Stefano Sabatini 07master:c0c06c1bba80: doc/texi2pod: fix warnings introduced in e7e14bc69a606a6bec82efef729263cd38f122d4 [15:23] <cone-258> ffmpeg.git 03Anton Khirnov 07master:adf0110d878d: eatgq: cosmetics, reformat. [15:23] <cone-258> ffmpeg.git 03Anton Khirnov 07master:edb2426b75a1: dxa: return meaningful error codes. [15:23] <cone-258> ffmpeg.git 03Anton Khirnov 07master:9221c0af772e: pngdec: cosmetics, reformat. [15:23] <cone-258> ffmpeg.git 03Anton Khirnov 07master:04e12496094a: iff: drop ff_ prefix from a static function. [15:23] <cone-258> ffmpeg.git 03Anton Khirnov 07master:a0cabd0a2758: mimic: cosmetics, reformat [15:23] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:8ab97a60ef94: Merge commit 'a0cabd0a27587525e90a44660c795d40d2f44fe2' [15:28] <someone-noone> Hello! I'm developing own container-format and have problems while decoding h264(other codecs work fine). When I try to decode my file I always receiving "no frame!" error. Looks like I doesn't provide some options for h264. Which one? [15:33] <Compn> someone-noone : check how nut or libnut contains h264 ? [15:33] <Compn> maybe ti will give you hint on the missing parser or other missing thing [15:33] <someone-noone> Compn, thanks [15:33] <Compn> since nut is a simple container. maybe it will be easy [15:33] <Compn> :) [15:34] <someone-noone> Compn, thanks.. Because, I was starting from avi... [15:34] <Compn> haha [15:34] <Compn> not supposed to put h264 in avi [15:35] <someone-noone> anyway I'm new in "containers" world. So I didn't know from where to start. :) [15:37] <Compn> why making a new container ? [15:38] <Compn> why not just use nut? its about as good as a new container, since very little supports it :) [15:40] <someone-noone> Compn, I need some specific container that can be dividable by fixed-size chunks (1300 bytes) [15:41] <someone-noone> for example: header - chunk1 - chunk2 - chunk3 [15:41] <durandal_1707> why 1300 bytes? [15:41] <someone-noone> and this container can be played in next situation: header - chunk2 - chunk3 [15:41] <someone-noone> 1300 minimum MTU [15:42] <someone-noone> not minimum, but nearly all has value greater than 1300 bytes [15:42] <iive> someone-noone: why not use mpeg-ts? [15:42] <someone-noone> iive, mpeg-ts can have fixed size chunks? [15:42] <nevcairiel> mpeg-ts is always 188 byte chunks [15:43] <someone-noone> hm [15:43] <nevcairiel> you can use 7, then you have 1316 [15:44] <someone-noone> nevcairiel, and where header is stored? In the beginning or in each chunk? How can I pass such stream to player (decoder) [15:45] <nevcairiel> there is a small mpeg-ts header in front of every chunk, and there is also pes headers inside the data, and data packets can also contain header-like tables, like pmt/pat/sdt, etc [15:46] <someone-noone> nevcairiel, so actually, I can play starting from any chunk, right? [15:46] <someone-noone> start playing * [15:46] <someone-noone> sorry for english :) [15:46] <nevcairiel> not any chunk, but it usually doesnt take long to find one where you can get all info you need [15:46] <cone-258> ffmpeg.git 03Anton Khirnov 07master:b965cb906bed: mimic: return meaningful error codes. [15:46] <cone-258> ffmpeg.git 03Anton Khirnov 07master:0ce033f88852: rawdec: cosmetics, reformat [15:46] <cone-258> ffmpeg.git 03Anton Khirnov 07master:7bcaeb408e3e: mjpegdec: fix indentation [15:46] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:8380fc88845e: Merge commit '7bcaeb408e3eb2d2f37a306009fa7fe7eb0f1d79' [15:47] <someone-noone> nevcairiel, thanks [15:47] <iive> mpeg-ts is used for DVB-S/C/T and the other non-eu standards for digital TV [15:47] <iive> they also include the usage of h264 for hdtv. [15:48] <someone-noone> hm& that makes reason for me [15:48] <iive> it is usually encapsulation format, where the video/audio is carried by mpeg-ps or mpeg-es [15:50] <someone-noone> anyway can't repack mp4 into mpeg-ts [15:50] <someone-noone> Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [15:51] <someone-noone> ./ffmpeg -i test.mp4 -vcodec copy -acodec copy -bsf h264_mp4toannexb test.ts [15:51] <cone-258> ffmpeg.git 03Anton Khirnov 07master:3f47d316cda9: mpegvideo_enc: do not modify the input frame. [15:51] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:c46943ec19df: Merge commit '3f47d316cda9037024ffbc76f789332e62b729bc' [16:37] <durandal_1707> doesnt this iff nonsense can be much smaller if one stop using single while{} instead of multiple for loops [18:40] <cone-258> ffmpeg.git 03Anton Khirnov 07master:04f4dbc2fa7c: mpegvideo_enc: fix indentation in load_input_picture() [18:40] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:5cb0782b952f: Merge remote-tracking branch 'qatar/master' [18:55] <Daemon404> i really wish i didnt keep finding functional changes introduced in merge commits [18:55] <Daemon404> its damn enar impossible to figure out the intent [18:55] <Daemon404> near* [18:57] <beastd> Seems michaelni is not hear at the moment. [18:58] <beastd> Daemon404: It can get hairy with conflicts. Besides conflict resolutions I agree it would be better to commit the merge. and then a set of patches and then push all together. [18:59] <Daemon404> beastd, these are not conflict fixe [18:59] <Daemon404> stheyre straight up functional changes [18:59] <Daemon404> for bug fixes [18:59] <Daemon404> there's no excuse for that. [18:59] <beastd> well there is one. bisectabilitly [18:59] <beastd> bisectability [19:00] <Daemon404> it's just as bisectable if you push a post-merge bug fix [19:00] <Daemon404> with a proper message [19:00] <beastd> Daemon404: a bug could go on and off and that could be hit when bisecting [19:01] <beastd> Daemon404: But I am in favor to do such things in separate commits because of communication and documentation benefits [19:04] <beastd> One could also opt for writing long merge commit messages, but that is currently not a strength of michaelni. [19:05] <beastd> So I still favor merge first, do one or more commits on top if needed. Then push out all in one go. [19:06] <beastd> We should tell michaelni though. As he does virtually all the merges it is not very productive to discuss without him- [19:28] <durandal_1707> why are refcounted buffers unaligned? [19:50] <nevcairiel> they are only unaligned if you use realloc, because many systems dont have an aligned realloc [19:51] <Daemon404> you could use je_reallocm or w/e [19:51] <Daemon404> :P [19:51] <nevcairiel> windows has an aligned realloc, so see if i care! [19:52] <Daemon404> lol [19:52] <Daemon404> durandal_1707, is on freebsd though [19:52] <Daemon404> and iric freensd's *alloc funcs are idneed je [19:53] <Daemon404> though you might need an m suffix [20:44] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:4e585f63257b: sws: use planarRgbToRgbWrapper only for 8bit per component [20:44] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:3c2ecfcc2476: sws: dont enable chrSrcHSubSample for planar RGB [20:44] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:02001a372de6: sws: add planar RGB formats to isAnyRGB [21:41] <beastd> oi michaelni [21:41] <beastd> Daemon404 complained about merge commits being hard to follow [21:41] <Daemon404> no i didnt [21:41] <Daemon404> i complained about adding functional changes and bgu commits in merge commits [21:43] <beastd> Daemon404: was about to complete that but didn't want to paste the whole discussion [21:43] <nevcairiel> some merge commits are impossible to read anyway, i never really know if some diff it shows is just because of a 3-way merge it performed due to differences between the versions, or if some manual changes went in <.< [21:43] <beastd> michaelni: maybe you can find it in channel log [21:44] <beastd> was about 3h ago [21:44] <michaelni> 3h ago my logs are full of " Disconnected (Connection reset by peer)." [21:45] <beastd> flood :) [21:46] <beastd> [18:55:26] <@Daemon404> i really wish i didnt keep finding functional changes introduced in merge commits [21:46] <beastd> [18:55:37] <@Daemon404> its damn enar impossible to figure out the intent [21:46] <beastd> [18:55:40] <@Daemon404> near* [21:46] <beastd> [18:57:52] <@beastd> Seems michaelni is not hear at the moment. [21:46] <beastd> [18:58:49] <@beastd> Daemon404: It can get hairy with conflicts. Besides conflict resolutions I agree it would be better to commit the merge. and then a set of patches and then push all together. [21:46] <beastd> [18:59:07] <@Daemon404> beastd, these are not conflict fixe [21:46] <beastd> [18:59:14] <@Daemon404> stheyre straight up functional changes [21:46] <beastd> [18:59:16] <@Daemon404> for bug fixes [21:46] <beastd> [18:59:21] <@Daemon404> there's no excuse for that. [21:46] <beastd> [18:59:34] <@beastd> well there is one. bisectabilitly [21:46] <beastd> [18:59:42] <@beastd> bisectability [21:46] <beastd> [19:00:01] <@Daemon404> it's just as bisectable if you push a post-merge bug fix [21:46] <beastd> [19:00:12] <@Daemon404> with a proper message [21:46] <beastd> [19:00:48] <@beastd> Daemon404: a bug could go on and off and that could be hit when bisecting [21:46] <beastd> [19:01:37] <@beastd> Daemon404: But I am in favor to do such things in separate commits because of communication and documentation benefits [21:46] <beastd> [19:04:11] <@beastd> One could also opt for writing long merge commit messages, but that is currently not a strength of michaelni. [21:46] <beastd> [19:05:17] <@beastd> So I still favor merge first, do one or more commits on top if needed. Then push out all in one go. [22:44] <durandal_1707> michaelni: the 4e585f63257bfacb4a9d is funny as prev line too [22:46] <durandal_1707> prior you added other gbrp to isAnyRGB... [22:47] <durandal_1707> also such checks like !=,==,&& .. can be simplified [22:50] <durandal_1707> Daemon404: you sure you did not miss some part of code in your patch? [22:50] <Daemon404> what patch [22:51] <Daemon404> i havent resubmitted my patch form months ago yet [22:52] <durandal_1707> michaelni: so no way to fix red warning? [23:23] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:f0d3a031150f: sws: include isRGB in isAnyRGB() so that future RGB formats wont be missed again [23:23] <cone-258> ffmpeg.git 03Michael Niedermayer 07master:6512405ce203: sws: disable yuv2rgb warning for planar rgb. [23:30] <durandal_1707> michaelni: that isAnyRGB is still weird [23:30] <durandal_1707> now it list bitstream rgb and pseudopal crap..... [23:38] <wm4> what is pseudo-pal at all? [23:40] <durandal_1707> wm4: pixel format that use pallete but do not need too ... [23:42] <wm4> so does a picture have a palette or not? [23:43] <durandal_1707> it have pseudo palette [23:43] <Daemon404> and wtf is a pseudo palette [23:44] <Daemon404> sure is ambiguous terms [23:44] <wm4> a "figure it our yourself moron" style of API [23:45] <durandal_1707> it is toy, why do you still play with toys? [23:45] <beastd> wm4. Daemon404: the pal is not stored in the file but generated by the decoder [23:45] <durandal_1707> swscale actually [23:46] <wm4> is that by libswscale can output AV_PIX_FMT_BGR4, but not take it as input? [23:48] <durandal_1707> only for bitstream variant of bgr4 [23:49] <durandal_1707> byte one have IO [23:49] <durandal_1707> wm4: you have nothing better to do? [23:49] <wm4> no [00:00] --- Sun Jan 27 2013

1 0

ffmpeg.log.20130126
by burek 27 Jan '13

27 Jan '13

[00:49] <RamajanSan> hello [00:49] <RamajanSan> Hello, I want to normalize audo in a *.mov file, how can i do this? [00:49] <RamajanSan> with ffmpeg? [00:50] <RamajanSan> or maybe you guys use a different tool? [00:58] <s2soul> hello does anyone know what is wrong with: ffmpeg -i original.MXF -vcodec mjpeg -qscale 1 -acodec pcm_s16le intermediate.avi [00:59] <s2soul> i get an error when trying to convert ac3 to pcm [00:59] <s2soul> it works though if i do: ffmpeg -i original.MXF -vcodec mjpeg -qscale 1 -acopy intermediate.avi [01:00] <s2soul> hope i got that right i am not in the OS where i do my video work right now... [01:01] <sacarasc> Paste the output to a pastebin type site. [01:01] <s2soul> hello....in that case i may have to reboot... but in passing...do you know anything about mencoder [01:02] <s2soul> are they very different in terms of quality? mencoder -vf harddup -demuxer lavf -oac pcm -ovc lavc -lavcopts vcodec=mjpeg:vhq:vbitrate=6000 -noskip -mc 0 -ofps 29.97 -o intermediate.avi original.mts [01:04] <s2soul> see those are the 2 recommended recipies for encoding mts to HQ mjpeg that i will need after working in Cinelerra using proxy only i thought the resulting file encoded with mencoder was low in mb.... [01:04] <s2soul> so i though stick with ffmpeg.... [01:04] <klaxa> both are using libavformat and libavcodec i don't think they should differ much [01:04] <s2soul> mencoder and ffmpeg both? [01:05] <klaxa> yeah [01:06] <s2soul> oh thanks well ok i might stick with what works right now but i was trying to adapt the formula set for fps 25 to my own footage 29.97 or 59.97 but i cant find how much to increase the bitrate! [01:06] <buhman> I'm wanting to capture audio output [01:06] <s2soul> vbitrate=6000 i changed to 8000 [01:07] <buhman> ffmpeg -f alsa -i hw:0,0 seems to work, but doesn't actually capture anything [01:07] <s2soul> but resulting file was exactly same mb! klaxa is that normal? [01:07] <klaxa> um... doesn't sound right [01:08] <s2soul> so i then could use ffmpeg recipe then if i can get to convert to pcm from ac3! [01:08] <s2soul> could my ffmpeg be too old? ubuntu 10.04 not that old dont know [01:09] <JEEBsv> that would be /quite/ old [01:09] <JEEBsv> in ffmpeg terms [01:09] <JEEBsv> (I consider everything older than fall 2011 old) [01:12] <s2soul> well i guess ac3 isnt so bad then as the audio for hq files i will convert my files also to dnXhd proxies for edititing in cinelerra [01:13] <buhman> anything older than 1.1 is ancient [01:13] <buhman> 1.1 is old [01:13] <s2soul> so i could just do ffmpeg -i original.MXF -vcodec mjpeg -qscale 1 -acopy intermediate.avi [01:17] <s2soul> uuum in some hours from now i will be back in said OS and i can paste the output of my issue just in case someone has an answer... [01:19] <s2soul> BTW does the latest version of ffmpeg have more DNxHD Supported Resolutions : 1080p / 29.7 DNxHD 45 1920 x 1080 8 29.97 45M [01:20] <s2soul> but there is no bitrate for 1080p 59.97 fps [01:20] <s2soul> nor is there 720p 29.97 [01:20] <s2soul> 720p / 59.94 DNxHD 220 1280x720 8 59.94 220M [01:21] <s2soul> its hard to guesstimate what bitrate to use when there is such a variation from 45M to 220M [01:27] <s2soul> ok one last question if my ffmpeg is so old i have the option of installing it on a latest version of Fedora (a remix) or ubuntu 12.04 with classic gnome desktop which is simplest and most effect distro to compile or install ffmpeg on?? [01:28] <burek> s2soul you need to spend more time reading the docs [01:28] <s2soul> this is on a quadcore amd laptop fairly recent [01:28] <llogan> either will be fine [01:28] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide [01:29] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide [01:30] <s2soul> thanks i saw the ubuntu compilation guide it looked time consuming and this project is laaate but then i could at least do something while the main computer is needed for otherwork [01:30] <llogan> it will take 20 minutes or less [01:30] <s2soul> ok well is it easier maybe on fedora (kororaa linux) [01:30] <s2soul> oh ok [01:31] <llogan> and you don't need to install all of the suggested dependencies depending on what you need [01:31] <s2soul> lol im slow thanks ill stick to barebones for now well i just want : ffmpeg -i original.MXF -vcodec mjpeg -qscale 1 -acodec pcm_s16le intermediate.avi to work [01:32] <s2soul> i just want my audio to convert succesfully as well as video [01:32] <llogan> without the console output we can only guess as to what the issue may be [01:34] <s2soul> ok you are right guys ok look if anyone is around later ill do that my partner now needs this computer for ardour audio work still waiting on me ok later! [01:34] <llogan> use ssh and you can both use it [01:50] <s2soul> ok back for only a couple of minutes [01:50] <s2soul> is anyone still there who wants to look at my pastebin output: http://pastebin.com/1vQtNSSV [01:51] <s2soul> i realise that of course i had ffmpeg installed here as well but i may not be able to use it while there is recording going on with firewire etc takes too much ressources [01:52] <s2soul> now i get audio! working in this version of ffmpeg but [01:52] <s2soul> the file converted from .mts 29.97 to 59.97 [01:53] <s2soul> i think its better for my project to have it 1080p but 29.97fps (even though some mts files were shot at 60fps) [01:55] <s2soul> ffmpeg -i original.MTS -vcodec mjpeg -qscale 1 -acodec pcm_s16le intermediate.avi [01:57] <s2soul> test file was 2.8 mb now 16.2 MB and fps is 60 rather then 30 ?? I need 30fps [01:57] <s2soul> http://pastebin.com/1vQtNSSV [01:58] <s2soul> ok ill stay on and check back from time to time [01:59] <llogan> s2soul: you're not using ffmpeg from FFmpeg [02:00] <llogan> use real ffmpeg or you can perhaps get help at #libav if you want to use that stuff [02:02] <s2soul> im back what is not real ffmpeg? [02:02] <s2soul> i dont understand it came with ubuntu studio 10.04 [02:04] <s2soul> what is problem with fork ? [02:05] <llogan> read the link [02:05] <s2soul> so i am using a libav fork? [02:05] <llogan> yes [02:05] <llogan> FFmpeg version 0.6.2-4:0.6.2-1ubuntu1.1~ppa1~lucid1, Copyright (c) 2000-2010 the Libav developers [02:06] <llogan> yes, it's a confusing mess. [02:06] <s2soul> oh no pls. tell me the formula i use is adequate for project it is not imperative to be perfect or top notch just acceptable HQ files for this time [02:06] <s2soul> i mean what i used now [02:06] <llogan> ask in #libav [02:06] <s2soul> oh [02:07] <s2soul> well if a file plays good and if i add the fps parameters i should be good ...? but ok sigh! later I will find help on libav irc [02:07] <s2soul> thanks anyways sigh! [02:09] <llogan> i can't give you an answer because i don't use libav [03:10] <lemonjelly> Hello. I'm writing/adapting a ffmpeg muxer inside libavformat. I'd like to get an estimated duration of the input but the 'duration' field of AVFormatContext is always set to 0, and I can't see any other useful fields. [03:12] <lemonjelly> I can see the ffmpeg command line estimates it using the input AVFormatContext (inside utils.c) but I can't get a reference to this from the output AVFormatContext [03:12] <lemonjelly> Is there any way to do this? Or perhaps an existing muxer that accomplishes this? (I haven't found any) [03:55] <ulatekh> @michaelni: I reproduced the problem I was having (i.e. lossless codecs don't preserve interlacing) with the latest git version of ffmpeg and submitted a bug report. Hopefully something happens. [04:11] <ulatekh> @michaelni: If you're around...I saw that you're available for ffmpeg consulting. There's a feature I've always wanted in ffmpeg, and if the cost isn't insane, I'd pay to have someone put it in there. [04:12] <michaelni> ulatekh, what feature ? [04:14] <ulatekh> Support for 3:2 pulldown, i.e. encoding a 24000:1001 fps source so that it plays at 30000:1001 fps, i.e. so that film can be put onto an NTSC DVD. I'd want to be able to transcode existing 3:2 pulldown videos to different bitrates, as well as transcoding existing 24000:1001 fps videos to 3:2 pulldown. I think this would be MPEG-1 and MPEG-2 codecs only. [04:14] <ulatekh> I don't think it's difficult, it just requires someone far more familiar with MPEG-1/2 than me. [04:25] <michaelni> ulatekh, it should be doable if iam not missing some issues [04:25] <ulatekh> I was trying to think of all the different use cases...those were the only 2 I could think of. [04:26] <ulatekh> Strangely, ffmpeg does a great job of converting 30000:1001 fps 3:2 pulldown video to 24000:1001. Go figure. [04:27] <michaelni> it should also do the opposit already but it does dumb frame duplication prior to encoding [04:27] <ulatekh> Yeah...3:2 pulldown would be smart frame duplication. :-) [04:28] <michaelni> field "lengthening" in the mpeg2 encoder [04:36] <hotwings> so the current ffmpeg git is giving me lots of video artifacts with mpeg2 video (i didnt test h264 however). i use vdpau btw. i was told this is a known problem in ffmpeg though, anyone know if thats the case? [04:42] <michaelni> hotwings, you use ffmpeg + mplayer or something else for vdpau [04:42] <michaelni> note though i have no VDPAU hardware so its unlikely i will be of much help [04:42] <hotwings> ffmpeg + mplayer2 [04:43] <hotwings> however, i wasnt using mplayer2 at the time.. i was watching live dvb-s tv using VDR and the softhddevice output plugin [04:43] <hotwings> i only use mplayer2 (via the mplayer VDR plugin) when playing media/not watching live tv [04:44] <michaelni> the vdpau problems probably began with remis patches [04:45] <hotwings> who/what is remis? what did his/her/its patches attempt to fix, assuming fixing something was the purpose? [04:48] <michaelni> moving stuff to hwaccel infrastructure [04:49] <michaelni> maybe you could ask him if he has time/interrest to look into the bug (his nick is courmisch i belive and he is on #videolan) [04:50] <hotwings> hopefully he'll find time/interest to fix it since he had time/interest to break it :\ [04:51] <hotwings> thanks, ill try him there [04:53] <hotwings> this reminds me of all the patches that were merged in v4l without proper (and in many cases, any!) testing.. [04:54] <michaelni> v4l2 ? [04:55] <hotwings> multimedia drivers for linux [04:57] <hotwings> for some insane reason, patches by people who dont even have the hardware to the drivers theyre "fixing" or screwing around with, get merged.. and of course break stuff.. then have to be reverted [04:57] <Youka> Is there already a solution to the c99wrap cl "C compiler test failed" problem of ffmpeg msvc configuration? i followed the instructions from the official ffmpeg documentations and are using msvc10 ex [04:57] <hotwings> why patches arent always tested _before_ theyre merged is beyond me [04:58] <Youka> google for it brings up more questions instead of answers [04:58] <ulatekh> @hotwings: I feel your pain...ffmpeg would probably win an award for the biggest unfulfilled potential, if there was such an award LOL [05:01] <michaelni> Youka, we have a msvc or more than one boxes at http://fate.ffmpeg.org/ which works, maybe looking at how its configured will help [05:04] <Youka> thx, i'll have a look [05:05] <michaelni> also look at config.log (its generated when you run configure) it might contain hints why things fail [05:10] <Youka> it was the linker :S [05:11] <Youka> cmd->msys sets preference to msys always, so i had to prepend msvc linker to PATH [12:34] <Orphis> While trying to compile ffmpeg for Windows using MSVC (and c99wrap) I get this error when running make: /c/somepath/ffmpeg/common.mak:138: *** missing separator. Stop. [12:34] <Orphis> Configure was a plain simple: configure --toolchain=msvc [12:34] <Orphis> Any idea what could be wrong? [12:35] <divVerent> Orphis: a stupid idea: tried using "the right make program", whatever it is? [12:35] <divVerent> i.e. if you were using nmake, try GNU make, and vice versa [12:36] <divVerent> or read the docs on which make to use :P [12:36] <Orphis> I'm using mingw make 3.81 [12:36] <divVerent> I don't have windows [12:36] <Orphis> Well, I don't think our Makefile is compatible with nmake :P [12:36] <divVerent> I only can tell you that this error means the "make" tool you used doesn't support the makefile [12:36] <divVerent> it typically comes when using bsdmake on a GNU makefile or vice versa ;) [12:36] <divVerent> so I'd suggest trying "some" other make tool [12:37] <Orphis> 3.81 should be working [12:37] <divVerent> is mingw make BTW a gnu or a bsd make? [12:37] <Orphis> I'm thinking of a crlf issue maybe [12:37] <Orphis> gnu [12:38] <divVerent> and this is git head? [12:38] <Orphis> With support for MinGW and Windows path [12:38] <Orphis> git head from a few days ago [12:38] <Orphis> I compiled it properly with a bsd make and another gnu make [12:38] <Orphis> But on Unix systems, so support is easier and compilation straightforward [12:38] <divVerent> 138 $(eval $(RULES)) [12:39] <Orphis> Yes [12:39] <Orphis> I tried a make -d to see what it would be doing, but it doesn't show much [12:39] <divVerent> no idea what that construct is for [12:39] <divVerent> i.e. why one can't just remove the lines 131, 136 and 138 [12:43] <Orphis> divVerent: It was a crlf problem, I've converted all *mak and Makefile files to unix format and it works [12:44] <Orphis> I think git converted them automatically to crlf when doing a clone [12:47] <JEEB> Orphis, https://help.github.com/articles/dealing-with-line-endings#platform-windows [12:47] <JEEB> set this to false [12:47] <Orphis> JEEB: I know, I will do it later [12:47] <JEEB> and then see the two first commands of re-normalizing endlines [12:47] <JEEB> :) [12:52] <Fjorgynn> Man what am I watching? [12:55] <Orphis> Wow. Compiling for Windows is über slow with the c99wrap :-/ [12:56] <Orphis> Not totally unexpected actually... [12:58] <Orphis> The setup for compiling ffmpeg is way too complicated and too long for the other devs working on the project, I guess I'll make a slimer build without all the features I don't need and distribute that with instructions on how to rebuild it [13:41] <divVerent> 12:43:33 Orphis | divVerent: It was a crlf problem, I've converted all *mak and Makefile files to unix format and it works [13:41] <divVerent> Orphis: looks like you could report this to the mingw-make guys [13:41] <divVerent> as it did support CRLF in the other lines [13:41] <divVerent> only not in this macro [13:55] <Orphis> What are the differences between all the mp3 decodes? [13:55] <Orphis> decoders* [13:56] <Orphis> mp3 and mp3float it's obvious. But what about mp3adu and mp3on4 ? [14:00] <durandal_1707> see ffmpeg -codecs output? [15:08] <Fjorgynn> e [15:31] <Orphis> Any particular reason why ffmpeg is compiled with /MT on Windows? [15:32] <someone-noone> because ffmpeg supports multi-thread de\en-coding!? [15:33] <Orphis> someone-noone: Well, MD is multithreaded too [15:33] <Orphis> MD is DLL runtime, MT is static runtime, both are multithreaded [15:35] <Mavrik> hmm, is compiling with MSVC even supported by ffmpeg? [15:35] <Orphis> Mavrik: It is, how surprising can that be [15:36] <Orphis> There's a tool that converts c99 code into code MSVC understands [15:36] <Mavrik> hmm, any reason to do it though? [15:36] <Mavrik> most people just compile windows builds with mingw [15:37] <Orphis> Mavrik: I want to statically link with ffmpeg [15:38] <Mavrik> ah [15:38] <Orphis> My software is only one binary, don't want to add too many files next to it for simplicity [15:38] <Mavrik> that's probably also the reason why ffmpeg is linked with static runtime :) [15:39] <Orphis> Meh :P I want static ffmpeg but dynamic runtime [15:43] <Orphis> Ah! "Command line compile will default to /MT " [15:43] <Orphis> I should be able to pass an extra cflags for that [15:54] <Plorkyeran> --extra-cflags=-MD does in fact work fine for me [15:54] <someone-noone> ./ffmpeg -i test.mp4 -vcodec copy -acodec copy test.ts [15:55] <someone-noone> Error: H.264 bitstream malformed, no startcode found, use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb) [15:55] <JEEB> yes [15:55] <JEEB> add -bsf h264_mp4toannexb after the -i [15:55] <someone-noone> ./ffmpeg -i test.mp4 -vcodec copy -acodec copy -bsf h264_mp4toannexb test.ts [15:55] <someone-noone> Error: Failed to open bitstream filter h264_mp4toannexb for stream 0 with codec copy: Invalid argument [15:55] <JEEB> :D [15:56] <someone-noone> after -i ? [15:56] <someone-noone> ok [15:56] <JEEB> that shouldn't matter tho as that one was after -i as well... [15:56] <JEEB> fun [15:56] <JEEB> so it's not copying the stream if you're doing a bsf :s [15:56] <Mavrik> actually [15:56] <Mavrik> as I remember [15:57] <Mavrik> the h264_mp4toannexb relies on grabbing PSS/SPS packets from x264 encoder data [15:57] <Mavrik> so it can't work with copy [15:57] <Mavrik> (talking from memory from source) [15:57] <someone-noone> Mavrik, so I need to compile with libx264? [15:57] <Mavrik> http://ffmpeg.org/doxygen/trunk/h264__mp4toannexb__bsf_8c_source.html#l00034 [15:57] <JEEB> no, it means it's borked [15:57] <Mavrik> it grabs the headers from extradata :\ [15:58] <someone-noone> So, how can I solve it? Reencode with x264? [16:01] <Mavrik> that would probably work yea [16:18] <Orphis> Alright, I'm down to 3 missing symbols when linking now: http://pastie.org/private/yfd483zrwvaiihnhqid9a [16:19] <Orphis> __imp__avpriv_snprintf, __imp__avpriv_vsnprintf and __imp__avpriv_strtod [16:19] <Orphis> From what I've found, they are build in compat/strtod.o compat/msvcrt/snprintf.o [16:20] <JEEB> yeah [16:20] <Orphis> Any reason why they aren't in any av*.a library? [16:22] <JEEB> seems like there's a ticket for that with -MD [16:22] <JEEB> https://ffmpeg.org/trac/ffmpeg/ticket/2049 [16:23] <Plorkyeran> oh, right [16:23] <Plorkyeran> I forceinclude a dumb workaround for that [16:24] <JEEB> lol [16:24] <JEEB> reminds me of my dumb workaround for mingw-w64 :P [16:24] <Plorkyeran> https://github.com/Aegisub/Aegisub/blob/master/aegisub/build/ffmpeg/dynamic… [16:25] <Plorkyeran> https://github.com/Aegisub/Aegisub/blob/master/aegisub/build/ffmpeg/ffmpeg.… has the full set of configure settings that works4me [16:26] <_raven> hi [16:26] <_raven> how to export every 5th frame of a video as jpeg image? [16:26] <Orphis> Plorkyeran: Thanks! [16:26] <Orphis> I'll have a look [16:32] <someone-noone> hello! trying to build ffmpeg with libx264 (both latest versions from git) and getting next link error: [16:32] <someone-noone> Undefined symbols for architecture x86_64: [16:32] <someone-noone> "_x264_encoder_open_129", referenced from: [16:32] <someone-noone> _X264_init in libavcodec.a(libx264.o) [16:33] <someone-noone> libx264 was configured with --enable-shared [16:33] <someone-noone> otherwise ffmpeg can't find static library [16:33] <someone-noone> where is a problem? [16:33] <JEEB> uhh [16:34] <JEEB> static library not being found sounds like a problem [16:34] <JEEB> but first of all, are you sure you have built both ffmpeg and libx264 with the same bitness? [16:35] <someone-noone> what does bitness mean? [16:44] <someone-noone> hm, found mistake& now ffmpeg sees static library of x264. But anyway linker error... [16:44] <someone-noone> LD ffplay_g [16:44] <someone-noone> Undefined symbols for architecture x86_64: [16:44] <someone-noone> "_x264_encoder_open_129", referenced from: [16:44] <someone-noone> _X264_init in libavcodec.a(libx264.o) [16:46] <someone-noone> btw, x264 --version prints next: [16:46] <someone-noone> x264 0.119.x [16:47] <JEEB> ..., sounds like you didn't build the x264 libraries [16:47] <JEEB> or your compiler isn't looking [16:47] <JEEB> or your x264cli is completely separate [16:47] <someone-noone> ls -l /usr/lib | grep x264 [16:48] <someone-noone> libx264.a [17:25] <juanmabc> local/lib? [18:08] <someone-noone> Hello! I have some mpeg-ts encoded(using libx264) from original mp4 file. In mpeg-ts I've skipped first 7*188 bytes. This video plays okay, but for the first ~5 seconds my video has less then 24fps speed. [18:09] <someone-noone> I'm receiving next error (ffplay): [18:09] <someone-noone> Context scratch buffers could not be allocated due to unknown size. [18:09] <someone-noone> [h264 @ 0x7fc47c089800] non-existing PPS 0 referenced [18:09] <someone-noone> Looks like I need some additional option to make PPS value appear more often in mpeg-ts chunks. How can I do it? [18:46] <mm3> hi, how can I send a rtmp stream without ffserver? [18:51] <someone-noone> mm3, you can do it with ffmprg [18:51] <someone-noone> ffmpeg [18:52] <someone-noone> mm3, something like this: ffmpeg -i rtmp://sourcehost rtmp://targethost [18:53] <someone-noone> you can send files with flv format (-f flv) [18:57] <mm3> someone-noone, I mean can ffmpeg listen to a rtmp and send a stream on request? [18:57] <mm3> open a rtmp port [18:58] <someone-noone> mm3, no you can't do that. You can just pull\push from\to existing rtmp server. [18:58] <someone-noone> mm3, you nead real flash server for that [18:58] <someone-noone> mm3, look at crtmpserver, it's free and do what you want [18:59] <someone-noone> ffmpeg is just tool for en-\de-coding, not server :) [19:00] <mm3> I don't need a server, just to provide 1 stream for 1 user [19:00] <someone-noone> mm3, anyway you need server for this [19:00] <mm3> yes, thank you someone-noone [19:02] <someone-noone> mm3, what is your source? (I mean format and file or stream) [19:05] <mm3> I guess mp4 [19:07] <someone-noone> mm3, is it a file or stream? [19:07] <mm3> I'd like to stream something, still didn't decide what [19:10] <mm3> someone-noone, do you think it's hard to write a small rtmp server? [19:11] <someone-noone> mm3, of course. It's very hard. Why not use open-source one? CRtmpServer - is very easy in setup [19:15] <someone-noone> How can I write ffmpeg output to named pipe? [20:02] <someone-noone> Hello! I want to make infinity stream to a pipe using mpeg-ts container. [20:02] <someone-noone> I have following script: [20:03] <someone-noone> while true [20:03] <someone-noone> do [20:03] <someone-noone> ./ffmpeg -re -i test.ts -vcodec copy -acodec copy -f mpegts - > test.pipe [20:03] <someone-noone> done [20:03] <someone-noone> then I try to play that file with: ./ffplay test.pipe [20:03] <someone-noone> And when second copy of ts-file is started to wrtiting, ffplay gets next error: [20:03] <someone-noone> [mpegts @ 0x7f8769054a00] PES packet size mismatchs [20:04] <someone-noone> can I fix it somehow? [20:15] <mm3> someone-noone, why not remove "- >" in your cmd? [20:15] <mm3> and mkfifo test.pipe [20:16] <someone-noone> mm3, I need uninterruptable input for ffplay [20:21] <beastd> someone-noone: you did create test.pipe with mkfifo , right? [20:23] <someone-noone> beastd, yes [20:27] <beastd> seems to work for me [20:28] <someone-noone> beastd, yes it works. But only for the first run [20:29] <someone-noone> on second run ffplay got errors in mpegts and h264 [20:29] <someone-noone> and I'm trying to make something like "infinity stream from file" [20:29] <beastd> someone-noone: multiple runs work here [20:30] <someone-noone> beastd, you're making something wrong. Do you run in loop? [20:30] <someone-noone> try this: [20:30] <someone-noone> while true [20:30] <someone-noone> do [20:30] <someone-noone> ./ffmpeg -re -i test.ts -vcodec copy -acodec copy -f mpegts - > test.pipe [20:30] <someone-noone> done [20:30] <someone-noone> and then ./ffplay test.pipe [20:30] <someone-noone> and wait when first video will end [20:32] <beastd> someone-noone: that is waht worked for me. i even used "./ffplay -autoexit test.pipe" [20:33] <someone-noone> beastd, and your video file is looping in ffplay, yes? [20:33] <someone-noone> does your video file has h264 codec? [20:33] <beastd> no [20:34] <beastd> will try h264 for fun now [20:37] <beastd> works too [20:39] <beastd> when your version with > shell redirection works correctly you could also just use -y as a global parameter and specify the file as output file as test.pipe directly as mm3 mentioned [20:41] <someone-noone> beastd, ok. Have you tried h264? :) Btw are you receiving: "PES packet size mismatch" warning ? [20:42] <beastd> [20:34:09] <beastd> will try h264 for fun now [20:42] <beastd> [20:37:52] <beastd> works too [20:42] <someone-noone> hm, beastd. Can you send your ts file to me? [20:43] Action: beastd does not get any messages from ffplay except analyze duration / bitrate estimation ones [20:45] <beastd> someone-noone: you can create is yourself: ffmpeg -f lavfi -i rgbtestsrc -t 30 -f mpegts -c:v h264 ./frag.ts [20:46] <someone-noone> beastd, thanks wil try it [20:49] <someone-noone> hm works for me too [20:51] <someone-noone> beastd, may I ask you to test my file? [20:54] <beastd> someone-noone: do you get that warning when you play the file once directly with ffplay? [20:54] <someone-noone> no, but I got anothers: [20:55] <someone-noone> reference picture missing during reorder [20:55] <someone-noone> mmco: unref short failure [20:55] <beastd> what does ffplay say if you you remux to mpegts and try to play the resulting file [20:56] <someone-noone> for the first "play" - no warnings, when second "play" comes, I got following: [20:56] <someone-noone> [mpegts @ 0x7feb0985b200] PES packet size mismatchsq= 0B f=0/0 [20:56] <someone-noone> [h264 @ 0x7feb0a068200] error while decoding MB 2 21, bytestream (-12) [20:59] <beastd> someone-noone: if you don't use the pipe but remux your input file to mpegps. then you play it directly, you get no warning? [20:59] <someone-noone> let me check [21:02] <someone-noone> beastd, no warnings [21:04] <beastd> hmm [21:14] <beastd> someone-noone: what happens if you "cat remuxed.ts remuxed.ts > 2.ts" and try to play that? [21:15] <someone-noone> beastd, one moment [21:19] <someone-noone> beastd, hm it doesn't produce any errors or warnings, but it also doesn't play (black screen) [21:20] <someone-noone> I mean it doesn't play second part [21:32] <someone-noone> beastd, btw ffmpeg -i shows that file duration is ~2 mins, but should be ~4 mins [21:39] <beastd> someone-noone: i think the duration thing is kind of normal. also timestamps will reset while you play it back [21:51] <someone-noone> beastd, I don't know how& but it's working for now [21:51] <someone-noone> beastd, I just reencoded file [21:57] <someone-noone> I know where is problem [21:57] <someone-noone> I encoded file with -g 12 option [21:57] <someone-noone> and it doesn't work [21:58] <someone-noone> but when I removed this option, piping files started to work [21:58] <someone-noone> but anyway I need keyframes&more often... [22:00] <beastd> someone-noone: did you try with different values for -g [22:01] <someone-noone> beastd, no I didn't [22:06] <beastd> someone-noone: maybe try some. i do not have time to look into this further. [22:07] <beastd> but maybe someone else here is more knowledgable than me about the topic [22:15] <Miesco> Should a regular .mp3 work on ios safari <audio> tag? [22:15] <Miesco> Im trying to get my bro to listen to my bands song on his iphone [22:18] <sacarasc> Send me an iPhone and I will test for you! [22:20] <sacarasc> Safari on iOS (including iPad) currently supports uncompressed WAV and AIF audio, MP3 audio, and AAC-LC or HE-AAC audio. HE-AAC is the preferred format. [22:20] <sacarasc> Yay! Google! [22:20] <Miesco> sacarasc: Thats what I read [22:20] <Miesco> sacarasc: Oh I thought it meanted uncompressed mp3 [22:20] <sacarasc> There is no uncompressed MP3. [22:21] <Miesco> like supports concompressed wav,mp3, etc [22:21] <Miesco> sacarasc: Yea thats why I was confused [22:30] <Tronic> Are there FFMPEG (not libav) packages for Ubuntu? [22:34] <JEEB> not sure if there are ppas, but unfortunately the official repos only have libav atm [22:45] <sacarasc> Would ffmpeg and libav be able to co-exist? [22:50] <durandal_1707> sacarasc: they can't eat each other [22:50] <sacarasc> I was just thinking, the libav* libraries might not like each other being installed. [22:56] <durandal_1707> why would you install both ? [22:56] <JEEB> the only problem is making all downstream apps who use the APIs being able to use both [22:56] <JEEB> without making multiple versions of them [22:57] <JEEB> just installing libav-tools or ffmpeg-tools should be simpler, with one of them being statically linked or whatever [22:57] <durandal_1707> API is same [22:57] <JEEB> mostly, yeah [22:58] <beastd> Tronic: PPA is listed here: http://ffmpeg.org/download.html#LinuxBuilds [22:59] <kms_> hello [22:59] <kms_> a have a question [23:00] <durandal_1707> go ahead [23:04] <durandal_1707> kms_: your question? [23:08] <kms_> i need crop video, and AFTER convert it and save to file.. command: avconv -i "filename" -vf "crop=in_w:in_w/16*9" -b:v 512k -aspect 16:9 -deinterlace -s:v 720x405 -ac 2 -aq 0 -y "out.webm" give me wrong result. Outputed video have right resolution but it is ni cropped but stretched horisontal. If i crop video and after by next command resize it - result is right [23:10] <durandal_1707> avconv is banned here [23:10] <kms_> i need crop top and bottom lines and resize video to 16:9 to smallest resolution [23:10] <cbsrobot> lol [23:10] <durandal_1707> actually you are on wrong channel if you ask help for avconv [23:10] <kms_> =O [23:10] <sacarasc> #libav for avconv. [23:12] <beastd> kms_: avconv is not from FFmpeg , ffmpeg from Libav is also not FFmpeg [23:12] <ulatekh> Hello all. Does anyone know how to get a list of pixel formats supported for raw video in QuickTime? The only one that produces accurate results (i.e. preserves interlacing) is rgb24, but that takes up a LOT of disk space. Every other format I've tried doesn't preserve interlacing. [23:13] <kms_> ok [23:13] <beastd> kms_: How to get FFmpeg is explained here http://ffmpeg.org/download.html [23:14] <durandal_1707> ulatekh: yuv420p = R420, yuv411 = R411 [23:15] <durandal_1707> ulatekh: those are from Radius DV for QuickTime [23:15] <durandal_1707> there are also other yuv packed variants [23:16] <durandal_1707> also raw formats DO take LOT of disk space [23:16] <ulatekh> @durandel_1707: I tried yuv420p, but it wouldn't decode...I'll try again to make darn sure. [23:17] <ulatekh> Yeah, I know they do...I'm just going to use them while editing...the final result will be encoded more efficiently. [23:17] <durandal_1707> hmm, it may not be supported by official qt... [23:17] <Tronic> beastd: Thanks. I actually ended up installing the git version because there were packages only for 0.10, not .11 :) [23:17] <ulatekh> Right now the files are in raw YUV format, but kdenlive doesn't support imported clips of that format. [23:17] <beastd> Tronic: OK. sometimes the packages might lag a bit behind. git-master is also fine :) Have fun! [23:19] <ulatekh> @durandel_1707: should I be using "-pix_fmt r420" and "-pix_fmt r411" in my "ffmpeg111 -i [input.yuv] -vcodec rawvideo -pix_fmt [pixfmt] [output.mov]" command line? [23:19] <ulatekh> And yes, I'm running latest git. [23:20] <durandal_1707> no, r420 is tag... [23:20] <ulatekh> As in "-vtag r420"? [23:20] <ulatekh> "ffmpeg111 -i karaoke-20130119-01-pass1-yuv420p.mov -f yuv4mpegpipe -pix_fmt yuv420p - | head -1" just gave me "[rawvideo @ 0x87bfd30] Pixel format was not specified and cannot be detected" [23:20] <durandal_1707> yes, but that should be set automagically [23:21] <durandal_1707> the -f part does not make sense... [23:21] <ulatekh> "-f yuv4mpegpipe" converts back to raw YUV format. [23:21] <durandal_1707> pastebin full command [23:22] <durandal_1707> ulatekh: i thought you wanted to save yuv into mov.... [23:22] <ulatekh> "ffmpeg111 -i [input.yuv] -vcodec rawvideo -pix_fmt [pixfmt] [output.mov]" was the full command..."pixfmt" works if it's "rgb24", but it just failed when set to "yuv420p" because it wouldn't decode. [23:22] <ulatekh> I need to save yuv into mov so that kdenlive will accept it as an imported clip. [23:23] <durandal_1707> ulatekh: if you want to compress lossless there is fffhuff [23:24] <ulatekh> Do those work in a .mov container? The whole problem is that the interlaced flag keeps getting lost. There's an uncommited change written yesterday that fixes it for raw video, but the patch author says I have to use a .mov container or else the interlaced flag gets lost. [23:24] <ulatekh> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/158164 to see the patch [23:26] <durandal_1707> hmm, only what mov allows and allow raw is png/jpeg2000 [23:27] <ulatekh> As in "-vcodec png"? Because that was rgb24 pixel-format too. [23:27] <ulatekh> So maybe rgb24 _is_ my only chance. Where did you find the list of pixel formats that .mov allows? [23:29] <durandal_1707> in libavcodec/raw.c [23:29] <durandal_1707> scroll to quicktime [23:30] <ulatekh> Yeah, around line 169 in latest git...but there's a big list there [23:31] <durandal_1707> it have rgb and yuv, and you will not save much with either... [23:31] <durandal_1707> (i ignore rgb555) [23:32] <durandal_1707> and gray... [23:32] <ulatekh> Hey, "-pix_fmt uyvy422 -vtag 2vuy" just produced a clip that imports into kdenlive! [23:32] <durandal_1707> yes, they packed is more supported [23:32] <durandal_1707> *packed pixel format variant [23:33] <ulatekh> yuv420p is listed there, but that produced a clip that wouldn't import or decode...so how do I tell which pixel-format in this list will actually work with raw video? [23:33] <ulatekh> And BTW, thank you very much for your help. [23:33] <durandal_1707> though, -vtag ... should not be needed as that one is default one ... [23:33] <ulatekh> I'll try it without that next. [23:34] <durandal_1707> ulatekh: planar could be also slow to use anyway [23:34] <ulatekh> As long as it's raw, and kdenlive accepts it as an imported clip, I don't care about speed/space...this is just for editing purposes. [23:34] <durandal_1707> ulatekh: i dunno what kdenlive supports, probably only packed formats [23:35] <durandal_1707> packed formats do _not_ have P in its name [23:35] <ulatekh> So far, it's accepted "ffmpeg111 -i [input.yuv] -vcodec rawvideo [pixfmt] [output.mov]" for pixfmt of "-pix_fmt rgb24" and "-pix_fmt uyvy422 -vtag 2vuy". [23:36] <durandal_1707> ahh yes, 2vuy is not really default, some other one is.... [23:36] <ulatekh> (ffmpeg111 is my latest-git version, renamed with --progs-suffix and --build-suffix so that it can coexist on my machine with the yum-repo version of ffmpeg, since several dozen packages on my system depend on that) [23:37] <ulatekh> "-pix_fmt yuv411p" didn't work. [23:37] <durandal_1707> default tag picked is UYVY , which is not optimal i guess for mov [23:37] <durandal_1707> ulatekh: do not try any planar any more, i bet 99.9999% it does not support any [23:38] <ulatekh> No problem...but what should I choose? There's a big list & only 2 have worked so far. Right now I'm trying "-pix_fmt uyvy422" without the "-vtag 2vuy". [23:39] <durandal_1707> ulatekh: unfortunately only remainig packed yuv is uyvy422 with different order : yuyv422 [23:40] <ulatekh> So I shouldn't bother trying the other ones? [23:41] <durandal_1707> you could try rgb565 for fun, it takes 66% of rgb24 [23:42] <ulatekh> OK, "-pix_fmt uyvy422" without the "-vtag 2vuy" worked. I'll try "rgb565" next. [23:42] <durandal_1707> or rgba if you want alpha [23:42] <durandal_1707> 565 is bad if you care for quality colors [23:43] <ulatekh> Maybe I should stick with uyvy422 then? [23:44] <durandal_1707> ulatekh: you could ask kdenlive [23:45] <ulatekh> This is more of an ffmpeg issue, though, i.e. which pixel-formats are allowed for raw video in a Quicktime container. [23:46] <ulatekh> OK, "-pix_fmt rgb565" produced a working file. [23:46] <durandal_1707> funny [23:46] <ulatekh> It's the exact same size as the uyvy422 file, too. [23:47] <durandal_1707> yup, they take same amount of bits: look -pix_fmts output [23:48] <durandal_1707> but probably one looks worse than other in most situations [23:48] <ulatekh> So unless you know which other pixel formats are worth trying, I think I have a working solution, and I can leave you in peace with my gratitude for your help. [23:49] <durandal_1707> ulatekh: ask kdenlive devs, it should support yuv444 [23:50] <durandal_1707> if not, it is not right tool... [23:50] <ulatekh> The issue is specifically with raw video in a Quicktime container...if ffmpeg can't decode a file, kdenlive can't either. [23:51] <durandal_1707> hmm, than you are limited with mov container [23:52] <ulatekh> That's fine, I just want to know what the limits are. Apparently only the .mov container is going to preserve the interlacing flag on raw video, so it's my only choice for a container. [23:53] <durandal_1707> hmm, nut should support that too IIRC [23:53] <ulatekh> As in "-f nut"? [23:54] <durandal_1707> also matroska should too [23:55] <ulatekh> I'll try .nut and .mkv output files next. [23:55] <durandal_1707> well interlaced flag is set per each frame when decoding [23:56] <durandal_1707> also ffv1 saves interlaced flag in bitstream [23:56] <durandal_1707> so you do not depend on container when using ffv1 [23:57] <ulatekh> Apparently I am...I can't use an AVI container...it loses the interlaced info. I'm trying .nut now, and next I'll try .mkv . [23:58] <durandal_1707> also other lossless video codecs like utvideo,huffyuv could store interlaced flag.... [23:58] <durandal_1707> ulatekh: you are apparently limited by version of lav* kdenlive use [23:58] <ulatekh> Yeah, the ffv1 and huffyuv video codecs were fixed by the patch I mentioned earlier...the issue now seems to be the container. [23:59] <ulatekh> I compiled my kdenlive and mlt on top of latest-git ffmpeg earlier today. [00:00] --- Sun Jan 27 2013

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