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July 2013

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ffmpeg-devel.log.20130710
by burek 11 Jul '13

11 Jul '13

[00:04] <saste> michaelni, what are dump_extra and remove_extra bsfs? [00:04] <saste> is there some way to remove the global header from a file via a bsf? [00:07] <michaelni> remove_extra removes the global header from the individual frames if it happens to be there [00:07] <saste> michaelni, dump_extra, is any different? [00:08] <saste> also how do we pass the args command by using the commandline? [00:08] <saste> or was it never supported? [00:09] <michaelni> dumo_extra adds the extradata in front of frames [00:10] <saste> uhm ok [00:10] <durandal_1707> saste: so i should use av_strtok only? [00:10] <saste> as for what regards the argument, do you confirm it can't be passed to the ff* tools right now? [00:10] <saste> durandal_1707, you should be able to use av_strtok only [00:11] <saste> but note that Nicholas added support to lists in opt.h [00:11] <saste> but i'm not sure they're good enough for you [00:14] <durandal_1707> i don't see them [00:16] <michaelni> saste, i dont see how it could be passed either [00:17] <saste> durandal: av_opt_set_int_list [00:23] <durandal_1707> i fail to see how i could use this [00:36] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:3aa2257d240a: doc/RELEASE_NOTES: add a note about AVISynth [00:36] <cone-395> ffmpeg.git 03Alexis Ballier 07master:c3ca70204bd8: codec_desc: fix typo in rv20 long_name. [00:39] <johnsmith_> Has anyone here successfully compiled the latest FFMPEG version on centos 5? [00:40] <durandal_1707> there is problem? [00:41] <johnsmith_> Been spending at least 10 hours trying to get the latest FFMPEG onto Centos 5 [00:41] <johnsmith_> the yum install ffmpeg ffmpeg-devel has deprecated functions only [00:42] <Compn> what is the problem [00:42] <Compn> build error? [00:42] <Compn> paste it to pastebin [00:43] <Compn> because a) none of us are using centos5 and b) we cant read minds, especially computer minds [00:43] <johnsmith_> Hmm.. [00:44] <johnsmith_> One of the symptoms I have is libavcodec.so.52 is actually missing within usr/local/libs [00:45] <johnsmith_> And another is problems getting references to av_parser from the util.c files [00:46] <johnsmith_> I think the it's really hard to pinpoint the exact problem of the build [00:47] <johnsmith_> I am following the centos compilation guide, tweaking the fdk-aac and omitting opus libs [00:47] <johnsmith_> then running ffmpeg [00:48] <johnsmith_> ffmpeg successfully does a make install operation [00:48] <johnsmith_> but gcc/eclipse yell at me with those problems mentioned above [00:48] <durandal_1707> libavcodec.so.52 is very old, 55 is latest one [00:50] <johnsmith_> Should it even exist within usr/local/libs in the first place? [00:52] <durandal_1707> if you have old one installed... [00:53] <johnsmith__> Here is the eclipse output [00:53] <johnsmith__> http://pastebin.com/5cVeNCea [00:53] <durandal_1707> this is not #eclipse [00:54] <johnsmith__> this is the output trying to compile it [00:54] <durandal_1707> why you use that to compile? [00:54] <durandal_1707> that is not supported way to compile ffmpeg [00:54] <johnsmith__> i can use gcc to do it as well [00:54] <johnsmith__> give me a sec [00:55] <johnsmith__> http://pastebin.com/9mCNHP39 [00:55] <johnsmith__> same thing [01:04] <cone-395> ffmpeg.git 03Marton Balint 07master:3886572a83bf: ffplay: do not update audio clock with old audio pts if current frame is AV_NOPTS_VALUE [01:04] <cone-395> ffmpeg.git 03Marton Balint 07master:f07cb53ab951: ffplay: assume 0 stream start time if start time is unset in duration check [01:04] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:49f441ea8991: MAINTAINERS: Update which releases are supported [01:04] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:9e214900ad31: Merge remote-tracking branch 'cus/stable' [01:15] <durandal11707> what why 2.0? [01:15] <durandal11707> what is wrong with 1.4 ? [01:18] <durandal11707> perhaps because of refcounting? [01:20] <cone-395> ffmpeg.git 03Eugene Dzhurinsky 07master:09f59d6adfa1: avformat/gif: Do not write GIF89a header if loop is set to -1. [02:06] <johnsmith> durandal, thanks for giving me the hint that the libavcodec.so.25 was an old file, using old methods required old files [02:09] <johnsmith> I can successfully compile the ffmpeg decoding encoding example on the website, but I realized that me calling avformat_find_stream_info(..) caused all sorts of reference issues [02:09] <johnsmith> is it because that method is deprecated? [02:38] <cone-395> ffmpeg.git 03Luca Barbato 07master:5eaed6d33689: atrac3: fix error handling [02:38] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:1ebb95d5f193: Update for 2.0 [02:41] <cone-395> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/heads/release/2.0': unknown revision or path not in the working tree. [02:41] <cone-395> Use '--' to separate paths from revisions [02:41] <cone-395> refs/heads/release/2.0:HEAD: Update for 2.0 [02:49] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:2b8b2ba19fe0: Changelog: replace next by 2.0 [02:49] <cone-395> ffmpeg.git 03Michael Niedermayer 07release/2.0:2b8b2ba19fe0: Changelog: replace next by 2.0 [03:12] <cone-395> ffmpeg.git 03Michael Niedermayer 07fatal: ambiguous argument 'refs/tags/n2.0': unknown revision or path not in the working tree. [03:12] <cone-395> Use '--' to separate paths from revisions [03:12] <cone-395> refs/tags/n2.0:HEAD: Changelog: replace next by 2.0 [03:43] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:a37e42b3ee42: Update back for git master [03:43] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:0574fe76b857: Changelog: add back the <next> placeholder for new changes [03:43] <cone-395> ffmpeg.git 03Piotr Bandurski 07master:ccf9211e29bd: avformat/utils: avformat_find_stream_info fix a crash in case of oom [08:42] <mark4o> Shouldn't the Changelog link on the download.html page link to the Changelog ( http://git.videolan.org/?p=ffmpeg.git;a=blob_plain;f=Changelog;h=n2.0 )? [08:42] <mark4o> or maybe two links, Changelog and git shortlog [08:43] <mark4o> oops I mean http://git.videolan.org/?p=ffmpeg.git;a=blob_plain;f=Changelog;hb=n2.0 [10:23] <Bor0> what is the best way to get current frame? currentpts*fps works but not too good [10:31] <av500> ? [10:50] <xlinkz0> saste: i am ready to receive my robe and wizard hat [10:58] <xlinkz0> can i disable HAVE_TERMIOS_H ? [11:00] <saste> xlinkz0, what? [11:00] <xlinkz0> you said yesterday that i can't possibly make ffmpeg into a library.. unless i was a wizard [11:01] <xlinkz0> and so it is done :D [11:01] <saste> xlinkz0, what did you do exactly? [11:02] <xlinkz0> saste: a library that facilitates this : http://codepad.org/wFaMWG3K [11:03] <xlinkz0> now i need to test it on ios/android to make sure it works but i don't see why it wouldn't [11:03] <xlinkz0> since the libav libs work.. [11:03] <saste> xlinkz0, so in android/ios you can't use system()? [11:03] <xlinkz0> no [11:04] <xlinkz0> you can not create processes unless you root the phone [11:04] <saste> uh i see [11:06] <xlinkz0> only one else that done this charges 400$ per licence lol [11:06] <xlinkz0> http://ffmpeg4android.netcompss.com/home/purchase [11:07] <xlinkz0> are they allowed to sell it? [11:08] <atiti> lol [11:08] <JEEB> as long as they comply with the software license [11:08] <atiti> theyr probably selling the app that runs the command [11:08] <xlinkz0> you can not launch other apps from inside your app [11:08] <atiti> which is basically a terminal app [11:09] <xlinkz0> no such thing as far as i understand [11:10] <atiti> xlinkz0: its not launching another app, its executing the ffmpeg binary [11:10] <atiti> at least as far as i can see on the screenshot [11:10] <xlinkz0> which is not possible from normal apps unless the user roots the phone [11:10] <atiti> http://stackoverflow.com/questions/2843250/how-to-run-terminal-command-in-a… [11:10] <xlinkz0> either this is false or my phone dev colleagues are n00bs :( [11:11] <xlinkz0> even if it works for android it won't work on ios [11:11] <atiti> sure its possible without rooting [11:11] <JEEB> the app just has to be in a place where it can have the execution bit, IIRC? [11:11] <atiti> all apps run as a user, so you can easily execute commands under that user [11:11] <xlinkz0> afaik you need to ship the ffmpeg binary with the app [11:11] <atiti> rooting is what the name stands for, how to get root on the phone :p [11:12] <xlinkz0> and when you ship it you must be able to chmod it which you can not.. [11:12] <atiti> yes, you can chmod it if your own user owns it [11:12] <xlinkz0> so you may be able to run ls which is already chmoded [11:12] <atiti> which it does, if it comes with your app [11:13] <xlinkz0> you tested it? [11:13] <JEEB> anyways, not sure how they do it but it sure looks like launching the command line interface in one way or another. Granted, I don't take out the possibility of herping a derp and actually making the CLI a "lib" :D [11:13] <atiti> xlinkz0: yes [11:13] <xlinkz0> well then i'll just go kill the android dev [11:14] <xlinkz0> hope this doesn't work on ios atleast, the ios dev said he can't call other processes [11:15] <atiti> im not familiar with the ios buildup so i cant comment on that [11:17] <xlinkz0> <Kanalia> xlinkz0: you can't run external executables/binaries from your app [11:17] <xlinkz0> mwhahaha [11:17] <xlinkz0> my work was not in vain [11:17] <xlinkz0> (ios) [12:24] <cone-59> ffmpeg.git 03Luca Barbato 07master:874c8a17ac9b: atrac3: fix error handling [12:24] <cone-59> ffmpeg.git 03Luca Barbato 07master:22e76ec635ba: atrac3: set the getbits context the right buffer_end [12:24] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:40e8967ee161: Merge commit '22e76ec635bafdd1d1ec35581a7ac09e69e3c43e' [12:36] <cone-59> ffmpeg.git 03Luca Barbato 07master:50cf5a7fb788: atrac3: Error on impossible encoding/channel combinations [12:36] <cone-59> ffmpeg.git 03Luca Barbato 07master:bbf6a4aa20bf: imc: Catch a division by zero [12:36] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:a3d9a2168355: Merge commit 'bbf6a4aa20bfe3d7869b2218e66063602dfb8aa7' [13:08] <cone-59> ffmpeg.git 03Luca Barbato 07master:12576afe206d: adpcm: Write the correct number of samples for ima-dk4 [13:08] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:067c9c3b6319: Merge commit '12576afe206d35231ccd61f9033c5fdab6a11e80' [13:48] <durandal_1707> michaelni: why 2.0 is not put into news? [13:49] <durandal_1707> and why "Nameless" as name, there are much better names.... [14:09] <cone-59> ffmpeg.git 03Paul B Mahol 07master:410192e36958: add crc24 tables [14:09] <cone-59> ffmpeg.git 03Paul B Mahol 07master:8491f4050053: tak: use crc table from lavu [14:09] <cone-59> ffmpeg.git 03Paul B Mahol 07master:4977e467a50a: lavf/takdec: check metadata blocks checksum [14:28] <kierank> ubitux: your r128 filter was presented today [14:38] <cone-59> ffmpeg.git 03Paul B Mahol 07master:884c890535c5: lavfi: add aecho filter [14:44] <durandal_1707> kierank: on cnn? [14:44] <kierank> at rmll [14:46] <durandal_1707> video? [14:47] <Compn> was it for ffmpeg talk or used in a different project ? [15:06] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:2c0ec526e2c3: fate/crc fix after master:410192e36958: add crc24 tables [15:14] <cone-59> ffmpeg.git 03Luca Barbato 07master:adadc3f2443d: qdm2: refactor joined stereo support [15:14] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:564909093a31: Merge commit 'adadc3f2443d25b375e21e801516ccfd78e0b080' [15:20] <cone-59> ffmpeg.git 03Paul B Mahol 07master:bc95b9428950: lavfi/aconvert: unbreak [15:37] <durandal_1707> michaelni: earwax filter overreads & overwrites [15:40] <durandal_1707> asetnsamples=17,earwax [16:07] <cone-59> ffmpeg.git 03Luca Barbato 07master:4ecdb5ed4459: qdm2: Conceal broken samples [16:07] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:2e6338b47276: Merge remote-tracking branch 'qatar/master' [16:30] <ghooman> Hi, i'm trying to make a custom audio filter [16:33] <ghooman> can i retrieve file duration from filter context& [16:35] <durandal_1707> ghooman: nope [16:35] <durandal_1707> why would you need it? [16:38] <ghooman> to calculate average volume on 1/1800 [16:38] <durandal_1707> 1/1800 ? [16:38] <ghooman> part of [16:39] <durandal_1707> you can count number of samples [16:41] <ghooman> i want to produce 1800(or some other number) of json values for http://www.waveformjs.org/ [16:43] <ghooman> i need that value at start, because i want to calculate size of 1/n segment [16:45] <durandal_1707> there is no way to get duration from filter context [16:45] <ghooman> thx [16:46] <durandal_1707> but as 1800 is fixed, when it does not need to be, which means you can get duration via other means and pass it to filter [16:47] <cone-59> ffmpeg.git 03Piotr Bandurski 07master:b0509563347b: avformat/utils: avformat_find_stream_info set value for ret in case of oom [16:48] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:fbe159e85079: avcodec/qdm2: store bits in an integer instead of float variable [16:48] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:8f09957194b8: avcodec/qdm2: initialize sign_bits [16:48] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:0a3a0edd52b9: avfilter/af_earwax: Fix out of array accesses on odd packets [17:03] <mateo`_> hi guys, anyone to review/comment my lavf/movenc patch ? [17:19] <ubitux> kierank: fun, any feedback? [17:20] <michaelni> ubitux, do you want to review mateo`_ patch or that webvtt patch ? [17:20] <ubitux> i'm sorry i can't currently [17:20] <ubitux> i'll be available full time in august for ffmpeg [17:20] <michaelni> ok [17:20] <ubitux> but nothing until then [17:21] <saste> michaelni, what about the names of psychedelic drugs? [17:23] <michaelni> saste, i dont think thats a good idea [17:24] <saste> michaelni, it doesn't have to be a good idea, must be fun [17:26] <ubitux> http://namingschemes.com/ [17:31] <durandal_1707> michaelni: for next releases, just do nameless II, nameless III and so on [17:31] <saste> nameless unreleased [17:32] <durandal_1707> http://namingschemes.com/Vagina_Synonyms [17:33] <ubitux> that will certainly bring a lot of female contributors [17:38] <durandal_1707> only this remain: http://namingschemes.com/Pokemon [17:46] <aballier> michaelni: have you considered dropping 1.1 branch and still keeping 1.0 maintained ? (I dont see any reason to use 1.1 but not 1.2 however 1.0 is nice for downstreams since it doesnt have the planar formats) [18:08] <michaelni> aballier, would be a possibility if thats what people prefer, but maintaining older releases is more work than more recent ones. [18:09] <michaelni> If someone wants to take over maintaince of a old release or wants to help that is welcome [18:14] <cptspiff> regarding topic. will it alternative libav/conflict libav or is the lib name pissing contest settled? former would require abi compat which sounds a bit.. challenging. or do you just want the fluff for third party repos? [18:15] <Compn> cptspiff : which topic ? which naming conflicts ? [18:15] <cptspiff> debian packaging [18:16] <Compn> you mean libav masquerading as 'ffmpeg' package ? its up to debian [18:16] <Compn> or ffmpeg trademark owner... [18:17] <j-b> FLV1 and MPEG-4 ASP have something in common? [18:17] <cptspiff> debian wont accept several packages with conflicting filenames unless they are alternatives [18:17] <Compn> j-b : i thought h263 , have to check what flv1 codec relies on [18:18] <Compn> cptspiff : well i think saste was trying to discuss with ubuntu/debian. i dont know outcome of that [18:18] <cptspiff> k. anyways just waned to offer my fu.. [18:19] <Compn> j-b : yes , flv links against mpegvideo and h263, same as mpeg4 asp ... [18:19] <michaelni> cptspiff, both variants, as alternative as well as with different soname could be made, its not that one of the 2 needs to be picked ad the other could then not be done [18:20] <j-b> Compn: thx [18:20] <michaelni> cptspiff, about debian/ubuntu also talk with beastd [18:26] <aballier> michaelni: I could try to help but i'm not sure if i'm competent enough; I was mainly asking because maintaining 1.1 seemed not worth the work [18:28] <aballier> michaelni: on a semi-related topic, are the recent jpeg2000 fixes needed for the old j2k code in 1.0 ? [18:31] <michaelni> hard to say, some possibly might be needed, but j2kdec was marked as experiemntal so it should not have been used by default but then some applications might have enabled experimental decoders by default on their own [18:32] <michaelni> about "compent" if you maintain 1.0 or help it will be in better shape than if you dont [18:33] <j-b> http://xhevc.com/en/downloads/downloadCenter.jsp based on ffmpeg 1.0.6 [18:33] <j-b> and there is source in it [18:37] <aballier> one possibility could be to disable j2kdec at build time and rely on openjpeg [18:37] <michaelni> yes [18:37] <aballier> michaelni: when did you stop backporting fixes to 1.0 ? [18:37] <aballier> (so that i know where to start from in e.g. 1.2 branch) [18:44] <michaelni> the last release from 1.0 was 1.0.7 from 2013-05-14, after that i thnk there are just some bugfixes backported by carl [18:45] <aballier> thanks, will have a look [19:04] <kierank> ubitux: true peak and hardcoded 48khz would be nice according to the guy [19:04] <ubitux> hardcoded 48khz ? [19:05] <ubitux> you mean removing the hardcoded 48khz to support other freq? [19:05] <kierank> yes [20:24] <cone-59> ffmpeg.git 03Carl Eugen Hoyos 07master:c9eb5c9751c8: lavf/utils.c: Avoid a null pointer dereference on oom after duration_error allocation. [20:24] <cone-59> ffmpeg.git 03Carl Eugen Hoyos 07master:1db88c33f2c0: Suggest recompilation with openssl or gnutls if the https protocol is not found. [20:24] <cone-59> ffmpeg.git 03Michael Niedermayer 07master:07206deed103: Merge remote-tracking branch 'cehoyos/master' [20:46] <mark4o> durandal_1707: [-]HH[:MM[:SS[.m...]]] ? should be [-][HH:]MM:SS[.m...], yes? [20:46] <mark4o> looks like other parts of doc have it wrong also [20:48] <ubitux> i would say [-][[HH:]MM:]SS[.m...] [20:48] <durandal_1707> patch welcome [20:48] <mark4o> well it also has [-]S+[.m...] [20:49] <mark4o> durandal_1707: a patch to your patch? [20:50] <durandal_1707> i can fix it in my patch [21:10] <durandal_1707> mark4o: i mean i will fix hhmmss in code i add [21:11] <cone-59> ffmpeg.git 03Paul B Mahol 07master:45119739019e: wmalosslessdec: do not set avctx->coded_frame [21:12] <cone-59> ffmpeg.git 03Derek Buitenhuis 07master:2ea60971dca9: fate: Always enable avresample [21:14] <cone-59> ffmpeg.git 03Paul B Mahol 07master:1e89f74902c6: lavfi/afade: use av_rescale() [21:24] <Daemon404> michaelni, when you say "more complex things than read a variable" for teh avio callback, what do you mean? [21:24] <Daemon404> i do something a bit more complex, but it may or may not be classified as "read a variable" [21:25] <Daemon404> https://github.com/dwbuiten/d2vsource/blob/master/core/decode.cpp#L44 and the following func, for referene [21:27] <cone-59> ffmpeg.git 03Paul B Mahol 07master:badbd2414d38: lavfi/aevalsrc: use AV_OPT_TYPE_DURATION [21:27] <cone-59> ffmpeg.git 03Paul B Mahol 07master:db4e191d8897: lavfi/aevalsrc: do not free AVOption variables in uninit() [21:59] <michaelni> Daemon404, i dont see any usage of the interrupt callback in there [22:00] <Daemon404> michaelni, uau already replied on the mL [22:02] <durandal_1707> that's not uau iirc [22:02] <nevcairiel> no its not [22:02] <Daemon404> woops [22:02] <Daemon404> youre right [22:02] <durandal_1707> s/that's/he's ... [23:23] <j-b> michaelni: is there a way to disable the VirtualAlloc/Free calls from swscale ? A config option? [23:26] <cehoyos> j-b: Did you try --disable-mmxext ? [23:28] <j-b> cehoyos: o_O [23:29] <cehoyos> Am I wrong? [23:31] <j-b> cehoyos: probably not [23:31] <cehoyos> (I just don't understand the emoticon and Google doesn't really help.) [23:34] <michaelni> j-b, i can add a configure option probably to disable it [23:35] <j-b> no, I just was wondering about it [23:35] <j-b> I'll find another way [23:38] <durandal_1707> is there reasonable way to automaticaly free pads that get dynamically allocated? [23:40] <durandal_1707> i could add some flag to each pad that get allocated.... [00:00] --- Thu Jul 11 2013

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ffmpeg.log.20130710
by burek 11 Jul '13

11 Jul '13

[00:06] <johnsmith_> Hi, has anyone ever had any success whatsoever installing the latest ffmpeg onto CENTOS 5/RHEL 5? [00:09] <ulkesh> johnsmith_: sorry, I installed it recently on a centOS 6 VM, but not tried on centOS 5 [00:27] <goldfndr> Zeranoe: the current ffmpeg.exe shared (http://ffmpeg.zeranoe.com/builds/win32/shared/ffmpeg-20130709-git-00b1401-w…) gives an error on WinXP SP3, "ffmpeg.exe - Unable To Locate Component", "This application has failed to start because libstdc++-6.dll was not found. Re-installing the application may fix this problem." [00:27] <goldfndr> http://ffmpeg.zeranoe.com/builds/win32/shared/ffmpeg-1.2-win32-shared.7z (from http://ffmpeg.zeranoe.com/builds/win32/shared/) doesn't have this error. [00:30] <tm512> Does anyone know how to stream with ffmpeg, without the audio and video being completely out of sync? [00:30] Action: goldfndr didn't see this documented/mentioned in the IRC logs at http://lists.ffmpeg.org/pipermail/ffmpeg-devel-irc/ [00:57] <curfont> When trying to compile something that uses -lswscale with gcc it works, but g++ doesn't find the reference [00:57] <curfont> Is there some option I am missing? [01:00] <curfont> g++ test.cpp -L/home/functor/ffmpeg_build/lib -I/home/functor/ffmpeg_build/include -lavformat -lavcodec -lswscale -lavutil -lm -D__STDC_CONSTANT_MACROS [01:00] <curfont> all I am doing in the test.cpp is calling sws_scale with 0,0,0,0 params, but its not finding it as a ref [01:00] <curfont> if I use gcc instead of g++ it works [01:01] <mark4o> curfont: did you put extern "C" { } around the #include? [01:02] <curfont> oh I see, let me try that [01:03] <curfont> extern "C" { #include <libswscale/swscale.h> } [01:03] <curfont> Like this? [01:03] <curfont> yeah it didn't work :/ [01:03] <mark4o> you need 3 lines [01:03] <curfont> complains about "#" being stray now [01:03] <curfont> ah ok [01:04] <curfont> yes that worked thanks [01:20] <curfont> whats the new version of av_set_parameters? [01:24] <durandal11707> there is no new version for that one. iirc you use av_set_opt [01:27] <curfont> I was reading that they use avformat_write_header instead now [01:27] <curfont> hmm [04:30] <tm512> Does anyone here know how to stream from ffmpeg without getting a/v desyncs? [04:45] <praveenmarkandu> hi, may i know what are the factors involved in the time it takes to transcode? [04:45] <praveenmarkandu> is it bitrate? codec? resolution? [09:17] <Crusher4> anyone around that can direct me to the docs on opencl for the 2.0 release? what platforms/configs it's supported in/etc? [09:34] <Mavrik> Crusher4, opencl support was just released, there isn't even an announcement on ffmpeg page yet. [09:34] <Mavrik> plus, what do you think OpenCL support does? [09:35] <Crusher4> saw it on phoronix. the shortlog wasnt specific in how it was integrated, but it was even less specific on which platforms ffmpeg runs on that opencl was supported on [09:36] <Crusher4> hence i came here, to see if anyone had more info other than "adds opencl support" [09:36] <Mavrik> yes, the framework has preliminary openCL support [09:36] <Mavrik> right now it's used by deshake and denoise filters [09:36] <Mavrik> that's about it. [09:37] <Crusher4> that's a start. on which platforms? [09:37] <Mavrik> no idea [09:37] <Mavrik> ones that support opencl I'd say. [09:37] <Mavrik> check the configure script. [09:38] <Crusher4> hm. good idea [09:41] <Crusher4> ah. requires OpenCL 1.2, so nothing on osx for awhile then [09:45] <sunjun> hi I have a problem about x86 compile [09:46] <sunjun> libavcodec/x86/mpegvideo_mmx_template.c:144:9: error: can't find a register in class 'GENERAL_REGS' [09:46] <Crusher4> Mavrik: thx for the help, laters [09:46] <sunjun> while reloading 'asm' [10:15] <Bor0> what is the best way to get current frame? currentpts*fps works but not too good [10:28] <someone-noone> hello! [10:28] <someone-noone> ./configure --enable-gpl --enable-libx264 --enable-libvpx --enable-libvorbis --enable-libopus --prefix=/usr [10:28] <someone-noone> ERROR: opus not found [10:28] <someone-noone> of course opus exists in /usr/include/opus * /usr/lib [10:28] <someone-noone> ffmpeg version N-54577-g3c5071d [10:28] <someone-noone> looks like a bug? [10:29] <someone-noone> in config.log: [10:29] <someone-noone> check_pkg_config opus opus_multistream.h opus_multistream_decoder_create [10:29] <someone-noone> ERROR: opus not found [10:30] <spaam> you forgot to install libopus-dev ? [10:30] <spaam> ah nvm [10:31] <someone-noone> I have /usr/include/opus/opus_multistream.h and there exists declaration of opus_multistream_decoder_create() [10:31] <someone-noone> ls /usr/lib | grep opus [10:31] <someone-noone> libopus.0.dylib [10:31] <someone-noone> libopus.a [10:31] <someone-noone> libopus.dylib [10:31] <someone-noone> libopus.la [10:31] <someone-noone> so, looks like opus is installed correctly [10:32] <saste> someone-noone, did you check config.log? [10:32] <someone-noone> yes, look above [10:32] <saste> someone-noone, how did you install libopus? [10:32] <saste> do you have the pkg-config files around? [10:33] <someone-noone> cd opus-1.0.2 ; ./configure prefix=/usr; make install [10:34] <someone-noone> I didn't do anything related to pkg-config [10:34] <someone-noone> looks like I should do? [10:38] <someone-noone> saste, how pkg-config files look like and where they should be located? I've never worked with pkg-config... [10:38] <saste> someone-noone, which system? [10:39] <saste> pkg-config libopus --version [10:39] <someone-noone> mac os x [10:40] <saste> usually you should find a lib/pkgconfig containing pkg-config metadata files [10:40] <someone-noone> hm, it printed: 0.26 . Looks like libopus is installed somewhere [10:40] <saste> which are looked after by pkg-config [10:41] <saste> someone-noone, weird... [10:41] <someone-noone> ? [10:49] <someone-noone> Okay, I've reinstalled libopus [10:50] <someone-noone> now I do: ls /usr/lib/pkgconfig/ | grep opus [10:50] <someone-noone> It outputs: opus.pc [10:50] <someone-noone> however: pkg-config opus --cflags [10:50] <someone-noone> Package opus was not found in the pkg-config search path. [10:50] <someone-noone> Perhaps you should add the directory containing `opus.pc' [10:50] <someone-noone> to the PKG_CONFIG_PATH environment variable [10:50] <someone-noone> No package 'opus' found [10:51] <saste> someone-noone, what's the value of PKG_CONFIG_PATH? [10:52] <someone-noone> Sorry for dummy question, but how can I know? echo $PKG_CONFIG_PATH is correct? [10:52] <saste> yes [10:52] <someone-noone> it's empty [10:52] <someone-noone> 0_0 [10:54] <saste> someone-noone, which should be fine, since it should look in /usr/lib/pkgconfig by default [10:55] <someone-noone> saste, where is a problem? [10:56] <mark4o> to show default package config path: pkg-config --variable pc_path pkg-config [10:56] <mark4o> should be same hierarchy as pkg-config itself [10:57] <someone-noone> mark4o, looks like you showed me the problem [10:57] <someone-noone> macports have installed pkg-config and pathes are: /opt/local/lib/pkgconfig:/opt/local/share/pkgconfig [10:58] <someone-noone> how should I add /usr/lib/pkgconfig path correctly ? [10:58] <mark4o> set PKG_CONFIG_PATH? [10:58] <mark4o> why do you want to install in /usr? [10:59] <someone-noone> I know that it's in general not correct [10:59] <mark4o> that is for the os, and may be overwritten on os update [11:05] <someone-noone> mark40, I'm just studying shell and some programs don't see "/usr/local". Now I know how to work with environment variables and I just need to forget about this silly prefix... [11:10] <someone-noone> finally it works :) thanks guys for help [11:13] <mark4o> great, did you install it in /usr/local or /opt? [11:14] <someone-noone> first of course, /opt is for macports [11:15] <mark4o> ok so I guess you didn't need pkg-config, cool [11:32] <ningujoan> hi there [11:33] <ningujoan> does anyone know if there is any version where the ffserver does work? I keep getting error messages same as everybody searching on google but no answers :/ [11:33] <ningujoan> it might be a version issue thoug, anyone has it running? [11:34] <Mavrik> ffserver works just fine. [11:37] <Bor0> what is the best way to get current frame? currentpts*fps works but not too good, especially for low fps vids [11:37] <ningujoan> it gives me a 404 when I try to connect: [11:37] <ningujoan> [tcp @ 0x25feec0] TCP connection to localhost:8090 failed: Connection refused [11:37] <ningujoan> [http @ 0x25f1380] HTTP error 404 Not Found [11:37] <ningujoan> http://localhost:8090/feed1.ffm: Input/output error [11:37] <ningujoan> and on the server: File '/feed1.ffm' not found [11:38] <ningujoan> it's so simple, don't know what i'm doing wrong then :/ [11:38] <Mavrik> Bor0, "get" where? [11:38] <Bor0> I want to retrieve current frame, let's say in ffplay [11:38] <Mavrik> Bor0, command-line? source? you using libraries? is't not clear. [11:38] <Bor0> in ffplay.c [11:39] <Mavrik> ningujoan, no, it's giving you connection refused [11:39] <Mavrik> ningujoan, which means it cannot connect to anything on port 8090 [11:39] <Bor0> I want to modify ffplay.c so that it'll display currentframe in console. [11:39] <Mavrik> which means you either have a firewall or ffserver isn't running on that port [11:39] <Bor0> currentpts * fps is not stable, especially for low-fps vids [11:39] <Mavrik> Bor0, hmm, counting frames would be better probably [11:39] <Bor0> when I press 'S' to frame step forward, it gives 1, 2, 3, 4, 4, 4, 8 or something [11:39] <Bor0> but how can I count them? what if the user seeks somewhere, how will the counting be handled? [11:40] <Mavrik> good question [11:40] <Mavrik> Bor0, anyway, currentpts * fps is totally wrong [11:40] <Mavrik> since PTS doesn't show time in seconds. [11:40] <Bor0> it works for videos with good fps [11:41] <Mavrik> current_time * fps would give you a good estimate [11:41] <Bor0> well this is the exact formula that I use: currentframe = (double)(pts_int * is->slotinfo->fps/(is->video_st->time_base.den)); [11:41] <Mavrik> but your problem is that videos don't have to have fps [11:41] <ningujoan> well, on the server it does connect: it gives a: File '/feed1.ffm' and then 127.0.0.1 - - [GET] "/feed1.ffm HTTP/1.1" 404 149 not found, and there are a number of questions like this one with no answer on google [11:41] <Mavrik> and that you can't really know how many frames you had in the middle of video [11:41] <Bor0> let's take out VBR for this case :) [11:41] <Bor0> and focus on constant rate [11:41] <d-fens_> hi, i need to generate a h264 from image series in a .net application, is there a recommened wrapper or way to archive this? [11:42] <JEEB> d-fens_, either use libavcodec and friends as a native C library, or use x264 itself as a native C library? [11:42] <Bor0> do you have any suggestions Mavrik ? or is this a tough thing to do [11:43] <Mavrik> Bor0, it's certanly not easy [11:43] <Mavrik> depends what your desired result is [11:43] <d-fens_> JEEB: ok but then i have to deal with all pinkvoke and stuff my myself [11:43] <Mavrik> with constant framerate you need to grab first PTS, calculate difference between current PTS and start PTS and then calculate number of frames with help of timebase [11:44] <JEEB> d-fens_, let's just say that if someone actually made you a wrapper if that person wasn't really motivated it would be out-of-date by now :) [11:44] <JEEB> there were lots of changes in APIs during 2011-2012 [11:44] <Mavrik> ningujoan, well, you DID define the feed1.ffm sink in ffserver configuration right_ [11:45] <d-fens_> ok, may be thera is a autogen-wrapper thingy hiden somewhere [11:45] <JEEB> most probably there isn't, though :) [11:45] <Bor0> so that's |pts_start - pts|/av_timebase against pts*currentframe/av_timebase ? [11:46] <d-fens_> i see, it just that i have to deep dive into libavcodec internals [11:46] <Bor0> pts*fps/av_timebase, sorry [11:46] <JEEB> well, another alternative is to use libx264 itself without the libavcodec around it [11:46] <d-fens_> which i would like to avoid [11:46] <Bor0> is the former an approximation as well? can I be certain that it will give the correct current frame? [11:46] <JEEB> and I don't think making AVFrames for an encoder nowadays is /that/ hard [11:47] <d-fens_> jepp adding some mp3 afterwards should be easy too, i just thought about wrapping the exe as a dirty approach [11:47] <mark4o> Bor0: don't you want to cast to double before the division? otherwise it is integer division and you'll lose precision [11:48] <JEEB> yes, just wrapping the exe and pushing stuff to stdout and piping to it is one hackjob way of doing it :) [11:48] <Bor0> I'm doing all the needed casts [11:48] <Mavrik> Bor0, don't convert to seconds until last second. [11:48] <Mavrik> otherwise you're just throwing away precision [11:48] <Bor0> you mean x = pts/av_timebase, then currentframe = fps * x ? [11:48] <Mavrik> (current_pts - start_pts) / stream_time_base * fps [11:48] <d-fens_> JEEB: thanks for your input i'll check and read up on my options now [11:49] <Bor0> what's an easy way to get the start_pts? [11:49] <mark4o> Bor0: your (double) cast is after the division, should be before like (double)(current_pts - start_pts) * fps / tb [11:50] <Bor0> what difference would start_pts make in that formula? isn't it a constant? [11:53] <Mavrik> yes, it is. and? [11:53] <Bor0> will not it also be an approximation? [11:56] <Mavrik> pts values are absolute. [11:59] <mark4o> Bor0: if you don't want to consider start_pts then this should be an improvement over what you have: currentframe = (double)pts_int * is->slotinfo->fps / is->video_st->time_base.den; [12:01] <Bor0> let me try the one with startpts and currentpts [13:42] <t4nk577> Hi, i'm trying to compile ffmpeg and x264 and got the following error in make [13:42] <t4nk577> libx264.o : error LNK2001: unresolved external symbol _x264_bit_depth libavcodec/avcodec-54.dll : fatal error LNK1120: 1 unresolved externals make: *** [libavcodec/avcodec-54.dll] Error 1 [13:42] <t4nk577> can some help me with this? [13:43] <Mattias> I have 'ffmpeg -i "filename" -y -ss "00:00:10" -s 720x460 -f image2 "thumb_filename"' but I want to have it save the aspect ratio and just go with the width [13:43] <Mattias> How can I do that? [13:46] <t4nk577> http://pastebin.com/RSeh09D3 [13:47] <t4nk577> this is my configure command and i have x264 builds in /home/username/x264-builds folder [13:47] <JEEB> can you pastebin your config.log? [13:49] <t4nk577> yes [13:52] <t4nk577> http://pastebin.com/A3HUvW4F [13:52] <t4nk577> this is configure log [13:54] <JEEB> ok [13:54] <JEEB> so configure doesn't even check for it :P [13:54] <JEEB> oh it does [13:56] <t4nk577> ok [14:04] <mark4o> Mattias: ffmpeg -i infile -y -ss 10 -vf scale=720:-1 -vframes 1 thumbfile [14:05] <Mattias> mark4o, thanks, got it to work now :) [16:49] <mrAlmond> Hi all [16:50] <mrAlmond> I neet to write custom metadata tags in an MP4 file muxed by my application in C code with ffmpeg [16:50] <mrAlmond> which is the best way to do this? [17:57] <juanmabc> 'tsup [19:35] <klaxa> https://gist.github.com/klaxa/5968374 [19:36] <klaxa> where do i put -strict -2 so it gets recognized? [19:36] <durandal_1707> after codec for which it is relevant [19:36] <klaxa> as you can see i put it there [19:38] <durandal_1707> than try it with before vp9 [19:39] <klaxa> nope [19:39] <klaxa> argument gets parsed correctly even wtf [19:39] <klaxa> >Reading option '-strict' ... matched as AVOption 'strict' with argument '-2'. [19:40] <klaxa> doesn't look like it's getting applied though [19:42] <durandal_1707> -strict:v ? [19:43] <klaxa> doesn't work either [19:44] <klaxa> https://gist.github.com/klaxa/5968451 [19:44] <klaxa> with -loglevel debug [19:56] <klaxa> filed bugreport http://www.ffmpeg.org/trac/ffmpeg/ticket/2772 [20:24] <tm512> does anyone at all know how to stream with ffmpeg, without getting desyncs? [20:33] <killown> ffmpeg now has opencl support? there is a tutorial for the opencl usage? [21:58] <humble_user> I'm looking for a little guidance regarding ffprobe. I'm wondering if it's feasible to use ffprobe to parse a transport stream of broadcast tv in order to get channel information (i.e. the name of the channel or call sign). Has anyone here ever done anything like that? [00:00] --- Thu Jul 11 2013

1 0

ffmpeg-devel.log.20130709
by burek 10 Jul '13

10 Jul '13

[01:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.10:f75964ad1fe7: mpeg12dec: avoid reinitialization on PS changes when possible. [01:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.11:c9f34c82604a: mpeg12dec: avoid reinitialization on PS changes when possible. [01:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.7:82c6b76d701a: mpeg12dec: avoid reinitialization on PS changes when possible. [01:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.8:f8c4dbe45ee2: mpeg12dec: avoid reinitialization on PS changes when possible. [01:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.9:d0a225d6c1a7: mpeg12dec: avoid reinitialization on PS changes when possible. [01:05] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:ec268c24a77e: mpeg12dec: avoid reinitialization on PS changes when possible. [01:05] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:18900381e2d3: mp3dec: detect CBR and use CBR axiom to seek [01:05] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:0f84286677b5: mpeg12dec: avoid reinitialization on PS changes when possible. [01:05] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:2e75c455935d: mp3dec: detect CBR and use CBR axiom to seek [01:05] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:6cb33e076323: mpeg12dec: avoid reinitialization on PS changes when possible. [01:56] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:d7fbe926d312: ffmpeg: move creation_time for ffserver setting up [01:56] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:d1e8d4b055bd: ffmpeg: Allow overriding ffservers input [02:32] <Foofie> just noticed that ffmpeg -h full does an infinite loop... is this a known issue? [02:33] <durandal_1707> Foofie: doesn't here [02:33] <durandal_1707> Foofie: what version? [02:33] <Foofie> git head [02:33] <Foofie> using latest git head of libvpx [02:34] <Foofie> it appears to continuously loop on libvpx related options [02:34] <durandal_1707> i don't have libvpx compiled [02:35] <durandal_1707> but something changed recently with libvpx stuff [02:35] <Foofie> it's very strange :/ [02:35] <durandal_1707> vp9 support have been added iirc [02:36] <Foofie> yes [02:36] <durandal_1707> Foofie: is this with MSVC? [02:36] <Foofie> git clone http://git.chromium.org/webm/libvpx.git ... it's a pretty easy ./configure && make && make install [02:36] <Foofie> nope, on mac osx [02:41] <Foofie> I have found that some options end up freezing execution (so when running -v debug), it will just wait forever when it's parsing an option, such as -g (and some others). My guess is that it's the same issue... some data pointing back to itself in a circular manner... ? (have not looked at the code though) [02:42] <Foofie> like I can make it hang forever if I use the -g option, but when I get rid of it, it does go on through. [02:43] <Foofie> afk for a bit [03:06] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:a20f049c2ab5: avcodec/libvpxenc: make class names unique [03:07] <michaelni> Foofie, should be fixed (i hope) [03:16] <cone-757> ffmpeg.git 03Matthew Heaney 07master:509642b4bd84: WebM muxer writes WebVTT subtitle track [05:32] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:00b1401df656: avcodec/h264_ps: try to support Ambarella AVC [09:43] <wm4> wow this webvtt muxing stuff is ugly [09:47] <superware> michaelni: hi. I wanted to thank you for fixing all issues with that hw stream! [09:56] <michaelni> wm4, anything that could be done better ? if so please reply in the thread [09:56] <wm4> it's probably too late and the format requires it [09:57] <michaelni> also theres a demuxer patch, if you want to review it ... [09:57] <michaelni> ubitux is busy AFAIK and wont review it quickly i suspect ... [09:58] <michaelni> BuxiNess, there are some jpeg2000 patches on ffmpeg-devel if you have time to review ... [10:07] <cone-395> ffmpeg.git 03Martin Storsjö 07master:a87a0acf9b5d: movenc: Make sure the RTP hint tracks have nondecreasing DTS [10:07] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:848295c2eeac: Merge commit 'a87a0acf9b5d27aad032e61eef4973e62a4a6830' [10:14] <cone-395> ffmpeg.git 03Martin Storsjö 07master:0677ddebc6e9: movenc: Rename the timestamp parameter to dts in write_hint_packets [10:14] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:e914a21ada12: Merge remote-tracking branch 'qatar/master' [10:36] <superware> is there anything I can try to reduce latency other than AVFMT_FLAG_NOBUFFER? [10:36] <JEEB> superdump, completely depends on what you're trying to do [10:37] <JEEB> for example the encoder you might be using and its settings are relatively relevant :P [10:39] <superware> JEEB: you are right. I can't touch the encoder, only the decoder. The stream is h.264/mpeg ts/udp.. [10:40] <b_jonas> yay, the ticket claims you fixed my bug! thanks for all the developers fixing it. [10:41] <JEEB> so you're just decoding a random stream that you're getting? well, can't help much there except for maybe setting the container/video|audio formats and making sure ffmpeg doesn't try to probe it first? [10:41] <JEEB> nothing else I can say :) [10:41] <superware> JEEB: probing time isn't related to later video-latency, right? [10:42] <BuxiNess> michaelni, yes I 'll get a look [10:42] <BuxiNess> I think I found the problem of perf according the number of threads. [10:42] <michaelni> BuxiNess, thanks! [10:42] <BuxiNess> I'll send a patch [10:42] <michaelni> ok [10:44] <JEEB> superdump, if you're just decoding the decoding should be going the usual speed, and it's mostly just latency in starting that decode? [10:45] <superdump> JEEB: superw<tab> [10:45] <JEEB> yes, mis-tabbed [10:45] <nevcairiel> or get a irc client that prefers people that have talked last :p [10:46] <superdump> although i just ruined such a feature [10:46] <superdump> su<tab><tab> should work too [10:46] <superdump> if your irc client cycles nicks [10:46] <superware> JEEB, I don't really care about probing latency, only video latency [10:47] <JEEB> yes, I just didn't cycle because I'm tired :P [10:47] <superware> JEEB, with AVFMT_FLAG_NOBUFFER I think the latency is about 200ms [10:48] <JEEB> I have no idea :P [10:56] <wm4> so do you have to call avcodec_decode_audio4() multiple times per packet or not? [10:56] <wm4> doc/examples/demuxing.c and ffplay.c disagree [10:57] <wm4> ffmpeg.c seems to do the same as ffplay.c [10:58] <wm4> basically, the question is whether this is still needed: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=ffplay.c;hb=HEAD#l2167 [10:59] <wm4> (not counting the need to finish decoding with null packets) [10:59] <nevcairiel> i believe you have to do that, it returns the number of consumed bytes and you have to handle that [10:59] <wm4> so doc/examples/demuxing.c is wrong? (or I read it wrong) [10:59] <nevcairiel> those examples are generally wrong or outdated :p [11:00] <wm4> ok my guess is nobody really knows [11:00] <nevcairiel> i'm quite sure i saw codecs where its mandatory [11:00] <nevcairiel> als i think [11:01] <nevcairiel> at least when playing a raw .als file where the frames arent pre-sliced [11:09] <wm4> hm from what I can see ffmpeg seems to support als in mp4 only [11:09] <nevcairiel> i'm certain i had a .als file once, but what do i know if it was secretly mp4 inside [11:10] <wm4> any other formats that could be affected? [11:11] <nevcairiel> realmedia codecs possibly [11:11] <nevcairiel> cook or atrac [11:12] <nevcairiel> i could quickly test a bunch of files if i were actually at home, but sadly that is not the case :d [11:17] <wm4> nevcairiel: tried some random rm samples, no "luck" [11:18] <wm4> randomly looking at some decoders in libavcodec I see how this could happen, though [11:29] <wm4> durandal_1707: since you seem to know about audio decoding, any clue whether this is still needed? http://git.videolan.org/?p=ffmpeg.git;a=blob;f=ffplay.c;hb=HEAD#l2167 [11:31] <durandal_1707> its for "retardet" codecs/containers [11:31] <durandal_1707> iirc used by shorten [11:31] <wm4> any samples? [11:32] <durandal_1707> and maybe adpcm in wav with block_align [11:32] <wm4> it's notable that doc/examples/demuxing.c seems to assume that a packet has to be decoded only once [11:43] <cone-395> ffmpeg.git 03Paul B Mahol 07master:3c5071db88b2: lavfi/psnr: rename 's' parameter to match documentation [11:52] <durandal_1707> michaelni: there are some bugs reported that you never fixed and now are in libav? [11:52] <durandal_1707> very "well" played [11:55] Action: michaelni doesnt understand [11:55] <durandal_1707> the security ones reported by google [11:57] <michaelni> we fix(ed) all security bugs [12:00] <durandal_1707> you must forgot some [12:00] <durandal_1707> or they are not sending samples too [12:04] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:0dd99628ea15: ff_lpc_calc_coefs: assert that the type is levinson or cholesky [12:04] <michaelni> durandal_1707, what you talk about ? [12:06] <durandal_1707> see http://lists.libav.org/pipermail/libav-devel/2013-July/048477.html [12:17] <wm4> durandal_1707: btw. if you know some samples that requires decoding a packet multiple times, that would be nice, and might even lead to either the demuxing.c example to be fixed, or maybe simplifying the API itself [12:18] <durandal_1707> wm4: you couldn't find shorted sample? [12:19] <durandal_1707> *shorten [12:19] <wm4> searching for it is kind of hard [12:19] <durandal_1707> it must be in fate [12:20] <durandal_1707> lossless-audio/luckynight-partial.shn [12:20] <durandal_1707> and if samples.ffmpeg.org/lossless something... [12:22] <wm4> hmmm [12:23] <wm4> indeed, this sample is affected [12:23] <wm4> thanks [13:25] <cone-395> ffmpeg.git 03Carl Eugen Hoyos 07master:9a1ec1f2f223: Force correct extradata for AVID AVCI50 in mov. [13:25] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:694550f18e02: Merge remote-tracking branch 'cehoyos/master' [14:10] <durandal_1707> michaelni: apad is not documented at all [14:57] <xlinkz0> god damn exit functions in ffmpeg tool [14:58] <xlinkz0> is it written in such a way that if i #define exit(x) it doesn't break? [14:59] <nevcairiel> read the code and find out? :) [14:59] <nevcairiel> what you're trying to do is not supported in any way, so don't expect it to be :D [15:00] <xlinkz0> well i got it compiled into a library [15:00] <xlinkz0> and it works.. once [15:00] <xlinkz0> because it then exit()s [15:01] <xlinkz0> 36 exits in ffmpeg.c [15:01] <xlinkz0> fml [15:56] <durandal_1707> michaelni: it looks like apad pads silence forever [15:57] <michaelni> durandal_1707, its intended to be used with -shortest so a audio stream can be padded to the langth of video [16:00] <durandal_1707> michaelni: shortest is ffmpeg only thing [16:15] <michaelni> durandal11707, yes, the filter was writen for the purpose of extending the audio to the same length as video with ffmpeg [16:16] <durandal11707> it should be extended to be less useless [16:26] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:7d9cb126893d: avcodec/wmaprodec: Check block_align [16:26] <cone-395> ffmpeg.git 03Luca Barbato 07master:87fb18c3e4fe: imc: Catch a division by zero [16:44] <xlinkz0> ffmpeg tool only cleans up after itself if no errors occur? [16:44] <xlinkz0> or does exit() by any chance trigger a callback? [16:50] <durandal_1707> michaelni: perhaps it still use stuff when calling .uninit [16:50] <durandal_1707> if i remove freep from it segv does not happen [16:50] <durandal_1707> i know uninit works as it does not segv on normal exit [16:52] <michaelni> xlinkz0, update your tree exit has been mostly replaced by exit_program() and yes ffmpeg used atexit() [16:52] <xlinkz0> was this in the last week? [16:54] <durandal_1707> michaelni: so you get imc report or not? [16:54] <durandal_1707> s/get/got [16:57] <xlinkz0> michaelni: thank you so much, it is like god looked upon my feeble attempts and granted me with a speckle of light [16:59] <xlinkz0> so i can just change exit(ret) from exit_program to pthread_exit [16:59] <xlinkz0> this.. can't be that easy [17:21] <cone-395> ffmpeg.git 03Paul B Mahol 07master:c5a1b18f183d: lavfi/drawtext: add support for printing frame metadata [17:23] <durandal_1707> now one can put psnr of videos with drawtext [19:05] <Compn> we should combine our network code with vlc [19:05] <Compn> so both projects dont have to reinvent the wheel every stream [19:05] <Compn> :P [19:10] <durandal_1707> Compn: everybody reinvents wheel [19:16] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:d5de1231954e: avfilter/graphparse: Fix build with --enable-incompatible-fork-abi [19:16] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:f9d7431272f4: avformat/AVFormatContext: Move fields down to match the fork [19:30] <michaelni> buxiness, do you see a problem with the jpeg2000dec patches ? [19:32] <durandal_1707> michaelni: what about questions I asked? [19:33] <michaelni> durandal_1707, about the division by 0 ? [19:33] <durandal_1707> about security issues that are still not fixed but reported [19:33] <michaelni> its possible it didnt happen for us because some other check stoped it before [19:33] <michaelni> iam not aware of any reported but not fixed issues [19:39] <cone-395> ffmpeg.git 03James Almer 07master:1e5bbbfcf303: lavf/flacdec: Share the function to parse cover art blocks [19:39] <cone-395> ffmpeg.git 03James Almer 07master:193782216fc8: vorbiscomment: Add support for embedded cover art [19:49] <cone-395> ffmpeg.git 03Piotr Bandurski 07master:aa74810fc6b9: bitstream: fix a crash in case of oom [19:54] <durandal_1707> saste: its just little confusing commit message [19:55] <saste> durandal_1707, i'm basically replacing the original text, and merging two sections in one [19:55] <saste> i have no problem to change the commit log [19:56] <saste> but the content as is is greatly confusing, the part about "thou shall publish the code" is a false boast [19:56] <durandal_1707> if they change something in lib itself, they MUST [20:02] <saste> durandal_1707, redistribute != punblish [20:02] <saste> you can modify and redistribute your copy only to the library user [20:02] <saste> you are not required to *publish* the code [20:03] <nevcairiel> if you make a build public, you need to make the changes in it public [20:03] <nevcairiel> what you do on your own PC is only your own business [20:03] <durandal_1707> no!!!!! [20:03] <saste> but you can have a single user [20:04] <durandal_1707> have GPLv3 fixed this? [20:04] <saste> for example I'm not required to publish all my experimental patches [20:04] <durandal_1707> saste: you must! [20:04] <iive> durandal_1707: not in gpl. [20:04] <saste> durandal_1707, that's absurd [20:04] <durandal_1707> all patches must be published [20:05] <saste> durandal_1707, no, that's not what LGPL states [20:05] <saste> also it is practically absurd, since you don't define what is "to publish" [20:05] <nevcairiel> you must publish source if you publish a binary [20:05] <iive> e.g. google doesn't have to publish its own in house code, and they use a lot of linux [20:05] <nevcairiel> this is basically all [20:05] <iive> nevcairiel: exactly. [20:05] <saste> you can encourage third parties who modifes the code to publish code (iff it is worthy to share that) [20:06] <durandal_1707> ah, so explain it in doc if they make stuff public they must comply and publish code too [20:06] <saste> comply?? [20:06] <saste> you can't enforce that [20:06] <nevcairiel> well technically you can [20:06] <nevcairiel> with lawyers [20:06] <saste> you even don't want to do that, also what is to publish? [20:07] <nevcairiel> its probably sufficient to supply the source on request, but IANAL [20:07] <saste> that's not LGPL, but funky LGPL [20:08] <saste> nevcairiel, so I add a change to print a count, I'm modifying the source code, so I must publish it? [20:08] <nevcairiel> it comes down to one major rule, if you distribute a binary, you must also make the source used to build this binary available [20:08] <saste> that's nonsense [20:08] <nevcairiel> thats how the (L)GPL works [20:09] <saste> i can modify the code and distribute it to my (single) customer [20:09] <saste> i'm required to provide him/her with the modified sourcecode, so he can change it [20:09] <saste> but i'm not required to publish the code to the known universe [20:10] <saste> nevcairiel, yes [20:10] <durandal_1707> saste: i'm going to hunt your patches [20:12] <nevcairiel> of course you're not required to publish it to the world, but anyone that has access to the binary can also ask for the source, more or less [20:12] <iive> and then he have the right to publish the source, if he wants. [20:13] <saste> nevcairiel, iive: yes [20:13] <michaelni> saste, about "thou shall publish the code", its not about legel req but what is the right thing to do, aka you improve it you should give back so all benefit [20:13] <saste> if i distribute a binary, i have to provide its source code [20:13] <saste> it's not like, i do a change, i must publish it => nonsense [20:14] <saste> michaelni, but don't confuse the license with ethics [20:14] <iive> I think that MPL is like that (Mozilla) [20:14] <saste> users are *encouraged* but not *enforced* to publish the modified source code (only if it makes sense) [20:15] <nevcairiel> you can't really enforce that and still be open-source, its also a bit about freedom of the developers [20:39] <johnsmith> Hi, I was wondering if anyone here has compiled/installed the latest version of FFMPEG onto CENTOS 5 (Work related reasons) [21:17] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:325411a687c4: snowenc: check minimum resolution [22:06] <wm4> how do I build the ffmpeg examples? FAQ says they are installed by default, but it installs the .c files [22:08] <saste> wm4, less README, make [22:08] <llogan> saste: ping on the PPC [22:09] <saste> llogan, ? i know nothing about PPC [22:10] <llogan> i guess you did time out then. i asked in IRC if you still wanted a PPC. i have an older, but still usable powerbook if you want it (if shipping isn't prohibitive). [23:16] <wm4> lol [23:17] <wm4> demuxing.c doesn't handle planar audio formats [23:17] <durandal_1707> example? [23:17] <wm4> how do you expect anyone to use this library if even the examples are highly buggy [23:30] <saste> wm4, demuxing was added before planar was introduced [23:30] <saste> so it was not even required back then [23:31] <wm4> does Libav not have examples? [23:31] <j-b> planar audio transition was the worse change ever done in libavcodec [23:31] <j-b> in terms of transition, not technical reason [23:32] <wm4> let's break all applications for fun! [23:32] <wm4> but the cause of the suck is deeper: decoders output whatever they want by design [23:32] <saste> wm4, the soluton is to use libswresample [23:32] <saste> in case of demuxing [23:32] <wm4> instead of returning the data in a format the user asks for [23:32] <wm4> i.e. too low level API [23:32] <saste> yet i feel the libswr API is still too much complex [23:33] <saste> wm4, i don't know we want that [23:33] <saste> at the same way decoders don't require the use of libswscale [23:35] <saste> eh two weeks is not "old" in the standard of ffmpeg [23:51] <durandal_1707> saste: using , as separator for delays need \\ escaping [23:51] <durandal_1707> i could use ' ' or '|' [23:51] <saste> durandal_1707, yes [23:51] <durandal_1707> so what separator? [23:52] <saste> | seems fine [23:52] <saste> not sure for the space, since it could imply other escaping issues [23:58] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:e9be4675b88a: doc/ffmpeg: document -override_ffserver [23:58] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:776155ca29e7: mjpeg_decode_scan: reset restart state [23:58] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:1c032e94d0a5: jpegdec: give handle_rstn() a return code that indicates if restart did happen [23:58] <cone-395> ffmpeg.git 03Michael Niedermayer 07master:233177a4ec68: jpegdec: clear EOBRUN on restarts [23:59] <saste> cehoyos: "fixed by you" is pretty useless [00:00] --- Wed Jul 10 2013

1 0

ffmpeg.log.20130709
by burek 10 Jul '13

10 Jul '13

[00:14] <semme> thanks got it running [00:24] <semme> hmm strange enough, but i don't see anything weird in the boxdumper output [00:31] <JEEBsv> even with the timestamp output? [00:31] <JEEBsv> see the timescale as well as the values [00:39] <semme> ohh i see it now [00:40] <semme> DTS = 0, CTS = 42 [00:40] <semme> but i have no idea how this value is so smal [00:41] <semme> *small [00:42] <JEEBsv> uhh [00:42] <JEEBsv> by itself just a small value says nothing [00:42] <JEEBsv> as I said, look at the timescale as well [00:47] <semme> Media timescale: 11776 [01:47] <semme> damn finally nailed it [04:07] <h6w> This might be a slightly OT question, but I have a webcam that I can connect to with v4l2 and supposedly outputs a raw mpeg2 stream. I also have an mpeg2 hardware decoder that does decode correctly as I've tested it with a VOB. [04:08] <h6w> However, when I try to pipe from one to the other, it tells me it sees the signal, but I just get a black screen. [04:09] <h6w> I get "Video codec omx-mjpeg width 640 height 480 profile -99 fps 30.000000". [04:36] <diegoviola> i tried to play a mkv file today in a home theater, and it didn't even read the file, so i converted it to avi, it read the file but told me "audio unsupported", so i converted the AAC audio to mp3, then it played the audio but not the video, so i converted the H.264 video to MPEG4, then it complained that the resolution was too big, then i just gave up [04:37] <diegoviola> why are these "home theater" manufacturers selling hardware with such limitations is beyond me [04:37] <diegoviola> maybe it's just some old device [04:37] <diegoviola> sorry for the off-topic [04:37] <klaxa> get a pi :X [04:37] <klaxa> or use an old laptop like i do [04:38] <klaxa> not too old though [04:38] <klaxa> either one with hardware accelerated playback or enough CPU power to decode videos in software [04:39] <diegoviola> i'll try with a pi, thanks :) [04:42] <klaxa> omxplayer doesn't handle softsubs /very well/ though [04:42] <klaxa> i think it handles them, but not so very well... something like that [04:44] <diegoviola> that's fine [04:44] <diegoviola> does mplayer or vlc runs fine on it? [04:44] <klaxa> not... really [04:44] <klaxa> only in software [04:44] <klaxa> that's why there is omxplayer [04:45] <diegoviola> oh.. [04:47] <diegoviola> ty [04:48] <klaxa> well i think it's okay if you only have .srt subs, with .ass with karaoke, dunno [04:49] <klaxa> hardware decoder decodes up to high profile of h264 [04:49] <klaxa> and if you buy the license MPEG2 too [05:02] <h6w> Yeah, but apparently the MPEG2 hardware decoding is in omxplayer and not in the libs, so you have to use it to get the hardware decoding. [05:02] <h6w> And omxplayer is almost completely undocumented. [05:26] <h6w> Does anyone know of an mpeg2 demo rtsp stream online somewhere? [05:56] <klaxa> how do i tell autoconf that i want to use a specific shared library for libx264 instead of the default one in /usr/lib ? [05:56] <klaxa> compiling ffmpeg [06:03] <relaxed> klaxa: --extra-cflags="-I/path/to/prefix/include" --extra-ldflags="-L/path/to/prefix/lib" [06:04] <klaxa> and it will prefer those over the ones in /usr/lib? [06:04] <relaxed> yes [06:04] <klaxa> awesome, thanks [06:17] <relaxed> klaxa: it will need to be in ld.so's search path or you can use LD_LIBRARY_PATH=/path/to/lib/dir [06:18] <klaxa> ah right, thanks would have forgotten about that and wondered why it wouldn't work ;) [06:21] <elkng> I use "ffmpeg version 1.2" and that command to convert file http://sprunge.us/VUUf and thats the output: http://sprunge.us/HigG, is something wrong with file ? I tryed mencoder, works without errors [06:22] <klaxa> >VUUf not found [06:24] <elkng> http://sprunge.us/JKKc that is what ffprobe tells about that video file [06:24] <elkng> http://sprunge.us/UMBM thats the command I use, working link [06:24] <elkng> klaxa: [06:25] <klaxa> you must be doing something wrong... >JKKc not found [06:26] <elkng> http://sprunge.us/SAaW that is what ffprobe tells about that video file, working link [06:26] <elkng> klaxa: [06:27] <klaxa> i... i'm sorry i tried with 2 browsers and wget but sprunge won't let me see that [06:27] <elkng> http://sprunge.us/SCPQ that should work, contains ffprobe output [06:27] <elkng> klaxa: [06:27] <klaxa> hmmm well that looks rather simple [06:28] <klaxa> oh wait [06:28] <klaxa> VP6A [06:28] <elkng> yes, what with that ? [06:28] <elkng> video codec ? [06:28] <klaxa> hmm... [06:28] <klaxa> >Unknown/unsupported AVCodecID V_MS/VFW/FOURCC. [06:28] <klaxa> can you run ffmpeg -codecs ? [06:28] <klaxa> although that might be not ffmpeg related but maybe a matroska issue [06:29] <elkng> klaxa: http://sprunge.us/XcQe output of "ffmpeg -codecs" [06:30] <klaxa> mencoder worked without problems you say? [06:32] <elkng> klaxa: yes, I use that command http://sprunge.us/QXEQ for mencoder [06:32] <klaxa> hmm... sorry, no clue, file a bugreport with a sample file maybe? [06:33] <elkng> I have mplayer-export-2012-04-22 and "mencoder --version" says: "MEncoder SVN-r34870-snapshot-4.4.4" [07:43] <t4nk829> hi [07:44] <t4nk829> Is it necessary to set format for ffmpeg when capture audio from usb sound card? [07:44] <t4nk829> for example: if my usb sound card support format:S24_3LE, is it necessary to tell ffmpeg the format S24_3LE? [07:46] <t4nk829> As I know the command :ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg [07:46] <t4nk829> it does not tell the format of audio from usb sound card [07:48] <klaxa> it does, it's alsa [07:49] <t4nk829> ? [07:49] <t4nk829> how ffmpeg know the format is S24_3LE? [07:49] <klaxa> alsa is the sound format of the audio stream from hw:1, which presumably is your usb soundcard [07:49] <t4nk829> yes, [07:50] <klaxa> magic? i would guess it reads it from the alsa framework at some point [07:50] <t4nk829> and in advance, how to set format in ffserver.conf? [07:51] <t4nk829> AudioCodec [07:51] <t4nk829> It seems AudioCodec can not set pcm_s24_3le [07:54] <klaxa> i not familiar with ffserver, it is poorly documented (afaik? if someone has good docs, they would be appreciated) and i have barely played around with it [07:56] <t4nk829> thanks, klaxa [08:51] <t4nk829> hi [08:52] <t4nk829> I got error when I try to open audio aound card [08:52] <t4nk829> the ffmpeg command line:ffmpeg -loglevel debug -f alsa -i hw:0 -ar 48000 -ac 2 -acodec copy -f video4linux2 -r 30 -s 1280x720 -input_format h264 -i /dev/video1 -vcodec copy http://localhost:8090/feed1.ffm [08:52] <t4nk829> the error is :cannot set sample format 0x10001 3 (Invalid argument) hw:0: Input/output error [08:52] <t4nk829> does anyone have ida? [08:53] <t4nk829> Is is necessary to assign format when I start ffmpeg? [09:49] <t4nk829> hi [09:49] <t4nk829> does anyone know how to set audio format for ffmpeg in command line? [09:50] <t4nk829> for example:I would like to set audio format is PCM S24_LE [09:50] <t4nk829> how to set the format in ffmpeg arguments? [09:53] <relaxed> t4nk829: -c:a pcm_s24le [09:54] <relaxed> `ffmpeg -codecs | less` [09:54] <relaxed> -f s24le (look at -formats too) [11:13] <Dheeraj_k> is it possible to pass duration to ffmpeg for how long it should record desktop? [11:17] Last message repeated 1 time(s). [11:44] <relaxed> Dheeraj_k: ffmpeg -i input -t $time ... [11:46] <someone-noone> ffmpeg -i test.mp4 -c:v libx264 -c:a copy -g 12 out.mp4 [11:46] <someone-noone> ^ ffmpeg always freezes on '-g' option [11:46] <someone-noone> what can it be? [11:46] <someone-noone> if I remove it, reencoding is fine [11:50] <mark4o> someone-noone: what version? [11:51] <someone-noone> N-49352-gc46943e [11:52] <mark4o> is it configured with git version of libvpx? [11:53] <someone-noone> quite right you're [11:53] <someone-noone> configuration: --enable-gpl --enable-libx264 --prefix=/usr --enable-libvpx --enable-libvorbis [11:54] <mark4o> does ffmpeg -h full go into an infinite loop also? michaelni checked in a fix earlier today, try git head [11:55] <someone-noone> ffmpeg -h works [11:55] <mark4o> -h full [11:55] <someone-noone> yes :) [11:55] <someone-noone> infinite loop [11:56] <someone-noone> is it fixed in yesterday version? [11:56] <someone-noone> or it was fixed today? [11:56] <mark4o> no, fixed 8 hours ago [11:56] <someone-noone> ok. thanks [12:16] <curfont> Hi guys, https://ffmpeg.org/trac/ffmpeg/ticket/2761#comment:4 - should I also open this bug for libav..? [12:17] <durandal_1707> if you want ... [12:18] <durandal_1707> and care it get fixed soner than later, but no promises... [12:18] <curfont> I guess people using zoneminder on ubuntu or debian would have the same issue otherwise [12:19] <durandal_1707> does it happens with libav too? [12:20] <curfont> yes [12:20] <curfont> "ffmpeg" in ubuntu is essentially libav now? [12:21] <JEEB> yes, debian-based distros use libav [13:38] <leandrosansilva> Hello. Is there in ffmpeg any tools which perform rgb/yuv -> hsl pixel conversion? I looked at ffmpeg source code but I couldn't find anything related. [13:55] <Mavrik> leandrosansilva, pixel conversion is done by libswscale [13:55] <Mavrik> check those calls. [13:57] <leandrosansilva> Mavrik, It seems libswscale doesn't support hsl :-( At least I couldn't find anything in pixfmt.h [13:58] <Mavrik> yeah, that would probably caused by non-existence of video formats supporting HSL. [14:02] <leandrosansilva> Ok, thx. I'll need to convert by hand or using openCV... Thx :-) [14:52] <mavetron> i am having trouble compiling ffmpeg from the compilation guide [14:53] <mavetron> the configure part of ffmpeg is not wroking [14:54] <mavetron> solved i just left out libopus [16:18] <beginner> hey all, I have a question... im streaming, but the resolution may changes, is there a way to "re-crop" somehow, or I must halt and restart the streaming anyway? [16:26] <beginner> then I ask in another way: can I stream video which automatic adapt the resoultion, even if it might changes? [16:41] <durandal_1707> michaelni: i get segv if i press q before ffmpeg finish filtering [16:42] <michaelni> durandal_1707, works fine here (even under valgrind) [16:43] <michaelni> but the exit/atexit change might be related to such bug [16:43] <durandal_1707> it happens with filter i work on, but not with apad [16:50] <michaelni> ahh ok [18:58] <DualC> Hi, i have a question.. i trie to compile ffmpeg on ubunto 11.04 after this explanation "http://trac.ffmpeg.org/wiki/UbuntuCompilationGuide". I have gone thru all the parts and starting with the configure script in ffmpeg dir, .. and it tells me it cant find OPUS, .. i can se the libopus.a / libopus.la files in ffmpeg_build/lib so what is wrong, ?? any ideas. [19:01] <curfont> did you put the correct configure? [19:03] <DualC> In ffmpeg u mean , .. sure i copyed it from the page.. [19:03] <DualC> guide* [19:05] <curfont> if you are using latest ubuntu DualC, use the package [19:05] <brontosaurusrex> relaxed, we have a problem. [19:06] <DualC> i use 11.04 [19:06] <DualC> but libopus compile well .. [19:07] <curfont> if all else fails install it to the normal place [19:07] <curfont> instead of the prefix [19:07] <curfont> :P [19:09] <DualC> Ah i will trie, .. what... do i need opus anyway ??, . i should install ffmpeg under serviio.. [19:16] <brontosaurusrex> trying to compile ffmpeg with x264, but keep getting "/home/b/Linux.Encoding.install/build.logs/ffmpeg.configure.log" [19:16] <brontosaurusrex> sorry, keep getting "ERROR: libx264 not found" [19:17] <brontosaurusrex> there is nothing special in x264 logs [19:18] <brontosaurusrex> and i've tryed at least 3 different approaches [19:23] <sasha-> O hai [19:23] <sasha-> Anyone know how to use ffmpeg to join two files together, transparently (no transcoding)? I tried ffmpeg -i File1.aac -i File2.mp4 copy FinalFile.mkv and it didn't work [19:24] <brontosaurusrex> sasha-, perhaps read on concat protocol [19:24] <Fjorgynn> perhaps do -c copy [19:25] <durandal_1707> sasha-: there is concat demuxer [19:25] <sasha-> http://pastebin.com/ZSnTRBJw [19:26] <sasha-> ^ that's the error I get [19:26] <durandal_1707> well first make sure that first file is not corrupted [19:27] <sasha-> oops& just found my mistake [19:28] <curfont> libavcodec/h264qpel.o is taking ages, is it suppose to? [19:28] <durandal_1707> taking ages what? [19:28] <sasha-> no, no, it's just that I exported the audio as mp4 and the video in aac [19:28] <curfont> to link [19:29] <durandal_1707> what os? [19:29] <curfont> make was going a long fine and its frozen at that object file [19:29] <curfont> ubuntu [19:29] <durandal_1707> what ffmpeg version? [19:30] <curfont> latest git [19:32] <durandal_1707> well it builds fine here... [19:32] <sasha-> Ok, I reextracted the audio and video files& still the same error [19:32] <curfont> might be a memory thing, looks full and swap is disabled [19:35] <curfont> there we go, stopped cassandra and its compliing now [19:35] <curfont> dat memory hog :( [19:35] <durandal_1707> cassandra? [19:37] <curfont> its a database thing that uses a lot of memory [19:38] <durandal_1707> database of what? [19:41] <zap0> data [19:42] Action: durandal_1707 *sigh [19:44] <curfont> durandal_1707: unrelated to ffmpeg, it's where I store CCTV logs and other stuff [19:50] <saste> ping on my doc/developer patch [19:50] <durandal_1707> saste: i replied to that [19:53] <tm512> does anyone know how to fix audio/video desyncs when recording with ffmpeg? [19:54] <tm512> I'm trying to stream, so I can't fix them after the fact [19:54] <tm512> -vsync, -async, -af aresample=async=1000, -copyts all do nothing [20:19] <rager> howdy folkks [20:19] <rager> is there a practical way to capture stderr off of some C that I'm running through NDK? [20:21] <tm512> freopen ("stderr.txt", "w", stderr); should workd, but what does that have to do with ffmpeg? [20:24] <rager> well, ffmpeg seems to be putting out progress data to stderr [20:24] <rager> or at least, it's making it into the logcat output [20:25] <johnsmith> Hi, I am having problems installing FFMPEG on CENTOS 5.9, and cannot work on CENTOS 6 for work related reasons, can anyone point me to the direction of how to install FFMPEG? http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide is for CENTOS 6, I am hoping someone has the resource to give me a modified version of this for CENTOS 5. [20:25] <rager> oh, poop [20:25] <rager> wrong place [20:25] <rager> johnsmith: ffmpeg is a nightmare on redhat-alikes [20:25] <rager> but it should be in the repos for 5.9? [20:25] <t4nk961> hi, i was trying to compile ffmpeg and x264 on mingw [20:25] <johnsmith> Where should I look for those repos? [20:26] <t4nk961> where exactly does ffmpeg look for x264 files [20:26] <rager> if you set the prefix, while configuring, then do make install [20:26] <t4nk961> i have my includes and libs in c/mingw/msys/1.0/local/ [20:26] <rager> and then set the prefix on ffmpeg, it should go to the given prefix to try to find dependencies [20:27] <t4nk961> okay, you mean while configuring x264 i need to set prefix [20:27] <t4nk961> and same the same prefix in ffmpeg configure right? [20:28] <rager> yes - that prefix being the toolchain environment you're building into, I'd expect [20:28] <rager> at least, that's my experience with android [20:29] <t4nk961> ok [20:30] <johnsmith> rager: Have you worked with RHEL5 FFMPEG before? Does FFMPEG for CENTOS 5 use the new methods (avcodec_open_input... etc)? [20:30] <rager> I've installed ffmpeg on amazon linux [20:30] <rager> both from centos 5.x repos and from source [20:31] <johnsmith> Does it use the new or deprecated methods? [20:31] <rager> I have no idea what those words mean in this context. [20:31] <johnsmith> i want to use the method [20:31] <rager> I'm missing an antecedent [20:31] <johnsmith> avcodec_decode_audio4(...) but it doesn't exist in a certain FFMPEG I am using (yum install ffmpeg ffmpeg-devel) [20:32] <johnsmith> Not sure if this is the most up to date version [20:32] <johnsmith> for CENTOS 5.9 [20:33] <rager> are you sure you need the most up-to-date stuff, or do you just "need" it because some newer tutorial says something about it? [20:33] <johnsmith> No, but I cannot find any previous iteration of avcodec_decode_audio4 [20:33] <rager> this is most of what I use to install ffmpeg [20:33] <rager> http://hastebin.com/lipubecume.hs [20:33] <rager> it probably wont work as is [20:33] <rager> but it's a basic start [20:34] <johnsmith> Seems to be the same as the centos compilation guide, but I'll give your script a run through [20:34] <rager> it pretty much is the exact same as that guide [20:34] <rager> and you're going to have to fix some pieces, I know for sure [20:41] <humbolt> when I am receiving an RTMP stream (or other streaming protocols) via ffmpeg, how can I make it to try to reestablish a connection if it is lost and to survive lost packages? [21:21] <llogan> johnsmith: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/APIchanges [21:22] <llogan> have you tried the Centos ffmpeg guide? maybe it will work on 5 just fine. [21:24] <llogan> rager: your script is not the same as the guide. [21:37] <tm512> anyone know how to sync audio and video while streaming? [21:37] <tm512> they seem to be out of sync at a random amount each time [22:11] <Fjorgynn> :D [22:13] <johnsmith> Does anyone know how to completely uninstall x264? [22:26] <llogan> johnsmith: depends on how you installed it. [22:26] <llogan> did you follow the Centos guide or that script from rager? [22:26] <rager> I wrote that script [22:27] <rager> after reading the centos guide [22:28] <rager> this is my more up-to-date one [22:28] <rager> http://hastebin.com/giyosuxebo.hs [22:29] <llogan> yours installs to the system. the guide "installs" to the user's home directory. i'm not sure if johnsmith noted this difference. [22:30] <llogan> also your old version was missing the --extra-libs="-ldl" which may have thwarted him [22:31] <rager> johnsmith: new script I found [22:31] <rager> bleh... just got done building for android [22:31] <rager> I'd like to figure a way to cut it down to just what I need, but I'm not sure what "just what I need" quite entails [22:31] <rager> since the app is now 30+mb [22:32] <llogan> cut what down? [22:35] <tm512> does anyone know how to stream with ffmpeg, without getting desyncs, or am I completely out of luck? [22:37] <tm512> I'm using http://hastebin.com/jisidabuba [23:14] <JodaZ> does anyone know a reasonably current (updated at least once or maybe twice a year xD) debian repo for a totally "unfree" ffmpeg ? [23:16] <durandal_1707> unfree can not be distributed, as its not legal [23:17] <durandal_1707> so you ask someone to distribute you undistributabel build on public channel [23:17] <Mavrik> that would be illegal? :) [23:17] <Mavrik> use static builds form download page for most current redistributable build [23:18] <JodaZ> download page ? [23:35] <beastd> http://ffmpeg.org/download.html [23:46] <llogan> JodaZ: !dl static [23:46] <llogan> duh... [00:00] --- Wed Jul 10 2013

1 0

ffmpeg-devel.log.20130708
by burek 09 Jul '13

09 Jul '13

[00:01] <Daemon404> guess ill tryy ffbisect [00:01] <Daemon404> inb4 it fails too [00:53] <cone-757> ffmpeg.git 03Paul B Mahol 07master:19436308b48b: dxa: make code independent of sizeof(AVFrame) [01:08] <cone-757> ffmpeg.git 03Paul B Mahol 07master:bd03f99d05d0: gif: do not set avctx->coded_frame [01:08] <cone-757> ffmpeg.git 03Paul B Mahol 07master:ec4e042f33f9: gif: remove unused AVFrame from encoder private context [01:17] <durandal_1707> michaelni: what release critical bugs remain? [01:29] <michaelni> durandal_1707, stuff marked as regression (minus unimportant ones) [01:29] <durandal_1707> all remaining are unimportant IMO [02:03] <cone-757> ffmpeg.git 03Paul B Mahol 07master:3bec6a44eab4: zmbvenc: do not set avctx->coded_frame [02:03] <cone-757> ffmpeg.git 03Paul B Mahol 07master:099e57bc38d7: zmbvenc: remove unused AVFrame from encoder private context [02:58] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:5312c319be98: avformat_find_stream_info: fallback to timestamps in analyzeduration check in more cases [03:51] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:5d876be87a11: avio: Handle AVERROR_EOF in the same way as the return value 0 [03:51] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:d35b6cd37754: rmdec: Use the AVIOContext given as parameter in rm_read_metadata() [03:51] <cone-757> ffmpeg.git 03Ronald S. Bultje 07master:6c86a63bad77: yuv4mpeg: Correctly round chroma up for odd luma sizes [03:51] <cone-757> ffmpeg.git 03Ronald S. Bultje 07master:46d208e1e097: vp8: Wait for prev_frame to parse segment_map before reading it [03:51] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:38e9585de993: Makefile: Remove stray tabs [03:51] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:7023715e8880: Merge commit '38e9585de993c32899588ab037180f2c930ce74c' [04:13] <cone-757> ffmpeg.git 03Martin Storsjö 07master:21732063a346: movenc: K&R formatting cosmetics [04:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:4cd066b69c08: Merge commit '21732063a346475eb22c332b27e8216b79f9ad4a' [04:47] <cone-757> ffmpeg.git 03Luca Barbato 07master:636ced8e1dc8: cmdutils: wrap exit explicitly [04:47] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:f982d006bb9e: Merge commit '636ced8e1dc8248a1353b416240b93d70ad03edb' [04:59] <cone-757> ffmpeg.git 03Luca Barbato 07master:c4abc9098cac: vqavideo: check the version [04:59] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:b5a87dc193d6: Merge commit 'c4abc9098cacb227dba39bac6aea16b2bceba0d0' [05:08] <cone-757> ffmpeg.git 03Luca Barbato 07master:76f5dfbfd902: westwood_vqa: do not free extradata on error in read_header [05:08] <cone-757> ffmpeg.git 03Luca Barbato 07master:f054e309c588: qdm2: use init_static_data [05:08] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:d28a8ad8f33d: Merge commit 'f054e309c58894450a5d18cce9799ef58aab9f14' [05:10] <highgod> Hi, I want to ask a quesion that I found a option in deshow of ffmpeg { "pixel_format", "set video pixel format", OFFSET(pixel_format), AV_OPT_TYPE_PIXEL_FMT, {.i64 = AV_PIX_FMT_NONE}, -1, AV_PIX_FMT_NB-1, DEC }, [05:10] <highgod> how can I input it? [05:10] <highgod> using command? [05:10] <highgod> thanks [05:11] <BBB> highgod: ffmpeg -pixel_format ... -i something [output options] outfile [05:16] <cone-757> ffmpeg.git 03Luca Barbato 07master:76efedeadb1f: qdm2: formatting cosmetics [05:16] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:e1f98f22313b: Merge commit '76efedeadb1f6bf79020c44a71dd0cee13d932ad' [05:27] <cone-757> ffmpeg.git 03Luca Barbato 07master:744a11c99664: qdm2: check and reset dithering index per channel [05:27] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:44d0e520f85a: Merge remote-tracking branch 'qatar/master' [07:08] <highgod> thanks [07:34] <highgod> Hi, I want to ask a question, I use 64bit ffmpeg libs and dlls, it cursh on my code, but I use the 32bit dlls and libs and using the same code ,the code run correct, can anyone give some help? thanks, my code is compiled by vs2010 [10:50] <michaelni> highgod, if something crashes, use gdb or another debugger to find out why/where [10:58] <highgod> Hi, michaelni, my systems problem, the same code run on other win7 64 is correct [11:22] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:efce76a80d26: ffmpeg_filter: replace exit() by exit_program() [11:22] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:eeddeb6579d8: ffmpeg_opt: replace exit() by exit_program() [11:23] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:5d3c3035b798: cmdutils: replace exit() by exit_program() [11:35] <xlinkz0> i have an AVCodecContext->extradata_size set to 475691104 [11:36] <xlinkz0> this is on ios, might this be because of the platform or maybe an older bug? [12:26] <cone-757> ffmpeg.git 03Carl Eugen Hoyos 07master:b86651a208ee: Check the return value of ff_rv34_decode_init() in rv30.c and rv40.c [13:16] <xlinkz0> how would i go about compiling the ffmpeg tool into a library? [13:17] <cptspiff> why on earth would you like to compile the tool as a library? [13:20] <xlinkz0> so i can use it on ios and android [13:21] <xlinkz0> sadly i can't understand complex makefiles :( [13:21] <xlinkz0> don't* [13:21] <cptspiff> well then you know where to start.. [13:23] <cptspiff> it will be a matter of writing a wrapper for main() [13:23] <cptspiff> plus probably renaming main.. [13:24] <saste> xlinkz0, don't do, unless you're a wizard [13:24] <saste> use system(..) instead [13:24] <cptspiff> no such thing on ios/droid. [13:24] <xlinkz0> indeed, that still relies on processes [13:25] <xlinkz0> i can't spawn processes [13:25] <xlinkz0> also wrapping it is only a matter of adding the thread local storage keyword in front of every global [13:25] <xlinkz0> shouldn't be hard [13:26] <xlinkz0> and compiling without threading support [13:26] <xlinkz0> cptspiff: but if i don't know how to compile it into an .so it is useless , any tips on that? [13:26] <cptspiff> uhm, -shared ? [13:27] <xlinkz0> what? [13:27] <cptspiff> http://gcc.gnu.org/onlinedocs/gcc/Link-Options.html [13:27] <xlinkz0> i need the full command ffmpegs makefile uses to compile ffmpeg.c [13:27] <cptspiff> make V=1 [13:28] <xlinkz0> sweet [13:28] <xlinkz0> thank you [14:24] <j-b> Good morning [14:24] <durandal_1707> good night [14:33] <durandal_1707> saste: are you porting any of mp filters? [14:33] <saste> durandal_1707, no [14:34] <durandal_1707> why? [14:35] <saste> why should i? [14:36] <durandal_1707> why should you not? [14:41] <durandal_1707> ubitux: fieldmatch is missing 440 [16:07] <ubitux> durandal_1707: please hold until august, or feel free to take over the filter until then [16:08] <durandal_1707> hold for what? [16:09] <durandal_1707> michaelni: diracdec is still using deprecated stuff [16:11] <durandal_1707> what's point of flatten attribute? it does not work for clang here and just spam [16:12] <durandal_1707> we will never get stable api [16:14] <durandal_1707> ubitux: stop watching anime [16:14] <ubitux> :( [16:15] <ubitux> honestly i'm not really available, and starting saturday i'll be even less [16:15] <ubitux> i come back in august for the vpx project, so back on ffmpeg [16:15] <ubitux> but until then that's not easy for me [16:18] <durandal_1707> saste: well then explain it to me why you using old drawutils in rotate filter? [16:19] <durandal_1707> there are 2 freaking drawutils and they are both suboptimal [16:26] <saste> durandal_1707, because I ported the code, and I didn't bother to port it to the new drawutils [16:29] <durandal_1707> saste: your AVERRROR(err); causes segv [16:30] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:1ba01d3d46e8: swscale: treat gray as a fullrage (0-255) format [16:30] <saste> that's because you have too many "R"s [16:32] <durandal_1707> :) no, that's typo above [16:33] <durandal_1707> why calling AVERROR() twice? [16:34] <saste> durandal_1707, err = AVERROR(errno); [16:34] <saste> now you call log, which could change errno [16:34] <saste> return err; [16:45] <durandal11707> for metadata i use lavfi.psnr.mse.r/g/b , lavfi.psnr.s.r/g/b , lavfi.psnr.mse & lavfi.psnr.s [17:03] <cone-757> ffmpeg.git 03Paul B Mahol 07master:6150bec3f873: lavfi: add psnr filter [17:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:e096283ea55b: mp3dec: detect CBR and use CBR axiom to seek [18:14] <cone-757> ffmpeg.git 03Alexandre Sicard 07master:70b922f371e2: avformat/mov: read stream bitrates from isml manifest [18:36] <durandal11707> saste: i move rotate to new drawutils [18:47] <saste> durandal11707, nice [18:48] <cone-757> ffmpeg.git 03Paul B Mahol 07master:d2dd1e075cf9: lavfi/pad: use FFSIGN [18:48] <cone-757> ffmpeg.git 03Paul B Mahol 07master:55c94f48ce5a: lavfi/drawutils: fix planar rgb [18:48] <cone-757> ffmpeg.git 03Paul B Mahol 07master:19c8f2271423: lavfi/rotate: switch to new drawutils [19:25] <durandal11707> saste: how should metadata be displayed with drawtext? [19:25] <durandal11707> some kind of expansion in text? [19:25] <saste> durandal11707, %{metadata:key} [19:26] <wm4> durandal11707: add lua scripting and let the scripts decide [19:28] <durandal11707> easier said than done [19:29] <wm4> true [19:29] <wm4> %{metadata:key} is a decent solution too... it's yet another kind of special syntax though [19:47] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:7dee9c22686f: tests/fate/filter-audio: update asyncts reference [19:56] <michaelni> did someone test if the msvc10 fate-filter-metadata-ebur128 is related to mmx/emms ? [20:00] <nevcairiel> its a bug in the compiler [20:00] <nevcairiel> the loop vectorizer does some wrong things when optimizing double variables in loops when sse2 is enabled [20:00] <nevcairiel> ...which you cant disable on x64 :) [20:02] <nevcairiel> MS ack'ed the bug, will be fixed sometime in the future [20:02] <nevcairiel> possibly VS2013 [20:06] <durandal11707> i can't get expr in drawtext expansion to use? [20:07] <durandal11707> and there are no such examples in doc [20:36] <durandal11707> text=%{localtime:%a %b %d %Y} [20:36] <durandal11707> doesn't work here [20:41] <mark4o> durandal11707: probably quoting issue [20:41] <mark4o> works: -vf "drawtext=fontfile=/Library/Fonts/DejaVuSans-Bold.ttf:fontcolor=red:text='%{localtime\:%a %b %d %Y}'" [20:42] <durandal11707> i use filter_script [20:42] <mark4o> then you don't need the shell quoting but still probably need backslash [20:47] <Daemon404> it's cause we use : in our parser [20:47] <Daemon404> it drives me nuts too (on windows) [20:47] <Daemon404> paths turn into a shitshoot [21:00] <wm4> quoting hell [21:02] <durandal11707> thing is metadata is added to frame in one filter, and in another filter its gone [21:04] <wm4> eh, AVFrame.metadata exists? [21:04] <wm4> weird [21:04] <durandal11707> why would that be weird? [21:04] <durandal11707> this worked when i did it manually in drawbox [21:14] <durandal11707> wtf AV_DICT_IGNORE_SUFFIX means? [21:31] <Daemon404> lets start that bisect again... [21:31] <Daemon404> maybe this time a transformer wont blow up near my house [21:44] Action: Daemon404 retries yet again [21:44] <Daemon404> finally figured out how to ue ffbisect [21:48] <wm4> huh, so ffbisect is a script created by another script? [21:48] <nevcairiel> its rather confusing [21:48] <Daemon404> wm4, it copies itself [21:49] <Daemon404> so you always have the proper version [21:49] <Daemon404> because it's also in git [21:49] <Daemon404> so a previosu commit may not have it [21:49] <Daemon404> you need to use it untracked [21:49] <Daemon404> hence the copy. [21:49] <wm4> that's... interesting [22:04] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:970c8df73528: mpeg12dec: avoid reinitializtation on PS changes when possible. [22:15] <cone-757> ffmpeg.git 03Paul B Mahol 07master:a03e79ed5a71: lavfi: fix broken logic in metadata handling [22:29] <durandal11707> i get this drawtext expansion working from script but from console there is no way [22:31] <durandal11707> Daemon404: didn't thing asyncts have be done in swr ? [22:32] <Daemon404> .. i cant parse that sentence [22:33] <Daemon404> seriously tho, you guys are a bunch of children. [22:34] <Daemon404> wm4, tl;dr to my email rpely to you ^ [22:34] <durandal11707> Daemon404: what asynts have? [22:34] <wm4> Daemon404: huh? [22:35] <Daemon404> wm4, re: why not enable avresample [22:35] <wm4> ah [22:35] <durandal11707> wm4 did replied? [22:35] <Daemon404> durandal11707, see my email [22:35] <Daemon404> i have explained what asyncts like 6 times [22:35] <Daemon404> and im fucking tired of it [22:35] <Daemon404> grep your goddamn logs [22:35] <Daemon404> ive explaiend it to YOU before [22:36] <durandal11707> i don't have logs, and you just act as giant douche [22:36] <Daemon404> because i can only explain somethin so many times [22:36] <Daemon404> before it gets on my nerves [22:36] <Daemon404> becaue everyone ha the memory of a goldfish [22:37] <Daemon404> what's that? i mention avresample? [22:37] <Daemon404> TIME TO INTERROGATE DEREK AGAIN [22:37] <Daemon404> every. single. time. [22:37] <durandal11707> i wanted to port it to swresample... [22:38] <Daemon404> if you recall, we collectively decided that was a bad idea [22:38] <Daemon404> and that aresample should improve instead [22:38] <Daemon404> fwiw, i see FAR more usage of avresample in the wild [22:38] <Daemon404> i actually know of no project using swresample [22:38] <Daemon404> even ffmpeg-only ones [22:39] <durandal11707> and michaelni added something to swresample, i think delay in the beginning of audio... [22:39] <durandal11707> i see no usage of avresample at all [22:39] <Daemon404> then you are blind [22:39] <Daemon404> lav filters, vlc, ffms2, mplayer2, mpv [22:40] <durandal11707> hilarious [22:40] <durandal11707> 3 of 5 don't use ffmpeg at all [22:40] <Daemon404> because clearly we shouldnt count those. [22:40] <Daemon404> cherry pick your stats etc [22:41] <wm4> I have ~35 lines of ifdeffery to make it work with both, but that's fugly [22:41] <durandal11707> since when we pick stuff becaues of "popularity" [22:41] <Daemon404> i am so tired of this fork sit [22:41] <Daemon404> shit* [22:41] <wm4> only now? [22:41] <Daemon404> wm4, if youre not on one side or the other [22:42] <Daemon404> the only option is to go insane [22:42] <Daemon404> and laugh at how childish everyone is [22:42] <wm4> heh [22:42] <Daemon404> also [22:42] <Daemon404> [16:40] <@durandal11707> 3 of 5 don't use ffmpeg at all <-- i think you mean 1 of 5 [22:42] <Daemon404> only vlc is libav only. [22:43] <Daemon404> at least get your facts straight. [22:43] <durandal11707> i see no facts [22:43] <wm4> lav filters is ffmpeg only, and I think mplayer2 is probably broken with recent git versions of libav [22:43] <Daemon404> ffms, mpv, mplayer2, and lav all can use ffmpeg. [22:44] <Daemon404> and in fact ffmpeg is the more supported of the two projects [22:44] <durandal11707> i don't see facts, i see only bad choices [22:44] <wm4> what does xbmc do? [22:44] <Daemon404> you fucking people -_- [22:44] <Daemon404> i am so fucking tired of thsi tribalistic bullshit [22:44] <wm4> lol [22:45] <iive> what is asyncts about? [22:46] Action: beastd could have predicted a quit [22:46] <durandal11707> stream timestamp syncronization [22:47] <durandal11707> it inserts silence in gaps iirc [22:48] <durandal11707> and it looks to me there is nothing similar when avresample is not enabled [22:49] <durandal11707> which is like bad, as this should not be lib limitation, even if it may appear as such [22:50] <durandal11707> and i don't see reason to have 2 libraries doing same thing enabled at same time [23:07] <michaelni> durandal11707, swresample should supports all that silence insert and time sync and stretching samples to compensate timestamp drift and ... [23:19] <durandal11707> should ... but does not ... [23:20] <michaelni> that would be a bug [23:28] <michaelni> also see "ffmpeg -help filter=aresample" for options to tune the async stuff with swresample [00:00] --- Tue Jul 9 2013

1 0

ffmpeg.log.20130708
by burek 09 Jul '13

09 Jul '13

[00:18] <Hafydd> Where are the -b, -vb, and -ab options of ffmpeg documented? [00:50] <brontosaurusrex> Hafydd, you mean -b:v and -b:a ? [00:59] <MachinaeWolf> Hmm tried someone's perfectly working ffmpeg on debian that they compiled and still an error: http://pastebin.com/g8mtuG62 [01:00] <djahandarie> Hmmm, does anyone know what controls this? "Aspect ratio is 1.38:1 - scaling to correct movie aspect." [01:01] <djahandarie> Because that results in some weird scaling, "VO: [xv] 1920x1080 => 1920x1394 Planar YV12", which I don't want. [01:01] <brontosaurusrex> djahandarie, aspect ratio flag of the input stream [01:01] <djahandarie> I crop and scale using -filter:v, does it not automatically correct the aspect ratio? [01:02] <djahandarie> (Those pastes were from mplayer on the output file.) [01:05] <djahandarie> Nevermind, error in the commandline, the crop wasn't getting applied properly. [01:10] <Hafydd> brontosaurusrex: I'm not sure, because I haven't seen the documentation yet. [01:11] <Hafydd> Let's just say I mean -b (or -b:a and -b:v, if you prefer), for now. [01:14] <brontosaurusrex> Hafydd, well, scroll around the docs for a while and you should spot it [01:15] <Hafydd> brontosaurusrex: which docs? [01:15] <brontosaurusrex> man ffmpeg [01:17] <Hafydd> I searched for "bitrate" and "bit rate" and "-b" in `man ffmpeg` and `man ffmpeg-filters` and `man ffmpeg-bitstream-filters`, to no avail. [01:17] <Hafydd> I'd be interested to see if you can find it there, and tell me what phrase to search for. [01:17] <Hafydd> (Because I also scrolled through all of the options and didn't see it) [01:22] <Hafydd> Or indeed anybody who knows were it's documented - my question still stands. [01:22] <brontosaurusrex> hmm, i have the html version of docs [01:23] <brontosaurusrex> and its on the main page [01:23] <brontosaurusrex> or the first page [01:24] <Hafydd> What does it say, brontosaurusrex? [01:25] <Hafydd> Ah, I found this ticket, actually: http://trac.ffmpeg.org/ticket/613 [01:25] <Hafydd> Maybe it was fixed in a later version than I have. [01:28] <brontosaurusrex> dunno, they keep changing cli all the time is all I know [01:28] <Hafydd> So it's in the output of ffmpeg -help, but not the manual page. [01:29] <Hafydd> How could such a basic option go poorly documented for so long. Maybe I should open another ticket. [01:33] <Hafydd> brontosaurusrex: what does your HTML documentation say about -b? [01:34] <brontosaurusrex> Hafydd, i have you a link [01:34] <brontosaurusrex> *gave [01:35] <Hafydd> brontosaurusrex: you did no. [01:35] <Hafydd> *not [01:37] <Hafydd> Oh, excuse me, my client didn't notify me of the notice. [01:37] <brontosaurusrex> relaxed, running your Linux.Encoding.install [01:38] <Hafydd> Where on that page does it describe the -b option, brontosaurusrex? [01:38] <brontosaurusrex> relaxed, it would be cool to add fdk-aac [01:38] <brontosaurusrex> Hafydd, no, the -b:v option and -b:a [01:39] <Hafydd> That's what I mean by "the -b options". :v and :a are additional arguments to an option name, [01:39] <Hafydd> So, where are they described? [01:41] <brontosaurusrex> Hafydd, wtf is wrong with you? [01:41] <brontosaurusrex> i told you 5 times allready [01:42] <Hafydd> All you've told me are a number of false statements. [01:42] <brontosaurusrex> Hafydd, ok, just put me on ignore, ok? [01:42] <Hafydd> I appreciate the effort to help, but I've put more effort into trying to find this than you think. [01:43] <Hafydd> So if there actually isn't any documentation, it's better to admit that. [01:43] <brontosaurusrex> Hafydd, iam not a ffmpeg dev, just a user like you [01:43] <brontosaurusrex> Hafydd, so i have really nothing to "admit" [01:44] <Hafydd> You can admit that you didn't even look in the documentation before wasting my time directing me to it. [01:49] <brontosaurusrex> i want my lawyer [02:25] <brontosaurusrex> trying to compile ffmpeg with x264 and i constantly get : ERROR: libx264 not found [02:25] <brontosaurusrex> i've tryed all different possible approaches, including relaxed superscript [02:25] <brontosaurusrex> always the same error [02:28] <brontosaurusrex> any clues? [02:28] <drv> check config.log and see why the test fails [02:32] <brontosaurusrex> ok [02:34] <brontosaurusrex> same thing at the bottom : ERROR: libx264 not found [02:34] <brontosaurusrex> i guess for some reason libx264.so is not generated [04:42] <mapreduce> Hi. I can't seem to build the most recent FFmpeg on msys (Windows) but it builds ok on Linux. [04:42] <mapreduce> make gives: common.mak:139: *** missing separator. Stop. [04:43] <mapreduce> line 139 of that file is $(eval $(RULES)), looks reasonable to me but I don't know make very well. Any suggestions? [04:59] <mapreduce> Nevermind, I think git config core.autocrlf false fixed it. [07:04] <nownot> trying to convert flac to m4a and I keep getting Input stream #0:0 frame changed from rate:44100 fmt:s16 ch:2 chl:stereo to rate:44100 fmt:s16p ch:2 chl:stereo [07:04] <nownot> using command for f in *.flac; do avconv -i "$f" -c:a alac "${f%.*}.m4a"; done & ideas? [07:32] <highgod> Hi, I want to ask a question, I use 64bit ffmpeg libs and dlls, it cursh on my code, but I use the 32bit dlls and libs and using the same code ,the code run correct, can anyone give some help? thanks [07:35] <robpitt> Hi, how can I use ffmpeg to capture audio from the microphone on android? [08:50] <igmrlm> g'day [08:51] <igmrlm> what would be the command line to generate random video noise with ffmpeg, like the snow from a tv set? [08:51] <igmrlm> apparently one could use the noise filter but i'm unsure the syntax [08:52] <igmrlm> i found a site that tells how to do this from /dev/urandom but i'm not on a linux machine at the moment [11:10] <mark4o> igmrlm: ffmpeg -f lavfi -i "nullsrc=s=640x480,geq=random(1)*255:128:128" ... [11:11] <mark4o> the filters are documented at http://ffmpeg.org/ffmpeg-filters.html [12:18] <robpitt> On android, how can I use ffmpeg to record the microphone? [14:31] <boys21> hi, Im trying to enable libx264 with ffmpeg in Mingw environment, but I keep getting error: libx264 not found [14:32] <boys21> i have x264 lib. include and bin folders in [14:32] <zap0> is your mingw build system really an ffmpeg problem? [14:33] <boys21> can someone help me with suggesstions [14:34] <boys21> no i guess, i compiled ffmpeg with out x264 more than thrice, but I get troubles enabling x264 [14:35] <Mavrik> boys21, check config.log [14:35] <Mavrik> and see why ffmpeg thinks you have no libx264 [14:37] <boys21> ffconf.wpjIoPmL.o_converted.c c99wrap link -L/usr/local/lib -nologo -out /tmp/ffconf.DdtrQknc.exe /tmp/ffconf.wpjIoPmL.o x264.lib psapi.lib advapi32.lib shell32.lib LINK : warning LNK4044: unrecognized option '/LC:/MinGW/msys/1.0/local/lib'; ignored LINK : fatal error LNK1181: cannot open input file 'x264.lib' ERROR: libx264 not found [14:37] <boys21> these were my last lines [14:41] <Mavrik> is x264.lib stored in your MinGW/msys/local/lib? [14:43] <boys21> yes i have x264.a [14:44] <boys21> i dont have x264.lib [14:44] <Mavrik> hmm [14:44] <Mavrik> .lib is usually extension for MSVC generated static libraries [14:44] <Mavrik> .a for gcc generated static libraries [14:45] <boys21> okay [14:45] <Mavrik> did you compile your x264 with mingw beforehand? [14:45] <Mavrik> and are you compiling ffmpeg with MSVC or mingw gcc? [14:45] <boys21> i compiled x264 before hand [14:45] <boys21> with mingw [14:45] <boys21> and now im compiling ffmpeg with msvc with x264 option [14:46] <Mavrik> ah [14:46] <Mavrik> compile it with gcc. [14:46] <Mavrik> and it'll look for the .a library [14:46] <boys21> okay [14:46] <boys21> how can i port them to msvc later? [14:49] <Mavrik> no idea [14:50] <boys21> ok, thanks anyways. I will give it a try in this direction. [14:50] <Mavrik> boys21, depends on how you want to use them [14:50] <boys21> it worked disabling the msvc [14:50] <Mavrik> I had no problems using gcc compiled ffmpeg/x264 .dll from MSVC and python projects [14:50] <boys21> ok [14:50] <Mavrik> for static linking you should probably compile everything with MSVC [14:51] <boys21> previously, i used the msvc tool chain option and used the staticlibs in the additional dependencies to compile ffmpeg projects [14:52] <boys21> i mean the additional dependencies of project properties in vs 2012 [15:02] <boys21> Thanks for suggestion, i wil try to recompile libx264 using msvc tool chain option [15:56] <robpitt> How can I record microphone audio in android? [16:08] <zap0> use android api [16:20] <robpitt> zap0 I'm not an android coder is there not some way i can do it on the command line with the ffmpeg binary? [16:21] <zap0> i dont know. i suspect you'd find it in a simple google query; if not, then probably not. [16:22] <robpitt> I've been googling for hours so maybe it can't be done. I see references to using a pipe but don't know how to make one [16:23] <zap0> you do know that access to hardware goes via the permissions system [16:24] <zap0> so unless its an obvious device you can open linux style, you likely need permissions to access the hardware [16:24] <Mavrik> robpitt, you cannot. [16:24] <Mavrik> microphone is especially protected on Android [16:25] <Mavrik> on most devices it's not possible to record conversations even with root access [16:25] <robpitt> Shame. I've got root and still can't do it [16:26] <zap0> apps can acquire permissions. so API's can access it. [16:28] <robpitt> There's a patched ffmpeg that has a device called lsfaudio but I really wanted it to work with common builds of ffmpeg [16:30] <zap0> it's not a ffmpeg issue. teh platform doesn't allow it [16:36] <robpitt> ffmpeg could do it though, the patched version proves that [16:38] <zap0> that doesn't prove `FFMPEG` is doing it... it proves some android specific API stuff has been grafted in. [16:38] <zap0> you said yourself, you wanted a `native` ffmpeg to work. not some franken-ffmepg [16:39] <JEEB> robpitt, if you want to see the feature get into ffmpeg, then feel free to grab the patch, see that its license is LGPL and not GPL, and then update it to the current ffmpeg master -> post on the mailing list [16:40] <JEEB> (and lurking around ffmpeg-devel is a good idea) [17:04] <robpitt> I don't have a PC so I tried to compile it on my phone but it requires the android ndk. Sigh. [17:55] <curfont> Hi I am trying to save an RTSP stream with ffmpeg, and it works in VLC but I can't find the arguments to use in actual ffmpeg [17:56] <curfont> It's an IP CAM streaming h264 over RTSP/554 [17:56] <Fjorgynn> ffmpeg -i ? [17:56] <curfont> Doesn't work, neither do examples I find online [17:56] <curfont> It says rtsp://192.168.0.220:554/live/h264/ch0: Invalid data found when processing input [18:51] <diegoviola> hi [18:54] <diegoviola> i have a home theater device that is not reading 720p mkv files, but it reads fine xvid videos with lower quality/format/res, is there a way to change the container format from the 720p files to avi, so the home theater would read them? [18:54] <diegoviola> without reencoding the audio/video in the process [18:56] <curfont> Ok guys, found the problem, this camera does not accept the "DESCRIBE" in the RTSP protocol that is sent by ffmpeg [18:58] <curfont> But it is fine with the one sent by LVC [18:58] <curfont> VLC* [19:03] <xlinkz0> diegoviola: -c copy out.avi ? [19:04] <diegoviola> can avi carry H.264 video and AAC audio? [19:05] <diegoviola> xlinkz0: thanks [19:05] <xlinkz0> one way of finding out is testing it :) [19:05] <diegoviola> right, thanks [19:05] <diegoviola> i'll give it a try [19:24] <curfont> Ok, so this specific camera needs a User-Agent in the DESCRIBE, is there any way to get ffmpeg to add a User-Agent to the RTSP packets? [19:58] <curfont> fyi https://ffmpeg.org/trac/ffmpeg/ticket/2761 [20:27] <diegoviola> crap [20:27] <diegoviola> this home theater won't accept H.264 also [20:27] <diegoviola> nor AAC [20:27] <diegoviola> not even mkv [20:27] <diegoviola> it seems to only accept avi/xvid/mp3 [20:28] <diegoviola> 2-pass encoding only affects file size or also quality? [20:28] <diegoviola> when it comes to xvid [20:28] <Fjorgynn> :( [20:29] <Plorkyeran> xvid doesn't have a good 1-pass mode, if that's what you're asking [20:29] <Fjorgynn> 2 pass affects the bitrate [20:30] <diegoviola> Plorkyeran: xvid is 2-pass encoding always? [20:31] <Plorkyeran> you can do 1-pass encodes, but it'll be lower quality at the same filesize as a 2-pass encode [20:31] <diegoviola> ok [20:32] <diegoviola> ty [20:33] <LithosLaptop> diegoviola: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20encode%20Xvid%20/%20DivX%20vi… [20:34] <diegoviola> is it best to use the xvid encoder or the native mpeg4 ffmpeg encoder? [20:35] <diegoviola> i'll try the ffmpeg encoder first [20:37] <Fjorgynn> :) [20:38] <diegoviola> ty folks [20:58] <MrE> Hey ive used this guide http://trac.ffmpeg.org/wiki/How%20to%20burn%20subtitles%20into%20the%20video for burning picture based dvb subtitles onto the video, it works quite good, but when I use a 720p hd source as input without rezise or anything else the subtitles is small and placed in the middle of the picture, using 1080p hd as source and resizing to 720p its even worse. Here is some samples : [20:58] <MrE> http://10only.explode.dk/screenshot003.png [20:58] <MrE> http://10only.explode.dk/screenshot004.png [20:59] <diegoviola> wth? [20:59] <diegoviola> Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height [20:59] <diegoviola> mpeg4/xvid requires a specific width/height? [20:59] <diegoviola> could i preserve the same 720p quality of h264? [20:59] <JEEBsv> see the lines before that [20:59] <MrE> Tried serveral options with overlay=main_w and all that, nothing seems to work, any help would be appreciated :) [20:59] <JEEBsv> that should contain the actual error [21:00] <JEEBsv> the "maybe incorrect parameters" thing just means that you will probably want to look at the previous lines for an error [21:10] <diegoviola> http://pastie.org/8122051 [21:11] <diegoviola> bitrate too low i guess? [21:11] <diegoviola> well it seems like it [21:11] <diegoviola> tried 800k and it works now [21:12] <diegoviola> but my first pass was with 555k [21:13] <diegoviola> i'll try again with 1k both passes [21:55] <someone-noone> Hello! [21:55] <someone-noone> ./configure --enable-gpl --enable-libx264 --prefix=/usr --enable-libopus --enable-libvpx [21:56] <someone-noone> ERROR: opus not found [21:56] <someone-noone> opus headers are in /usr/include/opus and /usr/include [21:56] <someone-noone> opus libs are in /usr/lib/ [21:56] <someone-noone> in config.log last 2 lines are: [21:56] <someone-noone> check_pkg_config opus opus_multistream.h opus_multistream_decoder_create [21:56] <someone-noone> ERROR: opus not found [21:57] <someone-noone> What is going on and how to fix it? [21:57] <someone-noone> ffmpeg version is latest from git [23:29] <semme> guys i've got a really strange problem: im remuxing an h264 f4v to mp4 (copying the streams) [23:29] <semme> and the output is ok (picture, sound), but the output's fps are 320fps [23:30] <semme> im doing r_frame_rate = av_d2q(_fps, INT_MAX); [23:31] <JEEBsv> you should be using PTS with such containers, really [23:31] <semme> i am [23:31] <semme> im copying the pts/dts values from the FLV, but thats not working either [23:32] <semme> is there a diff between MP4/FLV pts/dts? [23:33] <JEEBsv> there's o9k things you could be doing wrong there, and I have no idea what you might be doing wrong :P use L-SMASH's boxdumper or something to look at the CTS (almost PTS in "MP4" lingo, but not exactly) values of your samples [23:33] <JEEBsv> so you can see if you have the correct X/Y rate, as in if your PTS are correct on the timescale [23:34] <semme> ok thanks, im going to double check the pts/dts values [23:35] <JEEBsv> and as I said, you can use L-SMASH's boxdumper tool to dump a list of CTS values for the samples [23:35] <JEEBsv> of the output mov-like ("mp4" etc.) [23:39] <semme> might be a dumb question, but is l-smash tested on osx? [23:43] <semme> hmm guess not [23:44] <JEEBsv> it is [23:44] <JEEBsv> compiling it should be rather straightforward [23:45] <JEEBsv> it doesn't really require anything [23:45] <JEEBsv> ./configure with possibly maybe --cc=clang or something if you want to use clang [23:45] <JEEBsv> and then make [23:46] <mark4o> works for me on os x [23:47] <JEEBsv> yes, there have been people using OS X related to the L-SMASH development, and in general it's just so simple it'd be surprising if it'd broke [00:00] --- Tue Jul 9 2013

1 0

ffmpeg-devel.log.20130707
by burek 08 Jul '13

08 Jul '13

[00:00] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:9a6e814be7c0: oggparseskeleton: avoid header parsing failure [01:54] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:abce6dfd9ece: avcodec/x86/vp3dsp_init: move mmx functions under HAVE_MMX_INLINE [01:54] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:707b2135fda3: libavcodec/x86/mpegvideo: Move mmx functions under HAVE_MMX_INLINE [01:54] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:104dfaadaa0b: dct_unquantize_h263_intra_c: fix assert condition [02:03] <llogan> what's the status/plan of ffmpeg-devel(a)mplayerhq.hu? i noticed carl still sends to it, so it still works. [02:06] <Compn> is it the original mail addy ? [02:06] <Compn> i dont remember ... [02:06] <llogan> i believe so [02:06] <Compn> well then we leave it working in case someone wants to respond to a super old mail :P [02:07] Action: Compn has no idea really [02:07] <Compn> is there any reason to get rid of it ? [02:08] <llogan> i can't think of any good enough reasons. [02:08] Action: llogan is also ignorant [04:17] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:dc2a13aa802f: avformat/mov: Fix duration of fragmented mov [10:59] <cone-757> ffmpeg.git 03Luca Barbato 07master:8f689770548c: kmvc: use fixed sized arrays in the context [10:59] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:1aeddc3cbfe7: Merge commit '8f689770548c86151071ef976cf9b6998ba21c2a' [11:20] <cone-757> ffmpeg.git 03Luca Barbato 07master:4e7f0b082d8c: kmvc: Clip pixel position to valid range [11:20] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:2adf8a9c09c9: Merge commit '4e7f0b082d8c4b360312216b9241bec65ff63b35' [11:35] <cone-757> ffmpeg.git 03Luca Barbato 07master:73142e753381: fifo: K&R formatting cosmetics [11:35] <cone-757> ffmpeg.git 03Diego Biurrun 07master:6516632967da: tests: Only run noproxy test if networking is enabled [11:35] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:7bd417c22a90: Merge commit '6516632967da5e6bd7d6136e8678f826669ed26e' [11:42] <cone-757> ffmpeg.git 03Diego Biurrun 07master:cd7b6deeaaf1: tools: Wording and formatting cosmetics [11:42] <cone-757> ffmpeg.git 03Luca Barbato 07master:a4baf96abd64: avprobe: rename the prettyprint structures [11:42] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:b7ce51000327: Merge commit 'a4baf96abd64cda68e3e4692b182bdf60745a14d' [11:53] <cone-757> ffmpeg.git 03Luca Barbato 07master:f5e646a00ac2: wtv: Mark attachment with a negative stream id [11:53] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:b9d55a07f3cd: Merge remote-tracking branch 'qatar/master' [12:01] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:a71832f826bb: avformat/Makefile: remove stray tab [12:27] <superware> michaelni: hi [12:27] <michaelni> superware, hi [12:30] <superware> Kyle (zeranoe) is asking "Can you explain to me what aviocat does?" [12:54] <superware> michaelni: I know you're busy, but let me ask when will you have a few minutes to receive the udp stream itself? [12:55] <michaelni> superware, sure but whats the point if i dont have hours time to debug it ? [12:57] <michaelni> aviocat: "a tool that uses avio to read and write, doing a plain copy of data" (from log) [12:57] <superware> yeah, I just told him [13:00] <superware> Since the TS itself is playing normally, and since the stream source is sending quite large datagrams, the bug is in the way ffmpeg is handling incoming data/buffer. I don't think hours will be required, I just think it worth a shoot if you might be able to spot the problem in a few minutes. [13:20] <superware> michaelni: what are you suggesting? [13:21] <superware> by the way, I really appreciate your help so far. [13:22] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:4cc32e5cad98: avformat/mp3dec: read xing toc independant of usetoc, only skip filling index if requested [13:22] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:19d416333114: avformat/mp3dec: read TOC even if not all needed information is available [13:24] <michaelni> superware, like i said i dont really have time but if you want to send that udp stream i dont mind [13:25] <superware> great, can I please /msg you? [13:25] <michaelni> sure [16:25] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:c7cfd4cce98a: avformat_find_stream_info: use fps_dts as last resort to determine duration [16:25] <michaelni> superware, should be fixed [16:35] <superware> michaelni: you found it?!! [16:45] <michaelni> superware, yes [16:45] <superware> thanks, well done! where's the problem in code? [16:51] <superware> do you still need the stream? :) [16:55] <michaelni> superware, no you can stop the stream [16:56] <superware> ok. if you'll require it further to test the patch, please tell me. can you direct me to the problematic code [16:56] <superware> ? [17:05] <michaelni> superware, see last commit [17:24] <superware> ok, thanks [17:29] <durandal_1707> what? ffmpeg doesn't suppport 16bit PCM DVD audio? [18:59] <cone-757> ffmpeg.git 03Michael Niedermayer 07master:be25614177ef: mmsh: reimplement seeking [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.10:8514e3e08e23: rmdec: Pass AVIOContext to rm_read_metadata() [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.10:c25c89a53095: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.10:e445dc9237ae: avformat/mov: Fix duration of fragmented mov [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.11:139005452d0f: rmdec: Pass AVIOContext to rm_read_metadata() [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.11:cbbfdcf9ee1c: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.11:ec882942ce45: avformat/mov: Fix duration of fragmented mov [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.7:dc61d4484269: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.8:c997dcd38be7: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.9:0ca658a71659: rmdec: Pass AVIOContext to rm_read_metadata() [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.9:d6641c0bab21: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:5fa940d9c71e: rmdec: Pass AVIOContext to rm_read_metadata() [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:ee829bacdeb5: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:8c5c4adebbf6: libavcodec/x86/mpegvideo: Move mmx functions under HAVE_MMX_INLINE [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:e68b62a014b6: avformat/mov: Fix duration of fragmented mov [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:93fc80f8bfb3: avcodec/x86/dsputil_init: only use xvid idct for lowres=0 [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:af95e174c5d9: rmdec: Pass AVIOContext to rm_read_metadata() [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:b186a5d08cde: mpegts: only reopen pmt_cb filter if its different from the previous. [19:13] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:25ed0f05fdc3: libavcodec/x86/mpegvideo: Move mmx functions under HAVE_MMX_INLINE [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:ba8d68462228: avformat/mov: Fix duration of fragmented mov [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:99036565ca8b: avcodec/x86/dsputil_init: only use xvid idct for lowres=0 [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:b88bea1d6c8a: rmdec: Pass AVIOContext to rm_read_metadata() [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:cb340ecd7d96: mpegts: only reopen pmt_cb filter if its different from the previous. [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:5effd65312dd: avcodec/x86/vp3dsp_init: move mmx functions under HAVE_MMX_INLINE [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:5ce57e0248ec: libavcodec/x86/mpegvideo: Move mmx functions under HAVE_MMX_INLINE [19:14] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:627772d9886f: avformat/mov: Fix duration of fragmented mov [19:57] <Daemon404> sure is commit spam [19:58] <durandal11707> 8 concurent releases [20:06] <durandal11707> you must had lobotomy [20:36] <Daemon404> durandal_1707, +1 [20:36] <Daemon404> ive been complaining about it for ages [21:09] <durandal11707> wtf: https://trac.ffmpeg.org/attachment/ticket/2255/%C3%89N1.jpg [21:11] <durandal11707> michaelni: can you remove this "art" [21:11] <durandal11707> and ban that uploader with lack sense of humor [21:11] <Daemon404> whats wrong with it? [21:11] <durandal11707> its not ffmpeg art [21:11] <Daemon404> i guess i lack context [21:12] <durandal11707> look at it.... [21:12] <Daemon404> yes but an image along withotu context means nothing [21:12] <Daemon404> s/along/alone/ [21:12] <Daemon404> the ticket provides context [21:13] <Daemon404> which i have now read [21:13] <durandal11707> Daemon404: do you want pose for ffmpeg banner? [21:15] <Daemon404> im flattered [21:15] <Daemon404> how much does it pay? [21:18] <durandal11707> hmm, gonna ask saste [21:21] <durandal11707> michaelni: can same psnr code be used for audio? [21:22] <michaelni> in principle yes [21:22] <durandal11707> i'm thinking this filter could be improved a lot, added simd and made slice threading support, so it could be used in fate [21:22] <Daemon404> i dislike that [21:22] <Daemon404> it creates a hard dep on lavfi for FATE [21:23] <durandal11707> lavfi everywhere! [21:49] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/0.11:e452e4ca5cdb: mmsh: dont close context on seeking failure [21:49] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.0:40c521aec0ed: mmsh: dont close context on seeking failure [21:49] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.1:ce74b92c0981: mmsh: dont close context on seeking failure [21:49] <cone-757> ffmpeg.git 03Michael Niedermayer 07release/1.2:430ef8a7166b: mmsh: dont close context on seeking failure [23:39] <BBB> Daemon404: I think some people consider lavfi the best thing since sliced bread [23:40] <Daemon404> indeed [23:40] Action: Daemon404 continue his 4000-commit bisect into ffmpeg [23:41] <BBB> I bet you the resulting diff will be a massive merge commit [23:41] <BBB> gl figuring that out [23:41] <Daemon404> wouldnt be the first time [23:41] <Daemon404> im using a work server w/ 32 cores and 100 gb of ram... speeds things up a bit [23:42] <Daemon404> but configure is still dog slow [23:42] <Daemon404> whole lib builds instancely, except for dsputils [23:42] <Daemon404> which sits and crunches [23:42] <Daemon404> instantly* [23:55] <Daemon404> 1e8b9738fa70e20967ddb542d2f9d5552fc51ec6 is the first bad commit [23:55] <Daemon404> commit 1e8b9738fa70e20967ddb542d2f9d5552fc51ec6 [23:55] <Daemon404> Author: Hendrik Leppkes <h.leppkes(a)gmail.com> [23:55] <Daemon404> Date: Fri Mar 22 15:13:42 2013 +0100 [23:55] <Daemon404> what the fuck [23:55] <Daemon404> this doesnt make sense [23:55] <Daemon404> this is from libav [23:55] <Daemon404> and ths p roblem is ffmpeg-specific [23:55] <Daemon404> goddamn merge shit [23:56] <wm4> lol [23:57] <Daemon404> >_< [23:58] <wm4> I'm amazed that it still goes so well [23:58] <wm4> such a big codebase, maintained separately for such a long time [00:00] --- Mon Jul 8 2013

1 0

ffmpeg.log.20130707
by burek 08 Jul '13

08 Jul '13

[00:17] <miguel_> when I try to use ffmpeg to record, the audio and video recording start at different times [00:17] <miguel_> can I make it so it only starts after an amount of seconds? [03:35] <fling> Hello! [04:20] <silv3r_m00n> hi there, i am recording desktop with ffmpeg, using this command, ffmpeg -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -vcodec libx264 -preset ultrafast -crf 0 -threads 0 -y screencast_out.avi [04:20] <silv3r_m00n> how can i pause and resume the recording ? [04:29] <fling> this works great > ffmpeg -f concat -i list -c copy output.avi [04:29] <fling> ffmpeg rocks! [04:29] <fling> silv3r_m00n: try sigstop but I may be wrong [04:30] <fling> silv3r_m00n: or capture to separate files and join them together afterwards [04:48] <silv3r_m00n> seems like have to concat multiple files [05:35] <uersbob> hi. i have a question about ffmpeg. [05:36] <uersbob> test* [06:00] <Zeranoe> Does FFmpeg have any support for the amv format? [09:02] <userDavid> i'm not sure if anyone is on, but i will ask a question [09:05] <userDavid> i'm trying this command. it works, but im trying to cut off the first few seconds of both audio files. Instead, it cuts off the video each time, ive tried a lot of variations [09:05] <userDavid> ffmpeg -i video1.mp4 -c:v libx264 -i audio1.wav -ss 2 -i audio2.wav -ss 3 -filter_complex "amix=inputs=2" -c:a libvo_aacenc output.avi [09:26] <ubitux> userDavid: -ss 2 applies to audio2.wav and -ss 3 applies to the whole presentation [09:27] <ubitux> (output) [09:48] <userDavid> oh. i read that placing -ss after the inpute gets a more accurate result [09:49] <ubitux> "output seeking" is more accurate yes [09:49] <ubitux> but -ss 2 isn't a more accurate output seking on audio1.wav [09:50] <ubitux> it's an input option for audio2.wav [10:01] <hi117> is there a program that can record sound and stop recording after the volume is too low for a certain period? [10:02] <hi117> for linux and perferably command line [10:16] <hi117> found it, sox [12:40] <slystone> Hi! [12:40] <slystone> for i in *.mp4; do ffmpeg -i $i -vcodec mpeg2video -ar 44100 -s 1280x720 ${i%mp4}.mpg; done I've tried this command for a batch of videos. But it seems the quality is then appalling. [12:40] <slystone> Si I could use your help. [12:41] <slystone> What would you do differently please? [12:42] <LithosLaptop> why do you want to convert them to MPEG2? [12:44] <slystone> LithosLaptop: well, because some machines can only read mpeg2, that's why. [12:44] <slystone> I wouldn't bother otherwise. [12:47] <LithosLaptop> oh ok, well you will have to bump up the bitrate for MPEG2 [12:47] <LithosLaptop> I am not sure at what bitrates it will start looking okay [12:48] <slystone> Input #0, mpeg, from 'PSES2013-Bayart-FTTH_Fiber_to_the_home..mpg': [12:48] <slystone> Duration: 01:27:44.66, start: 1.000000, bitrate: 637 kb/s [12:48] <slystone> Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 104857 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc [12:48] <slystone> Stream #0.1[0x1c0]: Audio: mp2, 44100 Hz, stereo, s16, 128 kb/s [12:48] <slystone> this is what ffprobe has to tell. [12:48] <slystone> Could it work with -target dvd maybe? [12:49] <spaam> slystone: 1280x720 for dvd? :S [12:49] <spaam> dvd:s dont have that kind for size. [12:49] <LithosLaptop> also I assume all your input videos are the same? [12:49] <slystone> spaam: hum& right! [12:50] <slystone> LithosLaptop: I think so yes. [12:50] <LithosLaptop> can you ffprobe one of them? [12:51] <LithosLaptop> 637 kb/s is very low for 720p even for non MPEG2 codecs [12:52] <LithosLaptop> do those machines that can only read mpeg2 also support 720p resolutions? [12:52] <slystone> avprobe version 0.8.7-6:0.8.7-1, Copyright (c) 2007-2013 the Libav developers [12:52] <slystone> built on May 20 2013 12:25:19 with gcc 4.7.3 [12:52] <slystone> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'PSES2013-Zimmermann-Nos_machines_sont_elles_encore_nos_amies.mp4': [12:52] <slystone> Metadata: [12:52] <slystone> major_brand : isom [12:52] <slystone> minor_version : 512 [12:52] <slystone> compatible_brands: isomiso2avc1mp41 [12:52] <slystone> encoder : Lavf53.21.1 [12:52] <slystone> Duration: 00:38:11.92, start: 0.000000, bitrate: 1233 kb/s [12:52] <slystone> Stream #0.0(und): Video: h264 (Main), yuv420p, 1280x720, 1034 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc [12:52] <slystone> Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 191 kb/s [12:52] <slystone> this is for an input file. [12:53] <slystone> LithosLaptop: yes, they support 720p. [12:53] <LithosLaptop> ok [12:56] <slystone> LithosLaptop: so how would you change the bitrate at 637kb/s? [12:56] <LithosLaptop> ffmpeg -i $i -vb 2600k -vcodec mpeg2video -acodec libmp3lame -ab 192 -ar 44100 -s 1280x720 -fmt vob ${i%mp4}.mpg [12:56] <LithosLaptop> try that [12:57] <slystone> ok thanx! :) [12:58] <LithosLaptop> increase 2600 to something else if it still looks crap :) [12:58] <slystone> Yep, that part I've guessed. :) [12:58] <slystone> And what is -fmt vob for? [12:59] <LithosLaptop> MPEG 2 Program Stream. container format for DVDs [12:59] <slystone> Cool! :) [13:00] <slystone> Thanx a bunch LithosLaptop! [13:00] <LithosLaptop> no prob [13:00] <LithosLaptop> btw if it can't read the file try removing -fmt vob [13:01] <slystone> Hum, ok. :) [14:14] <kvz> the ffmpeg site mentions "FFmpeg Static Builds by Burek and FFmpeg Static Builds by Relaxed". I'm considering using one in production to convert a wide variety of formats. I'll run tests, but does anybody have good/bad experiences with either one of them? [14:14] <Mavrik> um, they should all work [14:14] <Mavrik> probably difference is in bundled libraries [15:22] <semme> guys i was wondering if there is example of on-the-fly remuxing with ffmpeg [16:50] <eduardoboucas> hi guys. I'm using ffmpeg to encode an MP3 from a wave file, but i'm getting an unexpected result. the audio is shifted for some reason [16:51] <eduardoboucas> I took a snapshot of both audio files on a DAW where the offset is very clear: http://snag.gy/lymi9.jpg [16:51] <eduardoboucas> does anyone have any idea of what can be causing this? [16:59] <brontosaurusrex> eduardoboucas, try lame decode your.mp3 and see if offset is gone [16:59] <brontosaurusrex> lame --decode my.mp3 [16:59] <eduardoboucas> ok [17:01] <eduardoboucas> yeah, the offset is gone [17:01] <eduardoboucas> what does this mean? [17:02] <brontosaurusrex> read http://www.hydrogenaudio.org/forums/index.php?showtopic=72327 [17:03] <eduardoboucas> thanks! [17:03] <brontosaurusrex> np [17:04] <eduardoboucas> wow, so there's no solution whatsoever? [17:04] <eduardoboucas> this will be a big problem for me :/ [17:05] <brontosaurusrex> use some other lossy format that doesnt have this? [17:05] <eduardoboucas> the thing is that I'm working on a web audio editor [17:05] <brontosaurusrex> aac perhaps [17:06] <eduardoboucas> and the formats supported by browsers for HTML5 Audio playback aren't that many [17:07] <eduardoboucas> the idea is that users record their tracks by plugging their instrument to the computer and the result is a wave file [17:07] <brontosaurusrex> the combo of vorbis and aac perhaps (but i'am not sure about offsets, you will have to test) [17:07] <eduardoboucas> but I can't afford to save wave files on the server, so I was planning on converting the files to MP3 [17:07] <eduardoboucas> but I can't lose sync, so this is no good for me [17:08] <eduardoboucas> but yeah, I'll make some tests. thanks! [17:08] <brontosaurusrex> here is some reading about lame http://lame.sourceforge.net/tech-FAQ.txt [17:09] <brontosaurusrex> looks like "--nores option in LAME" may do what you want [17:10] <eduardoboucas> I'll definitely check it out. thanks a lot! [17:13] <eduardoboucas> apparently, Vorbis does not introduce any offset [17:13] <eduardoboucas> I just tested [17:14] <LithosLaptop2> eduardoboucas: AAC is just as well supported as MP3 in HTML5 audio. In fact I didn't even know MP3 was widely supported [17:14] <LithosLaptop2> you will need to support Vorbis as well as AAC if you want to support most browsers [17:14] <eduardoboucas> you're right [17:15] <eduardoboucas> I'll try to encode to AAC to see if there is any offset, just in case :) [17:15] <brontosaurusrex> eduardoboucas, what OS are you on? [17:15] <LithosLaptop2> haha ok, just be carefull. there is only one good quality AAC encoder available with ffmpeg [17:16] <eduardoboucas> brontosaurusrex, a CentOS server [17:17] <brontosaurusrex> ok, then ffmpeg with fgh aac encoder is what you need or old nero linux encoder perhaps [17:17] <LithosLaptop2> fgh = libfdk_aac [17:17] <brontosaurusrex> --enable-libfdk-aac < yes [17:18] <brontosaurusrex> *fgh = fdk [17:19] <eduardoboucas> ok, so I assume I need to reinstall ffmpeg, right? [17:21] <brontosaurusrex> i have a static debian 64 compile from git, which i could share, but dunno if that would work on centOS [17:21] <brontosaurusrex> configuration: --enable-gpl --enable-libx264 --enable-libfdk-aac --enable-nonfree --enable-version3 --disable-shared --enable-static --extra-libs=-static --extra-cflags=--static [17:21] <eduardoboucas> hmm [17:22] <brontosaurusrex> anyway, you could build it from git. [17:22] <brontosaurusrex> adding the fdk part [17:23] <eduardoboucas> ok [17:25] <brontosaurusrex> and also vorbis and lame [17:25] <brontosaurusrex> is what you need [17:25] <eduardoboucas> yeah I have vorbis and lame working [17:26] <eduardoboucas> I remember I just followed some guide to download all those libraries and ffmpeg-source and then I installed everything [17:26] <eduardoboucas> so I guess I can just download this new library and repeat the whole process? [17:27] <brontosaurusrex> eduardoboucas, another thing, there are some js decoders out there btw [17:28] <brontosaurusrex> eduardoboucas, like http://libwebpjs.hohenlimburg.org/vp8/ogg-vorbis-javascript-decoder/ [17:28] <eduardoboucas> (yeah, I followed this, http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide, and for some reason I was stupid enough not to include the AAC library) [17:28] <eduardoboucas> hmm [17:28] <brontosaurusrex> dunno if that would be fast enough for what you are doing thought [17:30] <eduardoboucas> yeah, timing is a very delicate subject in my case [18:00] <semme> any idea how to copy a stream with libavcodec? [18:01] <semme> i have a/v stream coming from FLV over HTTP and i want to remux it to MP4 [18:01] <semme> how to copy the incoming streams to the remux MP4 [18:04] <eduardoboucas> brontosaurusrex, I'm getting an error when installing fdk-aac from Git. config.status: error: cannot find input file: Makefile.in [18:04] <eduardoboucas> do you have any idea of what might be happening? [18:05] <brontosaurusrex> eduardoboucas, take the tarball [18:05] <eduardoboucas> (still following this: http://ffmpeg.org/trac/ffmpeg/wiki/CentosCompilationGuide) [18:06] <Mavrik> semme, -codec copy [18:06] <semme> yeah the equivalent of acodec copy but through the libav API [18:06] <Mavrik> um [18:07] <Mavrik> just dont reencode the packet? [18:07] <semme> so i just memcopy the packet? [18:07] <Mavrik> mhm. [18:07] <Mavrik> I'm not even sure you have to memcpy it [18:08] <semme> thanks Mavrik, btw out of curiosity instead of remux a file, is there even faster to way to just change the header? [18:08] <Mavrik> from what? [18:11] <semme> i've got an F4V (h264/aac) but the program that is supposed to play it does not support FLV [18:11] <semme> but if i use MP4 header everything is fine [18:12] <semme> and i was thinking is there a way to just change the header and stream the body as is [18:16] <Mavrik> uh [18:16] <Mavrik> I doubt that would work. [18:16] <Mavrik> but I have no idea how FLV container is structured internally [18:24] <semme> ok thanks [18:24] <semme> seems diff than FLV [18:25] <semme> btw Mavrik, sorry if its too many questions, but do you have an idea where i can define the movflags [18:25] <semme> through the api [18:25] <semme> im talking flags like -movflags faststart, but defined through the API [18:25] <Mavrik> mhm [18:25] <Mavrik> I think it's av_option_set [18:25] <Mavrik> or something like that [18:25] <semme> ok thank you very much [18:25] <Mavrik> check mov.c source file on how the parameters are stored in the context [18:26] <semme> will do [18:40] <eduardoboucas> brontosaurusrex, I reinstalled FFmpeg, supposedly with fdk-acc, but it doesn't seem to be working :/ [18:41] <brontosaurusrex> eduardoboucas, some command lines http://www.hydrogenaudio.org/forums/index.php?showtopic=95989&st=25&p=80448… [18:41] <brontosaurusrex> and post your cli [18:43] <eduardoboucas> ok, but before I do that, is there any way to check if the library was correctly installed? [18:44] <durandal_1707> yes paste ffmpeg output [18:45] <eduardoboucas> http://pastebin.com/uEMXdW40 [18:46] <eduardoboucas> it says it was built on April [18:46] <brontosaurusrex> on debian "sudo ldconfig" will "fix" libraries [18:46] <brontosaurusrex> eduardoboucas, that doesnt have you aac encoder at all [18:47] <eduardoboucas> is it possible I'm running like the "old" ffmpeg version and not the one I just re-installed? [18:48] <brontosaurusrex> eduardoboucas, what does "which ffmpeg" says [18:48] <brontosaurusrex> eduardoboucas, acording to that centos build-link you are builing in user space [18:48] <brontosaurusrex> *building [18:49] <eduardoboucas> it says /usr/local/bin/ffmpeg [18:49] <brontosaurusrex> and your compile should be in /home/user/bin [18:50] <brontosaurusrex> or whatever is home folder in centos, --bindir="$HOME/bin" [18:51] <brontosaurusrex> try running ~/bin/ffmpeg [18:51] <eduardoboucas> http://pastebin.com/TLC1P3wY [18:51] <eduardoboucas> I guess this is the one I should be running [18:51] <brontosaurusrex> yep [18:52] <eduardoboucas> how can I fix this mess of having 2 ffmpegs? [18:52] <brontosaurusrex> how did you install the 1st one? [18:53] <eduardoboucas> I followed the same guide, so I don't know what could I have done differently [18:53] <brontosaurusrex> perhaps you have been using the version from repos without even knowing it? [18:54] <eduardoboucas> I don't know :S [18:54] <eduardoboucas> isn't there any way to just remove the first one? [18:54] <brontosaurusrex> use your apt-get remove [18:54] <brontosaurusrex> yum whatever [18:55] <eduardoboucas> I assume it's something more than just removing the binary [18:55] <brontosaurusrex> it is [18:55] <eduardoboucas> Package(s) ffmpeg available, but not installed. [18:56] <eduardoboucas> so I guess I wasn't using the one from yum [18:56] <brontosaurusrex> dunno :) [18:56] <brontosaurusrex> probably not [18:56] <eduardoboucas> gosh, what a mess [18:57] <brontosaurusrex> well, compare versions of /usr/local/bin/ffmpeg and the one in repos [18:57] <brontosaurusrex> and you should get a clue [18:58] <brontosaurusrex> if the repo version is older, than the one in /usr/local/bin/ffmpeg is something you "compiled in" [18:58] <brontosaurusrex> from there it gets yum related, and i dont know yum, so i will not comment on that [18:59] <eduardoboucas> ok thanks [19:02] <brontosaurusrex> looking at that guide its basically missing some sort of abstraction layer in the end [19:03] <brontosaurusrex> i try to compile shared stuff via checkinstall, which would produce *.deb file and my package manager would then also be aware of that [19:04] <brontosaurusrex> or i would build static and put ffmpeg bin to the specific spot manually [19:04] <brontosaurusrex> now, i'am far from compiling expert, just what i have noticed from absorbing various web guides .... [20:35] <rmzelnick> http://pastebin.com/sHxTTLFA [20:36] <rmzelnick> I got this problem yesterday, and now I can't play any video on VLC anymore =( [21:17] <mjrosenb> silly question-- what does this mean: Could not write header for output file #0 (incorrect codec parameters ?) [21:17] <mjrosenb> the audio and video codecs are 'copy' [21:18] <mjrosenb> is it talking about the container's codec? [21:19] <cbsrobot_> yes, not all containers support all codecs [21:20] <durandal11707> mjrosenb: what you are trying to do? [21:27] <mjrosenb> cbsrobot_: that I did not know. [21:27] <Mavrik> mjrosenb, usually there's an actual error written in previous ouput [21:27] <mjrosenb> durandal11707: I am trying to re-encode a .ts file as .mkv [21:27] <mjrosenb> mkvtoolnix can extract the video, but is clueless about the audio. [21:28] <mjrosenb> https://gist.github.com/5944597 is the full output. [21:28] <mjrosenb> mkv certainly supports h.264 [21:29] <mjrosenb> maybe it doesn't support aac? [21:29] <mjrosenb> that seems unlikely to me. [21:29] <Mavrik> that's wierd, ffmpeg shouldn't have any problems remuxing this [21:33] <mjrosenb> Mavrik: that is what I though, yet here I am :-p [21:36] <Mavrik> mjrosenb, can you try grabbing the latest static build? [21:38] <mjrosenb> Mavrik: sure thing, where do they live? [21:38] <Mavrik> uhh [21:39] <Mavrik> ffmpeg download site :P [21:39] <Mavrik> http://ffmpeg.gusari.org/static/64bit/ [21:40] <mjrosenb> looks like there haven't been any new ones for like two weeks? [21:41] <Mavrik> and you're using a build from previous year [21:41] <Mavrik> I think two weeks is fine ;) [21:41] <mjrosenb> touche. [21:42] <mjrosenb> Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input [21:42] <mjrosenb> well, that is certainly a different error. [21:46] <Mavrik> huh [21:47] <mjrosenb> although it doesn't say if it is in the container, audio or video stream. [21:48] <osmodiar> [mpegts @ 00000000023fa6e0] max_analyze_duration 5000000 reached at 5005000 microseconds [21:48] <osmodiar> Probably that. [21:50] <Mavrik> nah, that's not an error [21:50] <mjrosenb> so there is a bit of corruption, and the whole thing is falling over dead? [21:50] <mjrosenb> mplayer seems to be playing it without issue. [22:00] <mjrosenb> https://gist.github.com/5944693 is the output of running mplayer on the file. [22:17] <Mavrik> mjrosenb, did you try remuxing to mp4 just in case? [22:20] <Fjorgynn> :) [22:32] <mjrosenb> Mavrik: negative. [22:33] <mjrosenb> [mp4 @ 0x3a45e40] Malformed AAC bitstream detected: use audio bitstream filter 'aac_adtstoasc' to fix it ('-bsf:a aac_adtstoasc' option with ffmpeg) [22:33] <mjrosenb> av_interleaved_write_frame(): Operation not permitted [22:33] <mjrosenb> well, that was even more informative! [22:33] <Mavrik> hmm, that could explain some things :) [22:42] <mjrosenb> is there a reason that different containes give different errors on demuxing the input? [22:45] <Mavrik> mjrosenb, those errors are from the muxers [22:45] <mjrosenb> *boggle* [22:45] <Mavrik> and the reason is simply& because each muxer generates their own ;) [22:45] <mjrosenb> I added that line to the .mp4 output, and it muxed properly [22:45] <mjrosenb> I replaced .mp4 with .mkv [22:45] <mjrosenb> and it failed with the same cryptic error [22:45] <mjrosenb> but muxing the .mp4 to .mkv worked. [22:46] <Mavrik> ^^ [22:47] <LithosLaptop2> lol [22:47] <mjrosenb> Mavrik: is -bsf:a aac_adtstoasc an argument to the muxer? [22:47] <Mavrik> no, that's a bitstream filter [22:47] <Mavrik> it filters audio bytes before they go in the muxer [22:48] <mjrosenb> right, so presumably, it fixed whatever corruption was in the audio stream, so why would the mkv muxer still not like the stream? [22:49] <mjrosenb> nevermind. that muxing failed misreably. [22:49] <mjrosenb> muxing to .mp4 didn't work, evidently. [22:50] <MachinaeWolf> freaking errors http://pastebin.com/ZKVHbRfN [22:51] <Mavrik> mjrosenb, it seems your audio is encoded in ADTS format, not ASC without headers& AAC can have about 3 different bitstream formats [22:52] <Mavrik> some made for streaming, some for just "normal files" [22:52] <Mavrik> MachinaeWolf, good thing you're not running ffmpeg. [22:52] <MachinaeWolf> ?_? [22:52] <MachinaeWolf> wut [23:07] <MachinaeWolf> gah lame signal [23:07] <MachinaeWolf> <Mavrik> MachinaeWolf, good thing you're not running ffmpeg. [23:07] <MachinaeWolf> <MachinaeWolf> ?_? [23:07] <MachinaeWolf> <MachinaeWolf> wut [23:09] <osmodiar> Read that, MachinaeWolf. [23:09] <MachinaeWolf> k [23:11] <MachinaeWolf> Does it take the same options? [23:11] <mjrosenb> I saw "!fork" and thought that ffmpeg was forking, and executing something else that was spewing errors, needless to say, I was confused before reading the link. [00:00] --- Mon Jul 8 2013

1 0

ffmpeg-devel.log.20130706
by burek 07 Jul '13

07 Jul '13

[00:28] <cone-352> ffmpeg.git 03Michael Niedermayer 07master:390aabb2a1ad: tcp: Fix the default timeout [01:44] <cone-352> ffmpeg.git 03Michael Niedermayer 07master:efa9e6b42345: ff_network_wait_fd_timeout: do not break with timeout < 0 [01:44] <cone-352> ffmpeg.git 03Michael Niedermayer 07master:68b7b534be59: tcp: Use a default timeout of 5 sec for opening a connection but not for receiving packets [01:49] <durandal_1707> 3 [02:40] <durandal_1707> saste: i'm porting psnr filter [05:51] <cone-602> ffmpeg.git 03Michael Niedermayer 07master:0844630e6b01: avformat/mp3dec: Add usetoc option to allow dlsabling the use of the xing TOC [12:16] <cone-626> ffmpeg.git 03Luca Barbato 07master:4972e5a186fe: bitstream_filter: K&R formatting cosmetics [12:16] <cone-626> ffmpeg.git 03Michael Niedermayer 07master:98277fd1a0f7: Merge remote-tracking branch 'qatar/master' [12:35] <cone-626> ffmpeg.git 03Timothy Gu 07master:561e05136f1c: doc/encoders: add libopus encoder doc [12:35] <cone-626> ffmpeg.git 03Stefano Sabatini 07master:2aa2b4ac6555: examples/muxing: add support to audio resampling [13:39] <cone-626> ffmpeg.git 03James Almer 07master:3960992f0abf: oggparseskeleton: Fix fisbone header parsing [13:39] <cone-626> ffmpeg.git 03James Almer 07master:63d7684fefcf: oggparseskeleton: Replace avpriv_report_missing_feature() with a normal av_log() call [15:56] <saste> durandal_1707, what psnr filter? [15:56] <saste> the one on the archive? [15:56] <durandal_1707> the one from '11 [15:56] <saste> ok [16:19] <durandal_1707> perhaps i should use tiny_psnr instead? [16:33] <cone-626> ffmpeg.git 03Michael Niedermayer 07master:32de28053d23: avcodec/x86/cabac: factorize broken llvm/clang check out [16:33] <cone-626> ffmpeg.git 03Michael Niedermayer 07master:66537c7efd4e: avcodec/x86/cabac: Disable get_cabac_bypass_x86() on broken llvm/clang [16:39] <durandal_1707> what happens if shortest is set to 0 and repeatlast is set to 0 for overlay/haldclut? [17:10] <durandal_1707> michaelni: when muxing to nut and suspending ffmpeg, duration is not displayed for nut, when using ffprobe [17:24] <superware> michaelni, hi. I was really hoping that udp/mpegts issue is behind us, but please see http://pastebin.com/WVBkyZTv :| [17:25] <cone-626> ffmpeg.git 03Paul B Mahol 07master:96358062650a: lavfi/blend: use dual input helpers [17:29] <durandal_1707> what is AOB? [17:40] <michaelni> superware, so you have some hw that generates a ts over udp and that doesnt work with ffmpeg and you have a file that does work? [17:40] <michaelni> if so how was this file generated? [17:42] <Compn> durandal_1707 : like vob but aob ... so dvd audio file i think [17:43] <nevcairiel> yes, AOB is dvd audio [17:43] <nevcairiel> can contain PCM or MLP i think [17:44] <michaelni> durandal_1707, suspended nut duration works here [17:45] <superware> michaelni: VLC input dump [17:45] <durandal_1707> i tried with ffv1 only [17:45] <superware> michaelni: raw dump [17:51] <superware> michaelni: VLC -> Media -> Convert / Save -> Network -> udp://@:1234 -> Dump raw input [17:55] <michaelni> superware, and what does ffplay do ? it displays nothing? recognizreable video with heavy artifacts? good video with occasional artifacts ? [17:55] <superware> nothing [17:57] <superware> for some reason when I add -f h264 then after a few seconds the video opens up mostly grey, but in the right size [17:57] <superware> Can I please stream it to your IP? [17:58] <michaelni> superware, sure if you are on my LAN from outside it wont get past the router [17:59] <superware> port-forwarding :) [17:59] <michaelni> superware, iam not sure what the issue that remains is but you could try tools/aviocat [17:59] <michaelni> with aviocat you can dump the udp to a file [17:59] <michaelni> just to check if that works or not [18:00] <superware> the raw udp will be dumped, or the ts? [18:01] <michaelni> it will simply dump the packets as they are [18:02] <superware> ok, I think that's what VLC is doing when I "dump raw input" to test.ts, ffplay can play it [18:03] <superware> michaelni: if you add a port-forward for port 1234 to your intranet IP, it should go through [18:05] <superware> and if you already have a port forwarded to your machine, I can use that port instead [18:09] <michaelni> superware, ping me when you tested aviocat and please upload the resulting file from it (and vlc in case they behave differently) [18:10] <cone-626> ffmpeg.git 03Paul B Mahol 07master:d64f3b72e0ea: replace some deprecated defines [18:13] <michaelni> superware, also in case its unclear yes the person reporting a bug is supposed to take as much work of the developers shoulders as possible. All developers are quite busy most of the time ... [18:18] <superware> michaelni, do you know of an aviocat win32 build? if not, will a test.ts dump from VLC suffice? [18:19] <michaelni> well, one part iam trying to exclude is that vlc does more than dump raw udp [18:20] <superware> forget VLC, I can do my own raw dumping from udp, it will take a few minutes [18:29] <superware> michaelni, please try http://80.74.98.102/test.zip (this isn't a zip file, rename it to test.ts) [18:31] <durandal_1707> is mencoder finally dead.... [18:48] <superware> michaelni, got it? [19:06] <atiti> Too many slices, increase MAX_SLICES and recompile [19:06] <atiti> hm [20:43] <michaelni> superware, got the zip [20:46] <superware> great, rename to test.ts. Yet it will play with ffplay, so I guess it can't really help reproduce the issue [20:46] <michaelni> superware, this is not a raw udp dump or your hw uses ffmpeg [20:46] <michaelni> service_provider: FFmpeg [20:47] <superware> trust me it's a raw udp dump [20:47] <superware> maybe [20:47] <superware> I meant it might be FFmpeg [20:50] <michaelni> now the question is still, does it work with aviocat too [20:50] <michaelni> aviocat uses the same code as ffplay for receiving udp [20:54] <michaelni> superware, also if aviocat is not part of the stuff from zeranoe its probably a good idea to contact him, he might be unaware of it and the other tools [20:55] <michaelni> and the more trivial thing, i assume you waited a bt for ffplay to display something, it can for some cases of network streams take a few seconds [20:59] <superware> I waited [21:00] <superware> I've created a new udp socket and wrote all incoming data to a file, there's nothing more raw than that :) [21:01] <superware> I guess the issue/bug has something to do with the large datagrams being sent [21:08] <superware> michaelni: ok, emailed zeranoe [21:10] <superware> is there a chance you will be able to receive that stream? maybe there's a way to form a kind of VPN (supporting UDP)? [22:29] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:8ca5d277d8a8: avformat/utils: factor ff_find_last_ts() out of ff_gen_search() [22:29] <cone-826> ffmpeg.git 03Michael Niedermayer 07master:7f250502897e: nutdec: use ff_find_last_ts() [00:00] --- Sun Jul 7 2013

1 0

ffmpeg.log.20130706
by burek 07 Jul '13

07 Jul '13

[00:43] <b0fh_ua> Hello! How do I specify muxer option for gif muxer if the options with the same name is already used by image2pipe demuxer? [01:51] <durandal_1707> b0fh_ua: just as usual, but make sure it is last option [01:53] <durandal_1707> something like: ffmpeg -i in.gif -loop out.gif [05:22] <nice4u88m> hello\ [05:40] <voltagex> hi, how do I get access to the screen-capture-recorder dshow device? I don't have it in -list_devices [06:18] <Zeranoe> voltagex: did you install any directshow screen capture filters? [06:19] <voltagex> Zeranoe: thanks for your builds! I'm not sure, it appears I may have. I'm not sure how to uninstall them (came with a DV camera) [06:20] <Zeranoe> Please post the output of: ".\ffmpeg.exe -list_devices true -f dshow -i dummy" to http://paste.debian.net/ [06:21] <voltagex> http://paste.debian.net/14664/ [06:22] <voltagex> I've tried with 1.2 and the latest git build - both 32 bit shared. 64 bit builds can't enumerate DirectShow at all it seems [06:24] <Zeranoe> voltagex: While FFmpeg's support for dshow is growing, I still prefer another method. I go grf graphedit file>avisynth script>ffmpeg input. I use two scripts, one for audio, one for video. It is a lot more steps but it will give you outstanding quality. Don't forget a vsync though. If you want to just stick with FFmpeg, what device did you want to capture from that list? [06:28] <Zeranoe> voltagex: If you need any examples for that process let me know. [06:33] <voltagex> holy crap, I tried to uninstall that filter [06:33] <voltagex> and it hard locked my machine [06:33] <voltagex> i.e. mouse stopped and PC fell off the network [06:34] <voltagex> it's that the screen capture filter never even turns up [06:34] <Zeranoe> voltagex: Are you sure it is installed? [06:35] <voltagex> I couldn't find any instructions about what to install [06:35] <voltagex> I'd like to work out how to uninstall this other filter first [06:35] <voltagex> any ideas how to remove it manually? [06:35] <Zeranoe> Which one? [06:36] <voltagex> all Enosoft filters [06:37] <Zeranoe> It would be a registry thing, I haven't tired manually removing any [06:37] <voltagex> I'd be interested in this graphedit method, as it'd probably let me capture my TV card too [06:38] <Zeranoe> voltagex: http://www.videohelp.com/download/graphedit10-090724.zip [06:39] <voltagex> ...I'm actually going to do a Win8 Refresh & Reset first - this isn't the first odd hang I've had [06:40] <voltagex> but continue, as this IRC session is on a different machine so I'll still be able to read it [06:40] <voltagex> :P [06:42] <Zeranoe> voltagex: You'll need to install this dshow capture filter: http://www.umediaserver.net/bin/UScreenCapture.zip website is here: http://www.umediaserver.net/umediaserver/download.html [06:42] <voltagex> *right*, so that's the first dshow filter I saw mentioned on the mailiing list [06:44] <Zeranoe> voltagex: It has the most options, which you will be able to edit with graphedit [06:44] <voltagex> cool [06:44] <Zeranoe> voltagex: Best option I've found. I can record my 1080p screen without lag. [06:44] <voltagex> I wonder what happens if I load a 3D game though [06:45] <voltagex> oh. are multiple monitors going to mess it up? [06:45] <Zeranoe> voltagex: Not sure, I can record HD videos no problem which are using a little. [06:45] <Zeranoe> voltagex: Haven't tried that, it had options for multiple monitors, going to have to try. [06:46] <voltagex> this machine is fubar. [11:36] <kms> sorry, that now instead of a "-af" ? [11:49] <kms> hey, enybody here? [11:52] <kms> i need change audio volume [11:55] <saste> kms, -af volume [12:00] <kms> i have old build [12:01] <Mavrik> get a new one. [13:52] <silv3r_m00n> is it possible to record a specific window using such a command, avconv -f x11grab -s xga -r 30 -i :0 -c:v libx264 ~/session-recording.avi [15:35] <silv3r_m00n> i am recoding desktop video using avconv, is it possible to pause it ? [16:56] <kamal__> I am getting errors while compiling a simple code and the error seem to be linker-errors.. error output: [16:56] <kamal__> ffmpeg2/libavformat/rtmppkt.c:446: undefined reference to `av_log' [16:57] <kamal__> libavformat/id3v2.c:383: undefined reference to `av_dict_set' [16:57] <kamal__> and so on for tens of functions [16:58] <kamal__> My command was this: cc tut.c -lavcodec -lavutil -lavformat -lswscale [16:58] <kamal__> I wish someone could help [17:02] <saste> kamal__, move -lavutil towards the end [17:04] <kamal__> still it gives same linking errors and maybe i was incomplete in my mentioning of errors, these errors come for both libraries avutil and avformat [17:05] <kamal__> for e.g. libavutil/eval.c:304: undefined reference to `exp' and /libavformat/adxdec.c:51: undefined reference to `av_free_packet' [17:09] <kamal__> I am a novice and am in the middle of some important project, so if someone can spare a little time, plz guide me in my linking errors [17:22] <saste> kamal__, use pkg_config to compute the flags for ffmpeg [17:22] <saste> exp -> -lmath [17:22] <saste> check the Makefile in doc/examples [17:53] <silv3r_m00n> while encoding i press ctrl+c to stop it, is it possible to resume encoding in the same file ? [17:54] <silv3r_m00n> this is the command that is interrupted, ffmpeg -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -vcodec libx264 -preset ultrafast -crf 0 -threads 0 -y screencast_out.avi [17:54] <silv3r_m00n> basically i need to pause and resume recording [21:58] <Mavrik> hmm, is it possible to tell ffmpeg to decode only I-frames? [23:10] <mark4o> Mavrik: -skip_frame nokey [23:11] <mark4o> but it is broken for h264; you need to use -vf 'select=eq(pict_type\,I)' for that [23:11] <Mavrik> mhm, but select comes into effect after decoding [23:11] <mark4o> right, -skip_frame nokey for pre-decoding [23:12] <Mavrik> thanks [23:31] <miguel_> Im trying to record from my webcam with ffmpeg with the command "ffmpeg -f alsa -f video4linux2 -i /dev/video0 -r 30 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 0 cam_`date "+%d-%m-%Y$%Hh%Mmin%S$"`.mkv" [23:31] <miguel_> but when I run it, I cant type q to exit [23:31] <miguel_> is there a better way to record ? [23:44] <miguel_> Im trying with "ffmpeg -f alsa -i pulse -f video4linux2 -r 30 -i /dev/video0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 0 out.mkv", and it kinda works, but the audio is delayed [00:00] --- Sun Jul 7 2013

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