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February 2013

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ffmpeg-devel.log.20130227
by burek 28 Feb '13

28 Feb '13

[02:33] <michaelni> ubitux, fate sub-charenc seems failing on qemu boxes [02:34] <cone-286> ffmpeg.git 03Jean First 07master:2d7044683f3c: ffmpeg_opt: add -to option to specify stop time [03:10] <cone-286> ffmpeg.git 03Michael Niedermayer 07master:1672624ddc74: mpegvideo_enc: fix gray flag with 444 jpeg [03:10] <cone-286> ffmpeg.git 03Michael Niedermayer 07master:0dcecf45d80c: avcodec: mbd has a range of 0..2 [03:56] <cone-286> ffmpeg.git 03Michael Niedermayer 07master:5d2f2c76435e: oggdec: chained oggs have timestamp discontinuities [11:59] <cone-803> ffmpeg.git 03Martin Storsjö 07master:31a23a0dc663: x86: dsputil_mmx: Remove leftover inline assembly fragments [11:59] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:f2bbc2ffc338: Merge commit '31a23a0dc663bd42bf593275971b4277a479b73d' [12:03] <cone-803> ffmpeg.git 03Diego Biurrun 07master:096cc11ec102: x86: vc1dsp: Move ff_avg_vc1_mspel_mc00_mmxext out of dsputil_mmx.c [12:03] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:04ec796bda79: Merge commit '096cc11ec102701a18951b4f0437d609081ca1dd' [12:35] <cone-803> ffmpeg.git 03Diego Biurrun 07master:845cfc92f908: x86: dsputil: Drop aliasing of ff_put_pixels8_mmx to ff_put_pixels8_mmxext [12:35] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:cdb9752a0f96: Merge commit '845cfc92f908791714b8c4c8a49c91b8c64b685e' [12:40] <cone-803> ffmpeg.git 03Diego Biurrun 07master:ebc701993fec: x86: dsputil: Drop some unused function #defines [12:40] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:50c2738883b7: Merge remote-tracking branch 'qatar/master' [13:58] <ubitux> michaelni: yes but i don't have a good solution in mind [13:58] <ubitux> the issue is that some iconv setup seems to lack the cp1251 info [13:59] <ubitux> i thought about adding another flag specifically for that info in the configure, but it kind of sucks hard [14:03] <michaelni> ubitux, cant you check in configure that cp1251 is available and if not then not enable iconv [14:03] <michaelni> ? [14:04] <ubitux> some people might want the feature even without having the russian character encoding info [14:04] <ubitux> michaelni: can you send me the result of the iconv -l command on one of those machine? [15:09] <michaelni> ubitux, trying to execute the installed iconv through qemu results in "Error -1 while loading iconv" [15:10] <michaelni> and i dont think theres a target specific iconv excutable installed [15:10] <Daemon404> who runs the ffmpeg-devel thread archives? (http://ffmpeg.org/pipermail) [15:11] <Daemon404> its stopped being updated about 10 days ago [15:14] <Compn> Daemon404 : http://ffmpeg.org/pipermail/ffmpeg-devel/2013-February/date.html [15:14] <Compn> what ? [15:14] <Compn> i see posts from feb 27th... [15:14] <Compn> http://ffmpeg.org/pipermail/ffmpeg-devel/ [15:14] <Compn> same amount of messages in both thread and date [15:15] <Daemon404> hmmm [15:15] <Compn> you mean the downloadable tar.gz ? [15:15] <Daemon404> inb4 it's chrome's cache being rap [15:15] <ubitux> michaelni: arg... so isn't iconv just broken on those systems? [15:15] <Compn> ehe [15:15] <Compn> >using google software [15:16] <Daemon404> >using windows 2000 [15:18] <Compn> the irony being , chrome doesnt work on win2k [15:19] <Daemon404> i think you need to look up teh definition of irony. [15:20] <Compn> firefox dropped support as well [15:22] <ubitux> microsoft as well [15:22] <ubitux> (no?) [15:22] <Daemon404> microsoft ropped win2k support YEARS ago [15:22] <Daemon404> any years ago [15:22] <Daemon404> compn is basically botnet [15:22] <Compn> gotta do my part , for skynet! [15:23] <nevcairiel> cant wait for june 2014 when XP support finally drops [15:23] <Daemon404> ^ [15:23] Action: JEEB high-fives nevcairiel [15:24] <nevcairiel> i'm sure the millions that use XP right now still because they believe its better wont change their mind until then, but i stop caring then [15:25] <Daemon404> 13 years is a good run. [15:26] <michaelni> ubitux, no clue about iconv under user mode qemu [15:26] <JEEB> for a commercially supported piece of software used by end users as well as corps, it is very well (if you go for corps-only you will find software that has run for 20+ years, but that doesn't count) [15:26] <nevcairiel> even debian stable isnt that old =P [15:27] <Daemon404> 20+? [15:27] Action: Daemon404 knows of vax emus [15:27] <michaelni> ubitux, but if its broken it should not get enabled [15:27] <ubitux> michaelni: the configure says it's working [15:27] <ubitux> michaelni: it just seems cp1251 codepage is not available [15:27] <ubitux> so the particular fate test can not work [15:28] <michaelni> cant configure test for that codepage ? [15:28] <ubitux> yes sure it can, we can just add a "HAVE_ICONV_WITH_CP1251" just for fate [15:28] <ubitux> would that be fine? i find it a bit clumsy but well... [15:29] <michaelni> is there a point in enabling iconv at all if cp1251 isnt there ? [15:29] <ubitux> yes, maybe the iconv has various other convertion charsets [15:29] <ubitux> cp1251 is just for russian, it may have some other [15:29] <ubitux> (that's the reason i asked for a iconv -l to list them) [15:29] <Compn> what if its listed under windows 1251 ? :P [15:30] <ubitux> i have both here [15:30] <ubitux> - iconv -l|grep 1251 [15:30] <ubitux> CP1251// [15:30] <ubitux> WINDOWS-1251// [15:30] <ubitux> afaik they are the same [15:30] <Compn> ah [15:31] <michaelni> ubitux, id have to cross compile a iconv first before i can run qemu iconv -l [15:32] <michaelni> isnt there a easier way [15:32] Action: michaelni lazy [15:32] <ubitux> mmh [15:32] <ubitux> is there some stuff in /usr/share/i18n/charmaps ? [15:32] <Daemon404> that doesnt sound portable at all [15:33] <Daemon404> espeiallt to more unixy things like solaris [15:33] <ubitux> i'm not trying to ask a portable question :) [15:33] <ubitux> just trying to figure out a bit how this iconv is setup [15:33] <Daemon404> i thought all the cool kids used icu nowadays [15:35] <ubitux> the choice of iconv was purely arbitrary, i never tried icu [15:36] <durandal_1707> michaelni: your opinion on libswscale drop? [15:37] <ubitux> wat? [15:37] <JEEB> but icu has to be used together with something else [15:37] <Daemon404> oh [15:37] <Compn> durandal_1707 : thread or more detailed plan ? [15:37] <JEEB> also jesus christ I've read/heard too much Japanese to not see "icu" as the verb "iku" [15:38] <ubitux> oci [15:38] <ubitux> oic [15:38] <Compn> isnt iku japanese for .... [15:38] <JEEB> let's just say "to go" [15:38] <JEEB> and not take it further ;) [15:38] <durandal_1707> Compn: i want to use babl [15:38] Action: Compn seen too much "anime" too [15:39] <Compn> durandal_1707 : like everything, i'm sure ffmpeg would be ok with having two swscale setups. or a branch with a different sws library. if its faster, it will be approved... [15:40] <Compn> and merg-ed [15:43] <michaelni> durandal_1707, do you have a replacement that has same features and speed? [15:44] <michaelni> ubitux, i see /usr/share/i18n/charmaps but thats just the host i dont see target specific ones [15:52] <Compn> durandal_1707 : are you asking for michaelni's help in swapping in a new sws ? do you have a list of libs that can replace it ? [15:53] <durandal_1707> Compn: no i just dream about such library [15:55] <kierank> me too [15:55] <wm4> me too (please handle all corner cases) [15:56] <Daemon404> babl certainly is not this library though [15:58] <Compn> what about just replacing one scaler at a time ? whats the fastest rgb to yuv lib ? [15:59] <Daemon404> proably swscale. [16:00] <Compn> shh dont tell them that [16:00] <kierank> swscale can only do rgb to yuv with 601 [16:00] <Daemon404> ive personally been using a a diff lib, but it is slow [16:01] <Daemon404> and kind of mangled atm [16:01] <Daemon404> and teh author hates VCS [16:01] <Compn> kierank : you want rgb2yuv 702? [16:01] <Daemon404> no, but he might want 709 [16:01] Action: Compn forgets the numbers [16:01] <Compn> 709! [16:02] <Daemon404> whenever i do rgb<->yuv conversions for import into e.g. After effects, i use avisynth or dithertools [16:02] <Daemon404> for that very reason [16:03] <Daemon404> oh wait its called fmtconvert now [16:03] <wm4> yeah, I pointed durandal_1707 to it [16:04] <Daemon404> wm4, i feel like people around here could benefit from reading doom9 sometimes [16:04] <Daemon404> instead of being in an mplayer bubble [16:04] <Daemon404> me runs [16:04] <JEEB> wonder how easily you could first make fmtconv into a video filter to test it [16:04] <Daemon404> iirc its not so fun [16:04] <Daemon404> he mngled it in with teh plugin code [16:04] <Daemon404> and usd intrinsics [16:04] <Daemon404> used* [16:05] <Daemon404> sometimes with no C/C++ version... [16:05] <Daemon404> he's a classic avs dev. [16:05] <JEEB> so it seems [16:05] <wm4> and releasing source code by posting it as forum attachments too... interesting culture [16:05] <Compn> >blame mplayer forever [16:05] <Compn> :P [16:05] Action: Compn trolled [16:05] <Daemon404> Compn, well if the glove fits... [16:06] <Daemon404> wm4, most have oved to VCs (github) now [16:06] <JEEB> also regarding d9 [16:06] <Daemon404> or google code [16:06] <Daemon404> fmtconv guy is just a weirdo [16:06] <JEEB> someone just posted a damn swscale question in the H.264 forum [16:06] <kierank> oh it's that guy [16:06] <Compn> vlc uses swscale too ? [16:06] <kierank> yes [16:06] <Compn> i guess it has to [16:07] <wm4> doesn't vlc have some custom conversion code too? [16:07] <Compn> sure, they like to reinvent wheels [16:07] <Daemon404> practically every project in existence does [16:07] <Daemon404> precisely because fuck swscale [16:11] <durandal_1707> why not babl? [16:12] <durandal_1707> did we just get new kind of spamming with git format-patch ? [16:14] <Daemon404> maybe [16:14] <Daemon404> definitely. [16:15] <Compn> so what sws does imagemagick have ? [16:15] <Daemon404> oh wait nvm that wasnt git-send-email [16:15] <Daemon404> Compn, i dont know but it is hilariously slow [16:15] <Compn> we should help them [16:15] <Compn> give them swscale :D [16:16] <wm4> "give them deadly poison" [16:17] <wm4> well at least it's very fast [16:17] <Daemon404> i dont know if youve ever touched imagemagick [16:17] <Daemon404> but i ts p retty close to dealy poison itself [16:17] <wm4> everyone dies, best case /sarcasm [16:18] <michaelni> only safe way is to rewrite everything, never contribute never reuse. This is the 1st rule of FOSS ;) [16:19] <Compn> michaelni : should make it closed source, then they cant complain about what it looks like [16:19] <Daemon404> michaelni, i heard you like linux's audio subsystems [16:19] <wm4> I wonder how gstreamer's dynamic compiler thing fares from a technical point of view (I guess probably not much, but still interesting idea) [16:19] <Daemon404> has gstreamer ever done anything technical of note? [16:19] <Daemon404> ever [16:19] <michaelni> Daemon404, i dont like them because they tend not to "just work" from a user perspective [16:20] <durandal_1707> michaelni: just give bounty for rewrite of some part of swscale code [16:20] <Daemon404> michaelni, theyre also bad from a dev perspective :P [16:20] <michaelni> iam not planing to rewrite any or add more to the already existing number [16:20] <Daemon404> durandal_1707, money isnt enough for people to work on swscale [16:20] <Daemon404> hence the guy who wrote XYZ support put it in lavfi [16:20] <Compn> someone emailed projects@mphq requesting mplayer make ao oss priority ahead of ao alsa in the try audio outputs order... [16:21] <Daemon404> why u no pulse? [16:21] <durandal_1707> michaelni: also note that exr float conversions is making image outputs too underexposed [16:21] <Compn> because fuck ubuntu thats why [16:21] <Daemon404> >implying pulse<->ubuntu [16:22] <wm4> Daemon404: pulse adds additional bugs on top of alsa (really) [16:22] <Compn> what distro you running Daemon404 ? [16:22] <Daemon404> i run a distribution called windows 8 [16:22] <Daemon404> and also debian unstable [16:22] <Compn> did you install pulse or did debian ? [16:22] <Daemon404> neither [16:22] <Daemon404> beause its head;ess [16:22] <Daemon404> headless* [16:23] Action: Compn slaps himself [16:23] <Daemon404> ok, so i also have an ubuntu box for dev work sometimes (with fluxbox) ;) [16:23] <Daemon404> i guess that uses pulse [16:24] <Compn> but yes i'm implying no one would be using pulse if ubuntu didnt jump that shark [16:24] <Daemon404> michaelni, quick ping, re: swscale rgb patch -- libav finally OK'd it, so once ffmpeg Oks it i can finally let that 6 mnth old patch die [16:24] <Daemon404> Compn, i think you forget rhel and fedor [16:25] <Daemon404> who employ the guy wh wrote pulse [16:25] <Compn> its a conspiracy! [16:25] <Compn> but also i'm not the one who makes the decisions in mplayer [16:25] <Compn> so thats why i no pulse [16:25] <cone-803> ffmpeg.git 03David Favor 07release/1.1:d4d1f32e48d9: Slight bug building ffmpeg-1.1.3 on OSX + patch to fix [16:26] <cone-803> ffmpeg.git 03Michael Niedermayer 07release/1.1:cdbaaa4f001e: doc/ffmpeg: remove non ascii char [16:26] <wm4> as soon as systemd has a hard dependency on pulse, everyone will use it [16:26] <Daemon404> hahahaha [16:29] <ubitux> michaelni: http://pastie.org/private/isycwcxhp1opps2olfeew [16:29] <ubitux> i have no better idea in mind currently [16:30] <Daemon404> eh.. if check_exec does what i think it does [16:30] <Daemon404> then that oesnt work at all for cross compilation [16:32] <ubitux> i have no other mean to test this [16:32] <Daemon404> exactly. you cant. [16:33] <Daemon404> its up to teh target system [16:33] <ubitux> so what do you propose? [16:34] <Daemon404> a runtime check that fails gracefully [16:34] <Daemon404> as for fate -- probably nto feasible. [16:34] <ubitux> right, so the current solution is correct [16:35] <ubitux> (without this patch) [16:35] <Daemon404> it's not "correct" [16:35] <Daemon404> its actully outright WRONG [16:35] <Daemon404> oh [16:35] <Daemon404> without the patch [16:35] <ubitux> :) [16:35] <ubitux> maybe and iconv-test program can be built [16:36] <ubitux> and used before running the sub char enc test [16:40] <cone-803> ffmpeg.git 03Derek Buitenhuis 07master:c87c2d0d0287: swscale: Add support for unscaled 8-bit Packed RGB -> Planar RGB [17:16] <Compn> ehe [17:16] <Compn> i'd add "binary codec loader" to gsoc idea, but probably no one wants that :) [17:17] <Daemon404> you mean some ancient wine code copypasta'd? [17:21] <nevcairiel> native-only would be boring [17:21] <nevcairiel> need to load windows codecs! [17:21] <durandal_1707> Compn: this is not #mplayer [17:27] <gnafu> "Ain't nobody got time for that!" [17:29] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:21c2e201f6aa: vf_lut: correct color/comp permutation [17:29] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:5c924c9b7d6d: dv: Correctly identify CDVC profile [17:34] <cone-803> ffmpeg.git 03Paul B Mahol 07master:9774145fe5ae: exr: simplify decompression path [17:53] Action: durandal_1707 found that uploading images becomes extremly hard [17:55] Action: durandal_1707 translated adverts hurts my eyes [18:06] <durandal_1707> what compression is this: http://www.imagetoo.com/?v=omvm.png ? [18:07] <nevcairiel> that looks a bit like a wavelet transform [18:09] <Daemon404> doesnt look like a transform me me... more like some prediction [18:09] <Daemon404> but what do i know [18:10] <Daemon404> or dithered [18:13] <durandal_1707> its 16 bit haar wavelet [18:13] <nevcairiel> see, wavelet! [18:16] <Daemon404> ive never seen haar produce that sort of pattern [18:16] <Daemon404> (and only one image to boot) [18:18] <durandal_1707> that is what you get prior to calling wavelet decoding in exr with piz compression [18:18] <Daemon404> dont you need the lowpass info to reconstruct such a transform [18:21] <michaelni> ubitux, did you see the darwin iconv failures ? [18:22] <michaelni> (on fate) [18:22] <ubitux> arg [18:22] <ubitux> wtf is wrong here.. [18:24] <ubitux> maybe we should move to a config mode for iconv [18:25] <ubitux> this built-in vs lib mess is going to be a problem [18:43] <durandal_1707> should i use realloc for blocks of structures or there is higher api for it? [18:53] <cone-803> ffmpeg.git 03Frederic Jean 07master:c53e8d9029f5: Include fix for building ismindex under MinGW [18:53] <cone-803> ffmpeg.git 03Michael Niedermayer 07master:1fd04cac001f: Merge remote-tracking branch 'fredjean/master' [19:01] <cone-803> ffmpeg.git 03Nicolas George 07master:f102c24d9040: ffmpeg: free last sub when using -fix_sub_duration. [19:07] <cone-803> ffmpeg.git 03Nicolas George 07master:13811b19d6b4: lavc/libopusenc: report an error if global_quality is set. [19:26] <ubitux> trac looks down [19:27] <durandal_1707> not for me [19:27] <nevcairiel> seems fine here too [19:27] <ubitux> looks fixed [19:30] <Compn> ubitux : hows mplayer or vlc handle iconv ? [19:33] <ubitux> Compn: problem is not that much iconv support [19:34] <ubitux> afaik they don't have fate first [19:34] <ubitux> also, cross compiling mplayer is not known to be the most trivial op [19:35] <ubitux> anyway, it seems darwin will need some more advanced check (or maybe something is broken in the additional check_lib) [19:54] <llogan> http://i.stack.imgur.com/SxV5s.jpg [19:54] <llogan> used to illustrate an answer for ubuntu users about their "ffmpeg" [19:55] <gnafu> Hehe. [19:59] <Compn> lol [20:00] <Compn> llogan : why not make an infographic about it ? [20:05] <Compn> llogan : what does ubuntu ffmpeg now say? from the bug trac it looks like they changed it? or no? [20:09] <ubitux> nothing changed [20:10] <llogan> Compn: infographic would be interesting/fun, but meatspace work. [20:10] <llogan> ...is in the way [20:10] <llogan> only new thing that has happened is that the bizarro ffmpeg package has been dropped from Debian experimental, AFAIK. [20:11] <llogan> and i'm probably going to re-open the "deprecated" message bug. [20:11] <ubitux> afaik, on debian/sid, when you checkout ffmpeg, you still get the criminal "FFMPEG DEPRECATED" message [20:11] <Compn> llogan and the two gsoc 2013 pages [20:11] <Compn> s/pages/threads [20:11] <llogan> yeah, that was my fault. [20:12] <llogan> Drunky McDrunkerton [20:53] <burek> is there anyone here who managed to use ffmpeg for real-time voice streaming (like voice chat) [20:54] <durandal_1707> you mean ffmpeg as utility? [20:54] <burek> well, yes [20:55] <durandal_1707> you could use it, as there are some speech encoders [20:55] <burek> well, I'd like to make a working scenario and create a wiki for that [20:55] <Daemon404> libcelt or libopus [20:55] <Daemon404> are what you want likely [20:55] <burek> since I realized by accident that a lot of people are eager to know that [20:56] <Daemon404> i dont know if the cli util can... seems more suited to lib use [20:56] <burek> i see, and what containers would you recommend for those two [20:56] <Daemon404> opus is ogg ofc [20:56] <Daemon404> no idea what celt uses [20:56] <JEEB> I'm not sure what ffmpeg even supports for opus [20:56] <durandal_1707> same [20:56] <Daemon404> (ogg is crap at streaming) [20:56] <JEEB> to mux [20:56] <JEEB> I think it has a matroska mapping, which is EXPERIMENTAL [20:57] <JEEB> (and IIRC not the same that Mosu implemented) [20:57] <Daemon404> lol awesome [20:57] <Daemon404> 2 years from now, bugs galore [20:57] <JEEB> also ffmpeg IIRC didn't even mark it as experimental in the ID or whatever [20:57] <durandal_1707> JEEB: can you point at what exactly is broken? [20:58] <burek> it would be very cool if we could make just a basic example of using ffmpeg <-> ffplay for creating a simple p2p voice chat [20:58] <podman> Compn: hey, this is Adam from SproutVideo. Just saying hi. [20:58] <durandal_1707> ffmpeg mark it as experimental, libav still does not [20:58] <burek> no need for something professional and stuf for a start [20:58] <JEEB> funky [20:58] <JEEB> durandal_1707, not necessarily broken but since the spec for opus-in-mkv still isn't set... [20:58] <JEEB> lemme find mosu's post [20:59] <durandal_1707> JEEB: i read forums threads too [20:59] <JEEB> anyways, opus just shows a thing where matroska is limited (no pre-roll signaling support) [20:59] <JEEB> lulz [21:02] <cone-803> ffmpeg.git 03Clément BSsch 07master:2ecf564f94f8: build: fix iconv detection on some systems. [21:03] <durandal_1707> burek: theoreticaly it could be possible, you could try to setup something if you have two computers [21:03] <burek> no problem, i have a will [21:04] <burek> i could try with opus/speex/celt, and use ts format for a start [21:04] <burek> but since I'm considering using ffplay on the other side [21:04] <durandal_1707> first, ts cant be used with opus/celt [21:04] <burek> is there a better/easier way to not use some standard format, but some, i dunno, internal ffmpeg's format, like nut or so [21:05] <durandal_1707> second using ffplay on other side, means other side could only listen [21:05] <burek> the goal is for that to work, it doesn't have to be standard at all :) [21:05] <burek> durandal_1707 yes [21:05] <durandal_1707> for start you could use 8k alaw [21:05] <burek> and other side will also use ffmpeg -> ffplay for vice versa [21:05] <burek> if you get my idea [21:06] <burek> ok i'll write that down for tests [21:06] <durandal_1707> 8k - 8000 hz sample rate mono [21:06] <burek> yes yes i get it [21:09] <burek> is it possible that I use raw audio output from ffmpeg over udp for example [21:09] <burek> and to capture it on the other side with ffplay [21:10] <durandal_1707> raw over udp? that is not going to work at all [21:11] <durandal_1707> you just pick port and stream to it [21:13] <burek> hmh [21:13] <burek> the problem is i dont know what formats are suitable for which of those, to start from something [21:26] <Compn> podman : oh, hello [21:26] <podman> Compn: starting to build out my own transcoding stack so I figured I'd start lurking around here [21:29] <Compn> youtube-in-an-archive [21:29] <Compn> ehe [21:30] <podman> have a lot of learning to do but it'll pay off [21:30] <podman> should be a lot cheaper than using third party transcoders [21:30] <Compn> sure, yeah [21:31] <Compn> i think its only the corner cases that will give you trouble [21:31] <podman> yeah, luckily i know what a lot of them are since the other providers i used ran into them [21:31] <Compn> for those i'd run the official software under virtual machines. like quicktime + qt_tools project to script it [21:32] <podman> seems like certain WMVs have a lot of weird things with display aspect ratios and whatnot [21:33] <podman> and there seem to be a lot of issues with VFR videos of slide shows [21:35] <Daemon404> [15:31] <@Compn> for those i'd run the official software under virtual machines. like quicktime + qt_tools project to script it <-- except that violates apples licenses [21:35] <podman> i accidently found out that -vf fps seems to fix it, although i have no idea what that filter is or what it does. [21:44] <Compn> Daemon404 : which license ? i mean... what would be the violation? virtualization? or ? [21:46] <podman> i think apple doesn't like people running osx under virtualization [21:47] <Compn> says its ok on apple hardware :P [21:47] <Compn> i didnt say run it on non apple stuff [21:47] <Compn> mr Daemon404 [21:47] <podman> heh [21:47] <Compn> and mr podman [21:47] <Compn> buy a mac :P [21:47] <podman> looks like almost 65% of my uploads are AVC1 [21:48] <podman> so, things should be fairly simple [21:48] <Compn> got any super rare codecs you cant figure out ? :) [21:48] <Compn> thats my hobby, finding old rare codecs [21:50] <podman> I have a feeling FFMPEG covers most of them, but i'll see what i've got [21:51] <Daemon404> um [21:51] <Daemon404> you cant just use capples encoders/decoders for a paid service [21:51] <Daemon404> apple's* [21:51] <Daemon404> is what i meant [21:52] <Daemon404> (paid or not actually) [21:54] <podman> Compn: here is a weird one i just noticed: "Road Pizza" ? [21:54] <Compn> i think rpza [21:54] <Compn> its normal [21:55] <podman> silly name :) [21:56] <podman> I only have a corpus of roughly 100k videos, so I'm not sure I'll have too many odd ones. [22:00] <podman> the majority of uploads seem to be h264 anyway [00:00] --- Thu Feb 28 2013

1 0

ffmpeg.log.20130227
by burek 28 Feb '13

28 Feb '13

[00:00] <bunniefoofoo> Joda, the USB drive doesn't have the write caching checkbox (as the hard drives do), only has "optimize for performance" which has no effect [00:06] <JodaZ> doesn't the write cachign checkbox come avaiable if you uncheck the optimize for fast removal checkbox ? [00:06] <bunniefoofoo> nope [00:07] <JodaZ> hmn :/ [00:09] <bunniefoofoo> I am looking to patch ffmpeg now, or see if there is a flag somewhere. But I don't see an fopen() calls anywhere? [00:10] <JodaZ> just upping write buffer might not be enough, small individual writes might still hurt you [00:11] <bunniefoofoo> I shall soon find out, patching file_open in libavformat [00:19] <evil_core> hi all [00:20] <evil_core> How to compare two movies in quickest way. I got some "dupes", but one have correct audio, and another rip got better video [00:20] <evil_core> I want to know if after joining audio and video from other will not cause off-sync audio [00:22] <evil_core> dumping random frames (long from beginning) takes much time, and usually they are looking as identical [00:35] <bunniefoofoo> joda, seems to be slow because ffmpeg is constantly flushing its output buffer [00:37] <JodaZ> how do you know ? [00:37] <bunniefoofoo> I put a trace on aviobuff flush_buffer(), alternates between 200, 24K and 32K writes [00:38] <JodaZ> nop it ^^ [00:42] <bunniefoofoo> noop fail, nothing written ;-) [00:42] <Catoptromancy> moo [00:44] <Catoptromancy> i have an ogg, if I remaster it and export it as another ogg, does it loose quality? [00:45] <Catoptromancy> or if I save the remaster as a flac will it lose quality? [00:45] <Grublet> There is always a loss, the question is will you notice? [00:45] <Catoptromancy> the recording is bad enough already [00:45] <Catoptromancy> heh [00:45] <Grublet> If you save as a lossless format there won't be any further loss [00:46] <Catoptromancy> remastered ogg to flac will cause no additional loss [00:46] <Catoptromancy> hope its not a huge file [00:46] <Catoptromancy> 4 hours long [00:46] <Grublet> It will be much larger than the original ogg [00:47] <Catoptromancy> I will find out soon [00:47] <Catoptromancy> heh [00:55] <bunniefoofoo> joda, I increased IO_BUFFER_SIZE to 256k, and made flush_buffer() only write if it is a full buffer... speed is great now [03:31] <praveenmarkandu> hi, is HLS4 supported in ffmpeg [04:37] <Catoptromancy> im going to assume that saving a recording as flac, and then editing it later and resaving wont remove any quality? [04:38] <Catoptromancy> flac, edit, another flac [04:38] <Grublet> nope [04:38] <Grublet> no loss [04:38] <Catoptromancy> awesome [04:38] <Catoptromancy> i dont know how I just discoverd flac heh [04:38] <Catoptromancy> I knew of it but never used it [04:39] <Catoptromancy> already replaced my massive wav files [09:43] <mpfundstein> hello, after converting a mpeg2video(elementary stream) to h264(mp4 container), SOMETIMES the output files are corrupt even though the transcoder doesn't throw an error. When ffprobe'ing the output file, i encouter the following error: [mov,mp4,m4a,3gp,3g2,mj2 @ 0xa6f1fe0] moov atom not found. ffmpeg version N-48645-gf3c9d8d [09:47] <mpfundstein> fflogger: http://pastebin.com/cj5Wf0Ua [09:47] <mpfundstein> the command is pretty basic [09:51] <relaxed> "WARNING: library configuration mismatch" can't be good [09:54] <relaxed> medium is the default preset for libx264 [09:54] <relaxed> ffmpeg now takes -preset, instead of -vpre [09:56] <mpfundstein> relaxed: the config mismatch is because of --enable-static / --enable-shared [09:56] <relaxed> I would say, remove that build and make sure you uninstall the libs as well. Compile latest git and try again. Drop --enable-libfaac, since libfdk_aac is much better. [09:56] <relaxed> ok [09:57] <mpfundstein> relaxed: the weird thing is that 99% of the encoding works perfect [09:57] <mpfundstein> it just happens SOMETIMES [09:57] <mpfundstein> can it be because all videos are on a different NAS ? [09:57] <mpfundstein> they get uploaded and soon after upload i post a job to my transcoder queue. they pick it up with high prio, so MAYBE the transcoder doesnt hav full file information yet [09:58] <relaxed> could be, if the encode is in interrupted then the moov atom (index) won't be written. [09:58] <relaxed> -in [09:58] <relaxed> encode to a local disk and then move to the NAS [09:59] <mpfundstein> yeah ill probaly cp the file first [09:59] <relaxed> also, use libfdk_aac [10:00] <mpfundstein> i use [10:00] <relaxed> -c:a libfdk_aac [10:00] <mpfundstein> ah ok for the review not [10:00] <mpfundstein> but that doesnt matter [10:00] <mpfundstein> for the REAL videos i use libfdk_aac [10:00] <relaxed> ok [10:07] <relaxed> mpfundstein: do you test for a zero exit status? [10:10] <mpfundstein> yes of course [10:10] <mpfundstein> :-) [11:28] <DetachedScreen> how can add wmv support to wmv in configure time? [11:28] <DetachedScreen> wmv support to ffmpeg* [11:28] <Trashlord> get a lib that enables wmv support [11:29] <Trashlord> and include it in ./configure [11:29] <Trashlord> with --enable-libname [11:29] <DetachedScreen> do you know the libname? [11:30] <Trashlord> no [11:30] <Trashlord> but try wmv2 [11:30] <Trashlord> try to use that as the video codec [11:31] <DetachedScreen> when i use it it say your ffmpeg has no wmv support [11:31] <Trashlord> try --enable-wmv2 on ./configure [11:31] <Trashlord> see if it works [11:32] <DetachedScreen> Unknown option "--enable-wmv2" [11:32] <Trashlord> then I don't know [11:32] <DetachedScreen> thanks anyway [11:33] <Trashlord> why do you want wmv anyway? [11:33] <Trashlord> and try ffmpeg -i <file> <file.wmv> [11:33] <Trashlord> see if that works [11:34] <DetachedScreen> for my windows mpbile powered Pocket PC [11:35] <Trashlord> ok [11:35] <Trashlord> then just do like [11:36] <Trashlord> ffmpeg -i <file> -vcodec libx264 -acodec vorbis outputfile.wmv [11:36] <Trashlord> try that [11:36] <Trashlord> and see if it plays [11:36] <Trashlord> if vorbis doesn't work, try libaac [11:37] <Trashlord> if it says vorbis unknown codec, try libvorbis [11:37] <Trashlord> I'll bbs [11:39] <DetachedScreen> strangly it is encoding [11:39] <Trashlord> ok [11:39] <Trashlord> then try to play it on your device [11:39] <DetachedScreen> it takes time , i ll see [11:41] <relaxed> DetachedScreen: wmv is enabled by default. [11:42] <DetachedScreen> relaxed: in ffmpeg -formats it is not there [11:42] <relaxed> DetachedScreen: ffmpeg -i input -c:v wmv2 -q:v 3 -c:a wmav2 -b:a 128k -f asf output.wmv [11:43] <relaxed> ffmpeg -codecs | grep wm [11:43] <DetachedScreen> yess in codecs it is there [11:43] <relaxed> asf is the format [11:43] <DetachedScreen> ah [11:44] <relaxed> which is want *.wmv files are [11:44] <DetachedScreen> thanks [11:45] <Trashlord> yeah, thanks for that [11:45] <Trashlord> in the odd chance that I'll ever want to use wmv, now I know how to do it [11:46] <relaxed> you're welcome [11:56] <Yulth> Hi everyone! [11:56] <Yulth> In this line: video:0kB audio:27128kB subtitle:0 global headers:0kB muxing overhead 1.176652%, What exactly means 'muxing overhead'? [11:58] <JEEB> overhead by the container used [11:58] <sacarasc> When a stream is put into a container, the container needs to have some data itself. That is the data. [12:00] <Yulth> and these data needs 1'17 % of the file's size? [12:00] <JEEB> in this case, yes [12:00] <Yulth> I understand [12:01] <Yulth> is there any way to minimize this amount of data needed? [12:01] <JEEB> generally no [12:02] <JEEB> I mean you could compare between various containers, but in general the benefit of that will be rather miniscule [12:02] <JEEB> also it does depend on how big data you're putting there, and what kind of data [12:02] <JEEB> and so forth and so forth [12:02] <Yulth> ok, I note [12:08] <relaxed> Yulth: if you need a specific size use 2 pass encoding. [12:18] <Yulth> relaxed: very interesting! Could you give me some info please? [12:28] <relaxed> Yulth: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide [12:28] <relaxed> ^^ there's a section on it [12:29] <Yulth> ok, I need only encode to HE-AAC. Is on there this info? [12:30] <relaxed> Hmm, I don't think audio supports random bitrates. [12:31] <JEEB> set -b:a ? [12:31] <JEEB> relaxed, you should go take some coffee [12:31] <relaxed> well, it depends on the codec, does it not? [12:31] <JEEB> also this person was asking about muxing overhead and if he could minimize it [12:32] <JEEB> naturally [12:32] <JEEB> some formats are more limited, others aren't [12:53] <relaxed> does muxing overhead not correlate with size? [12:54] <JEEB> yes, in general the bigger bit rate you use, the less the overhead will be [12:54] <JEEB> but of course that depends on the container [13:10] <burek> test [13:11] <durandal_1707> SEGV [13:13] Action: Keshl flashes capslock, numlock and scrolllock on and off. [14:20] <code-guru> hi [14:20] <code-guru> how can initialize AVDictionary struct ? [14:20] <code-guru> except av_dict_set [15:42] <devios> hey all - using ffmpeg -ss (time) -an -vframes 1 to create a screenshot of an AVC (v_mpeg4/iso/avc) video in an MKV (v2) container, and the screenshot looks awful (it's all grey) - I know the video is OK - what would cause that? [15:46] <ubitux> code-guru: why would you want to do that? [15:57] <hughmanwho> Any in here have any development experience? I'm having trouble streaming H264 from RTSP. Here is a pastebin with my code: http://pastebin.com/mMSdFCAM [16:21] <hughmanwho> The pastebin is a little long let me know if you want me to further explain what's going wrong.. basically it tells me it's working but the frames it's outputting are garbage. [17:00] <Doxin> doing ffmpeg -i blag.mp3 -acodec mp3 -ab 16k oblah.mp3 gives me an error: "Unknown encoder 'mp3'", what do I do to get this to work? [17:02] <Trashlord> try aac [17:02] <Trashlord> libaac [17:02] <Trashlord> or use lame [17:03] <Trashlord> also, 16k? why so low? [17:04] <Trashlord> hm, try libmp3lame [17:04] <divVerent> if he WANTS mp3... isn't the acodec libmp3lame? [17:04] <divVerent> but 16k is REALLY very low [17:04] <Doxin> yesyes [17:04] <Doxin> I know [17:04] <divVerent> mp3 below 128k is aural suicide... at 128k it's aural torture, I wouldn't use below 192k for stereo. [17:05] <Doxin> still not working [17:05] <Doxin> libmp3lame is unknown [17:05] <divVerent> you need to compile ffmpeg with it enabled [17:05] <Doxin> aac won't work because it's experimental [17:05] <divVerent> --enable-libmp3lame is the compile option [17:05] <divVerent> aac can be enabled by -strict -2, but I advise against it [17:05] <divVerent> the builtin aac encoder is actually worse than mp3 in quality [17:05] <divVerent> (worse than libmp3lame, that is) [17:05] <Doxin> no matter, because it still won't work. [17:06] <divVerent> what is the error? [17:06] <Trashlord> I use libfdk_aac [17:06] <divVerent> yes [17:06] <divVerent> if you want aac, compile with libfdk_aac and use it [17:07] <Doxin> aac seems to be in there, but it keeps complaining about it being experimental [17:07] <Doxin> anyways, nvm, I'll dick around on my own some more. [17:07] <Doxin> thanks a bunch [17:07] <divVerent> -strict -2 turns off the experimental complaint [17:08] <divVerent> but the real solution is to recompile ffmpeg with a good set of codec libraries [17:46] <hughmanwho> I have a program that in theory says it is successfully pulling in H264 via RTSP and saving it as YUV frames but is actually outputting garbage. Any thoughts on what to check? Here is a pastebin of program if wanted: http://pastebin.com/mMSdFCAM [18:00] <Olive6767> Hi All :-) [18:03] <Olive6767> I'm using "-c:s mov_text" to add .srt soft subtitles to an mp4 video. Subs are added fine, however the video become lagguy when trying to fastforward... this seems to be the case whatever vid and subs I'm using. I suspect this is related to framerate although I'm not sure, and don't know anyway how I could fix it. Thus, any help is appreciated. [18:04] <Olive6767> lagguy = laggy [19:04] <kollapse> Hi. I have a video file at 59.94 frames per second that I am trying to play on a Smart TV, but the TV recognizes it as being 59 fps and fails due to "incompatible mode". Is there any way to encode the file so it has 60 fps sharp so the TV can't complain any longer ? [19:12] <iive> the framerate is probably the least problem to the tv. most likely the file uses too high profile/level. [19:13] <kollapse> iive, 1080/60p AVCHD 2.0 with "Video: h264 (High) (HDPR / 0x52504448), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 119.88 tbc" [19:45] <tg2> If i'm converting into 4 output formats from the same input, what is the best way to do 1 pass on the first then multiple second passes for each format in a way that wont be doing too much duplicate work. [19:46] <tg2> basically i just want to generate the mbtree the first time, then use that for each of the 4 second passes (different sizes and bitrates) [19:59] <cryptopsy> how do i find out if an encoder has multithreading capabilities ? [19:59] <cryptopsy> interested in .avi [20:16] <llogan> cryptopsy: avi is a(n outdated) container that can utilize a multitude of media formats produced by several encoders [20:16] <cryptopsy> avi encoder [20:16] <llogan> there is no avi encoder [20:17] <cryptopsy> mp3->avi [20:17] <llogan> ffmpeg -i audio.mp3 -c copy output.avi [20:17] <cryptopsy> what encoder is used?. [20:18] <Fjorgynn> lol? [20:18] <llogan> re-encoding is not used when stream copying [20:18] <llogan> as in my example [20:18] <cryptopsy> then why is taking like 30 minutes per 5 minutes of track? [20:18] <cryptopsy> pb what [20:19] <llogan> please use a pastebin site (like www.pastie.org or www.pastebin.com) to show your ffmpeg command and the complete console output [20:19] <tg2> @cryptopsy [20:20] <tg2> avi is a container [20:20] <tg2> you can put many things in it [20:20] <tg2> mp3 for audio, mpeg2 for video [20:20] <tg2> etc [20:20] <tg2> instead of reencoding the mp3 (whic is what you are doing), you should just take the mp3 and stuff it in an avi container [20:20] <cryptopsy> http://ompldr.org/vaGx3ag/lol [20:20] <tg2> whichi s the command that llogan put [20:20] <tg2> paste the entire command [20:20] <tg2> not just ffmpeg output [20:21] <cryptopsy> i'm using -acodec copy [20:21] <cryptopsy> $FFMPEG -threads 4 -loop_input -i $1 -i $2 -acodec copy -y -t $TIME $3 [20:21] <cryptopsy> $1 is an image, $2 is mp3, $3 is out file [20:21] <tg2> -acodec copy [20:21] <tg2> = its just copying the stream [20:21] <tg2> so there is no 'reencoding" happening [20:21] <tg2> 12 fps [20:22] <tg2> lol [20:22] <llogan> you are using the video encoder named "mpeg4" in this instance [20:22] <cryptopsy> how many fps do you need to look at a looping image? [20:22] <cryptopsy> llogan: aka the default [20:22] <tg2> you want a static image while the mp3 plays? [20:22] <llogan> and you're applying threads to the input (the decoder) [20:22] <Fjorgynn> 1? [20:22] <tg2> -threads 4 has to be after -i [20:22] <tg2> but thats nto the issue [20:23] <cryptopsy> mpeg4 encoder don't support multithread? [20:24] <tg2> what kind of computer are you runnin git on? [20:24] <tg2> $FFMPEG -threads 4 -loop_input -i $1 -i $2 -acodec copy -y -t $TIME $3 [20:24] <tg2> should be [20:25] <cryptopsy> modern shitbox [20:25] <tg2> $FFMPEG -loop_input -i $1 -threads 4 -i $2 -acodec copy -y -t $TIME $3 [20:25] <tg2> if it doesn't have multiple cores [20:25] <cryptopsy> 14:37:05 < cryptopsy> mpeg4 encoder don't support multithread? [20:26] <tg2> yes it does [20:26] <RSD> While compiling I have the following error: [20:26] <RSD> libavcodec/libavcodec.a(utils.o): In function `recode_subtitle': [20:26] <RSD> /usr/local/src/ffmpeg/ffmpeg/libavcodec/utils.c:1902: undefined reference to `libiconv_open' [20:26] <RSD> /usr/local/src/ffmpeg/ffmpeg/libavcodec/utils.c:1941: undefined reference to `libiconv_close' [20:26] <RSD> /usr/local/src/ffmpeg/ffmpeg/libavcodec/utils.c:1927: undefined reference to `libiconv' [20:26] <RSD> /usr/local/src/ffmpeg/ffmpeg/libavcodec/utils.c:1928: undefined reference to `libiconv' [20:26] <RSD> what can I do? [20:26] <tg2> but you're putting -threads before -i [20:26] <cryptopsy> use a pastebin ^ [20:26] <tg2> has to be after -i [20:26] <Guest4965> it is on centos 6.3 [20:26] <cryptopsy> tg2: where after -i ? [20:26] <Mavrik> Guest4965, you're not linking libiconv to your binary [20:26] <tg2> $FFMPEG -loop_input -i $1 -threads 4 -i $2 -acodec copy -y -t $TIME $3 [20:27] <cryptopsy> tg2: the first i is an image [20:27] <tg2> yes but that is also what is being "encoded" [20:27] <tg2> since you're telling it make a video with this image over and over [20:27] <tg2> its still encoding it [20:27] <cryptopsy> then i can place it after the second i too [20:27] <Guest4965> Ok thanks [20:27] <tg2> Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 600x849 [SAR 1:1 DAR 200:283], q=2-31, 200 kb/s, 25 tbn, 25 tbc [20:27] <tg2> no [20:28] <tg2> audio is just copy [20:28] <tg2> there is no threading on copy [20:28] <tg2> there doesn't need [20:28] <RSDRSDRSD> While compiling I have the following errors [20:28] <RSDRSDRSD> http://pastebin.com/tXQGrDX9 [20:28] <llogan> cryptopsy: refer to "ffmpeg -encoders". it tells you if an encoder supports multithreading (and if it is frame/slice multithreading) [20:28] <RSDRSDRSD> I am following the centoscompilationguide [20:28] <tg2> FMP4 is threaded though I think [20:28] <cryptopsy> llogan: tn [20:28] <cryptopsy> x [20:29] <RSDRSDRSD> tg2 what means !pb ? [20:29] <tg2> says -loop_input is depreciated, and to use -loop 1 [20:29] <cryptopsy> still not using more than 1 core [20:29] <cryptopsy> tg2: ^ [20:29] <tg2> set [20:29] <tg2> -threads 0 [20:29] <tg2> try that maybe [20:30] <tg2> do you ahve more than 1 cpu in that machine? [20:30] <cryptopsy> no [20:30] <tg2> or a dual core at least? [20:30] <cryptopsy> i have 4 cores [20:30] <tg2> ok [20:30] <burek> is there anyone here who managed to use ffmpeg for real-time voice streaming (like voice chat) [20:30] <llogan> RSDRSDRSD: that (!pb) is a shortcut to tell the #ffmpeg channel bot, fflogger, to tell you to "use a pastebin site (like www.pastie.org or www.pastebin.com) to show your ffmpeg command and the complete console output" [20:30] <RSDRSDRSD> oh ok thanks [20:30] <cryptopsy> tg2: nop [20:30] <RSDRSDRSD> sorry for that [20:30] <tg2> you can try using codec libxvid [20:31] <llogan> RSDRSDRSD: did you deviate from the guide at all? [20:31] <tg2> Multi-threading works fine here for the native FFmpeg MPEG-4 ASP encoder [20:31] <Fjorgynn> Jobba! [20:32] <tg2> try using -vcodec mpeg4 [20:32] <RSDRSDRSD> No I followed the guide [20:32] <tg2> then -threads 4 after that [20:32] <cryptopsy> what's the full command? [20:32] <cryptopsy> i gotta know where the i's go [20:32] <RSDRSDRSD> only the checkinstall wasn working for centos 6.3 [20:32] <RSDRSDRSD> so I did a make install [20:32] <RSDRSDRSD> for the aac encoder [20:33] <tg2> $FFMPEG -loop 1 -i $1 -i $2 -vcodec mpeg4 -threads 4 -acodec copy -y -t $TIME $3 [20:33] <tg2> try it [20:33] <tg2> and see if it works [20:33] <llogan> someone added that checkinstall section without trying it apparently. and that guide will eventually be converted to a non-system install [20:33] <RSDRSDRSD> what does that mean? [20:34] <RSDRSDRSD> instead of checkinstall I did a make install [20:34] <RSDRSDRSD> and it gave me no errors [20:34] <llogan> that's fine [20:34] <cryptopsy> tg2: nope [20:35] <RSDRSDRSD> but now ffmpeg is complaining about libiconv [20:35] <tg2> on what operating system crypto [20:35] <cryptopsy> shittux [20:35] <tg2> $FFMPEG -loop 1 -i $1 -i $2 -vcodec mpeg4 -qscale 2 -threads 4 -acodec copy -y -t $TIME $3 [20:35] <tg2> which version [20:35] <cryptopsy> what do you mean which version [20:35] <Fjorgynn> Hej [20:35] <tg2> distro [20:35] <cryptopsy> version GNU [20:35] <tg2> vanilla? [20:35] <cryptopsy> i don't understand [20:36] <cryptopsy> you mean ffmpeg version? [20:36] <tg2> no i mean on which operating system was it compiled [20:36] <RSDRSDRSD> This is my new pastebin with compile command for ffmpeg [20:36] <RSDRSDRSD> http://pastebin.com/QE2nCCuQ [20:36] <cryptopsy> tg2: linux [20:36] <tg2> if its compiled from source [20:37] <tg2> on vanilla linux [20:37] <tg2> assumin gyou used the right compiler [20:37] <tg2> it should work with threading in avi [20:37] <tg2> with mpeg4 encoder [20:37] <cryptopsy> since which version is this supported? [20:37] <tg2> reading docs from 2011 and its in there [20:37] <tg2> so at least 2 years [20:37] <cryptopsy> ill check later, i gotta write an exam in like 5mins [20:38] <tg2> :| [20:38] <tg2> noobs [20:38] <tg2> anywa [20:38] <RSDRSDRSD> no one has a clue [20:39] <llogan> RSDRSDRSD: be patient [20:39] <RSDRSDRSD> ok ;-) [20:39] <tg2> damn google already indexed that pastbin [20:39] <tg2> LOL [20:39] <tg2> i searched for the error and result 4 was that pastbin from 4 minutes ago [20:39] <tg2> :| [20:39] <tg2> gooogle u scary [20:40] <RSDRSDRSD> hehe that[ indeed really scary [20:40] <tg2> http://i.imgur.com/x220sRF.png [20:41] <RSDRSDRSD> yes, by me it is on the first place [20:41] <tg2> heh [20:43] <ubitux> RSDRSDRSD: are you on mac os? [20:43] <RSDRSDRSD> nope [20:43] <ubitux> what os? [20:44] <RSDRSDRSD> CentOS [20:44] <RSDRSDRSD> 6.3 [20:44] <ubitux> can you test something for me? [20:44] <RSDRSDRSD> yes [20:44] <ubitux> open the configure script with your editor [20:44] <ubitux> look for "check_func iconv || check_lib2 iconv.h iconv -liconv" [20:44] <ubitux> and replace it with "check_func iconv" [20:44] <ubitux> run the configure again, and try to build [20:46] <RSDRSDRSD> it is running [20:46] <RSDRSDRSD> still the same error [20:47] <ubitux> mmh. [20:47] <ubitux> can you paste your config.log somewhere? [20:49] <ubitux> it looks like the iconv availability check will need some more advanced check.. [20:49] <llogan> bah, whoever added fdk-aac to the centos guide didn't even test it [20:50] <durandal_1707> what happens? [20:50] <RSDRSDRSD> http://pastebin.com/X4dSk4zg [20:51] <RSDRSDRSD> a while ago I used that guide zo I thought it would work, but now it is broken unfortunately [20:52] <ubitux> i'm likely the one responsible from that breakage [20:52] <ubitux> sorry about that, i'm looking into fixing it [20:52] <RSDRSDRSD> iconv is installed on my system [20:52] <RSDRSDRSD> iconv: /usr/bin/iconv /usr/local/bin/iconv /usr/include/iconv.h [20:52] <ubitux> -check_func iconv || check_lib2 iconv.h iconv -liconv [20:52] <ubitux> +check_func_headers iconv.h iconv || check_lib2 iconv.h iconv -liconv [20:52] <ubitux> can you try this? [20:53] <durandal_1707> the iconv was added recently and that is causing breakage for you [20:56] <RSDRSDRSD> looks like it is working now [20:56] <ubitux> RSDRSDRSD: can you grep ICONV config.h [20:56] <ubitux> ? [20:57] <RSDRSDRSD> #define HAVE_ICONV 1 [20:57] <ubitux> and it links fine now? [20:57] <RSDRSDRSD> how do you mean? [20:58] <RSDRSDRSD> I can make and make install now [20:58] <RSDRSDRSD> yes [20:58] <RSDRSDRSD> configuration: --enable-gpl --enable-libfdk_aac --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libtheora --enable-nonfree --enable-gpl --enable-version3 [20:58] <beastd> ubitux: hmm, how it comes it says "libiconv_open". i thought that functions were prefixed with iconv_ [20:58] <ubitux> ok great, thx for your testing [20:59] <RSDRSDRSD> is this going to be added in git or do I have to write up the changes? [20:59] <ubitux> RSDRSDRSD: i'll push soon [20:59] <ubitux> give me a few minutes [20:59] <RSDRSDRSD> ok didn know that was possible ;-) [21:00] <ubitux> beastd: no idea, maybe some conflicts between the two [21:03] <ubitux> RSDRSDRSD: pushed [21:03] <RSDRSDRSD> ok great ;-) many thnx [21:03] <ubitux> thx for the testing [21:03] <RSDRSDRSD> that guide also misses the libtool [21:03] <ubitux> libtool is not required to build ffmpeg [21:04] <RSDRSDRSD> no not for ffmpeg, but for some codec [21:04] <RSDRSDRSD> on that guide [21:06] <RSDRSDRSD> I had to install libtool for fdk-aac [21:08] <RSDRSDRSD> have to go bye bye and many thanls [21:24] <qbasicer> Hey all, I seem to be having troubles with getting ffmpeg to put together a directory of still images. It seems to get to frame 730 regardless of codec and say input/output error. Here's the command line and output: http://pastebin.com/E9grQphW [21:24] <qbasicer> Can somebody give me a quick hand with this? There seems to be lots of disk free (40G) [21:25] <qbasicer> I tried deleting the pictures around the 730 mark (all 700-799), but it failed too [21:25] <qbasicer> I tried -v debug but was unable to see anything that could help me. If somebody thinks that'd be useful, I can pastebin that too. [21:25] <burek> qbasicer [21:26] <burek> your image sequence needs to be continous [21:26] <qbasicer> It is, img0001.jpg to img1124.jpg [21:26] <burek> no missing images, for example this wont work: img1234.jpg, img1235.jpg, img1237.jpg, img1238.jpg [21:26] <qbasicer> Ohhh I see what you mean [21:26] <burek> if you delete an image, you need to renumerate them [21:27] <burek> or [21:27] <burek> you could use cat *.jpg | ffmpeg -f image2 -i - .... [21:27] <burek> but, that won't have any order [21:27] <burek> wait, let me find you an example [21:28] <burek> there: http://ffmpeg.org/trac/ffmpeg/wiki/Create%20a%20video%20slideshow%20from%20… [21:28] <qbasicer> I was following a modified script from http://www.ffmpeg.org/faq.html#How-do-I-encode-single-pictures-into-movies_…, the one where they do the symlinks [21:29] <burek> oh [21:29] <burek> that creates a bunch of /tmp symlinks... [21:30] <burek> but ok, if it works, then ok [21:30] <burek> but i would rather go for something like: ls (sorting options) | cat | ffmpeg ... [21:31] <qbasicer> Sort of complicating it, I really only want every 10th image, which I guess I could do in a bash loop and pipe through it [21:37] <qbasicer> burek: Thanks for the tips! I think I got it working [21:38] <burek> :beer: :) [21:41] Action: qbasicer passes over 0.1 BTC :) [21:54] <Aaronds> Hi I'm converting wmv files for use as html5 video... I converted one video and it works fine, but the second video "Fails to decode". It plays fine on my desktop but not on the web... Anyone know what may be the problem? http://pastebin.com/wPcfkxnH [22:12] <burek> brb [22:14] <Aaronds> oh burek I'll add the console output hold on [22:17] <msmithng> is there anyway for me to tell ffprobe to skip stream 0, in the event of input that has no audio? [22:17] <msmithng> it's trying to get it [22:17] <msmithng> [mpegts @ 0x102009800] Could not find codec parameters for stream 0 (Unknown: none ([21][0][0][0] / 0x0015)): unknown codec [22:17] <msmithng> and suggests that I increase analyzedduration [22:17] <msmithng> but obviously that's not going to work [22:27] <svchost-> Hi. If I merge different MP4 movies (with different frame rate and data speed), might quality be lost? Are framerates etc. averaged (so frames might be interpolated from more frames) in output mp4? [23:28] <hughmanwho> I spent all day trying to figure out why my program wasn't working... it was, turns out the image viewer I was opening the file from wasn't I screwed with some settings! [23:46] <bunniefoofoo> is there a way to get ffmpeg to use buffered I/O on the output side... it seems there is little to no buffering at all [23:47] <bunniefoofoo> in particular, avio_flush() is called frequently [23:51] <bunniefoofoo> the issue presents itself when writing to an unbuffered medium like flash memory drive which typically do not have any write buffering due to data corruption concerns [23:51] <bunniefoofoo> of course I am referring to USB flash drives (thumb drives) and not SSD [00:00] --- Thu Feb 28 2013

1 0

ffmpeg-devel.log.20130226
by burek 27 Feb '13

27 Feb '13

[02:56] <cone-850> ffmpeg.git 03Michael Niedermayer 07master:bfcc21a4725d: libavcodec/motion-test: set the bitexact flag [02:56] <cone-850> ffmpeg.git 03Michael Niedermayer 07master:5bcb379ffe54: motion-test: fix warning: dsp_mask is deprecated [02:56] <cone-850> ffmpeg.git 03Michael Niedermayer 07master:f6fff8e54697: ac3enc_template: silence may be used uninitialized in this function warnings [03:39] <cone-850> ffmpeg.git 03Matt Wolenetz 07release/1.1:bc9d341be84e: x86: h264: Don't use redzone in AVX h264_deblock on Win64 [03:39] <cone-850> ffmpeg.git 03Reinhard Tartler 07release/1.1:88ae77cea4c1: update Changelog [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:b786ddc0f2ea: loco: check that there is data left after decoding a plane. [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:488ffb813514: mov: use the format context for logging. [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:8bce2c60b8eb: lagarith: avoid infinite loop in lag_rac_refill() [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:612b28194b4c: flicvideo: avoid an infinite loop in byte run compression [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:48fd461977f7: av_memcpy_backptr: avoid an infinite loop for back = 0 [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:e2cf32ca5f58: mlpdec: do not try to allocate a zero-sized output buffer. [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:a6403a3b6917: qtrle: add more checks against pixel_ptr being negative. [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:bb3f1cad171b: 4xm: check the return value of read_huffman_tables(). [03:39] <cone-850> ffmpeg.git 03Anton Khirnov 07release/1.1:77493bfd9762: cavs: initialize various context tables to 0 [03:39] <cone-850> ffmpeg.git 03Reinhard Tartler 07release/1.1:a991c0673f71: update Changelog [03:39] <cone-850> ffmpeg.git 03Reinhard Tartler 07release/1.1:670128ff13da: Prepare for 9.3 Release [03:39] <cone-850> ffmpeg.git 03Vicente Jimenez Aguilar 07release/1.1:b6ae41e7f425: doc: filters: Correct BNF FILTER description [03:39] <cone-850> ffmpeg.git 03Diego Biurrun 07release/1.1:dc745b76aa9a: doc: developer: Allow tabs in the vim configuration for Automake files [03:39] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:f64e4a8c9a19: Merge remote-tracking branch 'qatar/release/9' into release/1.1 [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:6e8ed38fabb5: aac: reconfigure output on pop [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:3348e66e2eb3: doc/APIchanges: fix odd .01 versions [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:ece16d91ee43: apichanges: fix date [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:4bde8c1369e7: apichanges: Use , instead of / to seperate multiple hashes [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:98e96652f11d: apichanges: fix 2 wrong hashes [05:00] <cone-850> ffmpeg.git 03Michael Niedermayer 07release/1.1:50ebb524cd54: doc/APIchanges: List merge commit hashes and version numbers [11:22] <durandal_1707> michaelni: the strange split for alpha is present in RGB48 case to [11:22] <durandal_1707> is that bug? [11:25] <durandal_1707> michaelni: also i set wrong alpha value, you did not notice it [11:32] <durandal_1707> i changed alpha to 1 in yuv2rgb and now i get segv [11:34] <durandal_1707> hmm, i think i get why segv happens, lets prove it... [11:35] <durandal_1707> michaelni: will add swscale rewrite task as gsoc and will mentor it if noone will not [12:01] <durandal_1707> i think i'm missing something in ff_yuv2rgb_c_init_tables [12:37] <cone-435> ffmpeg.git 03Diego Biurrun 07master:b58b00aeca21: configure: Separate "ln" command line arguments [12:37] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:2b277f2992d6: Merge commit 'b58b00aeca21de00ab6da2944684f784d9d6bc47' [12:43] <cone-435> ffmpeg.git 03Diego Biurrun 07master:b2d688ea9f9c: configure: Identify icc compiler with a less ambiguous pattern [12:43] <cone-435> ffmpeg.git 03Justin Ruggles 07master:e951b6d94c44: vorbisdec: cosmetics: rename variable avccontext to avctx [12:43] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:13fa07417326: Merge commit 'e951b6d94c441d46b396ef12da1428297d77251d' [12:56] <cone-435> ffmpeg.git 03Justin Ruggles 07master:09031b463966: vorbisenc: cosmetics: rename variable avccontext to avctx [12:56] <cone-435> ffmpeg.git 03Justin Ruggles 07master:699d02b839ff: libschroedinger: cosmetics: rename variable avccontext to avctx [12:56] <cone-435> ffmpeg.git 03Justin Ruggles 07master:e8da807537e3: cmdutils: only use libavresample when it is enabled [12:56] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:6fbddc80d63d: Merge commit 'e8da807537e314d74cb6d93598f1dcfb891fa655' [13:02] <cone-435> ffmpeg.git 03Justin Ruggles 07master:d925cca95f5d: avconv: remove an unused variable [13:02] <cone-435> ffmpeg.git 03Diego Biurrun 07master:76b19a398435: Fix a number of incorrect intmath.h #includes. [13:02] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:9b1a0c2ee82f: Merge commit '76b19a3984359b3be44d4f7e4e69b7b86729a622' [13:13] <cone-435> ffmpeg.git 03Diego Biurrun 07master:c242bbd8b693: Remove unnecessary dsputil.h #includes [13:13] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:a984efd104cc: Merge commit 'c242bbd8b6939507a1a6fb64101b0553d92d303f' [13:19] <cone-435> ffmpeg.git 03Diego Biurrun 07master:3a02b6884cda: configure: icc: Drop nonsense adding of cpuflags to LDFLAGS [13:19] <cone-435> ffmpeg.git 03Michael Niedermayer 07master:e907aa98cd27: Merge remote-tracking branch 'qatar/master' [13:35] <durandal_1707> i cant't find any posterize filter that saste is talking about [13:59] <saste> durandal_1707, elbg [14:02] <durandal_1707> and why it was not accepted? [14:02] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/101397/focus=101443 [14:03] <saste> durandal_1707, ^^ [14:04] <saste> Random-generator solution with flling palette and output with random entries would be even faster but produces images with worse quality [14:04] <saste> kostya reply :) [14:05] <durandal_1707> idea about sws is limited [14:05] <durandal_1707> pal8 use only 256 colors [14:05] <durandal_1707> and there is now way to control that to higher/lower number [14:06] <durandal_1707> or do different kind of posterization [14:06] <wm4> or do processing that yields output optimal for multiple frames? [14:07] <saste> durandal_1707, basically it needs to be updated to support more formats [14:07] <saste> but I suppose even an updated version of the patch would be acceptable in its current form [14:07] <saste> another thing that i wanted to implement [14:08] <saste> the possibility to select the palette [14:08] <saste> right now it computes the palette at each frame IIRC [14:08] <durandal_1707> it is implemeneted in swscale, something i'm not interested [14:09] <wm4> swscale is a rewrite candidate, patches welcome [14:17] <durandal_1707> wm4: want to help? [14:17] <wm4> patches welcome [14:20] <wm4> hm there was some other library which can do scaling and pixel conversion [14:20] <wm4> but I can't find it [14:20] <wm4> oh, there: http://forum.doom9.org/showthread.php?t=166504 [14:21] <durandal_1707> it is vapoursynth plugin [14:28] <durandal_1707> on positive side it is wtfpl license [14:30] <durandal_1707> wm4: have any other library in mind? [14:30] <wm4> not really... I also found pixman (has some new code for yuv conversions) and gstreamer shit (uses some sort of dynamic compiler...) [14:35] <durandal_1707> dynamic compiler? [14:36] <durandal_1707> i found babl [14:37] <durandal_1707> then there is pixfc-sse [14:39] <durandal_1707> babl lacks dithering and not multiple of 8 support [14:39] <wm4> durandal_1707: this is the dynamic compiler thing: http://liboil.freedesktop.org/ (link doesn't seem to work right now) [14:39] <durandal_1707> dunno how hard would be to add that to babl [14:40] <wm4> lol pixfc-sse employs similar techniques as swscale [14:42] <wm4> also it's GPL3 [14:45] <wm4> gstreamer also has a very old copy of some pre-swscale ffmpeg format conversion (?) [14:47] <Compn> wm4 : there should obviously be one in imagemagick [14:49] <wm4> durandal_1707: oh, here's gstreamer's conversion stuff: http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/gst/videoconver… [15:11] <durandal_1707> babl use lgpl 3 [15:25] <durandal_1707> wm4: "patch welcome" is all what you get here [15:25] <wm4> durandal_1707: patches welcome to fix that [16:56] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:c6d3b3be1555: aacsbr: Silence warning: max_qmf_subbands may be used uninitialized in this function [16:56] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:e4eedb983dbc: swscale-test: fix 3 pointer type warnings [19:11] <Compn> whoa [19:11] <Compn> xvba support?!? [19:12] <Compn> xbmc guys are working hard :) [19:18] <nevcairiel> why does linux not manage to create one hw interface that everyone supports? now we have vdpau, vaapi, xvba, and probably more :p [19:20] <Compn> nevcairiel : erm, you try getting intel, nvidia and ati to place nicely in this thing where everyone has patents on every little bit of 3d video tech from 20 years ago [19:21] <nevcairiel> works for windows [19:21] <Compn> i bet somewhere, some company has those 3dfx patents [19:21] <nevcairiel> nvidia bought 3dfx [19:21] <Compn> there you go [19:24] <nevcairiel> its still in their interest to support one API so more applications support their hardware [19:32] <Compn> well then you're talking about 80% windows and 10% linux [19:32] <Compn> so they spend time on 80% windows [19:32] <Compn> ignore 10% linux and move on [19:32] <Compn> drop some patches and run [19:32] <Compn> nvidia dedicated a guy to linux support, thats most of the reason vdpau was accepted in most applications [19:33] <Compn> plus a lot of devs had nvidia hardware ... [19:34] <Compn> like carl, uau and others [19:40] <cone-485> ffmpeg.git 03Michael Niedermayer 07master:997a36238ff0: mpeg12: Detect MXF essence stuff at the end of frames [20:15] <iive> for a long time xvba had no official documentation. I've heard the vaapi wrapper was written under NDA [00:00] --- Wed Feb 27 2013

1 0

ffmpeg.log.20130226
by burek 27 Feb '13

27 Feb '13

[01:01] <Mista_D> Can I overlay watermark_1 in the beginning of the video for 10 seconds, and watermark_2 at the end 10 seconds before end? [04:13] <PaulWay> Hi there! [04:13] <PaulWay> I'm trying to add metadata an MP4 file using ffmpeg, but the file I created didn't seem to get the metadata. [04:15] <PaulWay> The command I'm using is http://pastebin.com/KUL8yjni [07:14] <anonus> hello [07:16] <anonus> i have a live audio stream with a lot of awful noise in background, but silentdetet filter quite well detects silence intervals, but can i somehow make it to put blank audio instead of original noise when silence is detected? [07:20] <anonus> simply saying i need to reduce to 0 signal below some level [07:20] <anonus> is ffmpeg capable to do so? [07:32] <klaxa> anonus: i'm not sure whether or not ffmpeg supports it, but i know that sox can apply a noisegate filter [07:33] <klaxa> if you google for "sox noisegate" you should find some information on that [09:13] <Yulth> Hi everyone! [09:14] <Yulth> Could anybody give me please an example about how to encode to OPUS format? [09:30] <jeje2> Hi to all [09:31] <jeje2> I'm back with my problem of high CPU usage when I decompress several different H264 streams from IP camera [09:33] <jeje2> it seems using swscale take me a lot of CPU usage too [09:34] <jeje2> if I just decompress and not use swscale to transfrom from YUV420 to RGB32 and resize it seems to be better [09:36] <jeje2> Is there some condition to use several sw_Getcontext, sws_swscale and sws_freeContext in several threads? [10:09] <jeje2> I make a pastebin code http://pastebin.com/PGP4K6k0 [10:09] <jeje2> if someone can explain me [10:11] <jeje2> what's the matter: decompressing 1 stream 0%. Decompmressing 2 stream: 0-2%. Decompressing 3 stream is sometime 3% sometimes 10% depending when I open the decoder it seems. Decompressing 4 streams is 20-25% (a very grow up once) [11:02] <jeje2> If I play one more (5 decopressing at the same time) my CPU usage grow up to 40-50% [11:05] <durandal_1707> how many threads? [11:05] <durandal_1707> how many cpus? [11:06] <jeje2> I don't specify any thread_count in the FFMPEG codeccontext initialization (so it set to 0) [11:06] <durandal_1707> did you explored where it wastes most of time in such scenario? [11:06] <durandal_1707> jeje2: invalid, it is set to something, use -v debug to see it [11:07] <jeje2> I have a AMD Athlon X2 Dual Core (yes I know not very new) [11:07] <durandal_1707> and how much cores does it have? [11:08] <durandal_1707> also what OS? [11:08] <jeje2> When I add trace in my application after the avcodec_alloc_context3, the thread_count is set to 0 [11:09] <jeje2> In running on Windows XP 32 bits [11:10] <durandal_1707> what happens if you set -threads 1 [11:15] <jeje2> After the avcodec_open, my CodecContext threadcount is set to 8 [11:16] <jeje2> I 'll try with set thread to 1 [11:16] <durandal_1707> jeje2: experiment with -threads X where x is number between 1 and 8 [11:19] <jeje2> Please confirm me, to set the thread count of the codeccontext, I have to fix the number after avcodec_alloc_context3 and before avcodec_open2 (m_lpCodecCtx->thread_count = 1;) [11:20] <jeje2> because even if I set 1 to the thread count, I always have thread_count = 8 in the coedccontext after the avcodec_open2 [11:21] <durandal_1707> jeje2: please read documentation [11:22] <Dominic77> Hi [11:22] <Dominic77> I have a problem with ffmpeg converter [11:23] <durandal_1707> Dominic77: what problem? [11:23] <Dominic77> when I use lighttpd to streaming mp4 file [11:23] <Dominic77> with start, end arguments [11:24] <Dominic77> time duration has a little bit difference [11:25] <Dominic77> eg [11:25] <Dominic77> mysite.com/abc.mp4?start=0&end=10 [11:25] <Dominic77> but the player just played 7s [11:25] <ubitux> maybe because the accuracy of the lighty module isn't perfect (cutting at key frames) [11:26] <ubitux> and afaik it's not using ffmpeg as backend, it's a home made remuxing [11:27] <Dominic77> I tried another mp4 file [11:28] <Dominic77> with the same lighttpd [11:28] <Dominic77> it's perfect [11:28] <Dominic77> but with my mp4 file [11:28] <Dominic77> converted from ffmpeg [11:28] <jeje2> durandal_1707: sorry but I don't find how to set the thread_count in the documentation [11:29] <Dominic77> problem appear [11:29] <Dominic77> how can I fix it? [11:30] <ubitux> fix it in the module [11:30] <Dominic77> lighttpd module ? [11:30] <ubitux> yes [11:31] <ubitux> is this code shop ? [11:31] <ubitux> or they wrote a builtin module? [11:31] <Dominic77> it's h264 module [11:31] <Dominic77> code shop [11:31] <Dominic77> http://h264.code-shop.com/trac/wiki/Mod-H264-Streaming-Lighttpd-Version2 [11:32] <ubitux> (btw, you can also put a real flash player in front, being able to perform range requests) [11:32] <Dominic77> I tried with JW player too [11:32] <Dominic77> http://www.longtailvideo.com/jw-player/wizard/ [11:34] <Dominic77> You can try with this mp4 [11:34] <Dominic77> http://f26.stream.nixcdn.com/cb9276720daf3b1e394a4fabb493d6eb/512c8032/PreN… [12:25] <jeje2> to set the thread count, I set m_lpCodecCtx->thread_type =FF_THREAD_FRAME; and m_lpCodecCtx->thread_count = 2; but I have an error in FFMPEG: The maximum value for lowres supported by the decoder is 0 [12:26] <jeje2> I don't understand the reason [12:27] <jeje2> does I also need to set the lowres? [12:30] <jeje2> or do I neeed to use avcodec_thread_init [12:30] <jeje2> before using avcodec_open2 [12:30] <jeje2> ? [12:31] <jeje2> but it seems avcodec_thread_init is deprecated and also ported directly in the avcodec_open2 [12:32] <durandal_1707> avcodec_thread_init is not available [12:36] <jeje2> so why UI have this error: The maximum value for lowres supported by the decoder is 0 [12:48] <durandal_1707> jeje2: you are doing something obviously wrong [12:49] <Grublet> durandal_1707 the question is what is he doing wrong [12:49] <durandal_1707> i cant know without looking at his code [12:50] <Grublet> the eternal struggle [14:05] <jeje2> I sent a link to my code http://pastebin.com/PGP4K6k0 [14:08] <jeje2> I use dynamic link to the library but I hope it is'nt a problem [14:14] <jeje2> if someone can have a lokk at my code. Thks [14:17] <jeje2> I have this error evrytime I set the Codeccontext thread_type [14:29] <jeje2> if i don't set, the thread count is always set to 8 after avcodec_open2 [14:34] <durandal_1707> jeje2: where you allocate data for rgba? [14:38] <jeje2> My DIBdata is already allocate that's why I just use the avpicture_fill. In fact it's an DirectDrawSurfaceX (it's a windows application using DirectDraw) [14:40] <jeje2> But even if I don't call the part to use sw_scale (if (iframeFinished) I have the same error [14:45] <jeje2> The error come just calling avcodec_open2. The function return error -22 [14:46] <jeje2> To be sure, the only things I have to do to use the thread_count of the AVCodecContext, it's to initialize m_lpCodecCtx->thread_type = FF_THREAD_FRAME; and also m_lpCodecCtx->thread_count = 1; I'm right? [14:49] <jeje2> perhaps my configuration for compiling is bad! I use:./configure --prefix=/mingw/i686-pc-mingw32-last-win32threads --enable-shared --disable-static --enable-runtime-cpudetect --cpu=i686 --arch=x86 --disable-doc --disable-network --disable-devices --disable-avfilter --disable-filters --target-os=mingw32 --disable-encoders --disable-muxers --enable-w32threads --enable-debug (I just add the enable debug to see more) [15:04] <jeje2> I can't understand why setting the thread_count is modifying the lowres value [15:12] <durandal_1707> memory corruption [15:13] <ddhahn> durandal_1707: wanted to ask if you had any chance to look at my issue from yesterday? I was going to post to the mailing list, but wanted to check too..not sure if I should file a bug report.. not sure if I have enough info. [15:13] <jeje2> Yes I think so. Do I need to set the --enable-memalign-hack for the compilation? [15:13] <JEEB> that should be automatically enabled if the architecture needs it [15:14] <JEEB> so you shouldn't have to enable it manually :P [15:14] <jeje2> ok so my configuration to compile FFMPEG seems to be good (I disable a lot of think because I only need to have the H264 decoding part) [15:15] <JEEB> also lowres shouldn't even be supported with H.264 [15:15] <JEEB> so something's going quite wrong somewhere [15:16] <durandal_1707> ddhahn: i cant remmember [15:16] <jeje2> yes I'm trying to see where. I'm compiling ffmpeg with some av_log more to see the values of lowres during the avcodec_open2 function [15:20] <ddhahn> durandal_1707: had to do with transcoding from a wmalossless codec and a/v sync issues. Lost of DTS\PTS invalid, clipping in logs and "queue is backwards in time" coming from destination audio codecs. (libmp3lame and AAC). I'll post to the mailing list [16:03] <jeje2> after adding some traces in util.c in avcodec, I pass in the avcodec_open2 a AVCodecContext* with a value of 0 for lowres, inside the function the value has avctx->lowres=1. I try to find exactly where is the affectation [16:04] <jeje2> because this is why I have the error The maximum value for lowres supported by the decoder is 0 [16:10] <intracube> hi, can anyone provide usage examples of the -fps video filter? [16:11] <intracube> I tried; ffmpeg -i input.mpg -vf yadif=1,fps=5 -c:v libx264 -c:a libmp3lame -crf 18 output.mp4 [16:11] <intracube> to convert interlaced source to progressive @ 5fps [16:11] <intracube> but I get the error "Missing key or no key/value separator found after key '5'" [16:53] <jeje> After adding traces in the avcodec_open2 function, the lowres value is different in the avctx args of ther function than the codeccontext* value I pass to the function [16:53] <jeje> perhaps an alignment prob [16:55] <jeje> how FFMPEG is align? [17:01] <jeje> JEEB>Do you have any idea? [17:01] <JEEB> no, other than it works for most people seemingly [17:08] <relaxed> Was splitting the ffmpeg man page into 10 man pages supposed to simplify things? [17:09] <JEEB> that was probably the idea, they also split filters etc. off into separate pages on the site's docs [17:12] <relaxed> I know this is going to sound like crazy talk, but why not list those 9 man pages at the _top_ of the ffmpeg man page. [17:24] <relaxed> intracube: try quoting it [17:25] <saste> relaxed, why? [17:26] <saste> because that is just silly, that's not how MAN works [17:26] <saste> there is the SEE ALSO section for that [17:26] <intracube> relaxed: ffmpeg -i input.mpg -vf "yadif=1,fps=5" -c:v libx264 -c:a libmp3lame -crf 18 test.mp4 [17:26] <intracube> ^still doesn't work [17:35] <relaxed> intracube: That filter chain works here using latest git. [17:39] <intracube> relaxed: oh ok. I'm using 1.0.4 atm [17:46] <relaxed> saste: I've been reading man pages for 13 years- I'm well aware how they work. The top is the first thing users view, explain how that is silly. Except that it deviates from the norm. [17:47] <relaxed> Therefore I propose we move to info pages. They're the new man pages. [18:01] <creep> nah i grabbed a few motherboards and printer boards and stuff and unsoldered some connectors, ics [18:06] <jeje> when I call avcodec_open2, do I need to set an AVDictionary or can I pass a NULL struture to the function [18:06] <Mavrik> you can pass NULL [18:08] <jeje> I have an "windows" question because I use Visual Studio VC++ to use FFMPEG libraries. Is there any specification about the structure alignment (1 byte, 8bytes?) [18:34] <Mista_D> Is it possible to overlay watermark_1 in the beginning of the video for 10 seconds, and watermark_2 at the end 10 seconds before end using overlay filter? [18:43] <tclarke> is there a way (with ffmpeg.exe) to save raw I, P, and B h.264 frames prior to reconstructing them into complete frames? [18:44] <JEEB> you should be able to get a raw Annex B H.264 stream out of ffmpeg [18:44] <JEEB> ffmpeg -i welp.file -c:v copy -f h264 out.264 [18:44] <tclarke> ok, I'll try that..thanks [18:46] <jeje> it seems i need to have structure alignment to 8 using FFMPEG [18:47] <Mavrik> jeje, usually the av_malloc functions do that for you [18:47] <Mavrik> that's why they're always used :) [18:47] <jeje> I'm not sure [18:48] <jeje> If I use pragam pack (push,8) in my code, I doesn't have the error anymore [18:50] <JEEB> I don't think I've seen anyone really needed anything specific like that from MSVC [18:50] <jeje> so in fact if I use an alignment to 8 bytes before declaration of my AV* structure, it seems it work well [18:50] <JEEB> that said I have no effing idea what on earth you're doing [18:51] <jeje> only this one: [18:51] <jeje> #pragma pack( push, beforeffmpeg, 8) [18:51] <jeje> AVCodecContext* m_lpCodecCtx; [18:51] <jeje> AVCodec* m_lpCodec; [18:51] <jeje> AVFrame* m_lpFrame; [18:51] <jeje> AVPacket m_lpPacket; [18:51] <jeje> AVPicture m_lpPict; [18:51] <jeje> SwsContext* m_img_convert_ctx; [18:51] <jeje> #pragma pack(pop, beforeffmpeg) [18:51] <jeje> in my class.h [18:51] <JEEB> uhh, let's just say that I haven't seen anything like that with other code that is using the libav* libraries in ffmpeg [18:52] <JEEB> not even MSVC-based projects [18:52] <JEEB> either you are doing something very special [18:52] <JEEB> or you are doing something very wrong in general [18:52] <jeje> yes but perhaps other person used an 8 byte alignment per default, for me, I use 1 byte per default [18:53] <JEEB> I haven't seen any pragmas or compilation options setting that with MSVC-based projects that use ffmpeg [18:53] <JEEB> that's all [18:53] <jeje> if I use ffmpeg to decode a whole frame , I need to use thread_type = FF_THREAD_FRAME; can someone confirm me? [18:54] <JEEB> no, that just decides the threading mechanism used [18:54] <JEEB> some codecs use slice based threading, some use frame based threading [18:54] <jeje> and the difference between SLICE and FRAME please? [18:54] <JEEB> (and some have both slice and frame based threading, like H.264) [18:55] <jeje> ok so to make the init of my codeccontext I have to set the 2 ones SLICE|FRAME [18:56] <JEEB> you usually want one of those tho, not sure what happens with h264.c f.ex. if you set both :P [18:56] <JEEB> a slice is a slice, the exact definition depends on the format, but in general it means that the frame is cut vertically into X independently decode'able parts [18:56] <JEEB> that means that you can decode them independently in their own threads [18:56] <JEEB> frame based threading on the other hand means that you use multiple threads to decode multiple frames at the same time [18:57] <jeje> this is my case... thks for all explanations [20:47] <WmA> I have a corrupt ts file (avc/mp2) [20:48] <WmA> is it possible to correct PCR with ffmpeg? [20:48] <WmA> or reindex so that the timers are set correctly? [20:58] <Trashlord> hey guys, I'm trying to extract sound from a video file. I'm using -vcodec none, but it throws an error saying "no such codec" [20:59] <Fjorgynn> yeah? [20:59] <Fjorgynn> -vn [21:00] <Fjorgynn> test.mp4 -vn -c:a copy test.wav [21:00] <Fjorgynn> test.mp4 -vn -c:a copy test.aac [21:00] <Fjorgynn> maybe [21:00] <Trashlord> ah [21:00] <Trashlord> sec, I'll test [21:00] <Trashlord> that worked, thanks a lot :) [21:01] <Fjorgynn> :) [21:25] <burek> hi all :) [21:26] <burek> if we target quality/bitrate factor, is it better to use -crf option or 2-pass encoding with libx264/ffmpeg ? [21:27] <JEEB> average bit rate and crf pretty much have the same factor for that [21:27] <JEEB> so just use the one you need [21:27] <JEEB> if you just need a general quality level, use crf [21:27] <JEEB> if you just need a specific average bit rate, use the latter [21:28] <burek> i see [21:29] <burek> but does 2-pass also uses variable bitrate throughout the content like crf does? [21:29] <burek> use* [21:29] <JEEB> to get CBR from x264, you will have to try hard :P [21:29] <burek> i see :) great :) [21:29] <burek> it's no wonder it's the best video codec today :) [21:30] <burek> although ive been reading about something like h265 already.. [21:30] <JEEB> HEVC, yes [21:30] <JEEB> which got its ITU-T name "H.264" set on feb 25th [21:30] <JEEB> uhh [21:30] <JEEB> *H.265 [21:31] <burek> so, it's the same thing? [21:31] <burek> oh [21:31] <burek> ok :) [21:31] <JEEB> http://iphome.hhi.de/wiegand/assets/pdfs/2012_12_IEEE-HEVC-Overview.pdf [21:31] <JEEB> have an overview [21:32] <burek> thanks :) [21:32] <burek> :beer: :) [21:33] <JEEB> too bad, while HEVC does base on some of the things from AVC (CABAC for example, as well as many other ideas), it still changes enough that we can't just change x264 around a bit to create a good HEVC encoder [21:33] <JEEB> f.ex. the change from 16x16 blocks to tree units up to 64x64 of size [21:34] <burek> so.. back to the lab, eh? :) [21:34] <JEEB> yeah [21:34] <JEEB> the not open source stuff is gonna be the first to the cake, because they have the people to pay salaries for :) [21:34] <JEEB> but libavcodec should get a decoder in some time [21:36] <burek> it's in their interest also [21:36] <burek> since a lot of people on the internet are using ffmpeg/libav* [21:37] <JEEB> smarter has been working on a HEVC decoder at libav from around last may or so, and it's getting pretty good. ffmpeg will also get it through the usual merging process when it gets finished :) [21:40] <burek> btw, do we have fade in/out audio filter ? [21:40] <JEEB> not sure [21:40] <burek> i forgot if i created a ticket [21:40] <burek> but i think it would be useful [21:40] <burek> and volume audio filter might be used/upgraded for such purpose i guess [21:40] <JEEB> http://ffmpeg.org/ffmpeg-filters.html#afade [21:41] <burek> great :) [21:41] <JEEB> wouldn't this work? [21:41] <JEEB> :) [21:41] <burek> ffmpeg is like a magic box :) [21:41] <burek> whatever you think of, it's either already implemented or in the process of being implemented :) [22:04] <ilil> hi, is there a way to differ ffmpeg from its fork at compile time? [22:05] <ilil> something like #idef FFMPEG_DEFINE ... [22:05] <ilil> *#ifdef [22:06] <JEEB> see how ffms2 does it [22:07] <Plorkyeran> minor versions start at 100 for ffmpeg and 0 for libav [22:09] <ilil> Plorkyeran: is too unobvious, isn't it? [22:09] <ilil> JEEB: thanks, is checking out it [22:09] <Plorkyeran> a simple IS_FFMPEG define would be too hard [22:10] <ilil> Plorkyeran: ??? [22:10] <ilil> but why? politics again? [22:13] <Mavrik> the real ffmpeg please stand up -_- [22:14] <ilil> :)) [22:18] <ilil> oh no, the way ffms doing it: #if LIBAVFORMAT_VERSION_MICRO > 99 || LIBAVUTIL_VERSION_MICRO > 99 || LIBAVCODEC_VERSION_MICRO > 99 || LIBSWSCALE_VERSION_MICRO > 99 [22:43] <PaulWay> Does anyone know how to put correct metadata into an MP4 or WebM file using the -metadata option? [22:43] <PaulWay> I've tried it but VLC then shows nothing for the fields that I expect it to turn up in. [22:47] <JodaZ> wow wow wow, yesterday i read about that ffmpeg mutiny for the first time [22:57] <SubJunk> Is there a way to make FFmpeg create a thumbnail from a video within certain boundaries while keeping the aspect ratio? [22:57] <Mavrik> what does "within certain boundaries" mean? [22:58] <SubJunk> For example in MEncoder you use "-vf scale=320:-2,expand=:180" which means the width will be 320 and/or the height will be 180 [22:59] <Mavrik> hmm, there's a scale filter, but not an expand filter [23:00] <SubJunk> Would another filter achieve the same thing? [23:01] <SubJunk> I asked about a year ago and it wasn't possible so just checking if it's changed, no problem anyway [23:01] <beastd> SubJunk: You want to scale or you don't? [23:02] <SubJunk> If the input video is say 720p then yeah it will need to scale to fit 320x180 [23:41] <JodaZ> does one still need segmenter.c to live transcode to iOS devices or is there a way to do it directly ? [23:46] <bunniefoofoo> I am using ffmpeg (command line) to write directly to a USB drive. The writes are very slow (800 KB/s), compared to copying the file manually (6000KB/s) and no where near CPU bound... I believe this could be addressed by giving ffmpeg a larger file writing buffer on the output side, is it possible? [23:47] <hughmanwho> Trying to figure out why my code is not working, it's finally to the point where it is running but just shows the output as frames that are just vertical stripes of black green blue and red. Here is a pastebin: http://pastebin.com/mMSdFCAM Anyone have any thoughts? [23:47] <hughmanwho> Thanky ou! [23:48] <JodaZ> bunniefoofoo, on windows you can turn a switch to do this stuff globally i think [23:49] <JodaZ> bunniefoofoo, i think its propably also the disk cache, not the write buffer thats constraining things [23:49] <bunniefoofoo> well it is a USB flash drive. In my tests (using stdio), I have to issue writes in 256KB chunks to get good speed [23:50] <bunniefoofoo> I think stdio defaults to 32KB write buffer, unless ffmpeg is using fflush() ( I am assuming ffmpeg uses stdio ) [23:52] <bunniefoofoo> I don't see any fflush's in code besides stdout/stderr [23:53] <JodaZ> bunniefoofoo, this linux ? [23:53] <bunniefoofoo> windows [23:53] <bunniefoofoo> the switch.. is this an environment var that will set stdio buffer size in the CRT DLL ? [23:54] <JodaZ> bunniefoofoo, http://support.microsoft.com/kb/259716 [00:00] --- Wed Feb 27 2013

1 0

ffmpeg-devel.log.20130225
by burek 26 Feb '13

26 Feb '13

[00:29] <ezekiel> I spoke too soon - seems that one way the audio is always ahead of video, and the other way audio is always behind [04:49] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:e42028925bdd: ffmpeg: Force a first_pts of 0 for the first configuration of -async use [04:49] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:7f2ab129b19c: fate: force a first_pts=0 for the aresample test [04:49] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:35aaa306ac2d: swr: make the default of nopts for first_pts actually work [04:49] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:bb7bc3443dd9: af_biquads: memset(0) cache [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:fcd0e3235ad1: h264: Detect POC inconsistencies and try to handle them reasonably [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:cc64871bd86b: movenc: hotfix, dont store fiel for h264 / mpeg4-asp / dnxhd [05:41] <cone-697> ffmpeg.git 03Diego Biurrun 07release/1.0:02f241d1ed10: configure: Make warnings from -Wreturn-type fatal errors [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:97a740acc587: aac: reconfigure output on pop [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:c69709571af2: doc/APIchanges: fix odd .01 versions [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:5877811b0f13: apichanges: fix date [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:3a21007e4706: apichanges: Use , instead of / to seperate multiple hashes [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:ab70fa28f665: apichanges: fix 2 wrong hashes [05:41] <cone-697> ffmpeg.git 03Michael Niedermayer 07release/1.0:8cfd33fd3504: doc/APIchanges: List merge commit hashes and version numbers [10:50] <ubitux> durandal_1707: Unable to open iconv context with input character encoding "cp1251" [10:50] <ubitux> not sure how i'm supposed to disable the test if the iconv hasn't the cp1251 support [10:54] <ubitux> i could add a specific flag in the configure using iconv -l, but that's a bit ugly [10:58] <durandal_1707> why i cant run separate lavfi test? [10:58] <ubitux> what do you mean? [10:59] <durandal_1707> make fate-lavfi-lut [11:03] <durandal_1707> why simple padding filter depends on drawutils? [11:04] <durandal_1707> and why drawutils is designed to badly that it never cared for >8 bit colors [11:07] <ubitux> 10:59:16 <@durandal_1707> make fate-lavfi-lut // make fate-lavfi-{crop,field,...} work here [11:07] <ubitux> i don't see any lut test though [11:08] <ubitux> if you're adding one, depending on where you do it, you may need to re-run configure (or add it to a list, i think that changed a while ago) [11:54] <durandal_1707> michaelni: exr test is blocker because of missing rgba64 output [11:54] <durandal_1707> that is stupid [11:55] <durandal_1707> rgba64 is just rgb48 with alpha plane [11:59] <cone-644> ffmpeg.git 03James Almer 07release/1.0:e6324ccfbd8e: latmenc: Check for LOAS sync word [11:59] <cone-644> ffmpeg.git 03James Almer 07release/1.0:62e55034077d: lavc/bink: Chech for malloc failure [12:12] <cone-644> ffmpeg.git 03James Almer 07release/1.1:8d3bc52acd64: latmenc: Check for LOAS sync word [12:12] <cone-644> ffmpeg.git 03James Almer 07release/1.1:5fb5ac71488c: doc/Makefile: Fix make docclean [12:12] <cone-644> ffmpeg.git 03James Almer 07release/1.1:d92a7870d74e: lavc/bink: Chech for malloc failure [12:15] <durandal_1707> so alpha is just copied [12:30] <cone-644> ffmpeg.git 03Diego Biurrun 07master:3d035d5a6a91: dsputil_alpha.h: Add missing stddef.h header to fix standalone compilation [12:31] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:11dcecfcca0e: vorbisdec: Error on bark_map_size equal to 0. [12:31] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:cb72f698fe5d: Merge commit '11dcecfcca0eca1a571792c4fa3c21fb2cfddddc' [12:36] <cone-644> ffmpeg.git 03Luca Barbato 07master:fc386f2eea8d: vorbisdec: cosmetics [12:36] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:a0312a2e8531: Merge commit 'fc386f2eea8d93ecd4f81e1646c835d1645c56a0' [12:57] <cone-644> ffmpeg.git 03Luca Barbato 07master:5b47c19bfda9: vorbisdec: Add missing checks [12:57] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:f7dc6aa5fecf: Merge commit '5b47c19bfda92273ae49e83db26a565afcaed80a' [13:13] <cone-644> ffmpeg.git 03Luca Barbato 07master:23bd9ef4b209: vorbisdec: Accept 0 amplitude_bits [13:13] <cone-644> ffmpeg.git 03Michael Niedermayer 07master:875f88318506: Merge remote-tracking branch 'qatar/master' [13:58] <cone-644> ffmpeg.git 03Paul B Mahol 07master:eac93932b011: lavfi/geq: improve support for formats with alpha plane [14:13] <cone-644> ffmpeg.git 03Paul B Mahol 07master:e0ccb5fa38dd: lavfi/hflip: support more formats [14:19] <cone-644> ffmpeg.git 03Paul B Mahol 07master:b8f6912816d2: fate: update pixfmts_hflip [14:38] <bgmarete> Hello all. On GIT Tip, ffprobe ignore the "-ar" command line parameter while ffmpeg uses it. [14:41] <bgmarete> Is Stefano Sabatini online? [14:46] <ubitux> not now [14:46] <ubitux> -ar makes no sense for ffprobe [14:50] <bgmarete> ubitux: Why not? If you do -show_format you will frequently need "-ar" [14:51] <ubitux> i wonder why [14:51] <bgmarete> (Otherwise you could get the default smaple rate, for alaw files, for example (which is set at 44.1k), which is not correct. [14:52] <bgmarete> I mean not correct for the vast majority of alaw files (which are usually telcom-grade 8k samples). [14:53] <ubitux> mmh ok, i didn't have this in mind [14:54] <bgmarete> In a case such as this, in addition to mis-detecting the sample rate, ffprobe must also mis-detect the file length, giving a length that is 44.1/8 times to small. (This is all with regard to -show_format). [14:56] <bgmarete> ubitux: Are you familiar with the code-base so that you can point me in the direction of quickly adding "-ar" to ffprobe? [14:59] <ubitux> ffprobe.c :p [15:03] <Compn> bgmarete : can you upload a sample of a misdetected alaw file [15:03] <Compn> its possible we can fix, but also we can add -ar to ffprobe too [15:04] <bgmarete> Compn: As it happens, that' [15:04] <ubitux> afaik it's just pcm data so you can't actually detect anything [15:05] <bgmarete> Compn: As it happens, that's rather easy. All the asterisk ALAW samples in http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-… are mis-detected as 44.1k samples, while they are all 8k samples. [15:07] <bgmarete> ubitux: I think that's right. Another way to look at it is this: Does the 44.1k default make sense for alaw? In any case, ffrobe does need to respect "-ar". [15:26] <durandal_1707> isn't point of ffprobe to probe and not to need user input? [15:27] <nevcairiel> some raw formats just need user input to work [15:27] <nevcairiel> or it assumes defaults [15:30] <durandal_1707> than it is easy to change defaults for alaw files [15:31] <durandal_1707> currenly alaw files are not detected at all using extension (only "al" one works) [15:34] <durandal_1707> michaelni: gonna help/comment on that rgba64 patch? [16:25] <bgmarete> durandal_1707 and others: It should work like this: "-ar" and similar flags must be allowed for raw formats for which there is no hope of detection without this hint from the user. (A similar situation, this time with regard to size and framerate, occurs for raw video). But if FFprobe detects a more reliable value, then it should override this user input (this will never happen in the case of raw formats). [16:26] <bgmarete> durandal_1707: Changing the default does not solve the problem. There are valid alaw files with sample rate 44.1k. [18:10] <durandal_1707> i think than printf in code should be disabled again [18:17] <ubitux> i also agree with the macro being great to prevent from using the internal versions [18:22] <durandal_1707> ubitux: my intentions are more evil, less people would hack on code when they found that their printf hack cant compile [18:22] <ubitux> :D [18:22] <ubitux> #define printf segmentation_fault [18:38] <durandal_1707> this evil plane in lavfi is ugly [18:40] <cone-231> ffmpeg.git 03Michael Niedermayer 07master:a960f3b91876: pmpdec: fix integer overflow [18:40] <cone-231> ffmpeg.git 03Michael Niedermayer 07master:e6f27346b752: pmpdec: make i unsigned, avoid undefined behavior of i++ [18:40] <cone-231> ffmpeg.git 03Jean First 07master:af0e8144cd8b: ffmpeg_opt: cosmetics [18:41] <durandal_1707> michaelni: how evil will be merged? [18:44] <nevcairiel> "carefully" [19:19] <durandal_1707> can we just ban morons on libav-user@ ? [19:26] <Compn> ask lou [19:26] <nevcairiel> sadly, stupidity is usually not reason enough to ban [19:29] <durandal_1707> nevcairiel: :( [19:34] <Compn> killfile those posters ? [20:02] <saste> durandal_1707, ubitux, michaelni: comments on the partial overlay support patch? [20:04] <saste> it somehow affects performances, so i'd like someone to comment on it [20:04] <durandal_1707> saste: if it works, valgrind is happy, it should be ok [20:04] <saste> performances [20:04] <durandal_1707> saste: at what order performance is affected? [20:05] <saste> it loops on non overlayed pixels [20:05] <saste> i could avoid that, but that would complicate the logic [20:05] <durandal_1707> and how much that hurts? [20:05] <ubitux> instead of some "continue", can't you break to save some useless loops when it goes outbound? [20:06] <saste> since I would have to compute the non-clipped segments [20:06] <saste> ubitux, no [20:06] <ubitux> k [20:06] <saste> consider an overlay image bigger than the main input [20:08] <ubitux> if (y+i < 0 || y+i >= dst_height) continue; here it looks like you could break if(y+i>=dst_height) (since i will only increment) [20:08] <ubitux> and for the y+i<0 maybe start the loop later [20:08] <ubitux> or i'm missing something? [20:24] <durandal_1707> please avoid potato programming at all cost [20:27] <durandal_1707> what about this auto-vectorisation? does that mean we can finally ditch all this (inline) asm code? [20:28] <saste> durandal_1707, potato programming? [20:31] <Compn> durandal_1707 : dont look at all of those mips optimizations ;) [20:31] <durandal_1707> saste: extremly unreadabe/unmaintainable/unextendable/ugly code [20:31] <durandal_1707> Compn: i had enough of swscale today [20:32] <Compn> libav is thinking of rewriting it [20:32] <Compn> i wonder if michaelni would be interested in splitting it up ? or if that would help durandal_1707 ? [20:33] <durandal_1707> it is already splitted for no much gain [20:33] <Compn> i'm guessing an indent wouldnt help either [20:34] <Compn> just the mix of inline asm, no easy way to add formats ? and hard to decipher ? [20:34] <durandal_1707> it is already beautifully looking only on your screen [20:34] <Compn> i mean, what do you suggest be done to it ? [20:34] <Compn> change inline to seperate .asm ? [20:35] <durandal_1707> it's not about inline [20:35] <Compn> i'm not trolling, but i want to make a bugreport out of this , once and for all :) [20:36] <durandal_1707> it needs more structures and comments [20:38] <michaelni> yes, also some factorization in some areas, avoid overuse of macros [20:40] <michaelni> and various other things, it would be great if BBB would resume working on it [20:44] <saste> ubitux, gsoc tasks to propose? [20:44] <saste> maybe related to subtitles or filters? [20:45] <ubitux> saste: i've been wondering about it; i don't actually want to propose an impossible task such as "subtitles in lavfi" which will require quite a big amount of work [20:46] <ubitux> and making it a "work with other dev" thing will inspire the "dev will only give me the shit they don't want to do" [20:46] <ubitux> i've been thinking of adding formats though [20:46] <saste> ubitux, yes [20:46] <saste> we lack some image formats for example [20:46] <ubitux> like the .SUP thing (which is the bluray equivalent of VobSub afaik) [20:47] <saste> also some work on ffserver or telecine filters would be welcome [20:47] <saste> i don't think i'll mentor more than one task [20:47] <ubitux> yes i can propose such stuff, but i can't really be a mentor on it [20:48] <saste> ubitux, imho not much point in proposing projects one can't mentor [20:48] <saste> we already have enough of them [20:49] <ubitux> yes, that's the reason i didn't add it [20:49] <ubitux> i can mentor the port of some filters [20:49] <ubitux> typically, the port of eq filter [20:51] <cone-231> ffmpeg.git 03slhck 07master:188947c32ad6: libvpx: allow 0 as min quantizer value [20:51] <cone-231> ffmpeg.git 03slhck 07master:bfcc38ef4815: libvpx: check if CQ level is in correct bounds [20:52] <michaelni> ubitux, iam not sure gsoc allows "work with other dev" [20:52] <ubitux> ok, that closes the case then [20:53] <Compn> .sup is vobsub i think [20:53] <Compn> er [20:53] <Compn> i mean [20:53] <Compn> .sup is bluray subs i think [20:53] <michaelni> merbanan, do you want to mentor some tasks in ffmpeg gsoc ? [20:53] <michaelni> if so dont hesitate to add them to the wiki [20:54] <Compn> anyone interested in mentoring gotomeeting ? [20:54] <Compn> maybe j-b knows someone ? [20:54] <Compn> a gotomeeting decoder that is [20:55] <ubitux> 20:53:35 <@Compn> .sup is bluray subs i think // yeah, exactly what i said [20:55] <Compn> yeah i misread the whole thing :D [20:55] Action: Compn brain fart [21:00] <Skyler_> any chance I can be taken off the consulting page? I'm getting way too many emails and don't really have the time or sanity atm [21:02] <saste> Skyler_, send patch [21:02] <durandal_1707> :)) [21:03] <Skyler_> :/ [21:03] <durandal_1707> i could also mentor RE gsoc task (the one noone will pick) [21:03] <saste> Skyler_, btw who are you? [21:04] Action: Skyler_ Jason [21:04] <saste> ok [21:06] <Compn> Skyler_ : i can, but who am i removing ? [21:07] <durandal_1707> michaelni: ping on RGBA64 output, exr fate test is "blocked" because of this [21:08] <Compn> Skyler_ : oh wheres dark_shikari ? :P [21:10] <Compn> Skyler_ : removed. site updated [21:10] Action: Compn afk [21:12] <Skyler_> Thanks [21:26] <durandal_1707> what mjpegtools filters is worth porting? [21:28] <Compn> what filters are there to choose from ? [21:29] <Compn> got a list ? [21:35] <durandal_1707> nothing important that is not already in lavfi [21:36] <saste> durandal_1707, do you have a link to a list? [21:37] <durandal_1707> source code [21:37] <Compn> durandal_1707 : looks like most people want deinterlacing filters [21:38] <Compn> or ivtc i should say [21:38] <durandal_1707> than add it as main tasl [21:38] <durandal_1707> *task [21:38] <saste> lavplay [21:38] <saste> i didn't know mjpegtools had a tool named like that [21:43] <durandal_1707> saste: i have started writing jellyfish meter avf filter [21:44] Action: durandal_1707 wonders when vaf filter will appear [21:49] <Compn> theres a bunch of mplayer audio filters [21:49] <Compn> which ones are useful? pan, hrtf , volnorm , karaoke [21:49] <Compn> people want volume normalizer :P [21:49] <wm4> does libavformat give any indication whether the file format may employ wrapping timestamps or timestamp resets? [21:49] <wm4> Compn: none [21:50] <wm4> Compn: I think lavfi can do everything by now [21:50] <Compn> wm4 : you removed them from your fork then ? [21:50] <wm4> no [21:50] <Compn> ... [21:50] <wm4> no lavfi support yet [21:50] <wm4> waiting for the evil plan [21:50] <wm4> but I can assure you most of the mplayer code is very low quality [21:51] <wm4> :) [21:51] <Compn> do you know of any video filters that should be added ? [21:51] <Compn> to lavfi [21:51] <wm4> uh well ffmpeg ported almost all mplayer video filters [21:52] <wm4> other than that, they could look at the avinsynth universe [21:52] <durandal_1707> Compn: any gimp one [21:52] <wm4> *avisynth [21:52] <durandal_1707> the only big/important filter left is stereo3d [21:53] <durandal_1707> and emboss [21:53] <durandal_1707> posterize too [21:54] <wm4> how about a working ivtc filter [21:54] <wm4> or a deinterlacer that's not from the previous decade [21:55] <durandal_1707> than there almost completly unexplored field of image analysis [21:56] <durandal_1707> like for remote sense [21:56] <Compn> durandal_1707 : someone was asking about a skin color smoother or something :P [21:56] <Compn> so yeah, the ability to use gimp or photoshop filters on video would be fun [21:58] <Compn> sony working on 4k h265 ? [22:00] <durandal_1707> Compn: what "skin color smoother" means? [22:01] <ubitux> 21:49:33 <@Compn> people want volume normalizer :P // i'm somehow working on it [22:01] <ubitux> i need to get it done [22:01] <Skyler_> 'smart blur' probably? [22:02] <ubitux> smartblur is already ported afaik [22:03] <Compn> durandal_1707 : no idea , probably he meant airbrushing or something [22:03] <durandal_1707> but shape adaptive blur isn't [22:04] <Compn> i dont even know if people use sab/smartblur [22:05] <Skyler_> I use it on photos at least, it's good to get rid of ISO noise [22:05] <Skyler_> not in ffmpeg, but [22:05] <Compn> Skyler_ : any ideas for filters we should add to lavfi ? [22:06] <ubitux> ivtc [22:06] <ubitux> like asked everytime we wonder [22:06] <ubitux> (like a few minutes ago) [22:06] <Compn> lol i know :D [22:06] <nevcairiel> a proper adaptive ivtc with support for dynamically detecting the pattern would be neat [22:06] <Skyler_> just add avisynth dll support ;) [22:06] <Skyler_> then you don't need to any work [22:07] <Skyler_> Though, I guess that's not quite enough often, since a lot of avisynth 'filters' are complicated scripts that use multiple filters and may be really hard to implement in lavfi [22:07] <nevcairiel> avisynth api is annoying, imho [22:07] <durandal_1707> but isn't interlacing dying? [22:07] <nevcairiel> telecine is technically not interlacing [22:08] <nevcairiel> you can reproduce the progressive stream perfectly [22:09] <durandal_1707> ok so ivtc and scripting are on top of list [22:09] <wm4> isn't the avisynth API inverse to the lavfi API (avisynth has pull, lavfi push) - or maybe I'm misunderstanding how lavfi works [22:12] <durandal_1707> what would inverse mean? that its usability is inverse proportional of avisynth clones? [22:12] <wm4> inverse data flow [22:12] <durandal_1707> lavfi just lacks scripting [00:00] --- Tue Feb 26 2013

1 0

ffmpeg.log.20130225
by burek 26 Feb '13

26 Feb '13

[00:29] <ezekiel> I spoke too soon - seems that one way the audio is always ahead of video, and the other way audio is always behind [03:29] <p4plus2> Hey, quick problem, when recording with ffmpeg once I kill ffmpeg to stop the recording, I lose around 1 to 2 seconds of recording it seems to be lost [03:29] <p4plus2> placing "sleep 1" at the beginning of my kill script fixes it for the most part, but is there a better method? [06:57] <grepper> I wonder if it has to do with keyframes [10:11] <Guest1692> hi [14:37] <bgmarete> Hello all. On GIT Tip, ffprobe ignore the "-ar" command line parameter while ffmpeg uses it. [16:41] <lightbring3r> hi everyone [16:41] <lightbring3r> quick question: any idea how i set muxdelay programmatically? [17:02] <hughmanwho> How can I find the analyze duration? I'm getting an error message telling me that I should consider increasing the value for 'analyzeduration' and 'probesize' options [17:05] <klaxa> just add them as parameters i.e. ffmpeg -analyzeduration 9001 -probesize 4096 -i <input> <output> [17:06] <klaxa> hughmanwho: ^ [17:07] <jeje2> Hi to all [17:12] <jeje2> I use FFMPEG libraries (last released 1.1.2 I have recompiled) to decompress H264 video stream from IP camera. I'm facing a cpu usage problem. FFMPEG can decompress video stream like I want. But I have a limitation of simultaneous decompression. I can decompress 3 streams at 0-2% of CPU usage. But if I try to decompress an fourth at the same time, my CPU usage grows up to 20%. I really can't understand why... [17:12] <jeje2> I someone have an explanation (and a solution). Regards. [17:13] <jeje2> I can make a pastebin code if necessary [17:25] <hughmanwho> What is a good probesize? When it's at 5,000,000 which it was set to as default in code, then it takes 75 seconds to run 'avformat_find_stream_info', when it's twice that it takes 123 seconds, when it's at 4096 it crashes because it it's long enough.. I can definitely keep playing and narrow in on a good value but anyone have any experience there? [17:30] <hughmanwho> Actually that was probably faster to find than it was to ask [17:31] <hughmanwho> I went with 131072 (2 to the 17th), enough that it didn't crash after a few tries and also executed within 2 seconds [17:57] <jeje2> in fact I make a class in c++ (using Visual C++) to integrate the FFMPEG libraries on windows XP OS. I use dynamic pointers on the dll files. This class is used by my application to initialize the library, calling avregister once (I also call av_lockmgr_register because I'm in a multithreaded application). And after When receiving an RTP packet (I formatted from RTP H264 stream to understanding packet for FFMPEG). [17:57] <jeje2> Only the first time (by stream) I find the decoder m_lpCodec = lpfnavcodec_find_decoder(CODEC_ID_H264), I do my m_lpCodecCtx = lpfnavcodec_alloc_context3(m_lpCodec) , I open the decoder avcodec_open2( m_lpCodecCtx, m_lpCodec, &optionsDict); with optionsDict=NULL, I alocate my frame m_lpFrame = lpfnavcodec_alloc_frame(); After all these initialization, at each packet, I do lpfnav_init_packet(&m_lpPacket); [17:57] <jeje2> m_lpPacket.data = lpImageData;m_lpPacket.size = dwImageSize; then iRes = lpfnavcodec_decode_video2( m_lpCodecCtx, m_lpFrame, &iFrameFinished, &m_lpPacket); and lpfnav_free_packet(&m_lpPacket); [17:57] <jeje2> after decompression. [17:58] <jeje2> But I can't understand why CPU usage is so different between 3 and 4 simultaneous decompressions. I don't set any flags to the codeccontext and codec. [18:13] <jaffa4> hi [18:14] <jaffa4> How do I compile ffmpeg versoin that has mp3 enconding enabled that can be used by vlc to encode mp3? [18:31] <ddhahn> hey all.. using ffmpeg 1.1.2 on RHEL. Compiled from source with these options --extra-cflags=-O2 --enable-bzlib --disable-devices --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libschroedinger --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-avfilter --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libvpx --enable-libspeex --enable-postproc --enable-pthreads --disable-static [18:31] <ddhahn> nable-gpl --disable-debug --disable-optimizations --disable-stripping --extra-cflags=-fPIC --extra-ldflags=-fPIC --enable-nonfree --enable-version3 --libdir=/usr/local/lib [18:32] <ddhahn> some conversions log a lot of "que input is backward in time" from the audio encoder, and PTS, DTS issues.. The output files work, but I can't really tell if i should care about the messages. [18:36] <ddhahn> Here's a complete log from a conversion.. http://pastebin.com/n0mwUtqB [18:38] <ddhahn> Although the output works, in that in plays in a media player, there seems to be small jumps in the audio. [18:45] <ddhahn> command line was: ffmpeg -i /opt/kaltura/web//content/entry/data/4/267/0_6isiwutm_0_luyg7xky_2.wmv -vcodec libx264 -subq 2 -qcomp 0.6 -qmin 10 -qmax 50 -qdiff 4 -b 750k -s 528x352 -r 29.97 -g 60 -acodec libfaac -ab 96k -ar 44100 -f mp4 -y /opt/kaltura/tmp/convert/tmp_convert_51262dc767d03 >> "/opt/kaltura/tmp/convert/convert_0_6isiwutm_676be.log" 2>&1 [18:47] <ddhahn> for a little more background, I am using ffmpeg in the context of Kaltura 5.0, CE (kaltura.org) Needed to update ffmpeg from the included version to support the wmalossless codec. [19:10] <relaxed> ddhahn: why would you add these --disable-optimizations --extra-cflags=-O2 [19:11] <relaxed> use one of the libx264 presets [19:14] <Catoptromancy> moo [19:18] <ddhahn> I was mirroring how the ffmpeg was compiled that was distributed with kaltura is the only reason [19:18] <ddhahn> not really understang fully what all the options were, i tried to stick to that.. [19:22] <ddhahn> would you suggest removing those options and recompiling if that changes the behavior at all? [19:23] <ddhahn> as far as the command line options, I need to dig in a bit to kaltura to see where those options are set.. The system automatically generates a command line to use with ffmpeg. [19:27] <ddhahn> I am wondering if the issue lies somewhere in the wmalossless decoding.. I am testing some other videos that vary in codecs used to see if the same issue arises. [19:27] <relaxed> Amazing! I wouldn't have much faith in a company shipping out ffmpeg with --disable-optimizations. [19:28] <relaxed> ddhahn: probably, check out the latest source using git and see if you have still have the problem. [19:28] <durandal_1707> ddhahn: it could be bug in demuxer/decoder so please provide input sample [19:28] <relaxed> and, of course, remove --disable-optimizations --extra-cflags=-O2 [19:29] <ddhahn> relaxed: thanks, I'll do that [19:29] <ddhahn> durandal_1707: is there a preferred way to share input files? [19:30] <Catoptromancy> I am trying to get libavformat.so but compiling only seems to make libavformat.a [19:30] <Catoptromancy> i must be doing something wrong [19:30] <Catoptromancy> seems to compile flawlessly [19:31] <durandal_1707> ddhahn: just upload part(if it is very big) of it somewhere (make sure that part is playable firs) [19:31] <durandal_1707> Catoptromancy: --enable-shared [19:31] <relaxed> Catoptromancy: --enable-shared [19:31] <relaxed> I won [19:31] <ddhahn> Catoptromancy: did you configur with --enable--shared? [19:31] <Catoptromancy> thax! [19:31] <ddhahn> er.. --enable-shared [19:51] <ddhahn> durandal_1707: test file --> http://jabba.servebeer.com:51234/store/testfile.wmv [19:51] <ddhahn> that's the whole thing..~200MB or so.. not too bad [19:54] <durandal_1707> ddhahn: and when you get clips in audio? [19:58] <durandal_1707> does audio othewise plays fine wit ffplay/ or if you transcode it to flac? [19:58] <durandal_1707> from log errors it probably means dts/pts is causing libfaac encoder to misbehave [19:58] <ddhahn> have not tried transcoding to flac [19:59] <ddhahn> it plays mostly ok, but in vlc I noticed there is a small a/v sync problem [19:59] <ddhahn> So with that, and the errors, I am not sure if something is amiss or not [20:03] <ddhahn> durandal_1707: yea, small clips in the FLV output [20:04] <ddhahn> and in mp4, sorry [20:53] <ddhahn> Just tried compiling from fresh git clone.. same issue, though someone changed "que" to "queue" in the log files :) [21:19] <HorizonXP> hi there, I'm trying to use avconv on Ubuntu to capture frames from a video. I have this working [21:19] <Fjorgynn> and? [21:19] <HorizonXP> However, I'd like to name the output files such that they're named "screen_xxxx.png" [21:19] <HorizonXP> where xxxx = the timestamp of the frame [21:20] <HorizonXP> I'm wondering if this is possible via avconv [21:20] <HorizonXP> or perhaps I'd have to script ffmpeg via a python script [21:20] <Fjorgynn> this is ffmpeg [21:20] <durandal_1707> resistance is futile [21:21] <saste> HorizonXP, not at the moment, WIP and you should find a related ticket on trac [21:21] <saste> (check for words image2) [21:21] <saste> come back in a month and maybe it will be implemented [21:21] <durandal_1707> why so long? [21:22] <zoktar> Hello, im trying to caputure desktop and some games i play, but rather than showing the poor fps in a game for example, it speeds up to try and match the FPS, how do i record with a fixed fps regardless of the real fps ?. thanks. [21:22] <HorizonXP> saste: WIP to output timestamps? [21:22] <saste> HorizonXP, WIP to put the timestamp in the image filename [21:23] <saste> you have several tools to inspect the frames timestamps (showinfo filter, ffprobe, ffmpeg timestamp options) [21:23] <HorizonXP> saste: can you tell me what to search for in trac? [21:23] <HorizonXP> maybe I can implement it [21:23] <saste> HorizonXP, there is a discussion on ffmpeg-devel [21:24] <saste> i'm currently stuf since i have too much beef on my fire, and little time [21:24] <HorizonXP> saste: http://ffmpeg.org/pipermail/ffmpeg-devel/2012-January/118549.html ? [21:24] <HorizonXP> someone added a patch in January? [21:24] <saste> but the idea is that we should support some form of output pattern specification [21:26] <HorizonXP> saste: I would agree with that [21:26] <HorizonXP> sequential file naming and timestamps isn't the whole story [21:26] <HorizonXP> some kind of pattern would be nice [21:26] <HorizonXP> but it is a start [21:26] <HorizonXP> I'll try that patch out for now, since it will accomplish what I need [21:26] <saste> http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/158092/focus=158105 [21:26] <saste> ^^ [21:31] <llogan> relaxed: how often do you update the ffmpeg source on your static builds? [21:31] <HorizonXP> saste: thanks :) [00:00] --- Tue Feb 26 2013

1 0

ffmpeg-devel.log.20130224
by burek 25 Feb '13

25 Feb '13

[00:02] <cone-267> ffmpeg.git 03Reimar Döffinger 07master:50a37f9202a8: pmpdec: check for EOF while reading index. [00:45] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:b0bc0eb9789d: pmpdec: read index before creating audio streams [00:45] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:7276e9ea95a5: pmpdec: fix signedness [00:45] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:066739f6bc62: pmpdec: check packet sizes [02:44] Action: Compn types ./configure and cpu usage goes up. forgot how ffmpeg compiled silently [04:42] Action: Compn spends 3 hours tackling mingw [10:34] <durandal_1707> michaelni: have comments on exr slice threading? [11:28] <cone-697> ffmpeg.git 03Nicolas George 07master:bf0712c2f8ef: libavfilter/af_amerge: fix segfault if init fails. [11:55] <cone-697> ffmpeg.git 03Nicolas George 07master:3d7f4f87267c: lavf/avio: check that the protocol supports the open mode. [11:57] <durandal_1707> why is default thread type frame one? [12:07] <cone-697> ffmpeg.git 03Nicolas George 07master:2d98dd3d142b: lavfi: fix merging of formats and clarify exception. [12:07] <cone-697> ffmpeg.git 03Nicolas George 07master:e568d432b1d8: lavfi/formats: reindent after last commit. [12:20] <cone-697> ffmpeg.git 03Nicolas George 07master:ccc7bcc4fcaa: lavc: check return values consistency when decoding subtitles. [12:27] <cone-697> ffmpeg.git 03Anton Khirnov 07master:8097fc9a2dd4: 4xm: check the return value of read_huffman_tables(). [12:27] <cone-697> ffmpeg.git 03Anton Khirnov 07master:f935aca44c67: av_memcpy_backptr: avoid an infinite loop for back = 0 [12:27] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:4ba35194a9d7: Merge commit 'f935aca44c674d30e3ed940ef73bbad1228a5855' [12:31] <durandal_1707> ahh, frame threads are only by default used when probing [12:33] <durandal_1707> actually not [12:34] <durandal_1707> is there any way to force two decodings with single image? [12:34] <durandal_1707> *to force single decoding with one image input [12:35] <durandal_1707> currenly it is decoded twice [13:09] <cone-697> ffmpeg.git 03Anton Khirnov 07master:de6dfa2bb82d: lagarith: avoid infinite loop in lag_rac_refill() [13:09] <cone-697> ffmpeg.git 03Anton Khirnov 07master:067432c1c958: loco: check that there is data left after decoding a plane. [13:09] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:714ff44858a0: Merge commit '067432c1c95882c7221e694f33d9f3bdbe46de7f' [13:09] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:33796645dc35: loco: check the last plane too [13:14] <cone-697> ffmpeg.git 03Nicolas George 07master:b92c7a8f4b55: tools: add seek_print. [13:14] <cone-697> ffmpeg.git 03Nicolas George 07master:ea2de3d09614: lavf/concatdec: add the "duration" directive. [13:20] <cone-697> ffmpeg.git 03Anton Khirnov 07master:ddfe1246d98f: flicvideo: avoid an infinite loop in byte run compression [13:20] <cone-697> ffmpeg.git 03Anton Khirnov 07master:56daf10e0313: mov: use the format context for logging. [13:20] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:663ebae79a3e: Merge commit '56daf10e0313c5e36f43e773f457d2a99ff0df10' [13:28] <cone-697> ffmpeg.git 03Anton Khirnov 07master:0dff40bfb9a0: mlpdec: do not try to allocate a zero-sized output buffer. [13:28] <cone-697> ffmpeg.git 03Anton Khirnov 07master:4f3b058c84f5: cavs: initialize various context tables to 0 [13:28] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:9748cac5650a: Merge commit '4f3b058c84f570e261d743c7c22f865617fd28ac' [13:55] <cone-697> ffmpeg.git 03Anton Khirnov 07master:e10659244782: qtrle: add more checks against pixel_ptr being negative. [13:55] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:9ba38e62734e: Merge commit 'e10659244782b26061e7d52c06437de32a43a7af' [14:00] <cone-697> ffmpeg.git 03Anton Khirnov 07master:7b4f91155bd4: qtrle: cosmetics, reformat CHECK_PIXEL_PTR() macro [14:00] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:6f5f0671f3d6: Merge commit '7b4f91155bd4ef5a8d4e9af65c48b42bfa5b52c6' [14:09] <cone-697> ffmpeg.git 03Anton Khirnov 07master:d8a74d1d95a3: qtrle: use AV_LOG_ERROR in an error message. [14:10] <cone-697> ffmpeg.git 03John Van Sickle 07master:2f325a6fd442: libx264: change i_qfactor to use x264cli's default [14:10] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:a77a27a24b27: Merge commit '2f325a6fd4421c4dd4e26d7065e5d4bf26ed52f2' [14:32] <cone-697> ffmpeg.git 03Kostya Shishkov 07master:b5f536d24b5a: pnm: add high-bitdepth PGMYUV support for both encoder and decoder [14:32] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:198ea7a96f64: Merge commit 'b5f536d24b5ae360503935c34d5d59fa5181b94d' [14:32] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:5ecf8189c682: pnm: use av_pix_fmt_desc_get() [14:37] <cone-697> ffmpeg.git 03Diego Biurrun 07master:94ee7da08d2a: Remove pointless av_cold attributes in header files [14:37] <cone-697> ffmpeg.git 03Diego Biurrun 07master:040c565e5198: doc: developer: Allow tabs in the vim configuration for Automake files [14:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:2fc662ae63ab: Merge commit '040c565e51985477a8fa5e42d2ddfb26ebde6608' [14:42] <cone-697> ffmpeg.git 03Diego Biurrun 07master:7432e872066d: configure: Add print_3_columns helper function and use where appropriate [14:42] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:409890ca2f57: Merge commit '7432e872066d6960a9fbd31c51a94ebe6183389e' [14:50] <cone-697> ffmpeg.git 03Mans Rullgard 07master:0a8da1a3e5f0: configure: Do not redundantly list enabled hwaccel libs [14:50] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:74c92452541b: Merge commit '0a8da1a3e5f0b9329dfb89d17356ff5444c02351' [15:04] <cone-697> ffmpeg.git 03Diego Biurrun 07master:9840130edf3a: configure: Simplify VDA header and extralibs check [15:04] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:56c245f9219b: Merge commit '9840130edf3a969ec06dd0faa61dcf8d90c5f67a' [15:22] <cone-697> ffmpeg.git 03Diego Biurrun 07master:2aac411fd4c7: configure: Simplify VDPAU header check [15:22] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:42b411995caf: Merge commit '2aac411fd4c74e22b978525206f3b8257de1842b' [15:28] <cone-697> ffmpeg.git 03Diego Biurrun 07master:4cc4b33f71d3: build: Add proper infrastructure for adding and checking host CPPFLAGS [15:28] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:845bf99312d9: Merge commit '4cc4b33f71d3982df404fceb4405d656c538bc74' [15:33] <cone-697> ffmpeg.git 03Paul B Mahol 07master:1a08758e7c4e: exr: slice threading [15:33] <cone-697> ffmpeg.git 03Paul B Mahol 07master:645f96f129ad: exr: check color channel subsampling too [15:33] <cone-697> ffmpeg.git 03Paul B Mahol 07master:74a78bfe6c1b: exr: simplify filling channel_buffer[] [15:33] <cone-697> ffmpeg.git 03Paul B Mahol 07master:7b12554c5abc: exr: make sure that data_size is not bigger than expected [15:36] <durandal_1707> i need to wait 24h or 48h for fate sync? [15:39] <cone-697> ffmpeg.git 03Diego Biurrun 07master:215cdd35efd6: configure: Refactor dxva2api.h dependency declarations [15:39] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:60cc4a32b52f: Merge commit '215cdd35efd625ec28ef5846f1692b18f7c2c230' [15:46] <michaelni> durandal_1707, i dont know, it depends on how the clients are setup [15:47] <michaelni> none of my clients needs any delay [15:49] <cbsrobot_> whats the purpose of the last "codec" in Options ? [15:49] <cbsrobot_> { "c", HAS_ARG | OPT_STRING | OPT_SPEC, { .off = OFFSET(codec_names) }, "codec name", "codec" } [16:39] <cone-697> ffmpeg.git 03Diego Biurrun 07master:82ca17ac7a0a: configure: Fix vaapi/vda/vdpau dependency declarations [16:39] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:8cf9253aef19: Merge commit '82ca17ac7a0a08784cb6808384ee237ac28e8334' [16:39] <cone-697> ffmpeg.git 03Diego Biurrun 07master:fdd392ed27d8: configure: Simplify VDPAU header check [16:50] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:dbabea2f2378: configure: hwaccel autodetection has been removed in 82ca17ac7a0a08784cb6808384ee237ac28e8334 [16:58] <cone-697> ffmpeg.git 03Justin Ruggles 07master:d7c450436fcb: ac3dec: validate channel output mode against channel count [16:58] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:f7c4b76c488b: Merge commit 'd7c450436fcb9d3ecf59884a574e7684183e753d' [17:15] <cone-697> ffmpeg.git 03Justin Ruggles 07master:9f1223562e13: lavfi: connect libavresample options to af_resample via AVFilterGraph [17:15] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:71cf094e1be0: Merge commit '9f1223562e134bac6345a465870b9d56ff7d60cf' [17:18] <durandal_1707> michaelni: can you explain that merge? [17:23] <durandal_1707> nvm [17:42] <cone-697> ffmpeg.git 03Justin Ruggles 07master:5c7db097ebe1: avconv: pass libavresample options to AVFilterGraph [17:42] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:6db37c946805: Merge commit '5c7db097ebe1fb5c233cedd8846615074e7d6044' [17:49] <cone-697> ffmpeg.git 03Justin Ruggles 07master:50f4337a2fd3: lavr: Add "resample_cutoff" option as a duplicate of "cutoff" [17:49] <cone-697> ffmpeg.git 03Justin Ruggles 07master:b2eea615c077: lavr: allow setting internal_sample_fmt option by string [17:49] <cone-697> ffmpeg.git 03Diego Biurrun 07master:45235ac48836: configure: Move x11grab option to a more suitable place in the help output [17:49] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:5f8f9dc4362b: Merge commit '45235ac488363e3360bf2f2275102d1ec66eba0f' [18:01] <cone-697> ffmpeg.git 03Mans Rullgard 07master:3fc09b008118: configure: Move list of external libs to a separate variable [18:01] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:214cb30b72b9: Merge commit '3fc09b0081184f26edbb62d2d72ae89bf9e21768' [18:09] <cone-697> ffmpeg.git 03Mans Rullgard 07master:04cccb5fc109: configure: List external libs used using print_enabled() [18:09] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:036df234fb28: Merge remote-tracking branch 'qatar/master' [18:57] <durandal_1707> can somone disable lavfi if libavdevice is not compiled? [19:01] <ubitux> lavfi doesn't depend on lavd [19:03] <durandal_1707> ubitux: lavfi in lavd [19:49] <cone-697> ffmpeg.git 03Reimar Döffinger 07master:db6e2e848b21: hls: do not access pb->opaque for custom IO. [19:49] <cone-697> ffmpeg.git 03Reimar Döffinger 07master:c65eb7907c49: mpeg12: Fix non-hwaccel VDPAU decode. [19:54] <teratorn> can I ask why I see errors about non-monotonically increasing dts when I call av_interleaved_write_frame() ? isn't it the job of the interleaving to queue the packets and mux them in dts order? am I missing something ? :( [20:30] <cone-697> ffmpeg.git 03Paul B Mahol 07master:bc980d57caaf: fate: add animated gif demuxer test [20:30] <cone-697> ffmpeg.git 03Paul B Mahol 07master:8a7d177cf65d: fate: add animated gif decoder tests [20:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:2abbe6d07a34: swr: add duplicate cutoff for compatibility [20:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:c4c702b6d3b4: avfilter/avfiltergraph.h: Move public field out of the private fields [20:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:8b3affda87e4: swr: support a seperate output sample bits. [20:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:f3abdf4392a1: cmdutils: fix null pointer dereference [20:37] <cone-697> ffmpeg.git 03Michael Niedermayer 07master:ad899522ffa7: ffmpeg: use a AVDictionary instead of the context to move swr parameters around [22:47] <durandal_1707> ubitux: why is charenc test still failing? [23:46] <cone-697> ffmpeg.git 03Stefano Sabatini 07master:5085b46496f0: lavc: change type of AVFrame.channels field from int64_t to int [23:46] <cone-697> ffmpeg.git 03Stefano Sabatini 07master:b59cd089ffd6: lavfi/abuffersink: add sample_rates field to AVABufferSinkParams [23:46] <cone-697> ffmpeg.git 03Stefano Sabatini 07master:566560b85c37: lavfi/abuffersink: fix weird indent and spacing [23:46] <cone-697> ffmpeg.git 03Stefano Sabatini 07master:a3fa27e366cd: ffplay: set type for channel_layout AudioParams field to int64_t [23:46] <cone-697> ffmpeg.git 03Stefano Sabatini 07master:394130efe32f: ffplay: reindent and remove pointless cast in audio_decode_frame() code [23:57] <ezekiel> unexpected breakthrough today with synchronized audio+video capture with ffmpeg from git, any devs feel free to tell me if this is expected behavior: [23:57] <ezekiel> ffmpeg -y -f video4linux2 -i /dev/video0 -f alsa -i hw:CARD=U0x46d0x821,DEV=0 ~/tmp/output-file.mp4 #results in out-of-sync audio/video [23:57] <ezekiel> ffmpeg -y -f alsa -i hw:CARD=U0x46d0x821,DEV=0 -f video4linux2 -i /dev/video0 ~/tmp/output-file.mp4 #results in in-sync audio/video [23:57] <ezekiel> so - just changing the order of the inputs fixes audio sync - is that expected? [00:00] --- Mon Feb 25 2013

1 0

ffmpeg.log.20130224
by burek 25 Feb '13

25 Feb '13

[09:41] <Fuuzetsu> Greetings. I'm attempting to capture certain frames using ffmpeg from a mkv file into images. It's all nice and dandy but it doesn't seem to do subtitles which are in one of the streams in the file. [09:42] <Fuuzetsu> Is there a way to get the subtitles rendered onto the frame? Googling only brings up same issue without replies or different issues that aren't of help. [10:57] <IchGuckLive> hi all im getting double file size on ffmpeg -i rio.ts -target pal-dvd output.vob [10:57] <sacarasc> Hehe, IchGuckLive is watching riots. :D [10:58] <IchGuckLive> Thanks but this is the command [10:58] <IchGuckLive> you may want the input info and output info [10:58] <IchGuckLive> i think [10:58] <durandal_1707> i want full uncut output [11:01] <IchGuckLive> http://pastie.org/6327114 [11:02] <durandal_1707> that is not full uncut command output [11:02] <IchGuckLive> durandal_1707: http://g1.globo.com/rio-de-janeiro/carnaval/2013/fotos/2013/02/fotos-desfil… [11:05] <IchGuckLive> ok i did it again and here is the full workflow http://pastie.org/6327137 [11:06] <IchGuckLive> input is 62MB output 120MB [11:07] <IchGuckLive> output plays perfect [11:08] <Fuuzetsu> shouldn't ffmpeg -i file -filter_complex '[0:v][0:s]overlay[v]' -map [v] test.png produce an image with subtitles now visible on the video? [11:08] <Fuuzetsu> I can't get it to work& I always get just video. [11:09] <durandal_1707> Fuuzetsu: subtitles with that command? [11:09] <Fuuzetsu> I have subtitles at 0:2 [11:10] <durandal_1707> IchGuckLive: you are transcoding video, could use -vcodec copy [11:10] <Fuuzetsu> Stream mapping: Stream #0:0 (h264) -> overlay:main Stream #0:2 (ass) -> overlay:overlay overlay -> Stream #0:0 (png) suggests that it should work [11:10] <durandal_1707> IchGuckLive: and transcoding audio, dunno if that you want [11:10] <IchGuckLive> ok [11:10] <Fuuzetsu> you seem to have joined after I posted my issue& I'll repost (2 lines) [11:11] <Fuuzetsu> Greetings. I'm attempting to capture certain frames using ffmpeg from a mkv file into images. It's all nice and dandy but it doesn't seem to do subtitles which are in one of the streams in the file. [11:11] <Fuuzetsu> Is there a way to get the subtitles rendered onto the frame? Googling only brings up same issue without replies or different issues that aren't of help. [11:13] <IchGuckLive> durandal_1707: means -vcodec copy -acodec copy ? [11:14] <durandal_1707> IchGuckLive: dunno, input may be incompatible with pal-dvd target you want [11:14] <Fjorgynn> -c:v copy -c:a copy ? [11:14] <durandal_1707> or just -c copy [11:15] <IchGuckLive> ok [11:15] <durandal_1707> this should not increase size at all [11:15] <durandal_1707> but dunno if it will be playable on stuff that expects pal-dvd [11:17] <durandal_1707> otherwise if size is too big you try to lower quality, again i really do not know what are requirements for pal-dvd, so maybe output you get is expected result [11:18] <IchGuckLive> thanks [11:18] <durandal_1707> Fjorgynn: i fail to see how using test.png will burn subtitles [11:19] <durandal_1707> there are examples on wiki [11:20] <durandal_1707> https://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20burn%20subtitles%20into%20th… [11:25] <Fuuzetsu> durandal_1707: you meant to highlight me [11:26] <durandal_1707> yes, fail [11:26] <Fuuzetsu> I've been staring at that very page for a while now [11:26] <Fuuzetsu> I don't see how to make the jump from mkv with saa subs embedded into a png [11:26] <Fuuzetsu> The examples assume I have a subtitle file already and the last one is exactly what I'm doing now. [11:26] <durandal_1707> so your subtitles in mkv are bitmap? [11:27] <Fuuzetsu> I get a sub2video: non-bitmap subtitle [11:27] <Fuuzetsu> Stream #0:2(eng): Subtitle: ssa (default) [11:28] <durandal_1707> so you could try ass filter example [11:29] <durandal_1707> but i guess you need to extract it first from mkv [11:29] <durandal_1707> because i dunno if there is shorter way [11:30] <Fuuzetsu> Hmm, I see. [11:32] <Fuuzetsu> [AVFilterGraph @ 0x82023b0] No such filter: 'ass' [11:32] <Fuuzetsu> ;_; [11:32] <durandal_1707> you need build with libass support [11:33] <Fuuzetsu> Ah, yes, I just noticed that I don't have a USE flag for it [11:33] <durandal_1707> libass actually renders subtitles for ffmpeg [11:34] <Fuuzetsu> At least extracting the subtitles doesn't take a long time. I'll be happy to get it working at all. [11:34] <Fuuzetsu> The night has now turned into 10:30am& [12:10] <Fuuzetsu> durandal_1707: I got it to work. Thank you so, so, so much. [12:35] <hendry> how does one convert a Macromedia Flash data (compressed), version 11 to mkv? [12:36] <hendry> i tried ffmpeg, but i got "could not find codec parameters" as an error [12:39] <hendry> durandal_1707: http://sprunge.us/KKVV [12:40] <durandal_1707> hendry: upload file somewhere and post link to it [12:43] <hendry> durandal_1707: http://s.natalian.org/2013-02-24/test.swf [12:48] <durandal_1707> hendry: it is some binary stuff it does not have video/audio [12:50] <Fuuzetsu> When using -ss, the video is properly captured from the time I seek to but the subtitles always start from the beginning. Is there a way to remedy this? [12:54] <durandal_1707> Fuuzetsu: no yet, because -ss seek mkv only and there is no way to seek with filter [12:54] <durandal_1707> but you could try to extract subtitle with -ss .... [12:55] <durandal_1707> or could try to change pts of subtitle packets .... [12:56] <durandal_1707> ^hmm not possible with subtitle streams [12:56] <Fuuzetsu> I see. You say no yet though. Does this mean that it's a planned feature? [12:59] <durandal_1707> maybe, also not need to extract subtitle is even more useful feature [13:00] <Fuuzetsu> I agree& [13:00] <durandal_1707> so you could create ticket(s) [13:00] <durandal_1707> for this of for anything you need and should be present [13:01] <Fuuzetsu> I might actually do that in a bit. Bed first. Thanks for all the help. [13:02] <shevy> hmm how many of the people here contribute code to ffmpeg? [13:03] <Fuuzetsu> I'm guessing anyone who sits around in #ffmpeg-devel [13:31] <mpfundstein> #join #vlc-devel [13:34] <Fjorgynn> why? [13:40] <ubitux> durandal_1707, Fuuzetsu: at least two tickets exists for this [13:40] <ubitux> but the subtitles injection in lavfi is not simple [13:49] <ubitux> Fuuzetsu: also, if you're just remuxing subtitles, afaik it works [13:50] <Fuuzetsu> My ultimate goal is to be able to take screencaps at some intervals. [13:50] <ubitux> i just checked [13:51] <ubitux> it works [13:51] <Fuuzetsu> I clearly don't want to re-encode the whole video just for that purpose [13:51] <Fuuzetsu> How? [13:51] <ubitux> i tried ffmpeg -i in.mkv -ss 10:00 -t 3:00 -c copy -map 0 -y out.mkv [13:51] <ubitux> and it just works [13:52] <ubitux> i've fixed this a while ago [13:52] <ubitux> make sure you're up to date [13:52] <Fuuzetsu> ffmpeg version 1.0.4 [13:52] <Fuuzetsu> built on Feb 24 2013 10:39:56 with gcc 4.6.3 (Gentoo 4.6.3 p1.11, pie-0.5.2) [13:52] <ubitux> i've fixed this month ago, iirc before 1.0 but i'm not sure [13:53] <ubitux> you should try git/HEAD or at least 1.1 [13:53] <Fuuzetsu> OK will do [13:53] <ubitux> if you have an available sample around i can have a look [13:53] <Fuuzetsu> but are you saying that the subtitles should seek along with the video without any additional magic? [13:54] <ubitux> yes [13:54] <Fuuzetsu> I'll try head [13:54] <ubitux> what doesn't is when you're trying to hardsub subtitles with libavfilter (-vf subtitles=...) [13:54] <ubitux> in that particular case, the seek won't work [13:54] <ubitux> otherwise it should [13:58] <Fuuzetsu> is something horribly wrong with 1.1.x? My distro seems to have it masked [13:58] <Fuuzetsu> (Gentoo, using portage) [14:00] <ubitux> the api changes may break some apps built maybe [14:00] <ubitux> but i'm not a gentoo user, no idea [14:01] Action: Fuuzetsu prays, gritts teeth and crosses fingers while he starts building [14:01] <Fuuzetsu> I still haven't recovered from last night breakage [14:01] <JEEB> when I last looked at gentoo's stuff marked as "stable" I had to laugh :P [14:02] <JEEB> basically the newest two or three versions of firefox of the current release branch were labled nonstable and thus you can guess which version people set up on "stable" were using :P [14:02] <JEEB> this was back with 3.6.x or so [14:02] <Fuuzetsu> JEEB: we're all daredevils on ~ [14:02] <JEEB> tl;dr it really felt like that stamp of "stable" didn't really mean anything >_> [14:03] <Fuuzetsu> it just tends to mean old and tested [14:03] <JEEB> yeah, which just got really laughable with a browser that gets security updates :P [14:05] <Fuuzetsu> Hey, if you want security then you won't be using firefox in the first place [14:06] <JEEB> if you want security you don't stick your machine into a network [14:06] <Fuuzetsu> (in fact, if you want securty then it probably will be on a box without a Real Browser") [14:24] <hendry> I'm trying to limit my capture to 5seconds, but ffmpeg -t 5 -threads auto -f x11grab -s 1366x768 -i :0.0 -f alsa -i hw:0,0 -acodec pcm_s16le -vcodec ffvhuff test.mkv # does not work :/ what am I missing? [14:25] <Tchico> Hi every body [14:25] <Tchico> please when you have time, look at my topic on the forum [14:25] <Tchico> http://ffmpeg.gusari.org/viewtopic.php?f=11&t=840&p=1657#p1657 [14:36] <Keyboard_Warrior> Tchico, take a look at qt-faststart [14:37] <Keyboard_Warrior> i asume its becasue your mp4's moov-atom is at the end [14:37] <Keyboard_Warrior> so, you have to download the entire video file before you can play it in the browser [14:38] <Tchico> thx I will see [14:38] <Tchico> and test [14:38] <Keyboard_Warrior> it comes with ffmpeg [14:38] <Keyboard_Warrior> or can atleast [14:47] <ubitux> -movflags +faststart [14:58] <Keyboard_Warrior> ubitux, you can do that these days? :P [14:58] <Keyboard_Warrior> ubitux, i really should re-learn ffmpeg [14:58] <ubitux> yes you can [14:59] <durandal_1707> Keyboard_Warrior: there is documentation [14:59] <Keyboard_Warrior> durandal_1707, there wasnt much good info back when i was using it [14:59] <Keyboard_Warrior> 2007ish :P [14:59] <Keyboard_Warrior> heck, most of the info was even wrong :P [14:59] <Keyboard_Warrior> well not most [15:00] <Keyboard_Warrior> but, there were plenty in the official man and help stuff, that was just blatantly wrong [15:00] <Keyboard_Warrior> durandal_1707, so yeah, im still stuck with how ffmpeg worked in the long long time agoes :P [15:00] <Keyboard_Warrior> i havent used it at all since 2010 or so [15:04] <ubitux> the documentation is split and maintained actively, at least for new stuff [15:04] <ubitux> (this might not have been the case in the past yes) [15:13] <Tchico> no, qt-faststart don't hlp me :( [15:13] <Tchico> http://gnagn.olympe.in/nancyCamera.php look with chrome or ie this page you will understand [15:21] <Tchico> The videos don't load ... but if I put a video of big_bunny or any other mp4 it's works ... [15:27] <Tchico> I update my post, on the forum: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=840&p=1658#p1658 [15:27] <Tchico> Check when you have time please [15:31] <ubitux> a little surprising but maybe it only supports a few h264 levels [15:46] <Tchico> h264 levels ? [15:46] <Tchico> When I convert a video to a h264 video, it works but not when I create the video from pictures with ffmpeg [17:17] <Tchico> Anybody here ? [17:21] <cbsrobot_> show the output for both files please [17:24] <Tchico> ok [17:51] <Tchico> this is the encoding of the video from the pictures [17:51] <Tchico> http://pastebin.com/kAe3R3PY [17:53] <Tchico> this is the qt-faststart [17:53] <Tchico> http://pastebin.com/bcRnFEvM [17:55] <ubitux> SVN-r26402 [17:55] <ubitux> bitch plz [17:55] <ubitux> :D [17:55] <ubitux> come on, please upgrade :) [17:58] <Tchico> ok ^^ [17:58] <cbsrobot_> Tchico: my guess would be yuvj420p is not working on your device, but ubitux is right - please upgrade [17:59] <cbsrobot_> and btw. next time show the output of the file that works and the file that does not work .... [17:59] <Tchico> I am encoding the file qhich work [18:00] <Tchico> still encoding [18:00] <Tchico> when he finishes I send you his pastebin and go upgrade ffmpeg [18:00] <cbsrobot_> Tchico: just show the output of: ffmpeg -i file_working.ext [18:00] <cbsrobot_> and: ffmpeg -i file_not_working.ext [18:01] <Tchico> oh, ok [18:01] <Tchico> sorry doesn't know you want that [18:02] <cbsrobot_> Tchico: well *I* don't want it that badly - but that way you can compare the two files [18:03] <Tchico> And the pastebin of the file which work: http://pastebin.com/RYcpXyfC [18:05] <Tchico> the video which normally works, doesn't work this time - -' [18:06] <Tchico> so I update ffmpeg and I retest [19:41] <teratorn> can I ask why I see errors about non-monotonically increasing dts when I call av_interleaved_write_frame() ? isn't it the job of the damn interleaving to queue the packets and mux them in dts order? [19:41] <teratorn> or am I misunderstanding something? :( [19:57] <spinx60> +history [19:57] <spinx60> is there a way to see chat history [19:58] <durandal_1707> spinx60: yes [19:58] <spinx60> how ? :) [19:59] <Mavrik> teratorn, it interleaves them over several streams [19:59] <Mavrik> teratorn, you still have to make sure yourself that DTS increases monotonicaly within a stream [20:00] <teratorn> Mavrik: aye - I believe I am doing that. the error seems to be because I encode several audio packets very quickly, but my video codec delays several frames before it emits any packets. When I write the first video packet, I get the dts non-monotonically increasing error [20:01] <teratorn> the error shows it comparing against the latest audio packet [20:01] <teratorn> the dts's that is [20:01] <teratorn> I'm not sure what this means. I had the idea that if I just wrote a video packet *first*, then it would queue the audio properly.. but I don't know what is really wrong [20:01] <Mavrik> teratorn, hmhmhm, that shouldn't happen [20:02] <Mavrik> teratorn, do your packets have correct stream indexes set? [20:02] <teratorn> I believe so. I should double check [20:02] <teratorn> I know that stream_index must be set right.. so I'll double check that [20:04] <Mavrik> teratorn, also sometimes if DTS difference of audio and video is too big you get there errors [20:04] <teratorn> ffs. I was setting the audio packets to the video index [20:04] <teratorn> and I spent already an hour writing a damn type that would queue and interleave the packets to "fix" this error... [20:04] <Mavrik> been there, done that, killed a day to debug that :P [20:05] <teratorn> I don't think I was cut out for this kind of work :( [20:05] <teratorn> timestamps and freaking stream muxing [20:05] <teratorn> *sigh* - thanks [20:06] <durandal_1707> spinx60: ffmpeg.org/contact.html [20:26] <Tchico> a [20:35] <Tchico> ok [20:35] <Tchico> ffmpeg upgraded, that's the pastbin of the 2 videos recreated : http://pastebin.com/f8JqkFpH [20:35] <Tchico> they still don't work :( [20:39] <durandal_1707> Tchico: what does not work? [20:44] <Tchico> the video that I create from pictures cant be visible on html5 video balise for chrome and IE [20:45] <durandal_1707> Tchico: what command you use [20:46] <Tchico> http://ffmpeg.gusari.org/viewtopic.php?f=11&t=840&p=1661#p1661 [20:46] <Tchico> I put all details in this post [20:47] <Tchico> ffmpeg -i %d.jpg -vcodec libx264 -preset slow -profile baseline -b 250k -bt 50k -an -g 30 -s 640x360 -r 24 -y res.mp4 [20:48] <Tchico> durandal_1707 : its the new command I use [20:53] <durandal_1707> and what pix fmt are jpgs? [20:54] <Tchico> JFIF [20:55] <durandal_1707> added -movflags +faststart ? [20:56] <Tchico> when ? [20:56] <Tchico> I call qt-faststart after on my video [20:57] <durandal_1707> qt-faststart is not needed, latest ffmpeg can do it itself with above mentioned args [20:57] <Tchico> I am encoding a video with -movflags +faststart and -threads 0 options [21:01] <durandal_1707> perhaps something is unexpected [21:01] <durandal_1707> please post full uncut output of conversion that does not work [21:04] <Tchico> k [21:18] <durandal_1707> Tchico: you do not need to wait it to complete [21:19] <Tchico> ? [21:19] <durandal_1707> you could also limit it to encode only first 10 sec with -t 10 [21:20] <Tchico> http://pastebin.com/VWUvc3WL [21:20] <Tchico> thx, next time I will do it ;) [21:23] <durandal_1707> perhaps you should use -provile:v instead of just -profile [21:23] <durandal_1707> same for -b:v instead of -b [21:24] <Tchico> ok, cause its still not working :( [21:24] <durandal_1707> dunno, maybe it needs audio [21:25] <Tchico> :( [21:32] <Tchico> vlc give me no details of the codecs now [21:36] <foonix> is there a way to compare two inputs ? [21:36] <foonix> i mean something like psnr [21:41] <Tchico> I update my post: http://ffmpeg.gusari.org/viewtopic.php?f=11&t=840&p=1666#p1666 if someone have an idea :( [21:51] <Tchico> I have a good news :) it's working on IE :) but still not in chrome - -' [22:02] <tbo> I compiled ffmpeg git today and vda_h264 is no longer being included? intentional? [22:11] <tbo> --enable-vda to the ffmpeg configure fixed it, just weird that it needed that [22:12] <Fjorgynn> night [22:18] <ubitux> tbo: http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=dbabea2f23782c09106765… [22:18] <tbo> ah okay, thanks [22:18] <ubitux> sorry about that :) [22:19] <tbo> np, I had to change the homebrew formular and compile it again. [22:19] <tbo> hwaccel is annoying anyway [22:29] <Jiyuu|Alt> can anyone help me figure out why ffmpeg is crashing for me whenever i try to encode to h264 [22:30] <Jiyuu|Alt> ffmpeg -i "video.mkv" -c:v h264 -c:a copy -s 800:480 output4.mp4 [22:31] <JEEB> is it crashing or just telling you that there is no such encoder? [22:31] <JEEB> h264 is just a decoder in libavcodec, the only H.264 encoder in libavcodec is libx264 [22:31] <Jiyuu|Alt> crash [22:31] <Jiyuu|Alt> im on win 64 btw [22:31] <Jiyuu|Alt> in case its a problem [22:32] <Jiyuu|Alt> my mistake [22:32] <JEEB> hm? [22:32] <Jiyuu|Alt> ffmpeg -i "video.mkv" -c:v h264 -c:a copy -s 800:480 output4.mp4 [22:32] <Jiyuu|Alt> ah [22:32] <Jiyuu|Alt> i did copy correctly [22:32] <Jiyuu|Alt> nvm [22:32] <Jiyuu|Alt> anyway [22:32] <JEEB> hmm, reminds me of a bug that was around for a moment for win64 and H.264 decoding [22:32] <Jiyuu|Alt> yeah its still crashing [22:33] <ubitux> it's not supposed to [22:33] <Jiyuu|Alt> should i try win32 [22:33] <JEEB> ubitux, wasn't there some red zone thingy regarding H.264 decoding lately? [22:33] <Jiyuu|Alt> would there be any diff as far as performance goes? [22:33] <JEEB> on win64 [22:33] <ubitux> JEEB: maybe, i'm not following this stuff [22:33] <JEEB> Jiyuu|Alt, nothing too big, a few % I would guess [22:33] <JEEB> I remember BBB pushing a fix for it [22:33] <ubitux> Jiyuu|Alt: is this a recent ffmpeg anyway? [22:33] <ubitux> JEEB: same here [22:34] <ubitux> 82a4a4e7 too [22:34] <Jiyuu|Alt> ok it is working with win32 [22:34] <JEEB> then it's most probably that if the build is new [22:34] <ubitux> make sure you have a recent version of ffmpeg [22:35] <JEEB> (or well, relatively new) [22:35] <ubitux> (basically git/master/HEAD) [22:35] <Jiyuu|Alt> http://ffmpeg.zeranoe.com/builds/ [22:36] <Jiyuu|Alt> Download FFmpeg git-066739f 64-bit Static [22:37] <JEEB> if you do ffmpeg -i input.mkv -c:v rawvideo -f null NUL , does it still crash? [22:41] <Jiyuu|Alt> i can encode to mpeg4 at least [22:41] <Jiyuu|Alt> could [22:41] <JEEB> interesting [22:41] <JEEB> x264 should haven't any problems with its current normal source tree on win64 [22:41] <JEEB> what if you use the proper encoder name libx264 instead of h264? [22:42] <Jiyuu|Alt> tried it and also crashed [22:42] <Jiyuu|Alt> it does detect the codec btw [22:42] <JEEB> k [22:42] <Jiyuu|Alt> one sec [22:42] <Jiyuu|Alt> i'll show you the log [22:43] <JEEB> yeah, ffmpeg nowadays probably has a way of going from "h264" on the encoding side to libx264 [22:43] <JEEB> just like "aac" goes to one of the aac encoders available in the build [22:43] <JEEB> so yeah, seems like zeranoe failed at building 64bit x264? [22:46] <Jiyuu|Alt> http://pastebin.com/8WRwEhGb [22:48] <JEEB> if you wanna see where it exactly crashes [22:48] <JEEB> http://x264.fushizen.eu/builds/gdb/gdb-7.5.7z [22:48] <JEEB> grab this [22:49] <JEEB> then use the 64bit gdb like this: "gdb ffmpeg.exe" [22:49] <JEEB> although wait... that thing probably doesn't contain symbols ^^; [22:54] <Jiyuu|Alt> gdb works on windows? lol [22:58] <JEEB> yeah [23:03] <elkng> there are 3 formats of the same video on youtube flv,mp4,webm and they are all of "640x360" resolution, which one should be in better quality ? [23:03] <elkng> flv ? [23:03] <durandal_1707> higher bitrate [23:04] <durandal_1707> or actually visually inspecting output [23:07] <Tchico> Now I have a picture on IE [23:07] <Tchico> only one picture of all video, but he recognise the format [23:48] <ezekiel> unexpected breakthrough today with synchronized audio+video capture with ffmpeg from git, any devs feel free to tell me if this is expected behavior: [23:48] <ezekiel> ffmpeg -y -f video4linux2 -i /dev/video0 -f alsa -i hw:CARD=U0x46d0x821,DEV=0 ~/tmp/output-file.mp4 #results in out-of sync audio/video [23:49] <ezekiel> ffmpeg -y -f alsa -i hw:CARD=U0x46d0x821,DEV=0 -f video4linux2 -i /dev/video0 ~/tmp/output-file.mp4 #results in in-sync audio/video [23:49] <ezekiel> so - just changing the order of the inputs fixes audio sync - is that expected? [00:00] --- Mon Feb 25 2013

1 0

ffmpeg-devel.log.20130223
by burek 24 Feb '13

24 Feb '13

[00:15] <ubitux> saste: you forgot to update fate after the time parsing patch [00:15] <saste> ubitux, how is it affected? [00:15] <ubitux> -12:34 -> error [00:15] <ubitux> +12:34 -> +754000000 [00:16] <ubitux> parseutils [00:16] <ubitux> fate is staining toward yellow [00:18] <gnafu> Ew, "yellow stains." [00:19] <cone-345> ffmpeg.git 03Stefano Sabatini 07master:e95637841c0a: tests: fix parseutils test after 12a269a5229d3a37be0743fc9655f743ebc44b6e [00:25] <ubitux> btw [00:26] <ubitux> is 30:12.123 supported? :) [00:27] <ubitux> mmh seems it is [00:27] <ubitux> great [00:28] <cone-345> ffmpeg.git 03Clément BSsch 07master:0ac71f9a320d: fate/subtitles: add character encoding conversion test. [00:42] <saste> uhm... dynamic expression evaluation in volume filter [00:48] <saste> how should I handle NaN? [00:48] <saste> NaN = 0? [01:39] <Compn> saste : like if a pointer returns NaN ? [01:39] Action: ubitux is lost with the format negociation thing [01:39] Action: Compn no clue :) [01:40] <Compn> ubitux : audio or video ? [01:40] <ubitux> audio [01:40] <ubitux> Compn: NaN out of eval() i guess [01:40] <ubitux> saste: NaN = 1 i'd say (unchanged output) [01:41] <wm4> ubitux: libavfilter format negotiation? [01:41] <ubitux> wm4: yes [01:41] <wm4> fun stuff, I guess... [01:41] <ubitux> saste: i don't need to set anything for the output? [01:41] <ubitux> removing the format thing from query_formats in ebur128 seems to solve stuff [01:41] <ubitux> (for the output) [01:42] <Compn> michaelni : i didnt get reply from laurent (fenrir) about dxva2 patch [02:05] <cone-345> ffmpeg.git 03Matt Wolenetz 07release/1.1:5bed920971c5: Fix Win64 AVX h264_deblock by not using redzone on Win64 [02:30] <michaelni> Compn, did you (or someone else) test that dxva2 patch ? [02:32] <Compn> i cant find anyone to test it , either [03:40] <cone-345> ffmpeg.git 03Clément BSsch 07fatal: ambiguous argument 'refs/tags/n1.1.3': unknown revision or path not in the working tree. [03:40] <cone-345> Use '--' to separate paths from revisions [03:40] <cone-345> refs/tags/n1.1.3:HEAD: fate/subtitles: add character encoding conversion test. [03:48] <michaelni> av_interleaved_write_frame(): No space left on device [03:48] <michaelni> once again one of my fate boxes ate up all its space ... [03:49] <Compn> michaelni : it would be nice if someone came up with a list of cards that support dxva2 as well [03:49] <Compn> at least then we can try to find someone [03:49] <Compn> with the right card [03:50] <michaelni> yes [03:51] <Compn> i guess my card supports dxva 2.0 [03:51] <Compn> radeon 5830 [03:51] <Compn> i could test it [04:00] <michaelni> if you find bugs in it then the author could fix them and improve the patch, so its surely a good idea to test [04:02] <Compn> michaelni : still, someone has to review it [04:02] <Compn> michaelni : what about including that vid.stab filter ? [04:03] <Compn> that the author has asked if he should work on including a wrapper for a 3rd party dll or include the whole thing in libavfilter ? [04:03] <Compn> or its up to saste ? [04:05] <michaelni> did the author state exactly what he would prefer ? [04:06] <michaelni> did someone object ? [04:11] <Compn> michaelni : theres a lot of 'ffmpeg policy this ' and 'ffmpeg policy that' [04:11] <Compn> i dont see any actual objections [04:12] <Compn> everyones waiting for the great dictator to make his judgement ;) [04:12] <Compn> so itd be cool to have your opinion if you could write a few words on the subj [04:12] <Compn> or your official decree [04:12] <Compn> your royal proclamation [04:15] <Compn> i think he stated that his filter is faster (by a factor of 3-4) and has more features than ours [04:15] <Compn> are we going to add in every video filter until we port them? or keep them seperate until we port them ? are we going to port every filter? so many qustions [04:20] <Compn> thanks for replying :) [04:30] <tonsofpcs> still no answer about dts? [04:34] <michaelni> tonsofpcs, dts=pts when theres no reordering (or the file is broken or the code is buggy) [04:37] <tonsofpcs> michaelni - why is this done? [10:38] <cahit> i have used swresample instead of patching aacdec for S16 sample format, now i get audio values like: 186 2 0 0 75 255 0 0 221 254 0 0 243 254 0 0 173 0 0 0 234 1 0 0 82 255 0 0 80 255 0 0 87 5 0 0 182 8 0 0 153 5 0 0 160 2 0 0 101 3 0 0 [10:38] <cahit> lots of zeros inbetween [10:39] <nevcairiel> looks like stereo with one silent channel [10:40] <cahit> yes, but the audio is stereo [10:40] <cahit> i init resample context like this: [10:40] <cahit> SwrContext* resampleCtx = swr_alloc_set_opts(NULL,av_get_default_channel_layout(pCodecCtx->channels), AV_SAMPLE_FMT_S16, pCodecCtx->sample_rate, av_get_default_channel_layout(pCodecCtx->channels), pCodecCtx->sample_fmt,pCodecCtx->sample_rate, 0, 0); [10:40] <cahit> it should be safe as i don't change channels etc.. [10:44] <cahit> i think i give the input correct: int dataSize = swr_convert(resampleCtx, &resampledOut, AVCODEC_MAX_AUDIO_FRAME_SIZE,((const uint8_t**) &aligned_samples), length / (pCodecCtx->channels * 2)); [10:44] <cahit> do you have any ideas related to what i am missing here? [10:45] <cahit> (if i print audio values before resample, i don't get zeroes) [10:45] <nevcairiel> the decoder probably outputs planar audio, and you're only providing the first plane [10:46] <nevcairiel> the "in" parameter to swr_convert should simply be AVFrame->data [10:46] <cahit> ov! [10:49] <cahit> so i should give it const uint8_t *in[] { a,b } [10:50] <nevcairiel> since you obviously use avcodec for decoding, why not simply use the "data" member of the AVFrame you get from the decode function, and be done with it? :) [10:50] <nevcairiel> dont need to reshuffle data [10:50] <nevcairiel> or pointers [10:50] <nevcairiel> but yes, it needs an "array" of pointers to the different planes [10:52] <cahit> i am using avcodec_decode_audio3 function and manually expanding the data pointer until all the data in that av_read_frame packet. at the end of every read_frame loop i push decoded data to another buffer. [10:52] <cahit> so after avcodec_decode_audio3 i only have a single uint pointer [10:54] <nevcairiel> i would suggest to do the S16 conversion inside this loop and push the final S16 data into your buffer [10:54] <nevcairiel> would probably simplify it a bit [10:57] <cahit> and i think i have to use avcodec_decode_audio4, or i have to reverse current audio3 response to generate other other plane from returned data... [10:58] <nevcairiel> using the latest version is always a good idea, the older ones will go away eventually [11:00] <cahit> thank you for your help, it's a lot cleaner now [12:28] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:3c14c82b7e01: avfilter: Silence warning: passing argument 3 of av_image_copy from incompatible pointer type [12:28] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:cfcab4c50747: vf_overlay: silence warning: X may be used uninitialized in this function [12:29] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:23c9180c001c: aacdec: Fix warning: initialization from incompatible pointer type [12:31] <durandal_1707> michaelni: uploaded gif.tar to incoming (for animation gif coverage) [12:38] <michaelni> durandal_1707, uploaded [13:28] <durandal_1707> michaelni: uploaded exr.tar.bz2 to incoming (for some exr coverage) [13:57] <michaelni> durandal_1707, uploaded [14:50] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:5270cb39ba7c: bmp: Fix warning X may be used uninitialized in this function [14:50] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:c10c2aed47da: h264: Silence warning: comparison of distinct pointer types lacks a cast [14:52] <durandal_1707> "libav is more or less only the more API-stable version of ffmpeg. " [15:10] <ubitux> durandal_1707: lol [15:11] <durandal_1707> ubitux: it is one of comments on debian community on g+ [15:11] <cahit> int dataSize = swr_convert(resampleCtx, &resampledOut, AVCODEC_MAX_AUDIO_FRAME_SIZE,(const uint8_t**)decoded_frame.extended_data, (decoded_frame.linesize[0]+decoded_frame.linesize[1]) / (pCodecCtx->channels * 2)); [15:11] <cahit> i am doing something wrong (: [15:12] <cahit> (trying to convert planar pcm to s16 for wav) [15:12] <nevcairiel> isnt there a decoded_frame.samples or something for the number of samples in a frame [15:13] <nevcairiel> besides, the number of samples is defined by one plane, you dont have to add both [15:15] <cahit> and when i'm pushing it to queue, i do this: av_fifo_generic_write(fifoPlayback,(uint8_t *)resampledOut,dataSize*4 [15:16] <cahit> i didn't understand the conversion, so i found out that i have to multiply with 4 by chance. try&fail [15:20] <durandal_1707> cahit: did you explored other usages of swr_convert() in source code? [15:21] <cahit> yes, i actually tried different parts from swresample-test.c [15:30] <cone-978> ffmpeg.git 03Paul B Mahol 07master:df63e0c8bbf3: doc/filters: add forgotten sentence for blend filter examples [15:30] <cahit> i think i have to multiply by 4 as i have devided the length with pCodecCtx->channels * 2, right? [15:30] <durandal_1707> cahit: have you read documentation? [15:31] <cahit> i really did, i think i am not good at understanding differences these days :-/ [15:32] <durandal_1707> https://ffmpeg.org/doxygen/trunk/group__lswr.html#gaa5bb6cab830146efa8c760f… [15:33] <durandal_1707> so 1 sample in s16 case is 2 bytes [15:35] <cahit> i see [15:36] <cahit> thank you for your patience, really.. i got it now. [16:40] <wm4> durandal_1707: looking at this, I wonder what "Convertion will run directly without copying whenever possible." is supposed to mean? [17:01] <durandal_1707> wm4: ? [17:02] <wm4> durandal_1707: doesn't the API copy the sample data from one place to another? [17:02] <durandal_1707> wm4: what API? [17:02] <wm4> swr [17:05] <durandal_1707> michaelni, this sure can be better worded [17:07] <durandal_1707> wm4: i think that means extra copying, but you could try pass in = out and see what happens [17:16] <michaelni> durandal_1707, feel free to change any wording to be better [17:17] <michaelni> iam not the best at wording things ... [17:19] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:02ac3398eb52: rtmpproto: Check APP_MAX_LENGTH [17:29] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:da8ef5ac2f1b: rtmpproto: increase APP_MAX_LENGTH [17:45] Action: durandal_1707 hate potato programmers [19:33] <cone-978> ffmpeg.git 03Michael Niedermayer 07master:73fce258b774: vf_mp: Set pseudo pal [22:39] <cone-267> ffmpeg.git 03Michael Niedermayer 07release/1.1:1f9073f41be1: vf_mp: Set pseudo pal [23:16] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:285485ac5f89: matroskaenc: fix cue tracknum off by 1 error [23:16] <cone-267> ffmpeg.git 03Michael Niedermayer 07master:82d79289db03: avformat: Allocate duration_error separately [00:00] --- Sun Feb 24 2013

1 0

ffmpeg.log.20130223
by burek 24 Feb '13

24 Feb '13

[00:01] <exutux> can someone try espeak command in my paste and convert file into mp3? please :| [00:07] <hughmanwho> Anyone have any suggestions for what to look for if av_read_frame is failing? The error code I get is the negative of context [00:08] <exutux> bye [00:18] <hughmanwho> Having a lot of problems with av_read_frame.. anyone have experience there? [01:21] <NoReflex> hello! I'm trying to record a Sopcast stream to a file using ffmpeg using -c copy. The problem is that the resulting file has problems with seeking. Should I try to re-encode the stream? [01:27] <alyssa_> how can I create thumbnails from one video file at 1 thumbnail per 10 seconds? The thumbnails all being in one image file instead of separate thumbnail files. [01:37] <[locke]> how can i remove the ffmpeg encoding profile from the output file? also same Q including ffmpeg binary information? [01:55] Last message repeated 1 time(s). [11:08] <benkaiser> Hey I am having trouble trying to screencast with ffmpeg. I can do a recording just fine and the audio syncs up however it only comes through on the right audio channel. How can I fix it? (It doesn't matter if the solution is to mirror the right channel to the left). Here is the gist for my screencast with variables in <> https://gist.github.com/benkaiser/5005081 [11:29] Last message repeated 1 time(s). [11:30] <relaxed> Maybe it's an issue with pulse. You could have a channel muted or something. [11:35] <benkaiser> How can I check / view levels for channels? I am running Ubuntu 12.04 [11:36] <hendry> benkaiser: maybe try this line instead https://github.com/kaihendry/recordmydesktop2.0/blob/master/x11capture#L69 [11:36] <relaxed> benkaiser: pavucontrol [11:47] <relaxed> hendry: for prg in xdpyinfo ffmpeg; do type -P $prg || die [11:48] <relaxed> &>>/tmp/r2d2.log [12:01] <benkaiser> relaxed: I checked pavucontrol but it has all the inputs marked as left and right both at 100% [12:05] <spinx60> whats the command for specifying Bitrate in ffmpeg i want to make it to around 600-750 KBs [12:06] <spinx60> im trying to tweak my command for streaming to twitch.tv but im getting really low fps like 16-17 fps conatant need to be able to get above 20 some how [12:06] <juha__> hi. any good ideas on how high a bitrate can be in xvid video converted for hardware mpeg4 player? [12:06] <juha__> i didnt find more specs for the player than just mpeg4 supported ;) obviously not a h264... bought in 2008 [12:06] <spinx60> http://pastebin.com/7rRu0qNj [12:06] <spinx60> command im using [12:07] <durandal_1707> spinx60: what version? [12:07] <spinx60> ffmpeg version git-2013-02-22-db05f7a [12:08] <JEEB> spinx60, when you start the encoding, does libx264 show that is using asm optimizations, in other words does it show SSE2 and so forth used? [12:08] <JEEB> also which preset are you using [12:08] <JEEB> setting a bit rate will not make the encoding go faster [12:09] <spinx60> heres what comes when i stream [12:09] <JEEB> although you will need to use -maxrate and -bufsize when you are outputting to a bandwidth-limited medium, such as networks [12:09] <spinx60> http://pastebin.com/t7FdvSPT [12:09] <JEEB> your streaming provider should tell you what kind of maxrate and how much buffer are set on its player [12:10] <JEEB> ok, so libx264 is built with asm optimizations, great [12:10] <JEEB> what is the preset you're using? [12:10] <spinx60> i dont know :P [12:11] <JEEB> how can you not know [12:11] <JEEB> -preset "$QUAL" [12:11] <spinx60> its all in my command that im using.. mabee i schould add something to it ? [12:11] <JEEB> you set that variable to something [12:11] <JEEB> what are you setting that variable to? [12:12] <spinx60> look here. thats the thing im pretty noob when it comes to this.. but id like to learn http://pastebin.com/K5pr4e1B [12:13] <JEEB> ... [12:13] <JEEB> that doesn't show what you set that variable to [12:13] <JEEB> as I already said [12:13] <spinx60> then where do i set the variable ? [12:13] <JEEB> -preset "$QUAL" , and that means that you set that $QUAL to something before that [12:14] <JEEB> because you are passing the content of that variable to the setting -preset, which controls the speed vs compression setting in libx264 [12:14] <spinx60> yeah im not. hence probably my im having problems.. i had a scripe before but i kinda stopped working and i dont have it anymore. but i got it to stream with only running the command :) [12:14] <JEEB> ... [12:14] <spinx60> so now im trying to figure out how to set it right [12:15] <spinx60> yeah i know ...... im a noob :) [12:15] <JEEB> then I wonder why ffmpeg isn't giving you a warning that it couldn't parse what you were setting -preset to, which means that there is SOMETHING there [12:15] <JEEB> you're not running that command as-is, right? [12:15] <spinx60> yeah [12:16] <spinx60> running directly from shell [12:16] <JEEB> do you have a system variable called QUAL set? [12:17] <spinx60> no [12:17] <JEEB> are you running that command as-is or are you running it off a script? [12:17] <JEEB> because if you are running it as-is it doesn't make any effing sense [12:19] <spinx60> im running it as.is [12:19] <JEEB> I hate x264 --fullhelp for not showing the default settings :| (what preset medium sets) Because now I don't know if you're actually setting -preset to 'fast' and lying to me, or if that just gets set to nothing, and thus it is set to the default medium [12:19] <JEEB> it surely isn't faster or any of the faster presets [12:19] <JEEB> because those have a shorter default rc-lookahead [12:20] <JEEB> anyways, I will guess that just becomes -preset "" [12:20] <JEEB> which I have no idea why it doesn't warning out in ffmpeg [12:20] <spinx60> okey well cant i change the "$QUAL" to a fast/faster/normal preset ? [12:20] <spinx60> or it doesnt work like that ? [12:20] <JEEB> YES, THAT IS WHY I WAS ASKING WHICH FUCKING PRESET YOU WERE USING [12:20] <JEEB> http://mewiki.project357.com/wiki/X264_Settings#preset <- have a list [12:20] <spinx60> Okey chill out man :) il change it and look at the outcome [12:21] <JEEB> just make it f.ex. -preset veryfast [12:21] <JEEB> also as I said, go look up your streaming provider's documentation [12:22] <JEEB> they should tell you the maximum rate allowed and the buffer size there [12:22] <spinx60> yeah will do... by the way 1 more dumb question schould it be like "fast" or just fast ? whitout bunny ears ? [12:23] <JEEB> should work without just fine [12:23] <JEEB> spinx60, you have both $STREAM_KEY and $QUAL in that "command line" of yours btw [12:23] <JEEB> which means that those are fucking variables [12:23] <JEEB> thus I was trying to ask you what the fuck those were set to [12:23] <JEEB> thus you saying that you don't know what those are set to does not make any sense [12:24] <spinx60> to be honest i was amaized as to why i worked running this command aswell [12:24] <JEEB> try echo $STREAM_KEY [12:24] <JEEB> in terminal [12:24] <JEEB> does it give you that? [12:25] <JEEB> because without that key being set to something it doesn't work, naturally [12:25] <JEEB> or [12:25] <spinx60> Wierd it give me nothing http://pastebin.com/LiYFBZS4 [12:26] <JEEB> THEN YOU ARE FUCKING NOT DOING WHAT YOU ARE TELLING ME YOU ARE DOING [12:26] <spinx60> well it gets my key from somewhere thats for sure [12:26] Action: JEEB sighs [12:26] <JEEB> just fucking paste what you are ACTUALLY doing [12:26] <JEEB> you can clear the fucking key manually before pastebin'ing [12:26] <JEEB> I don't give a flying fuck about that [12:26] <JEEB> I just want to know EXACTLY what you are doing [12:27] <JEEB> instead of trying to guess [12:27] <spinx60> okey il paste exactly what im executing [12:27] <spinx60> http://pastebin.com/weFB7TyE [12:28] <spinx60> nothing else am i doing 100% sure. [12:28] <JEEB> and you tell me that in that exact terminal echo $STREAM_KEY gives you nothing? [12:28] <JEEB> also you just pastebin'd your key [12:28] <JEEB> might want to remove that paste [12:29] <spinx60> btw i also do have my twitch key in a file name .twitch_key in my home directory [12:29] <JEEB> anyways, that shows that -preset is set to fast [12:29] <spinx60> doesnt matter i can re create a new key [12:30] <JEEB> if that is not fast enough, pick a faster preset [12:30] <spinx60> yeah now it is becouse i changed that thansk to you [12:30] <spinx60> still get bad fps [12:31] <JEEB> everything but "ultrafast" should be usable from libx264's side [12:31] <spinx60> ok [12:33] <JEEB> naturally, this is all under the thought that libx264 is your actual bottleneck [12:33] <JEEB> if it isn't, you can't really do jack about it [12:34] <spinx60> okey. [12:34] <JEEB> because libx264 is almost the last thing in the chain [12:34] <spinx60> well setting preset to fast/faster or whatever doesnt change anything fps wise at least [12:35] <LithosLaptop> try a different codec maybe? [12:35] <JEEB> spinx60, what if you go more further down the line? [12:36] <spinx60> how you mean further down the line ? [12:36] <JEEB> fast and faster are just the two things to the faster side of medium (default) [12:37] <JEEB> you might as well test out ultrafast and see if it gets faster [12:37] <JEEB> if it doesn't [12:37] <JEEB> then it's something else [12:37] <JEEB> and not libx264 [12:37] <JEEB> that is limiting you [12:37] <JEEB> although wait... [12:37] <JEEB> what CPU is this? [12:38] <spinx60> hmmm ultrafast gave me even less fps [12:38] <JEEB> that makes no sense [12:38] <JEEB> how are you checking fps btw? [12:38] <spinx60> my cpu is a AMD Pheonom II x4 965 BE [12:39] <spinx60> frame= 200 fps= 13 q=32766.0 Lsize= 590kB time=00:00:16.34 bitrate= 295.6kbits/s [12:39] <JEEB> interesting... I would have thought it'd have more capabilities usable [12:39] <spinx60> thats line [12:39] <JEEB> also it becoming slower with a faster preset makes no sense ^^; [12:39] <JEEB> something else is holding you up [12:39] <spinx60> yeah it doesnt [12:39] <JEEB> so let's leave that aside for now [12:40] <JEEB> for streaming [12:40] <JEEB> you need to set vbv maxrate and bufsize [12:40] <JEEB> for ANY streaming [12:40] <spinx60> yeah well i have looked around the net and my bandwith aint the best but its enough i have 1 mb upload but im gonna negotiate for more from my provider [12:40] <JEEB> uhh [12:40] <JEEB> not that you dummy [12:40] <JEEB> now go to your streaming provider(s) and find out what maximum bit rate and what buffering size/length they want [12:40] <spinx60> nahh i was just saying it cant be that [12:41] <JEEB> I know it can't be that [12:41] <JEEB> but you will need to set that in ANY CASE [12:41] <JEEB> so might as well get this done with now [12:41] <JEEB> (read: I have no idea what the fuck is slowing you down and I had started explaining vbv before, so migth as well go back to that for now) [12:42] <JEEB> also it's essential for streaming so you need those values in any case [12:42] <spinx60> where schould i get those values from Twitch.tv ? [12:42] <JEEB> their documentation [12:43] <JEEB> they should tell you what kind of maximum bit rate they want, as well as the buffer size [12:43] <spinx60> lemme check [12:43] <JEEB> also, I would guess that you don't really need to limit -threads to 4 [12:43] <JEEB> you can just remove that [12:46] <spinx60> well around max bitrate 600-750 for my connection its all connection based [12:47] <spinx60> and buffer size around 1500-2000kbs [12:47] <spinx60> okey il remove it [12:47] <JEEB> uhh, that streaming provider has no limits for maximum bit rate? Or do you mean that the streamer's maximum bitrate is bigger than your bandwidth? [12:48] <JEEB> also, "around"? Excuse me? [12:48] <JEEB> it's not an "around" kind of thing [12:50] <spinx60> yeah [12:50] <spinx60> it is [12:50] <spinx60> its alot bigger than my bandwith [12:51] <spinx60> they accept up to 10000kbs bitrate [12:51] <JEEB> ugh, the fact that you're not being specific makes me ache like hell [12:52] <JEEB> just link the documentation :P [12:52] <JEEB> and I'll quickly go through it [12:53] <spinx60> http://www.justin.tv/p/api [12:53] <spinx60> go there [12:56] <spinx60> brb 1 min btw [12:58] <spinx60> back [13:14] <spinx60_> JEEB im back now and i called my internet provider and he boosted my upload to nearly 2-2.5mb upload intsead of 1mb :) [13:14] <spinx60_> so thats alot better [13:16] <spinx60_> im going away a few more mins i need some coca cola :)) [17:40] <ambro718> if I have an MPEG-TS stream (udp multicast), how do I repack it to a form that can be served via an HTTP server and understood by common players? [17:41] <ambro718> this is a technical question, not "what program can do that". Udpxy does exactly that for example. [17:43] <ambro718> I presume it is not ok to just concatenate the udp payloads? [17:43] <ambro718> if one udp packet is cut short or deformed that could confuse the player and make it lose track of message boundaries? [17:58] <defaultro> can ffmpeg pan an image as video? [17:59] <defaultro> or maybe zoom in/out [18:12] <durandal_1707> defaultro: there is crop, pad and scale filter [18:13] <defaultro> so maybe scale would be for zoom [18:15] <durandal_1707> zooming in removes outer stuff [18:15] <defaultro> yup [18:15] <durandal_1707> so there is no simple way to do pan/zoom but there are tools [19:17] <taqattack> i'm trying to use ffmpeg to stream to rtmp server [19:17] <taqattack> i'm using the following command [19:17] <taqattack> ffmpeg.exe -i pipe -vcodec copy -acodec copy -f flv rtmp://url [19:18] <taqattack> if my network speed is too low, the stream sort of lags behind over time [19:18] <taqattack> is there anyway to drop/skip frames to keep it as real-time as possible? [20:56] <mm3> hi, anyone here knows about crtmpserver? [21:51] <shevy> hmm when cutting a .mp3 file with ffmpeg [21:51] <shevy> I specify something like this: [21:51] <shevy> -ss 00:00:36 -t 00:00:44 [21:51] <Tchico> Hi every body [21:51] <shevy> I thought -t specifies the end point, instead it specifies the duration. is there an end-point specification possible, similar to -ss ? [21:51] <JEEB> shevy, not as far as I know [21:52] <shevy> hmm ok, then I will calculate the duration on my own and then pass it to ffmpeg, thanks JEEB [21:52] <Tchico> Did ffmpeg crete good x264 videos from jpg ? Cause when I encode a video from pictures, my chrome browser and IE9 cant read it, but when I do the same encode but from an ogv movies, it works ... [21:53] <Tchico> I do [21:53] <taqattack> can someone help me out with the livestreaming issue? [21:54] <Tchico> ffmpeg -i %d.jpg -vcodec libx264 -vpre slow -vpre baseline -b 250k -bt 50k -an -g 30 -s 640x360 -r 24 -y res.mp4 [21:55] <Tchico> thats create a video which can be read by vlc or windows media player but not from video html5 tag [21:55] <Tchico> and when I take the big_buck_bunny.ogv and I do [21:55] <Tchico> ffmpeg -i big_buck_bunny.ogv -vcodec libx264 -vpre slow -vpre baseline -b 250k -bt 50k -an -g 30 -s 640x360 -r 24 -y res.mp4 [21:56] <Tchico> the result video works fine on all supports ... [21:57] <Tchico> and I am searching since 1 week ... [21:57] <Tchico> I recompile all ffmpeg with all codecs ... [21:58] <Tchico> And Vlc say me that the 2 videos are H264 - MPEG-4 AVC (part10) (avc1) [21:58] <Tchico> with 1 stream for the 2 [22:03] <Tchico> ? [22:11] <Tchico> anybody ? [22:15] <mm3> when I play a live stream on crtmpserver I get a big delay, why? [22:40] <Tchico> nobody ? [22:42] <Fjorgynn> nope [22:43] <Tchico> I will post on the forum so [00:00] --- Sun Feb 24 2013

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